Merge "post proc : volume listener : fix effect release crash"
diff --git a/Android.mk b/Android.mk
index 273c9cb..42f202b 100644
--- a/Android.mk
+++ b/Android.mk
@@ -16,6 +16,7 @@
include $(MY_LOCAL_PATH)/visualizer/Android.mk
include $(MY_LOCAL_PATH)/audiod/Android.mk
include $(MY_LOCAL_PATH)/post_proc/Android.mk
+include $(MY_LOCAL_PATH)/qahw_api/Android.mk
endif
endif
diff --git a/configs/msm8937/audio_policy_configuration.xml b/configs/msm8937/audio_policy_configuration.xml
index 238c49e..44abe28 100644
--- a/configs/msm8937/audio_policy_configuration.xml
+++ b/configs/msm8937/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
@@ -245,13 +245,13 @@
<!-- route declaration, i.e. list all available sources for a given sink -->
<routes>
<route type="mix" sink="Earpiece"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Speaker"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Wired Headset"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Wired Headphones"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Line"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="HDMI"
@@ -268,8 +268,6 @@
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
- <route type="mix" sink="Telephony Tx"
- sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
<route type="mix" sink="voice_rx"
sources="Telephony Rx"/>
</routes>
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index 1202aba..7e5c90f 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -41,7 +41,7 @@
AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
BOARD_USES_SRS_TRUEMEDIA := true
DTS_CODEC_M_ := true
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
MM_AUDIO_ENABLED_FTM := true
MM_AUDIO_ENABLED_SAFX := true
TARGET_USES_QCOM_MM_AUDIO := true
@@ -219,3 +219,7 @@
#Enable HW AAC Encoder by default
PRODUCT_PROPERTY_OVERRIDES += \
qcom.hw.aac.encoder=true
+
+#flac sw decoder 24 bit decode capability
+PRODUCT_PROPERTY_OVERRIDES += \
+flac.sw.decoder.24bit.support=true
diff --git a/configs/msm8953/audio_platform_info_extcodec.xml b/configs/msm8953/audio_platform_info_extcodec.xml
index cf68190..ac0eabc 100644
--- a/configs/msm8953/audio_platform_info_extcodec.xml
+++ b/configs/msm8953/audio_platform_info_extcodec.xml
@@ -47,10 +47,13 @@
<usecase name="USECASE_VOICEMMODE1_CALL" type="out" id="35"/>
<usecase name="USECASE_VOICEMMODE2_CALL" type="in" id="36"/>
<usecase name="USECASE_VOICEMMODE2_CALL" type="out" id="36"/>
+ <usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="37"/>
<usecase name="USECASE_QCHAT_CALL" type="in" id="42"/>
<usecase name="USECASE_QCHAT_CALL" type="out" id="42"/>
</pcm_ids>
<config_params>
+ <param key="spkr_1_tz_name" value="wsatz.11"/>
+ <param key="spkr_2_tz_name" value="wsatz.12"/>
<param key="native_audio_mode" value="src"/>
<param key="input_mic_max_count" value="4"/>
</config_params>
diff --git a/configs/msm8953/audio_policy_configuration.xml b/configs/msm8953/audio_policy_configuration.xml
index 238c49e..44abe28 100644
--- a/configs/msm8953/audio_policy_configuration.xml
+++ b/configs/msm8953/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
@@ -245,13 +245,13 @@
<!-- route declaration, i.e. list all available sources for a given sink -->
<routes>
<route type="mix" sink="Earpiece"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Speaker"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Wired Headset"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Wired Headphones"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Line"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="HDMI"
@@ -268,8 +268,6 @@
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
- <route type="mix" sink="Telephony Tx"
- sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
<route type="mix" sink="voice_rx"
sources="Telephony Rx"/>
</routes>
diff --git a/configs/msm8953/mixer_paths.xml b/configs/msm8953/mixer_paths.xml
index 91544a1..a20c6cf 100644
--- a/configs/msm8953/mixer_paths.xml
+++ b/configs/msm8953/mixer_paths.xml
@@ -54,10 +54,10 @@
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="RX2 Digital Volume" value="84" />
<ctl name="RX3 Digital Volume" value="84" />
- <ctl name="IIR1 INP1 Volume" value="84" />
- <ctl name="IIR1 INP2 Volume" value="84" />
- <ctl name="IIR1 INP3 Volume" value="84" />
- <ctl name="IIR1 INP4 Volume" value="84" />
+ <ctl name="IIR1 INP1 Volume" value="53" />
+ <ctl name="IIR1 INP2 Volume" value="53" />
+ <ctl name="IIR1 INP3 Volume" value="53" />
+ <ctl name="IIR1 INP4 Volume" value="53" />
<ctl name="ADC1 Volume" value="4" />
<ctl name="ADC2 Volume" value="4" />
<ctl name="ADC3 Volume" value="4" />
diff --git a/configs/msm8953/mixer_paths_mtp.xml b/configs/msm8953/mixer_paths_mtp.xml
index 42a9e68..d618169 100644
--- a/configs/msm8953/mixer_paths_mtp.xml
+++ b/configs/msm8953/mixer_paths_mtp.xml
@@ -57,10 +57,10 @@
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="RX2 Digital Volume" value="84" />
<ctl name="RX3 Digital Volume" value="84" />
- <ctl name="IIR1 INP1 Volume" value="84" />
- <ctl name="IIR1 INP2 Volume" value="84" />
- <ctl name="IIR1 INP3 Volume" value="84" />
- <ctl name="IIR1 INP4 Volume" value="84" />
+ <ctl name="IIR1 INP1 Volume" value="53" />
+ <ctl name="IIR1 INP2 Volume" value="53" />
+ <ctl name="IIR1 INP3 Volume" value="53" />
+ <ctl name="IIR1 INP4 Volume" value="53" />
<ctl name="ADC1 Volume" value="4" />
<ctl name="ADC2 Volume" value="4" />
<ctl name="ADC3 Volume" value="4" />
@@ -442,13 +442,22 @@
<ctl name="QUIN_MI2S_RX Audio Mixer MultiMedia7" value="1" />
</path>
+ <path name="compress-offload-playback2 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 usb-headphones">
+ <path name="compress-offload-playback2 afe-proxy" />
+ </path>
+
<path name="compress-offload-playback2 speaker-and-hdmi">
<path name="compress-offload-playback2 hdmi" />
<path name="compress-offload-playback2" />
</path>
- <path name="compress-offload-playback2 afe-proxy">
- <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
+ <path name="compress-offload-playback2 speaker-and-usb-headphones">
+ <path name="compress-offload-playback2 usb-headphones" />
+ <path name="compress-offload-playback2" />
</path>
<path name="compress-offload-playback transmission-fm">
@@ -1105,6 +1114,11 @@
<path name="headphones" />
</path>
+ <path name="wsa-speaker-and-headphones">
+ <path name="wsa-speaker" />
+ <path name="headphones" />
+ </path>
+
<path name="usb-headphones">
</path>
@@ -1119,6 +1133,11 @@
<path name="usb-headphones" />
</path>
+ <path name="wsa-speaker-and-usb-headphones">
+ <path name="wsa-speaker" />
+ <path name="usb-headphones" />
+ </path>
+
<path name="voice-rec-mic">
<path name="handset-mic" />
</path>
@@ -1270,4 +1289,8 @@
<path name="speaker-and-headphones" />
</path>
+ <path name="wsa-speaker-and-line">
+ <path name="wsa-speaker-and-headphones" />
+ </path>
+
</mixer>
diff --git a/configs/msm8953/mixer_paths_qrd_sku3.xml b/configs/msm8953/mixer_paths_qrd_sku3.xml
index 0d68a71..1edb0b4 100644
--- a/configs/msm8953/mixer_paths_qrd_sku3.xml
+++ b/configs/msm8953/mixer_paths_qrd_sku3.xml
@@ -2028,6 +2028,11 @@
<ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
</path>
+ <path name="wsa-speaker-and-headphones">
+ <path name="wsa-speaker" />
+ <path name="headphones" />
+ </path>
+
<path name="usb-headphones">
</path>
@@ -2045,6 +2050,11 @@
<path name="usb-headphones" />
</path>
+ <path name="wsa-speaker-and-usb-headphones">
+ <path name="wsa-speaker" />
+ <path name="usb-headphones" />
+ </path>
+
<path name="speaker-and-hdmi">
<path name="wsa-speaker" />
<path name="hdmi" />
@@ -2089,7 +2099,7 @@
<ctl name="DMIC MUX7" value="DMIC0" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC1" />
+ <ctl name="DMIC MUX8" value="DMIC2" />
<ctl name="SLIM_0_TX Channels" value="Two" />
</path>
@@ -2101,7 +2111,7 @@
<ctl name="DMIC MUX7" value="DMIC0" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC1" />
+ <ctl name="DMIC MUX8" value="DMIC2" />
<ctl name="SLIM_0_TX Channels" value="Two" />
</path>
@@ -2148,7 +2158,7 @@
<ctl name="DMIC MUX7" value="DMIC0" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC1" />
+ <ctl name="DMIC MUX8" value="DMIC2" />
</path>
<path name="dmic-broadside">
@@ -2248,4 +2258,9 @@
<path name="speaker-and-line">
<path name="speaker-and-headphones" />
</path>
+
+ <path name="wsa-speaker-and-line">
+ <path name="wsa-speaker" />
+ <path name="headphones" />
+ </path>
</mixer>
diff --git a/configs/msm8953/mixer_paths_qrd_skuh.xml b/configs/msm8953/mixer_paths_qrd_skuh.xml
index aa70a90..ebf9331 100644
--- a/configs/msm8953/mixer_paths_qrd_skuh.xml
+++ b/configs/msm8953/mixer_paths_qrd_skuh.xml
@@ -57,10 +57,10 @@
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="RX2 Digital Volume" value="84" />
<ctl name="RX3 Digital Volume" value="84" />
- <ctl name="IIR1 INP1 Volume" value="84" />
- <ctl name="IIR1 INP2 Volume" value="84" />
- <ctl name="IIR1 INP3 Volume" value="84" />
- <ctl name="IIR1 INP4 Volume" value="84" />
+ <ctl name="IIR1 INP1 Volume" value="53" />
+ <ctl name="IIR1 INP2 Volume" value="53" />
+ <ctl name="IIR1 INP3 Volume" value="53" />
+ <ctl name="IIR1 INP4 Volume" value="53" />
<ctl name="ADC1 Volume" value="6" />
<ctl name="ADC2 Volume" value="6" />
<ctl name="ADC3 Volume" value="6" />
diff --git a/configs/msm8953/mixer_paths_qrd_skuhf.xml b/configs/msm8953/mixer_paths_qrd_skuhf.xml
index 84b95bd..1ece99d 100644
--- a/configs/msm8953/mixer_paths_qrd_skuhf.xml
+++ b/configs/msm8953/mixer_paths_qrd_skuhf.xml
@@ -57,10 +57,10 @@
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="RX2 Digital Volume" value="84" />
<ctl name="RX3 Digital Volume" value="84" />
- <ctl name="IIR1 INP1 Volume" value="84" />
- <ctl name="IIR1 INP2 Volume" value="84" />
- <ctl name="IIR1 INP3 Volume" value="84" />
- <ctl name="IIR1 INP4 Volume" value="84" />
+ <ctl name="IIR1 INP1 Volume" value="53" />
+ <ctl name="IIR1 INP2 Volume" value="53" />
+ <ctl name="IIR1 INP3 Volume" value="53" />
+ <ctl name="IIR1 INP4 Volume" value="53" />
<ctl name="ADC1 Volume" value="6" />
<ctl name="ADC2 Volume" value="6" />
<ctl name="ADC3 Volume" value="6" />
diff --git a/configs/msm8953/mixer_paths_qrd_skui.xml b/configs/msm8953/mixer_paths_qrd_skui.xml
index aa70a90..ebf9331 100644
--- a/configs/msm8953/mixer_paths_qrd_skui.xml
+++ b/configs/msm8953/mixer_paths_qrd_skui.xml
@@ -57,10 +57,10 @@
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="RX2 Digital Volume" value="84" />
<ctl name="RX3 Digital Volume" value="84" />
- <ctl name="IIR1 INP1 Volume" value="84" />
- <ctl name="IIR1 INP2 Volume" value="84" />
- <ctl name="IIR1 INP3 Volume" value="84" />
- <ctl name="IIR1 INP4 Volume" value="84" />
+ <ctl name="IIR1 INP1 Volume" value="53" />
+ <ctl name="IIR1 INP2 Volume" value="53" />
+ <ctl name="IIR1 INP3 Volume" value="53" />
+ <ctl name="IIR1 INP4 Volume" value="53" />
<ctl name="ADC1 Volume" value="6" />
<ctl name="ADC2 Volume" value="6" />
<ctl name="ADC3 Volume" value="6" />
diff --git a/configs/msm8953/mixer_paths_qrd_skum.xml b/configs/msm8953/mixer_paths_qrd_skum.xml
index 8343847..d504456 100644
--- a/configs/msm8953/mixer_paths_qrd_skum.xml
+++ b/configs/msm8953/mixer_paths_qrd_skum.xml
@@ -57,10 +57,10 @@
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="RX2 Digital Volume" value="84" />
<ctl name="RX3 Digital Volume" value="84" />
- <ctl name="IIR1 INP1 Volume" value="84" />
- <ctl name="IIR1 INP2 Volume" value="84" />
- <ctl name="IIR1 INP3 Volume" value="84" />
- <ctl name="IIR1 INP4 Volume" value="84" />
+ <ctl name="IIR1 INP1 Volume" value="53" />
+ <ctl name="IIR1 INP2 Volume" value="53" />
+ <ctl name="IIR1 INP3 Volume" value="53" />
+ <ctl name="IIR1 INP4 Volume" value="53" />
<ctl name="ADC1 Volume" value="6" />
<ctl name="ADC2 Volume" value="6" />
<ctl name="ADC3 Volume" value="6" />
@@ -366,6 +366,28 @@
<ctl name="INTERNAL_FM_RX Audio Mixer MultiMedia4" value="1" />
</path>
+ <path name="compress-offload-playback2 hdmi">
+ <ctl name="QUIN_MI2S_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 usb-headphones">
+ <path name="compress-offload-playback2 afe-proxy" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-hdmi">
+ <path name="compress-offload-playback2 hdmi" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-usb-headphones">
+ <path name="compress-offload-playback2 usb-headphones" />
+ <path name="compress-offload-playback2" />
+ </path>
+
<path name="compress-offload-playback3">
<ctl name="PRI_MI2S_RX Audio Mixer MultiMedia10" value="1" />
</path>
@@ -795,6 +817,11 @@
<path name="headphones" />
</path>
+ <path name="wsa-speaker-and-headphones">
+ <path name="wsa-speaker" />
+ <path name="headphones" />
+ </path>
+
<path name="usb-headphones">
</path>
@@ -809,6 +836,11 @@
<path name="usb-headphones" />
</path>
+ <path name="wsa-speaker-and-usb-headphones">
+ <path name="wsa-speaker" />
+ <path name="usb-headphones" />
+ </path>
+
<path name="voice-rec-mic">
<path name="handset-mic" />
</path>
@@ -934,4 +966,8 @@
<path name="speaker-and-headphones" />
</path>
+ <path name="wsa-speaker-and-line">
+ <path name="wsa-speaker-and-headphones" />
+ </path>
+
</mixer>
diff --git a/configs/msm8953/mixer_paths_skuk.xml b/configs/msm8953/mixer_paths_skuk.xml
index 98a1bab..1538275 100644
--- a/configs/msm8953/mixer_paths_skuk.xml
+++ b/configs/msm8953/mixer_paths_skuk.xml
@@ -57,10 +57,10 @@
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="RX2 Digital Volume" value="84" />
<ctl name="RX3 Digital Volume" value="84" />
- <ctl name="IIR1 INP1 Volume" value="84" />
- <ctl name="IIR1 INP2 Volume" value="84" />
- <ctl name="IIR1 INP3 Volume" value="84" />
- <ctl name="IIR1 INP4 Volume" value="84" />
+ <ctl name="IIR1 INP1 Volume" value="53" />
+ <ctl name="IIR1 INP2 Volume" value="53" />
+ <ctl name="IIR1 INP3 Volume" value="53" />
+ <ctl name="IIR1 INP4 Volume" value="53" />
<ctl name="ADC1 Volume" value="6" />
<ctl name="ADC2 Volume" value="6" />
<ctl name="ADC3 Volume" value="6" />
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index e646646..2917f9d 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -41,7 +41,7 @@
BOARD_USES_SRS_TRUEMEDIA := true
DTS_CODEC_M_ := true
#AUDIO_FEATURE_ENABLED_MULTIPLE_TUNNEL := true
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
MM_AUDIO_ENABLED_FTM := true
MM_AUDIO_ENABLED_SAFX := true
TARGET_USES_QCOM_MM_AUDIO := true
@@ -219,3 +219,7 @@
#Enable HW AAC Encoder by default
PRODUCT_PROPERTY_OVERRIDES += \
qcom.hw.aac.encoder=true
+
+#flac sw decoder 24 bit decode capability
+PRODUCT_PROPERTY_OVERRIDES += \
+flac.sw.decoder.24bit.support=true
diff --git a/configs/msm8996/audio_policy_configuration.xml b/configs/msm8996/audio_policy_configuration.xml
index 56848ad..e8d4cd0 100644
--- a/configs/msm8996/audio_policy_configuration.xml
+++ b/configs/msm8996/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
@@ -139,7 +139,7 @@
<mixPort name="voip_rx" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_VOIP_RX">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ samplingRates="8000,16000,32000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
</mixPort>
<mixPort name="primary input" role="sink">
@@ -256,13 +256,13 @@
<!-- route declaration, i.e. list all available sources for a given sink -->
<routes>
<route type="mix" sink="Earpiece"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Speaker"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Wired Headset"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Wired Headphones"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Line"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="HDMI"
@@ -281,8 +281,6 @@
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="record_24"
sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
- <route type="mix" sink="Telephony Tx"
- sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
<route type="mix" sink="voice_rx"
sources="Telephony Rx"/>
</routes>
diff --git a/configs/msm8996/mixer_paths_tasha.xml b/configs/msm8996/mixer_paths_tasha.xml
index 5741192..9f63413 100644
--- a/configs/msm8996/mixer_paths_tasha.xml
+++ b/configs/msm8996/mixer_paths_tasha.xml
@@ -1975,8 +1975,7 @@
</path>
<path name="handset-mic-sbc">
- <path name="adc5" />
- <ctl name="ADC5 Volume" value="12" />
+ <path name="dmic3" />
</path>
<path name="three-mic">
diff --git a/configs/msm8996/msm8996.mk b/configs/msm8996/msm8996.mk
index 3b83c24..306fa97 100644
--- a/configs/msm8996/msm8996.mk
+++ b/configs/msm8996/msm8996.mk
@@ -41,7 +41,7 @@
AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
BOARD_USES_SRS_TRUEMEDIA := true
DTS_CODEC_M_ := true
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
MM_AUDIO_ENABLED_FTM := true
MM_AUDIO_ENABLED_SAFX := true
TARGET_USES_QCOM_MM_AUDIO := true
@@ -50,6 +50,7 @@
#DOLBY_DDP := true
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
+AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
##AUDIO_FEATURE_FLAGS
#Audio Specific device overlays
@@ -196,3 +197,7 @@
use.qti.sw.alac.decoder=true
PRODUCT_PROPERTY_OVERRIDES += \
use.qti.sw.ape.decoder=true
+
+#flac sw decoder 24 bit decode capability
+PRODUCT_PROPERTY_OVERRIDES += \
+flac.sw.decoder.24bit.support=true
diff --git a/configs/msmcobalt/aanc_tuning_mixer.txt b/configs/msmcobalt/aanc_tuning_mixer.txt
index 35743ff..5639bd2 100644
--- a/configs/msmcobalt/aanc_tuning_mixer.txt
+++ b/configs/msmcobalt/aanc_tuning_mixer.txt
@@ -8,10 +8,10 @@
AIF1_CAP Mixer SLIM TX8:1
SLIM TX7 MUX:DEC7
ADC MUX7:DMIC
-DMIC MUX7:DMIC3
+DMIC MUX7:DMIC4
SLIM TX8 MUX:DEC8
ADC MUX8:DMIC
-DMIC MUX8:DMIC2
+DMIC MUX8:DMIC0
SLIM_0_TX Channels:Two
MultiMedia1 Mixer SLIM_0_TX:1
@@ -35,13 +35,13 @@
SLIM TX7 MUX:DEC7
ADC MUX7:ANC_FB_TUNE1
ADC MUX10:DMIC
-DMIC MUX10:DMIC2
+DMIC MUX10:DMIC0
SLIM TX8 MUX:DEC8
ADC MUX8:ANC_FB_TUNE2
ADC MUX12:DMIC
-DMIC MUX12:DMIC2
-ANC0 FB MUX:ANC_IN_EAR
-ANC EAR Enable Switch:1
+DMIC MUX12:DMIC0
+ANC0 FB MUX:ANC_IN_EAR_SPKR
+ANC OUT EAR SPKR Enable Switch:1
SLIM_0_TX Channels:Two
MultiMedia1 Mixer SLIM_0_TX:1
@@ -56,7 +56,7 @@
DMIC MUX10:ZERO
DMIC MUX12:ZERO
ANC0 FB MUX:ZERO
-ANC EAR Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
#ANC_TEST_E_PATH_MIC_STEREO Capture
@@ -70,13 +70,13 @@
SLIM TX7 MUX:DEC7
ADC MUX7:ANC_FB_TUNE1
ADC MUX10:DMIC
-DMIC MUX10:DMIC3
+DMIC MUX10:DMIC4
SLIM TX8 MUX:DEC8
ADC MUX8:ANC_FB_TUNE2
ADC MUX12:DMIC
-DMIC MUX12:DMIC3
-ANC0 FB MUX:ANC_IN_EAR
-ANC EAR Enable Switch:1
+DMIC MUX12:DMIC4
+ANC0 FB MUX:ANC_IN_EAR_SPKR
+ANC OUT EAR SPKR Enable Switch:1
SLIM_0_TX Channels:Two
MultiMedia1 Mixer SLIM_0_TX:1
@@ -91,7 +91,7 @@
DMIC MUX10:ZERO
DMIC MUX12:ZERO
ANC0 FB MUX:ZERO
-ANC EAR Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
#ANC_TEST_S_PATH_HANDSET_SPKR_ANC_MONO
@@ -103,19 +103,23 @@
ANC Function:ON
SLIM RX0 MUX:AIF_MIX1_PB
SLIM_0_RX Channels:One
-RX INT0_1 MIX1 INP0:RX0
-RX INT0 DEM MUX:CLSH_DSM_OUT
-RX0 Digital Volume:87
+RX INT7_1 MIX1 INP0:RX0
+SpkrLeft SWR DAC_Port Switch:1
ANC Slot:7
-EAR PA Gain:G_6_DB
+ANC OUT EAR SPKR Enable Switch:1
+ANC SPKR PA Enable Switch:1
+SpkrLeft WSA PA Gain:G_6_DB
SLIMBUS_0_RX Audio Mixer MultiMedia1:1
disable
SLIMBUS_0_RX Audio Mixer MultiMedia1:0
+SpkrLeft WSA PA Gain:G_0_DB
+ANC SPKR PA Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
+SpkrLeft SWR DAC_Port Switch:0
ANC Slot:0
SLIM RX0 MUX:ZERO
-RX INT0_1 MIX1 INP0:ZERO
-RX0 Digital Volume:0
+RX INT7_1 MIX1 INP0:ZERO
ANC Function:OFF
#ANC_TEST_E_PATH_HANDSET_SPKR_ANC_MONO
@@ -127,17 +131,21 @@
ANC Function:ON
SLIM RX0 MUX:AIF_MIX1_PB
SLIM_0_RX Channels:One
-RX INT0_1 MIX1 INP0:RX0
-RX INT0 DEM MUX:CLSH_DSM_OUT
-RX0 Digital Volume:87
+RX INT7_1 MIX1 INP0:RX0
+SpkrLeft SWR DAC_Port Switch:1
ANC Slot:8
-EAR PA Gain:G_6_DB
+ANC OUT EAR SPKR Enable Switch:1
+ANC SPKR PA Enable Switch:1
+SpkrLeft WSA PA Gain:G_6_DB
SLIMBUS_0_RX Audio Mixer MultiMedia1:1
disable
SLIMBUS_0_RX Audio Mixer MultiMedia1:0
+SpkrLeft WSA PA Gain:G_0_DB
+ANC SPKR PA Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
+SpkrLeft SWR DAC_Port Switch:0
ANC Slot:0
SLIM RX0 MUX:ZERO
-RX INT0_1 MIX1 INP0:ZERO
-RX0 Digital Volume:0
+RX INT7_1 MIX1 INP0:ZERO
ANC Function:OFF
diff --git a/configs/msmcobalt/aanc_tuning_mixer_tavil.txt b/configs/msmcobalt/aanc_tuning_mixer_tavil.txt
new file mode 100644
index 0000000..78156d3
--- /dev/null
+++ b/configs/msmcobalt/aanc_tuning_mixer_tavil.txt
@@ -0,0 +1,151 @@
+#ANC_TEST_P_PATH_MIC_STEREO Capture
+acdb_dev_id:85
+!Capture
+Txdevice:0
+
+enable
+AIF1_CAP Mixer SLIM TX7:1
+AIF1_CAP Mixer SLIM TX8:1
+CDC_IF TX7 MUX:DEC7
+ADC MUX7:DMIC
+DMIC MUX7:DMIC4
+CDC_IF TX8 MUX:DEC8
+ADC MUX8:DMIC
+DMIC MUX8:DMIC0
+SLIM_0_TX Channels:Two
+MultiMedia1 Mixer SLIM_0_TX:1
+
+disable
+MultiMedia1 Mixer SLIM_0_TX:0
+AIF1_CAP Mixer SLIM TX7:0
+AIF1_CAP Mixer SLIM TX8:0
+CDC_IF TX7 MUX:ZERO
+CDC_IF TX8 MUX:ZERO
+DMIC MUX7:ZERO
+DMIC MUX8:ZERO
+
+#ANC_TEST_S_PATH_MIC_STEREO Capture
+acdb_dev_id:88
+!Capture
+Txdevice:0
+
+enable
+AIF1_CAP Mixer SLIM TX7:1
+AIF1_CAP Mixer SLIM TX8:1
+CDC_IF TX7 MUX:DEC7
+ADC MUX7:ANC_FB_TUNE1
+ADC MUX10:DMIC
+DMIC MUX10:DMIC0
+CDC_IF TX8 MUX:DEC8
+ADC MUX8:ANC_FB_TUNE2
+ADC MUX12:DMIC
+DMIC MUX12:DMIC0
+ANC0 FB MUX:ANC_IN_EAR_SPKR
+ANC OUT EAR SPKR Enable Switch:1
+SLIM_0_TX Channels:Two
+MultiMedia1 Mixer SLIM_0_TX:1
+
+disable
+MultiMedia1 Mixer SLIM_0_TX:0
+AIF1_CAP Mixer SLIM TX7:0
+AIF1_CAP Mixer SLIM TX8:0
+CDC_IF TX7 MUX:ZERO
+CDC_IF TX8 MUX:ZERO
+ADC MUX7:DMIC
+ADC MUX8:DMIC
+DMIC MUX10:ZERO
+DMIC MUX12:ZERO
+ANC0 FB MUX:ZERO
+ANC OUT EAR SPKR Enable Switch:0
+
+
+#ANC_TEST_E_PATH_MIC_STEREO Capture
+acdb_dev_id:91
+!Capture
+Txdevice:0
+
+enable
+AIF1_CAP Mixer SLIM TX7:1
+AIF1_CAP Mixer SLIM TX8:1
+CDC_IF TX7 MUX:DEC7
+ADC MUX7:ANC_FB_TUNE1
+ADC MUX10:DMIC
+DMIC MUX10:DMIC4
+CDC_IF TX8 MUX:DEC8
+ADC MUX8:ANC_FB_TUNE2
+ADC MUX12:DMIC
+DMIC MUX12:DMIC4
+ANC0 FB MUX:ANC_IN_EAR_SPKR
+ANC OUT EAR SPKR Enable Switch:1
+SLIM_0_TX Channels:Two
+MultiMedia1 Mixer SLIM_0_TX:1
+
+disable
+MultiMedia1 Mixer SLIM_0_TX:0
+AIF1_CAP Mixer SLIM TX7:0
+AIF1_CAP Mixer SLIM TX8:0
+CDC_IF TX7 MUX:ZERO
+CDC_IF TX8 MUX:ZERO
+ADC MUX7:DMIC
+ADC MUX8:DMIC
+DMIC MUX10:ZERO
+DMIC MUX12:ZERO
+ANC0 FB MUX:ZERO
+ANC OUT EAR SPKR Enable Switch:0
+
+
+#ANC_TEST_S_PATH_HANDSET_SPKR_ANC_MONO
+acdb_dev_id:86
+!Playback
+Rxdevice:0
+
+enable
+ANC Function:ON
+SLIM RX0 MUX:AIF1_PB
+SLIM_0_RX Channels:One
+RX INT7_1 MIX1 INP0:RX0
+SpkrLeft SWR DAC_Port Switch:1
+ANC Slot:7
+ANC OUT EAR SPKR Enable Switch:1
+ANC SPKR PA Enable Switch:1
+SpkrLeft WSA PA Gain:G_6_DB
+SLIMBUS_0_RX Audio Mixer MultiMedia1:1
+
+disable
+SLIMBUS_0_RX Audio Mixer MultiMedia1:0
+SpkrLeft WSA PA Gain:G_0_DB
+ANC SPKR PA Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
+SpkrLeft SWR DAC_Port Switch:0
+ANC Slot:0
+SLIM RX0 MUX:ZERO
+RX INT7_1 MIX1 INP0:ZERO
+ANC Function:OFF
+
+#ANC_TEST_E_PATH_HANDSET_SPKR_ANC_MONO
+acdb_dev_id:89
+!Playback
+Rxdevice:0
+
+enable
+ANC Function:ON
+SLIM RX0 MUX:AIF1_PB
+SLIM_0_RX Channels:One
+RX INT7_1 MIX1 INP0:RX0
+SpkrLeft SWR DAC_Port Switch:1
+ANC Slot:8
+ANC OUT EAR SPKR Enable Switch:1
+ANC SPKR PA Enable Switch:1
+SpkrLeft WSA PA Gain:G_6_DB
+SLIMBUS_0_RX Audio Mixer MultiMedia1:1
+
+disable
+SLIMBUS_0_RX Audio Mixer MultiMedia1:0
+SpkrLeft WSA PA Gain:G_0_DB
+ANC SPKR PA Enable Switch:0
+ANC OUT EAR SPKR Enable Switch:0
+SpkrLeft SWR DAC_Port Switch:0
+ANC Slot:0
+SLIM RX0 MUX:ZERO
+RX INT7_1 MIX1 INP0:ZERO
+ANC Function:OFF
diff --git a/configs/msmcobalt/audio_effects.conf b/configs/msmcobalt/audio_effects.conf
index 5738cf9..d643592 100644
--- a/configs/msmcobalt/audio_effects.conf
+++ b/configs/msmcobalt/audio_effects.conf
@@ -48,6 +48,9 @@
volume_listener {
path /system/lib/soundfx/libvolumelistener.so
}
+ audiosphere {
+ path /system/lib/soundfx/libasphere.so
+ }
}
# Default pre-processing library. Add to audio_effect.conf "libraries" section if
@@ -252,6 +255,10 @@
library volume_listener
uuid 0b776dde-0590-11e5-81ba-0025b32654a0
}
+ audiosphere {
+ library audiosphere
+ uuid 184e62ab-2d19-4364-9d1b-c0a40733866c
+ }
}
# additional effect from vendor
diff --git a/configs/msmcobalt/audio_output_policy.conf b/configs/msmcobalt/audio_output_policy.conf
index 67d79bf..e60c664 100644
--- a/configs/msmcobalt/audio_output_policy.conf
+++ b/configs/msmcobalt/audio_output_policy.conf
@@ -42,14 +42,14 @@
direct_pcm_24 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
- sampling_rates 44100|48000|96000|192000
+ sampling_rates 44100|48000|96000|176400|192000|352800
bit_width 24
app_type 69940
}
compress_passthrough_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING|AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
- formats AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_E_AC3_JOC|AUDIO_FORMAT_DTS|AUDIO_FORMAT_DTS_HD
- sampling_rates 32000|44100|48000|88200|96000|176400|192000
+ formats AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_E_AC3_JOC|AUDIO_FORMAT_DTS|AUDIO_FORMAT_DTS_HD|AUDIO_FORMAT_DSD
+ sampling_rates 32000|44100|48000|88200|96000|176400|192000|352800
bit_width 16
app_type 69941
}
diff --git a/configs/msmcobalt/audio_platform_info.xml b/configs/msmcobalt/audio_platform_info.xml
index 72ed9f3..07839fd 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -53,6 +53,11 @@
<usecase name="USECASE_AUDIO_PLAYBACK_FM" type="in" id="34"/>
<usecase name="USECASE_AUDIO_SPKR_CALIB_RX" type="out" id="5"/>
<usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="35"/>
+ <usecase name="USECASE_AUDIO_PLAYBACK_AFE_PROXY" type="out" id="6"/>
+ <usecase name="USECASE_AUDIO_RECORD_AFE_PROXY" type="in" id="7"/>
+ <usecase name="USECASE_AUDIO_RECORD_LOW_LATENCY" type="in" id="17" />
+ <usecase name="USECASE_AUDIO_PLAYBACK_ULL" type="out" id="17" />
+ <usecase name="USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE" type="out" id="27" />
</pcm_ids>
<config_params>
<param key="spkr_1_tz_name" value="wsatz.13"/>
@@ -68,9 +73,12 @@
<backend_names>
<device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_ANC_HEADSET" backend="headphones" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_SPEAKER_AND_LINE" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_VOICE_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_ANC_HEADSET" backend="headphones" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
diff --git a/configs/msmcobalt/audio_policy.conf b/configs/msmcobalt/audio_policy.conf
index dd827fe..70ab311 100644
--- a/configs/msmcobalt/audio_policy.conf
+++ b/configs/msmcobalt/audio_policy.conf
@@ -26,21 +26,21 @@
sampling_rates 44100|48000
channel_masks AUDIO_CHANNEL_OUT_STEREO
formats AUDIO_FORMAT_PCM_16_BIT
- devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE
+ devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY
}
raw {
sampling_rates 48000
channel_masks AUDIO_CHANNEL_OUT_STEREO
formats AUDIO_FORMAT_PCM_16_BIT
- devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE
+ devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
flags AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW
}
deep_buffer {
sampling_rates 44100|48000
channel_masks AUDIO_CHANNEL_OUT_STEREO
formats AUDIO_FORMAT_PCM_16_BIT
- devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE
+ devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_FM|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
flags AUDIO_OUTPUT_FLAG_DEEP_BUFFER
}
compress_passthrough {
@@ -58,17 +58,24 @@
flags AUDIO_OUTPUT_FLAG_DIRECT
}
direct_pcm {
- sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
+ sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000|352800
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
- devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE
+ devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
}
compress_offload {
sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
- devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE
+ devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES|AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
+ flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
+ }
+ dsd_compress_passthrough {
+ sampling_rates 2822400|5644800
+ channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO
+ formats AUDIO_FORMAT_DSD
+ devices AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
}
incall_music {
@@ -108,14 +115,6 @@
}
}
a2dp {
- outputs {
- a2dp {
- sampling_rates 44100
- channel_masks AUDIO_CHANNEL_OUT_STEREO
- formats AUDIO_FORMAT_PCM_16_BIT
- devices AUDIO_DEVICE_OUT_ALL_A2DP
- }
- }
inputs {
a2dp {
sampling_rates 44100|48000
diff --git a/configs/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index a3876ef..66b7d17 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -86,13 +86,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
@@ -118,6 +118,21 @@
<profile name="" format="AUDIO_FORMAT_AAC_HE_V2"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_DTS"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_DTS_HD"
+ samplingRates="32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_WMA"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
@@ -137,6 +152,12 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
</mixPort>
+ <mixPort name="dsd_compress_passthrough" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+ <profile name="" format="AUDIO_FORMAT_DSD"
+ samplingRates="2822400,5644800"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+ </mixPort>
<mixPort name="voice_tx" role="source">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
@@ -144,7 +165,7 @@
<mixPort name="voip_rx" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_VOIP_RX">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
</mixPort>
<mixPort name="primary input" role="sink">
@@ -228,6 +249,22 @@
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
+ <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="USB Device Out" type="AUDIO_DEVICE_OUT_USB_DEVICE" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
<devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
@@ -257,19 +294,25 @@
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
</devicePort>
+ <devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
+ samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
</devicePorts>
<!-- route declaration, i.e. list all available sources for a given sink -->
<routes>
<route type="mix" sink="Earpiece"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic"/>
- <route type="mix" sink="Speaker"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
- <route type="mix" sink="Wired Headset"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
- <route type="mix" sink="Wired Headphones"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
- <route type="mix" sink="Line"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+ <route type="mix" sink="Speaker"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+ <route type="mix" sink="Wired Headset"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
+ <route type="mix" sink="Wired Headphones"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
+ <route type="mix" sink="Line"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,dsd_compress_passthrough,voip_rx"/>
<route type="mix" sink="HDMI"
sources="primary output,raw,deep_buffer,multichannel,direct_pcm,compressed_offload,compress_passthrough"/>
<route type="mix" sink="Proxy"
@@ -278,27 +321,69 @@
sources="primary output"/>
<route type="mix" sink="BT SCO All"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
+ <route type="mix" sink="USB Device Out"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Telephony Tx"
sources="voice_tx"/>
+ <route type="mix" sink="voice_rx"
+ sources="Telephony Rx"/>
<route type="mix" sink="primary input"
- sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
+ sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,Telephony Rx"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="record_24"
sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
- <route type="mix" sink="Telephony Tx"
- sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
- <route type="mix" sink="voice_rx"
- sources="Telephony Rx"/>
+ <route type="mix" sink="BT A2DP Out"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+ <route type="mix" sink="BT A2DP Headphones"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
+ <route type="mix" sink="BT A2DP Speaker"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload"/>
</routes>
</module>
- <!-- A2dp Audio HAL -->
- <xi:include href="a2dp_audio_policy_configuration.xml"/>
+ <!-- A2DP Audio HAL -->
+ <module name="a2dp" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="a2dp input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+
+ <devicePorts>
+ <devicePort tagName="BT A2DP In" type="AUDIO_DEVICE_IN_BLUETOOTH_A2DP" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+
+ <routes>
+ <route type="mix" sink="a2dp input"
+ sources="BT A2DP In"/>
+ </routes>
+ </module>
<!-- Usb Audio HAL -->
- <xi:include href="usb_audio_policy_configuration.xml"/>
+ <module name="usb" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="usb_accessory output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="USB Host Out" type="AUDIO_DEVICE_OUT_USB_ACCESSORY" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="USB Host Out"
+ sources="usb_accessory output"/>
+ </routes>
+ </module>
<!-- Remote Submix Audio HAL -->
<xi:include href="r_submix_audio_policy_configuration.xml"/>
diff --git a/configs/msmcobalt/graphite_ipc_platform_info.xml b/configs/msmcobalt/graphite_ipc_platform_info.xml
new file mode 100644
index 0000000..f6775be
--- /dev/null
+++ b/configs/msmcobalt/graphite_ipc_platform_info.xml
@@ -0,0 +1,47 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!--- Copyright (c) 2016, The Linux Foundation. All rights reserved. -->
+<!--- -->
+<!--- Redistribution and use in source and binary forms, with or without -->
+<!--- modification, are permitted provided that the following conditions are -->
+<!--- met: -->
+<!--- * Redistributions of source code must retain the above copyright -->
+<!--- notice, this list of conditions and the following disclaimer. -->
+<!--- * Redistributions in binary form must reproduce the above -->
+<!--- copyright notice, this list of conditions and the following -->
+<!--- disclaimer in the documentation and/or other materials provided -->
+<!--- with the distribution. -->
+<!--- * Neither the name of The Linux Foundation nor the names of its -->
+<!--- contributors may be used to endorse or promote products derived -->
+<!--- from this software without specific prior written permission. -->
+<!--- -->
+<!--- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED -->
+<!--- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF -->
+<!--- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT -->
+<!--- ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS -->
+<!--- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR -->
+<!--- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF -->
+<!--- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR -->
+<!--- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, -->
+<!--- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE -->
+<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
+<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -->
+<graphite_ipc_platform_info>
+ <no_of_glink_channels value="4">
+ </no_of_glink_channels>
+ <!-- channel 1 configuration -->
+ <glink_channel name="g_glink_ctrl" latency_in_us="5000"
+ no_of_intents="1" intents_size="1024">
+ </glink_channel>
+ <!-- channel 2 configuration -->
+ <glink_channel name="g_glink_persistent_data_ild" latency_in_us="30000"
+ no_of_intents="0">
+ </glink_channel>
+ <!-- channel 3 configuration -->
+ <glink_channel name="g_glink_persistent_data_nild" latency_in_us="30000"
+ no_of_intents="0">
+ </glink_channel>
+ <!-- channel 4 configuration -->
+ <glink_channel name="g_glink_audio_data" latency_in_us="10000"
+ no_of_intents="2" intents_size="4096, 4096">
+ </glink_channel>
+</graphite_ipc_platform_info>
diff --git a/configs/msmcobalt/mixer_paths_dtp.xml b/configs/msmcobalt/mixer_paths_dtp.xml
index 9bcf15b..a6c61e4 100644
--- a/configs/msmcobalt/mixer_paths_dtp.xml
+++ b/configs/msmcobalt/mixer_paths_dtp.xml
@@ -138,6 +138,8 @@
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
@@ -617,7 +619,7 @@
</path>
<path name="audio-ull-playback">
- <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback speaker-protected">
@@ -634,11 +636,11 @@
</path>
<path name="audio-ull-playback hdmi">
- <ctl name="HDMI Mixer MultiMedia3" value="1" />
+ <ctl name="HDMI Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback bt-sco">
- <ctl name="AUX_PCM_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback bt-sco-wb">
@@ -652,7 +654,7 @@
</path>
<path name="audio-ull-playback afe-proxy">
- <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="multi-channel-playback hdmi">
<ctl name="HDMI Mixer MultiMedia2" value="1" />
@@ -1103,11 +1105,11 @@
</path>
<path name="low-latency-record">
- <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
</path>
<path name="low-latency-record bt-sco">
- <ctl name="MultiMedia5 Mixer AUX_PCM_UL_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer AUX_PCM_UL_TX" value="1" />
</path>
<path name="low-latency-record bt-sco-wb">
@@ -1116,11 +1118,11 @@
</path>
<path name="low-latency-record usb-headset-mic">
- <ctl name="MultiMedia5 Mixer AFE_PCM_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer AFE_PCM_TX" value="1" />
</path>
<path name="low-latency-record capture-fm">
- <ctl name="MultiMedia5 Mixer TERT_MI2S_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer TERT_MI2S_TX" value="1" />
</path>
<path name="fm-virtual-record capture-fm">
diff --git a/configs/msmcobalt/mixer_paths_skuk.xml b/configs/msmcobalt/mixer_paths_skuk.xml
new file mode 100644
index 0000000..24499b9
--- /dev/null
+++ b/configs/msmcobalt/mixer_paths_skuk.xml
@@ -0,0 +1,2415 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!-- Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. -->
+<!-- -->
+<!-- Redistribution and use in source and binary forms, with or without -->
+<!-- modification, are permitted provided that the following conditions are -->
+<!-- met: -->
+<!-- * Redistributions of source code must retain the above copyright -->
+<!-- notice, this list of conditions and the following disclaimer. -->
+<!-- * Redistributions in binary form must reproduce the above -->
+<!-- copyright notice, this list of conditions and the following -->
+<!-- disclaimer in the documentation and/or other materials provided -->
+<!-- with the distribution. -->
+<!-- * Neither the name of The Linux Foundation nor the names of its -->
+<!-- contributors may be used to endorse or promote products derived -->
+<!-- from this software without specific prior written permission. -->
+<!-- -->
+<!-- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED -->
+<!-- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF -->
+<!-- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT -->
+<!-- ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS -->
+<!-- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR -->
+<!-- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF -->
+<!-- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR -->
+<!-- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, -->
+<!-- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE -->
+<!-- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
+<!-- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -->
+<mixer>
+ <!-- These are the initial mixer settings -->
+ <ctl name="Voice Rx Device Mute" id="0" value="0" />
+ <ctl name="Voice Rx Device Mute" id="1" value="-1" />
+ <ctl name="Voice Rx Device Mute" id="2" value="20" />
+ <ctl name="Voice Tx Mute" id="0" value="0" />
+ <ctl name="Voice Tx Mute" id="1" value="-1" />
+ <ctl name="Voice Tx Mute" id="2" value="500" />
+ <ctl name="Voice Rx Gain" id="0" value="0" />
+ <ctl name="Voice Rx Gain" id="1" value="-1" />
+ <ctl name="Voice Rx Gain" id="2" value="20" />
+ <ctl name="Voip Tx Mute" id="0" value="0" />
+ <ctl name="Voip Tx Mute" id="1" value="500" />
+ <ctl name="Voip Rx Gain" id="0" value="0" />
+ <ctl name="Voip Rx Gain" id="1" value="20" />
+ <ctl name="Voip Mode Config" value="12" />
+ <ctl name="Voip Rate Config" value="0" />
+ <ctl name="Voip Evrc Min Max Rate Config" id="0" value="1" />
+ <ctl name="Voip Evrc Min Max Rate Config" id="1" value="4" />
+ <ctl name="Voip Dtx Mode" value="0" />
+ <ctl name="TTY Mode" value="OFF" />
+ <ctl name="SLIMBUS_0_RX Port Mixer SLIM_0_TX" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_6_RX Port Mixer SLIM_0_TX" value="0" />
+ <ctl name="SLIMBUS_4_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="SLIMBUS_4_RX Audio Mixer MultiMedia2" value="0" />
+ <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="0" />
+ <ctl name="MultiMedia5 Mixer AFE_PCM_TX" value="0" />
+ <ctl name="MultiMedia5 Mixer SLIM_8_TX" value="0" />
+ <ctl name="MultiMedia5 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia1 Mixer SLIM_0_TX" value="0" />
+ <ctl name="MultiMedia1 Mixer SLIM_4_TX" value="0" />
+ <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_4_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
+ <ctl name="HDMI Mixer MultiMedia1" value="0" />
+ <ctl name="HDMI Mixer MultiMedia2" value="0" />
+ <ctl name="HDMI Mixer MultiMedia3" value="0" />
+ <ctl name="HDMI Mixer MultiMedia4" value="0" />
+ <ctl name="HDMI Mixer MultiMedia5" value="0" />
+ <ctl name="HDMI Mixer MultiMedia7" value="0" />
+ <ctl name="HDMI Mixer MultiMedia9" value="0" />
+ <ctl name="HDMI Mixer MultiMedia10" value="0" />
+ <ctl name="HDMI Mixer MultiMedia11" value="0" />
+ <ctl name="HDMI Mixer MultiMedia12" value="0" />
+ <ctl name="HDMI Mixer MultiMedia13" value="0" />
+ <ctl name="HDMI Mixer MultiMedia14" value="0" />
+ <ctl name="HDMI Mixer MultiMedia15" value="0" />
+ <ctl name="HDMI Mixer MultiMedia16" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia6" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia8" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia2" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia2" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia3" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="USB_AUDIO_RX Channels" value="One" />
+ <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
+ <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
+ <ctl name="USB_AUDIO_TX Channels" value="One" />
+ <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
+ <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
+ <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
+ <ctl name="SLIM_2_RX Format" value="UNPACKED" />
+ <ctl name="SLIM_2_RX SampleRate" value="KHZ_48" />
+ <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
+ <ctl name="SLIM_0_RX Channels" value="One" />
+ <ctl name="SLIM_5_RX Channels" value="One" />
+ <ctl name="SLIM_6_RX Channels" value="One" />
+ <ctl name="SLIM_2_RX Channels" value="One" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="SLIM_1_TX Channels" value="One" />
+ <ctl name="AIF1_CAP Mixer SLIM TX9" value="0"/>
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="0"/>
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="0" />
+ <ctl name="AIF1_CAP Mixer SLIM TX6" value="0" />
+ <ctl name="AIF1_CAP Mixer SLIM TX5" value="0"/>
+ <ctl name="AIF1_CAP Mixer SLIM TX4" value="0" />
+ <ctl name="AIF1_CAP Mixer SLIM TX3" value="0"/>
+ <ctl name="AIF1_CAP Mixer SLIM TX2" value="0" />
+ <ctl name="AIF1_CAP Mixer SLIM TX1" value="0"/>
+ <ctl name="AIF1_CAP Mixer SLIM TX0" value="0"/>
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="HDMI Mixer MultiMedia4" value="0" />
+ <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="ZERO" />
+ <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="ZERO" />
+ <ctl name="VI_FEED_TX Channels" value="Two" />
+ <ctl name="AIF4_VI Mixer SPKR_VI_1" value="0" />
+ <ctl name="AIF4_VI Mixer SPKR_VI_2" value="0" />
+ <ctl name="SLIM_4_TX Format" value="UNPACKED" />
+ <ctl name="AANC_SLIM_0_RX MUX" value="ZERO" />
+ <!-- HFP start -->
+ <ctl name="HFP_PRI_AUX_UL_HL Switch" value="0" />
+ <ctl name="SLIMBUS_0_RX Port Mixer SLIM_7_TX" value="0" />
+ <!-- HFP end -->
+ <!-- echo reference -->
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="None" />
+ <!-- usb headset -->
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="MultiMedia1 Mixer AFE_PCM_TX" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
+ <!-- usb headset end -->
+ <!-- fm -->
+ <ctl name="SLIMBUS_8 LOOPBACK Volume" value="0" />
+ <ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="0" />
+ <ctl name="SLIMBUS_DL_HL Switch" value="0" />
+ <ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="0" />
+ <ctl name="SLIMBUS6_DL_HL Switch" value="0" />
+ <ctl name="MultiMedia1 Mixer SLIM_8_TX" value="0" />
+ <ctl name="MultiMedia2 Mixer SLIM_8_TX" value="0" />
+ <!-- fm end -->
+
+ <!-- Multimode Voice1 -->
+ <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="0" />
+ <!-- Multimode Voice1 HDMI -->
+ <ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="0" />
+ <!-- Multimode Voice1 BTSCO -->
+ <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="0" />
+ <!-- Multimode Voice1 USB headset -->
+ <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer AFE_PCM_TX_MMode1" value="0" />
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="0" />
+ <!-- Miltimode Voice1 end-->
+
+ <!-- Multimode Voice2 -->
+ <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="0" />
+ <!-- Multimode Voice2 HDMI -->
+ <ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="0" />
+ <!-- Multimode Voice2 BTSCO -->
+ <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="0" />
+ <!-- Multimode Voice2 USB headset -->
+ <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="VoiceMMode2_Tx Mixer AFE_PCM_TX_MMode2" value="0" />
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="0" />
+ <!-- Multimode Voice2 end-->
+
+ <!-- Voice external ec. reference -->
+ <ctl name="VOC_EXT_EC MUX" value="NONE" />
+ <ctl name="AIF3_CAP Mixer SLIM TX1" value="0" />
+ <ctl name="AIF3_CAP Mixer SLIM TX2" value="0" />
+ <!-- Voice external ec. reference end -->
+
+ <!-- RT Proxy Cal -->
+ <ctl name="RT_PROXY_1_RX SetCalMode" value="CAL_MODE_NONE" />
+ <ctl name="RT_PROXY_1_TX SetCalMode" value="CAL_MODE_NONE" />
+ <!-- RT Proxy Cal end -->
+
+ <!-- Incall Recording -->
+ <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
+ <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+ <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="0" />
+ <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="0" />
+ <!-- Incall Recording End -->
+
+ <!-- Incall Music -->
+ <ctl name="Incall_Music Audio Mixer MultiMedia2" value="0" />
+ <!-- Incall Music End -->
+
+ <!-- compress-voip-call start -->
+ <ctl name="SLIM_0_RX_Voice Mixer Voip" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer Voip" value="0" />
+ <ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="0" />
+ <ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
+ <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
+ <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
+ <ctl name="Voip_Tx Mixer AFE_PCM_TX_Voip" value="0" />
+ <ctl name="USB_AUDIO_RX_Voice Mixer Voip" value="0" />
+ <ctl name="Voip_Tx Mixer USB_AUDIO_TX_Voip" value="0" />
+ <!-- compress-voip-call end-->
+
+ <!-- Audio BTSCO -->
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia6" value="0" />
+ <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+ <!-- audio record compress-->
+ <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer AFE_PCM_TX" value="0" />
+ <!-- audio record compress end-->
+
+ <!-- split a2dp -->
+ <ctl name="BT SampleRate" value="KHZ_8" />
+ <ctl name="AFE Input Channels" value="Zero" />
+ <ctl name="SLIM7_RX ADM Channels" value="Zero" />
+ <!-- split a2dp end-->
+
+ <!-- ADSP testfwk -->
+ <ctl name="SLIMBUS_DL_HL Switch" value="0" />
+ <ctl name="SLIMBUS6_DL_HL Switch" value="0" />
+ <!-- ADSP testfwk end-->
+
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
+
+ <!-- Codec controls -->
+ <!-- WSA controls -->
+ <ctl name="SpkrLeft COMP Switch" value="0" />
+ <ctl name="SpkrRight COMP Switch" value="0" />
+ <ctl name="SpkrLeft BOOST Switch" value="0" />
+ <ctl name="SpkrRight BOOST Switch" value="0" />
+ <ctl name="SpkrLeft VISENSE Switch" value="0" />
+ <ctl name="SpkrRight VISENSE Switch" value="0" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="0" />
+ <ctl name="SpkrRight SWR DAC_Port Switch" value="0" />
+ <ctl name="SpkrLeft WSA PA Gain" value="G_0_DB" />
+
+ <!-- Volume controls -->
+ <ctl name="LINEOUT1 Volume" value="13" />
+ <ctl name="LINEOUT2 Volume" value="13" />
+ <ctl name="HPHL Volume" value="20" />
+ <ctl name="HPHR Volume" value="20" />
+ <ctl name="EAR PA Gain" value="G_6_DB" />
+ <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
+
+ <ctl name="RX0 Digital Volume" value="84" />
+ <ctl name="RX1 Digital Volume" value="84" />
+ <ctl name="RX2 Digital Volume" value="84" />
+ <ctl name="RX3 Digital Volume" value="84" />
+ <ctl name="RX4 Digital Volume" value="84" />
+ <ctl name="RX5 Digital Volume" value="84" />
+ <ctl name="RX6 Digital Volume" value="84" />
+ <ctl name="RX7 Digital Volume" value="84" />
+ <ctl name="ADC1 Volume" value="12" />
+ <ctl name="ADC2 Volume" value="12" />
+ <ctl name="ADC3 Volume" value="12" />
+ <ctl name="ADC4 Volume" value="12" />
+ <ctl name="DEC0 Volume" value="84" />
+ <ctl name="DEC1 Volume" value="84" />
+ <ctl name="DEC2 Volume" value="84" />
+ <ctl name="DEC3 Volume" value="84" />
+ <ctl name="DEC4 Volume" value="84" />
+ <ctl name="DEC5 Volume" value="84" />
+ <ctl name="DEC6 Volume" value="84" />
+ <ctl name="DEC7 Volume" value="84" />
+ <ctl name="DEC8 Volume" value="84" />
+
+ <!-- Compander controls -->
+ <ctl name="COMP1 Switch" value="1" />
+ <ctl name="COMP2 Switch" value="1" />
+ <ctl name="COMP7 Switch" value="0" />
+ <ctl name="COMP8 Switch" value="0" />
+
+ <!-- ADC, DMIC and AMIC controls -->
+ <ctl name="ADC MUX0" value="AMIC" />
+ <ctl name="ADC MUX1" value="AMIC" />
+ <ctl name="ADC MUX2" value="AMIC" />
+ <ctl name="ADC MUX3" value="AMIC" />
+ <ctl name="ADC MUX4" value="AMIC" />
+ <ctl name="ADC MUX5" value="AMIC" />
+ <ctl name="ADC MUX6" value="AMIC" />
+ <ctl name="ADC MUX7" value="AMIC" />
+ <ctl name="ADC MUX8" value="AMIC" />
+ <ctl name="ADC MUX10" value="AMIC" />
+ <ctl name="ADC MUX11" value="AMIC" />
+ <ctl name="ADC MUX12" value="AMIC" />
+ <ctl name="ADC MUX13" value="AMIC" />
+ <ctl name="DMIC MUX0" value="ZERO" />
+ <ctl name="DMIC MUX1" value="ZERO" />
+ <ctl name="DMIC MUX2" value="ZERO" />
+ <ctl name="DMIC MUX3" value="ZERO" />
+ <ctl name="DMIC MUX4" value="ZERO" />
+ <ctl name="DMIC MUX5" value="ZERO" />
+ <ctl name="DMIC MUX6" value="ZERO" />
+ <ctl name="DMIC MUX7" value="ZERO" />
+ <ctl name="DMIC MUX8" value="ZERO" />
+ <ctl name="DMIC MUX10" value="ZERO" />
+ <ctl name="DMIC MUX11" value="ZERO" />
+ <ctl name="DMIC MUX12" value="ZERO" />
+ <ctl name="DMIC MUX13" value="ZERO" />
+ <ctl name="AMIC MUX0" value="ZERO" />
+ <ctl name="AMIC MUX1" value="ZERO" />
+ <ctl name="AMIC MUX2" value="ZERO" />
+ <ctl name="AMIC MUX3" value="ZERO" />
+ <ctl name="AMIC MUX4" value="ZERO" />
+ <ctl name="AMIC MUX5" value="ZERO" />
+ <ctl name="AMIC MUX6" value="ZERO" />
+ <ctl name="AMIC MUX7" value="ZERO" />
+ <ctl name="AMIC MUX8" value="ZERO" />
+ <ctl name="AMIC MUX10" value="ZERO" />
+ <ctl name="AMIC MUX11" value="ZERO" />
+ <ctl name="AMIC MUX12" value="ZERO" />
+ <ctl name="AMIC MUX13" value="ZERO" />
+
+ <!-- CDC_IF and SLIM controls -->
+ <ctl name="SLIM RX0 MUX" value="ZERO" />
+ <ctl name="SLIM RX1 MUX" value="ZERO" />
+ <ctl name="SLIM RX2 MUX" value="ZERO" />
+ <ctl name="SLIM RX3 MUX" value="ZERO" />
+ <ctl name="SLIM RX4 MUX" value="ZERO" />
+ <ctl name="SLIM RX5 MUX" value="ZERO" />
+ <ctl name="SLIM RX6 MUX" value="ZERO" />
+ <ctl name="SLIM RX7 MUX" value="ZERO" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="CDC_IF RX1 MUX" value="SLIM RX1" />
+ <ctl name="CDC_IF RX2 MUX" value="SLIM RX2" />
+ <ctl name="CDC_IF RX3 MUX" value="SLIM RX3" />
+ <ctl name="CDC_IF RX4 MUX" value="SLIM RX4" />
+ <ctl name="CDC_IF RX5 MUX" value="SLIM RX5" />
+ <ctl name="CDC_IF RX6 MUX" value="SLIM RX6" />
+ <ctl name="CDC_IF RX7 MUX" value="SLIM RX7" />
+ <ctl name="CDC_IF TX0 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX1 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX2 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX3 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX4 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX5 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX6 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX7 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX8 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX9 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX10 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX13 MUX" value="ZERO" />
+
+ <!-- Interpolator chain controls -->
+ <ctl name="RX INT0_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT0_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT0_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT2_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT2_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT2_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT7_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT7_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT8_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT8_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT8_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT0_2 MUX" value="ZERO" />
+ <ctl name="RX INT1_2 MUX" value="ZERO" />
+ <ctl name="RX INT2_2 MUX" value="ZERO" />
+ <ctl name="RX INT7_2 MUX" value="ZERO" />
+ <ctl name="RX INT8_2 MUX" value="ZERO" />
+ <ctl name="RX INT1_1 NATIVE MUX" value="OFF" />
+ <ctl name="RX INT2_1 NATIVE MUX" value="OFF" />
+ <ctl name="RX INT1_2 NATIVE MUX" value="OFF" />
+ <ctl name="RX INT2_2 NATIVE MUX" value="OFF" />
+ <ctl name="ASRC0 MUX" value="ZERO" />
+ <ctl name="ASRC1 MUX" value="ZERO" />
+ <ctl name="RX INT1 SEC MIX HPHL Switch" value="0" />
+ <ctl name="RX INT2 SEC MIX HPHR Switch" value="0" />
+ <ctl name="DSD_L IF MUX" value="ZERO" />
+ <ctl name="DSD_R IF MUX" value="ZERO" />
+ <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="0" />
+ <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="0" />
+ <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
+
+ <!-- Headphone Default mode - uLP -->
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
+
+ <!-- IIR/voice anc -->
+ <ctl name="IIR0 Band1" id ="0" value="268435456" />
+ <ctl name="IIR0 Band1" id ="1" value="0" />
+ <ctl name="IIR0 Band1" id ="2" value="0" />
+ <ctl name="IIR0 Band1" id ="3" value="0" />
+ <ctl name="IIR0 Band1" id ="4" value="0" />
+ <ctl name="IIR0 Band2" id ="0" value="268435456" />
+ <ctl name="IIR0 Band2" id ="1" value="0" />
+ <ctl name="IIR0 Band2" id ="2" value="0" />
+ <ctl name="IIR0 Band2" id ="3" value="0" />
+ <ctl name="IIR0 Band2" id ="4" value="0" />
+ <ctl name="IIR0 Band3" id ="0" value="268435456" />
+ <ctl name="IIR0 Band3" id ="1" value="0" />
+ <ctl name="IIR0 Band3" id ="2" value="0" />
+ <ctl name="IIR0 Band3" id ="3" value="0" />
+ <ctl name="IIR0 Band3" id ="4" value="0" />
+ <ctl name="IIR0 Band4" id ="0" value="268435456" />
+ <ctl name="IIR0 Band4" id ="1" value="0" />
+ <ctl name="IIR0 Band4" id ="2" value="0" />
+ <ctl name="IIR0 Band4" id ="3" value="0" />
+ <ctl name="IIR0 Band4" id ="4" value="0" />
+ <ctl name="IIR0 Band5" id ="0" value="268435456" />
+ <ctl name="IIR0 Band5" id ="1" value="0" />
+ <ctl name="IIR0 Band5" id ="2" value="0" />
+ <ctl name="IIR0 Band5" id ="3" value="0" />
+ <ctl name="IIR0 Band5" id ="4" value="0" />
+ <ctl name="IIR0 Enable Band1" value="0" />
+ <ctl name="IIR0 Enable Band2" value="0" />
+ <ctl name="IIR0 Enable Band3" value="0" />
+ <ctl name="IIR0 Enable Band4" value="0" />
+ <ctl name="IIR0 Enable Band5" value="0" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="IIR0 INP0 MUX" value="ZERO" />
+ <ctl name="IIR0 INP1 MUX" value="ZERO" />
+ <ctl name="IIR0 INP2 MUX" value="ZERO" />
+ <ctl name="IIR1 INP0 MUX" value="ZERO" />
+ <ctl name="IIR1 INP1 MUX" value="ZERO" />
+ <ctl name="IIR1 INP2 MUX" value="ZERO" />
+
+ <!-- anc related -->
+ <ctl name="ANC Slot" value="0" />
+ <ctl name="ANC Function" value="OFF" />
+ <ctl name="ANC0 FB MUX" value="ZERO" />
+ <ctl name="ANC1 FB MUX" value="ZERO" />
+ <ctl name="ANC OUT EAR Enable Switch" value="0" />
+ <ctl name="ANC OUT EAR SPKR Enable Switch" value="0" />
+ <ctl name="ANC SPKR PA Enable Switch" value="0" />
+
+ <!-- vbat related data -->
+ <!-- vbat related data end -->
+
+ <!-- Codec controls end -->
+
+ <!-- These are audio route (FE to BE) specific mixer settings -->
+ <path name="gsm-mode">
+ <ctl name="GSM mode Enable" value="ON" />
+ </path>
+
+ <path name="echo-reference speaker-vbat-mono">
+ </path>
+
+ <path name="echo-reference speaker-vbat">
+ </path>
+
+ <path name="echo-reference">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
+ </path>
+
+ <path name="echo-reference headphones">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX" />
+ </path>
+
+ <path name="echo-reference headphones-44.1">
+ </path>
+
+ <path name="deep-buffer-playback">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-protected">
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback hdmi">
+ <ctl name="HDMI Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-hdmi">
+ <path name="deep-buffer-playback hdmi" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-display-port">
+ <path name="deep-buffer-playback display-port" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="deep-buffer-playback bt-sco" />
+ </path>
+
+ <path name="deep-buffer-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-usb-headphones">
+ <path name="deep-buffer-playback usb-headphones" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-headphones">
+ <path name="deep-buffer-playback headphones" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="low-latency-playback">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback speaker-protected">
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback hdmi">
+ <ctl name="HDMI Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="low-latency-playback bt-sco" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-hdmi">
+ <path name="low-latency-playback hdmi" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-display-port">
+ <path name="low-latency-playback display-port" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-usb-headphones">
+ <path name="low-latency-playback usb-headphones" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-headphones">
+ <path name="low-latency-playback headphones" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="audio-ull-playback">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback speaker-protected">
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="audio-ull-playback headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-headphones">
+ <path name="audio-ull-playback" />
+ <path name="audio-ull-playback headphones" />
+ </path>
+
+ <path name="audio-ull-playback hdmi">
+ <ctl name="HDMI Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="1" />
+ </path>
+
+ <path name="audio-ull-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-ull-playback bt-sco" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-hdmi">
+ <path name="audio-ull-playback hdmi" />
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-display-port">
+ <path name="audio-ull-playback display-port" />
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="audio-ull-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="multi-channel-playback hdmi">
+ <ctl name="HDMI Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="multi-channel-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="multi-channel-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="compress-offload-playback">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback speaker-protected">
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback hdmi">
+ <ctl name="HDMI Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="silence-playback hdmi">
+ <ctl name="HDMI Mixer MultiMedia9" value="1" />
+ </path>
+
+ <path name="compress-offload-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="silence-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="1" />
+ </path>
+
+ <path name="compress-offload-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-hdmi">
+ <path name="compress-offload-playback hdmi" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-display-port">
+ <path name="compress-offload-playback display-port" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-usb-headphones">
+ <path name="compress-offload-playback usb-headphones" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-headphones">
+ <path name="compress-offload-playback headphones" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback2">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 hdmi">
+ <ctl name="HDMI Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback2 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-hdmi">
+ <path name="compress-offload-playback2 hdmi" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-display-port">
+ <path name="compress-offload-playback2 display-port" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback2 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-usb-headphones">
+ <path name="compress-offload-playback2 usb-headphones" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback2 headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-headphones">
+ <path name="compress-offload-playback2 headphones" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback3">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 hdmi">
+ <ctl name="HDMI Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback3 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-hdmi">
+ <path name="compress-offload-playback3 hdmi" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-display-port">
+ <path name="compress-offload-playback3 display-port" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback3 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-usb-headphones">
+ <path name="compress-offload-playback3 usb-headphones" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback3 headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-headphones">
+ <path name="compress-offload-playback3 headphones" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback4">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 hdmi">
+ <ctl name="HDMI Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback4 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-hdmi">
+ <path name="compress-offload-playback4 hdmi" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-display-port">
+ <path name="compress-offload-playback4 display-port" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+
+ <path name="compress-offload-playback4 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-usb-headphones">
+ <path name="compress-offload-playback4 usb-headphones" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback4 headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-headphones">
+ <path name="compress-offload-playback4 headphones" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback5">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 hdmi">
+ <ctl name="HDMI Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback5 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-hdmi">
+ <path name="compress-offload-playback5 hdmi" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-display-port">
+ <path name="compress-offload-playback5 display-port" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback5 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-usb-headphones">
+ <path name="compress-offload-playback5 usb-headphones" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback5 headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-headphones">
+ <path name="compress-offload-playback5 headphones" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback6">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 hdmi">
+ <ctl name="HDMI Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback6 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-hdmi">
+ <path name="compress-offload-playback6 hdmi" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-display-port">
+ <path name="compress-offload-playback6 display-port" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback6 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-usb-headphones">
+ <path name="compress-offload-playback6 usb-headphones" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback6 headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-headphones">
+ <path name="compress-offload-playback6 headphones" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback7">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 hdmi">
+ <ctl name="HDMI Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback7 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-hdmi">
+ <path name="compress-offload-playback7 hdmi" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-display-port">
+ <path name="compress-offload-playback7 display-port" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback7 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-usb-headphones">
+ <path name="compress-offload-playback7 usb-headphones" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback7 headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-headphones">
+ <path name="compress-offload-playback7 headphones" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback8">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 hdmi">
+ <ctl name="HDMI Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback8 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-hdmi">
+ <path name="compress-offload-playback8 hdmi" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-display-port">
+ <path name="compress-offload-playback8 display-port" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback8 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-usb-headphones">
+ <path name="compress-offload-playback8 usb-headphones" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback8 headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-headphones">
+ <path name="compress-offload-playback8 headphones" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback9">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 hdmi">
+ <ctl name="HDMI Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback9 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-hdmi">
+ <path name="compress-offload-playback9 hdmi" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-display-port">
+ <path name="compress-offload-playback9 display-port" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="compress-offload-playback9 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-usb-headphones">
+ <path name="compress-offload-playback9 usb-headphones" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="compress-offload-playback9 headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 headphones-44.1">
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-headphones">
+ <path name="compress-offload-playback9 headphones" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="audio-record">
+ <ctl name="MultiMedia1 Mixer SLIM_0_TX" value="1" />
+ </path>
+
+ <path name="audio-record usb-headset-mic">
+ <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="audio-record bt-sco">
+ <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="audio-record bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-record bt-sco" />
+ </path>
+
+ <path name="audio-record capture-fm">
+ <ctl name="MultiMedia1 Mixer SLIM_8_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress">
+ <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress bt-sco">
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-record-compress bt-sco" />
+ </path>
+
+ <path name="audio-record-compress usb-headset-mic">
+ <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="low-latency-record">
+ <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
+ </path>
+
+ <path name="low-latency-record bt-sco">
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="low-latency-record bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="low-latency-record bt-sco" />
+ </path>
+
+ <path name="low-latency-record usb-headset-mic">
+ <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="low-latency-record capture-fm">
+ <ctl name="MultiMedia8 Mixer SLIM_8_TX" value="1" />
+ </path>
+
+ <path name="fm-virtual-record capture-fm">
+ <ctl name="MultiMedia2 Mixer SLIM_8_TX" value="1" />
+ </path>
+
+ <path name="play-fm">
+ <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+ <ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="1" />
+ <ctl name="SLIMBUS_DL_HL Switch" value="1" />
+ </path>
+
+ <path name="play-fm headphones">
+ <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+ <ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="1" />
+ <ctl name="SLIMBUS6_DL_HL Switch" value="1" />
+ </path>
+
+ <path name="incall-rec-uplink">
+ <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+ </path>
+
+ <path name="incall-rec-uplink bt-sco">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink bt-sco-wb">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink usb-headset-mic">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink afe-proxy">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink-compress">
+ <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="1" />
+ </path>
+
+ <path name="incall-rec-uplink-compress bt-sco">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-compress bt-sco-wb">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-compress usb-headset-mic">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-compress afe-proxy">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink">
+ <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
+ </path>
+
+ <path name="incall-rec-downlink bt-sco">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink bt-sco-wb">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink usb-headset-mic">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink afe-proxy">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink-compress">
+ <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="1" />
+ </path>
+
+ <path name="incall-rec-downlink-compress bt-sco">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink-compress bt-sco-wb">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink-compress usb-headset-mic">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink-compress afe-proxy">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink">
+ <path name="incall-rec-uplink" />
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink bt-sco">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink bt-sco-wb">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink usb-headset-mic">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink afe-proxy">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress">
+ <path name="incall-rec-uplink-compress" />
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress bt-sco">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress bt-sco-wb">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress usb-headset-mic">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress afe-proxy">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="hfp-sco">
+ </path>
+
+ <path name="hfp-sco headphones">
+ </path>
+
+ <path name="hfp-sco-wb">
+ <path name="hfp-sco" />
+ </path>
+
+ <path name="hfp-sco-wb headphones">
+ <path name="hfp-sco headphones" />
+ </path>
+
+ <path name="compress-voip-call">
+ <ctl name="SLIM_0_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="1" />
+ </path>
+
+
+ <path name="compress-voip-call bt-sco">
+ <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-voip-call bt-sco" />
+ </path>
+
+ <path name="compress-voip-call afe-proxy">
+ <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer AFE_PCM_TX_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call usb-headphones">
+ <ctl name="USB_AUDIO_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer USB_AUDIO_TX_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call voice-speaker-vbat">
+ <path name="echo-reference speaker-vbat-mono" />
+ <path name="compress-voip-call"/>
+ </path>
+
+ <path name="voicemmode1-call">
+ <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call hdmi">
+ <ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call bt-sco">
+ <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="voicemmode1-call bt-sco" />
+ </path>
+
+ <path name="voicemmode1-call afe-proxy">
+ <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer AFE_PCM_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call usb-headphones">
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call voice-speaker-vbat">
+ <path name="echo-reference speaker-vbat-mono" />
+ <path name="voicemmode1-call"/>
+ </path>
+
+ <path name="voicemmode2-call">
+ <ctl name="SLIM_0_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call hdmi">
+ <ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call bt-sco">
+ <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="voicemmode2-call bt-sco" />
+ </path>
+
+ <path name="voicemmode2-call afe-proxy">
+ <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer AFE_PCM_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call usb-headphones">
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call voice-speaker-vbat">
+ <path name="echo-reference speaker-vbat-mono" />
+ <path name="voicemmode2-call"/>
+ </path>
+
+ <path name="spkr-rx-calib">
+ <ctl name="SLIMBUS_DL_HL Switch" value="1" />
+ </path>
+
+ <path name="spkr-vi-record">
+ </path>
+
+ <!-- These are actual sound device specific mixer settings -->
+ <path name="amic1">
+ <ctl name="AIF1_CAP Mixer SLIM TX6" value="1"/>
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+ <ctl name="ADC MUX6" value="AMIC" />
+ <ctl name="AMIC MUX6" value="ADC1" />
+ <ctl name="IIR0 INP0 MUX" value="DEC6" />
+ </path>
+
+ <path name="amic2">
+ <ctl name="AIF1_CAP Mixer SLIM TX0" value="1"/>
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+ <ctl name="ADC MUX0" value="AMIC" />
+ <ctl name="AMIC MUX0" value="ADC2" />
+ <ctl name="IIR0 INP0 MUX" value="DEC0" />
+ </path>
+
+ <!-- For Tavil, DMIC numbered from 0 to 5 -->
+ <path name="dmic1">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC0" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="dmic2">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="dmic3">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="dmic4">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC3" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="dmic5">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC4" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="dmic6">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC5" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="speaker">
+ <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="SLIM_0_RX Channels" value="One" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+ <ctl name="COMP7 Switch" value="1" />
+ <ctl name="SpkrLeft COMP Switch" value="1" />
+ <ctl name="SpkrLeft BOOST Switch" value="1" />
+ <ctl name="SpkrLeft VISENSE Switch" value="1" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ </path>
+
+ <path name="speaker-mono">
+ <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="SLIM_0_RX Channels" value="One" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+ <ctl name="COMP7 Switch" value="1" />
+ <ctl name="SpkrLeft COMP Switch" value="1" />
+ <ctl name="SpkrLeft BOOST Switch" value="1" />
+ <ctl name="SpkrLeft VISENSE Switch" value="1" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ </path>
+
+ <path name="speaker-fluid">
+ <path name="speaker-mono" />
+ </path>
+
+ <path name="speaker-liquid">
+ <path name="speaker" />
+ </path>
+
+ <path name="speaker-vbat-mono">
+ <path name="speaker-mono" />
+ </path>
+
+ <path name="speaker-vbat">
+ <path name="speaker" />
+ </path>
+
+ <path name="sidetone-iir">
+ <ctl name="IIR0 Enable Band1" value="1" />
+ <ctl name="IIR0 Enable Band2" value="1" />
+ <ctl name="IIR0 Enable Band3" value="1" />
+ <ctl name="IIR0 Enable Band4" value="1" />
+ <ctl name="IIR0 Enable Band5" value="1" />
+ </path>
+
+ <path name="sidetone-headphones">
+ <path name="sidetone-iir" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="RX INT1 MIX2 INP" value="SRC0" />
+ <ctl name="RX INT2 MIX2 INP" value="SRC0" />
+ </path>
+
+ <path name="sidetone-handset">
+ <path name="sidetone-iir" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="RX INT7 MIX2 INP" value="SRC0" />
+ </path>
+
+ <path name="speaker-mic">
+ <path name="dmic3" />
+ </path>
+
+ <path name="speaker-mic-liquid">
+ <path name="dmic3" />
+ </path>
+
+ <path name="speaker-mic-sbc">
+ </path>
+
+ <path name="speaker-protected">
+ <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+ <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
+ <path name="speaker" />
+ <ctl name="VI_FEED_TX Channels" value="One" />
+ <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
+ </path>
+
+ <path name="voice-speaker-protected">
+ <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+ <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
+ <path name="speaker-mono" />
+ <ctl name="VI_FEED_TX Channels" value="One" />
+ <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
+ </path>
+
+ <path name="vi-feedback">
+ </path>
+
+ <path name="speaker-protected-vbat">
+ <path name="speaker-protected" />
+ </path>
+
+ <path name="voice-speaker-protected-vbat">
+ <path name="voice-speaker-protected" />
+ </path>
+
+ <path name="handset">
+ <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="SLIM_0_RX Channels" value="One" />
+ <ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
+ <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="EAR PA Gain" value="G_6_DB" />
+ </path>
+
+ <path name="handset-mic">
+ <path name="dmic1" />
+ </path>
+
+ <path name="handset-mic-db">
+ </path>
+
+ <path name="handset-mic-cdp">
+ <path name="amic1" />
+ </path>
+
+ <path name="handset-mic-sbc">
+ </path>
+
+ <path name="three-mic">
+ <ctl name="AIF1_CAP Mixer SLIM TX0" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX1" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX2" value="1" />
+ <ctl name="SLIM_0_TX Channels" value="Three" />
+ <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+ <ctl name="ADC MUX0" value="DMIC" />
+ <ctl name="DMIC MUX" value="DMIC0" />
+ <ctl name="CDC_IF TX1 MUX" value="DEC1" />
+ <ctl name="ADC MUX1" value="DMIC" />
+ <ctl name="DMIC MUX1" value="DMIC1" />
+ <ctl name="CDC_IF TX2 MUX" value="DEC2" />
+ <ctl name="ADC MUX2" value="DMIC" />
+ <ctl name="DMIC MUX2" value="DMIC2" />
+ </path>
+
+ <path name="anc-handset">
+ <ctl name="ANC Function" value="ON" />
+ <ctl name="ANC Slot" value="6" />
+ <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="SLIM_0_RX Channels" value="One" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ <ctl name="ANC OUT EAR SPKR Enable Switch" value="1" />
+ <ctl name="ANC SPKR PA Enable Switch" value="1" />
+ <ctl name="SpkrLeft WSA PA Gain" value="G_6_DB" />
+ </path>
+
+ <path name="headphones">
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="Two" />
+ <ctl name="RX INT1_2 MUX" value="RX2" />
+ <ctl name="RX INT2_2 MUX" value="RX3" />
+ </path>
+
+ <path name="headphones-44.1">
+ <ctl name="SLIM RX4 MUX" value="AIF3_PB" />
+ <ctl name="SLIM RX5 MUX" value="AIF3_PB" />
+ <ctl name="SLIM_5_RX Channels" value="Two" />
+ <ctl name="CDC_IF RX4 MUX" value="SLIM RX4" />
+ <ctl name="CDC_IF RX5 MUX" value="SLIM RX5" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX4" />
+ <ctl name="RX INT2_1 MIX1 INP0" value="RX5" />
+ <ctl name="RX INT1_1 NATIVE MUX" value="ON" />
+ <ctl name="RX INT2_1 NATIVE MUX" value="ON" />
+ <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
+ <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
+ </path>
+
+ <path name="asrc-mode">
+ <ctl name="RX INT1_2 NATIVE MUX" value="ON" />
+ <ctl name="RX INT2_2 NATIVE MUX" value="ON" />
+ <ctl name="ASRC0 MUX" value="ASRC_IN_HPHL" />
+ <ctl name="RX INT1 SEC MIX HPHL Switch" value="1" />
+ <ctl name="ASRC1 MUX" value="ASRC_IN_HPHR" />
+ <ctl name="RX INT2 SEC MIX HPHR Switch" value="1" />
+ </path>
+
+ <path name="headphones-dsd">
+ <ctl name="SLIM RX6 MUX" value="AIF2_PB" />
+ <ctl name="SLIM RX7 MUX" value="AIF2_PB" />
+ <ctl name="SLIM_2_RX Channels" value="Two" />
+ <ctl name="DSD_L IF MUX" value="RX6" />
+ <ctl name="DSD_R IF MUX" value="RX7" />
+ <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="1" />
+ <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="1" />
+ <ctl name="SLIM_2_RX Format" value="DSD_DOP" />
+ </path>
+
+ <path name="hph-highquality-mode">
+ <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
+ </path>
+
+ <path name="hph-lowpower-mode">
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
+ </path>
+
+ <path name="true-native-mode">
+ <ctl name="SLIM RX2 MUX" value="AIF3_PB" />
+ <ctl name="SLIM RX3 MUX" value="AIF3_PB" />
+ <ctl name="CDC_IF RX2 MUX" value="SLIM RX2" />
+ <ctl name="CDC_IF RX3 MUX" value="SLIM RX3" />
+ <ctl name="RX INT1 NATIVE MUX" value="ON" />
+ <ctl name="RX INT2 NATIVE MUX" value="ON" />
+ <ctl name="SLIM_5_RX Channels" value="Two" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX2" />
+ <ctl name="RX INT2_1 MIX1 INP1" value="RX3" />
+ <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
+ <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="COMP1" value="1" />
+ <ctl name="COMP2" value="1" />
+ </path>
+
+ <path name="line">
+ <path name="headphones" />
+ </path>
+
+ <path name="headset-mic">
+ <path name="amic2" />
+ </path>
+
+ <path name="headset-mic-liquid">
+ <path name="amic2" />
+ </path>
+
+ <path name="voice-handset">
+ <path name="handset" />
+ </path>
+
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
+ <path name="voice-speaker">
+ <path name="speaker-mono" />
+ </path>
+
+ <path name="voice-speaker-fluid">
+ <path name="speaker-fluid" />
+ </path>
+
+ <path name="voice-speaker-mic">
+ <path name="speaker-mic" />
+ </path>
+
+ <path name="voice-speaker-vbat">
+ <path name="speaker-vbat-mono" />
+ </path>
+
+ <path name="voice-headphones">
+ <path name="headphones" />
+ </path>
+
+ <path name="voice-line">
+ <path name="voice-headphones" />
+ </path>
+
+ <path name="voice-headset-mic">
+ <path name="headset-mic" />
+ </path>
+
+ <path name="speaker-and-headphones">
+ <path name="headphones" />
+ <path name="speaker" />
+ </path>
+
+ <path name="speaker-and-line">
+ <path name="speaker-and-headphones" />
+ </path>
+
+ <path name="speaker-and-headphones-liquid">
+ <path name="headphones" />
+ <path name="speaker" />
+ </path>
+
+ <path name="speaker-and-line-liquid">
+ <path name="speaker-and-headphones-liquid" />
+ </path>
+
+ <path name="usb-headphones">
+ </path>
+
+ <path name="afe-proxy">
+ </path>
+
+ <path name="anc-headphones">
+ <path name="headphones" />
+ </path>
+
+ <path name="speaker-and-anc-headphones">
+ <path name="anc-headphones" />
+ <path name="speaker" />
+ </path>
+
+ <path name="anc-fb-headphones">
+ <path name="anc-headphones" />
+ </path>
+
+ <path name="speaker-and-anc-fb-headphones">
+ <path name="anc-fb-headphones" />
+ <path name="speaker" />
+ </path>
+
+ <path name="voice-anc-headphones">
+ <path name="voice-headphones" />
+ </path>
+
+ <path name="voice-anc-fb-headphones">
+ <path name="voice-headphones" />
+ </path>
+
+ <path name="speaker-and-anc-headphones-liquid">
+ <path name="anc-headphones" />
+ </path>
+
+ <path name="hdmi">
+ </path>
+
+ <path name="display-port">
+ </path>
+
+ <path name="speaker-and-usb-headphones">
+ <path name="speaker" />
+ <path name="usb-headphones" />
+ </path>
+
+ <path name="speaker-and-hdmi">
+ <path name="speaker" />
+ <path name="hdmi" />
+ </path>
+
+ <path name="speaker-and-display-port">
+ <path name="speaker" />
+ <path name="display-port" />
+ </path>
+
+ <path name="voice-rec-mic">
+ <path name="handset-mic" />
+ </path>
+
+ <path name="camcorder-mic">
+ <path name="handset-mic" />
+ </path>
+
+ <path name="hdmi-tx">
+ <path name="handset-mic" />
+ </path>
+
+ <path name="bt-sco-headset">
+ </path>
+
+ <path name="bt-sco-mic">
+ </path>
+
+ <path name="bt-sco-headset-wb">
+ </path>
+
+ <path name="bt-sco-mic-wb">
+ </path>
+
+ <path name="usb-headset-mic">
+ </path>
+
+ <path name="capture-fm">
+ </path>
+
+ <path name="aanc-handset-mic">
+ <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX9" value="1" />
+ <ctl name="SLIM_0_TX Channels" value="Three" />
+ <ctl name="AANC_SLIM_0_RX MUX" value="SLIMBUS_0_TX" />
+ <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+ <ctl name="ADC MUX6" value="DMIC" />
+ <ctl name="DMIC MUX6" value="DMIC2" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC4" />
+ <ctl name="CDC_IF TX9 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC0" />
+ <ctl name="IIR0 INP0 MUX" value="DEC6" />
+ <ctl name="ADC MUX10" value="DMIC" />
+ <ctl name="DMIC MUX10" value="DMIC4" />
+ <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
+ </path>
+
+ <!-- Dual MIC devices -->
+ <path name="handset-dmic-endfire">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC4" />
+ <ctl name="SLIM_0_TX Channels" value="Two" />
+ </path>
+
+ <path name="speaker-dmic-endfire">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC4" />
+ <ctl name="SLIM_0_TX Channels" value="Two" />
+ </path>
+
+ <path name="dmic-endfire">
+ <path name="handset-dmic-endfire" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="dmic-endfire-liquid">
+ <path name="handset-dmic-endfire" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="handset-stereo-dmic-ef">
+ <path name="handset-dmic-endfire" />
+ </path>
+
+ <path name="speaker-stereo-dmic-ef">
+ <path name="speaker-dmic-endfire" />
+ </path>
+
+ <path name="voice-dmic-ef-tmus">
+ <path name="dmic-endfire" />
+ </path>
+
+ <path name="voice-dmic-ef">
+ <path name="dmic-endfire" />
+ </path>
+
+ <path name="voice-speaker-dmic-ef">
+ <path name="speaker-dmic-endfire" />
+ </path>
+
+ <path name="voice-rec-dmic-ef">
+ <path name="dmic-endfire" />
+ </path>
+
+ <path name="voice-rec-dmic-ef-fluence">
+ <path name="dmic-endfire" />
+ </path>
+
+ <path name="handset-stereo-dmic-ef-liquid">
+ <path name="handset-dmic-endfire" />
+ </path>
+
+ <path name="speaker-stereo-dmic-ef-liquid">
+ <path name="speaker-dmic-endfire" />
+ </path>
+
+ <path name="voice-dmic-ef-liquid-liquid">
+ <path name="dmic-endfire-liquid" />
+ </path>
+
+ <path name="voice-speaker-dmic-ef-liquid">
+ <path name="dmic-endfire-liquid" />
+ </path>
+
+ <path name="voice-rec-dmic-ef-liquid">
+ <path name="dmic-endfire-liquid" />
+ </path>
+
+ <path name="voice-rec-dmic-ef-fluence-liquid">
+ <path name="dmic-endfire-liquid" />
+ </path>
+
+ <path name="speaker-dmic-broadside">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="SLIM_0_TX Channels" value="Two" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC2" />
+ </path>
+
+ <path name="dmic-broadside">
+ <path name="speaker-dmic-broadside" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="voice-speaker-dmic-broadside">
+ <path name="dmic-broadside" />
+ </path>
+
+ <!-- Quad MIC devices -->
+ <path name="speaker-qmic">
+ <ctl name="AIF1_CAP Mixer SLIM TX5" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="SLIM_0_TX Channels" value="Four" />
+ <ctl name="CDC_IF TX5 MUX" value="DEC5" />
+ <ctl name="ADC MUX5" value="DMIC" />
+ <ctl name="DMIC MUX5" value="DMIC1" />
+ <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+ <ctl name="ADC MUX6" value="DMIC" />
+ <ctl name="DMIC MUX6" value="DMIC0" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC4" />
+ </path>
+
+ <path name="speaker-qmic-liquid">
+ </path>
+
+ <path name="voice-speaker-qmic">
+ <path name="speaker-qmic" />
+ </path>
+
+ <path name="quad-mic">
+ <path name="speaker-qmic" />
+ </path>
+
+ <path name="voice-speaker-qmic-liquid">
+ <path name="speaker-qmic-liquid" />
+ </path>
+
+ <path name="quad-mic-liquid">
+ <path name="speaker-qmic-liquid" />
+ </path>
+
+ <!-- TTY devices -->
+
+ <path name="tty-headphones">
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="One" />
+ <ctl name="RX INT1_2 MUX" value="RX2" />
+ </path>
+
+ <path name="voice-tty-full-headphones">
+ <ctl name="TTY Mode" value="FULL" />
+ <path name="tty-headphones" />
+ </path>
+
+ <path name="voice-tty-vco-headphones">
+ <ctl name="TTY Mode" value="VCO" />
+ <path name="tty-headphones" />
+ </path>
+
+ <path name="voice-tty-hco-handset">
+ <ctl name="TTY Mode" value="HCO" />
+ <path name="handset" />
+ </path>
+
+ <path name="voice-tty-full-headset-mic">
+ <path name="amic2" />
+ </path>
+
+ <path name="voice-tty-hco-headset-mic">
+ <path name="voice-tty-full-headset-mic" />
+ </path>
+
+ <path name="voice-tty-vco-handset-mic">
+ <path name="dmic1" />
+ </path>
+
+ <path name="unprocessed-handset-mic">
+ <path name="handset-mic" />
+ </path>
+
+ <path name="unprocessed-mic">
+ <path name="unprocessed-handset-mic" />
+ </path>
+
+ <!-- Added for ADSP testfwk -->
+ <path name="ADSP testfwk">
+ <ctl name="SLIMBUS_DL_HL Switch" value="1" />
+ </path>
+
+ <path name="bt-a2dp">
+ <ctl name="BT SampleRate" value="KHZ_48" />
+ <ctl name="AFE Input Channels" value="Two" />
+ <ctl name="SLIM7_RX ADM Channels" value="Two" />
+ </path>
+
+ <path name="speaker-and-bt-a2dp">
+ <path name="speaker" />
+ <path name="bt-a2dp" />
+ </path>
+
+ <path name="deep-buffer-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="low-latency-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="compress-offload-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="audio-ull-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-bt-a2dp">
+ <path name="deep-buffer-playback bt-a2dp" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-bt-a2dp">
+ <path name="compress-offload-playback bt-a2dp" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-bt-a2dp">
+ <path name="low-latency-playback bt-a2dp" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback2 bt-a2dp" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback3 bt-a2dp" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback4 bt-a2dp" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback5 bt-a2dp" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback6 bt-a2dp" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback7 bt-a2dp" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback8 bt-a2dp" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback9 bt-a2dp" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-bt-a2dp">
+ <path name="audio-ull-playback bt-a2dp" />
+ <path name="audio-ull-playback" />
+ </path>
+</mixer>
diff --git a/configs/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index 860d014..efd275d 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -77,11 +77,10 @@
<ctl name="DEC8 Volume" value="84" />
<ctl name="COMP1 Switch" value="1" />
<ctl name="COMP2 Switch" value="1" />
- <ctl name="COMP7 Switch" value="1" />
- <ctl name="COMP8 Switch" value="1" />
+ <ctl name="COMP7 Switch" value="0" />
+ <ctl name="COMP8 Switch" value="0" />
<ctl name="RX HPH Mode" value="CLS_H_LP" />
<ctl name="SLIMBUS_3_RX Port Mixer MI2S_TX" value="0" />
- <ctl name="HDMI_RX Port Mixer MI2S_TX" value="0" />
<ctl name="SLIMBUS_0_RX Port Mixer SLIM_0_TX" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
@@ -139,6 +138,22 @@
<ctl name="HDMI Mixer MultiMedia14" value="0" />
<ctl name="HDMI Mixer MultiMedia15" value="0" />
<ctl name="HDMI Mixer MultiMedia16" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia6" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia8" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia2" value="0" />
@@ -190,12 +205,6 @@
<ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
- <ctl name="USB_AUDIO_RX Channels" value="One" />
- <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
- <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
- <ctl name="USB_AUDIO_TX Channels" value="One" />
- <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
- <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
<ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
<ctl name="IIR0 INP0 MUX" value="ZERO" />
<ctl name="IIR0 INP1 MUX" value="ZERO" />
@@ -301,8 +310,6 @@
<ctl name="SPL SRC3 MUX" value="ZERO" />
<ctl name="RX INT1 SPLINE MIX HPHL Switch" value="0" />
<ctl name="RX INT3 SPLINE MIX LO1 Switch" value="0" />
- <ctl name="RX INT1 SPLINE MIX HPHL Native Switch" value="0" />
- <ctl name="RX INT2 SPLINE MIX HPHR Native Switch" value="0" />
<ctl name="RX INT2 SPLINE MIX HPHR Switch" value="0" />
<ctl name="RX INT4 SPLINE MIX LO2 Switch" value="0" />
<ctl name="RX INT5 SPLINE MIX LO3 Switch" value="0" />
@@ -339,6 +346,7 @@
<ctl name="SLIM RX4 MUX" value="ZERO" />
<ctl name="SLIM RX5 MUX" value="ZERO" />
<ctl name="EAR PA Gain" value="G_6_DB" />
+ <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
<ctl name="SpkrLeft COMP Switch" value="0" />
<ctl name="SpkrRight COMP Switch" value="0" />
<ctl name="SpkrLeft BOOST Switch" value="0" />
@@ -360,6 +368,10 @@
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_6_RX Port Mixer AUX_PCM_UL_TX" value="0" />
<ctl name="HDMI Mixer MultiMedia4" value="0" />
+ <ctl name= "RX INT1_1 NATIVE MUX" value="OFF" />
+ <ctl name= "RX INT2_1 NATIVE MUX" value="OFF" />
+ <ctl name= "RX INT3_1 NATIVE MUX" value="OFF" />
+ <ctl name= "RX INT4_1 NATIVE MUX" value="OFF" />
<!-- HFP start -->
<ctl name="HFP_PRI_AUX_UL_HL Switch" value="0" />
<ctl name="SLIMBUS_0_RX Port Mixer SLIM_7_TX" value="0" />
@@ -381,7 +393,7 @@
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
<!-- usb headset end -->
<!-- fm -->
- <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+ <ctl name="SLIMBUS_8 LOOPBACK Volume" value="0" />
<ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="0" />
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
<ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="0" />
@@ -513,6 +525,9 @@
<ctl name="ANC0 FB MUX" value="ZERO" />
<ctl name="ANC1 FB MUX" value="ZERO" />
<ctl name="ANC EAR Enable Switch" value="0" />
+ <ctl name="ANC OUT EAR SPKR Enable Switch" value="0" />
+ <ctl name="ANC SPKR PA Enable Switch" value="0" />
+ <ctl name="SpkrLeft WSA PA Gain" value="G_0_DB" />
<!-- anc handset end -->
<ctl name="ANC Function" value="OFF" />
<ctl name="ANC HPHL Enable Switch" value="0" />
@@ -548,6 +563,11 @@
<ctl name="LSM8 MUX" value="None" />
<ctl name="SLIMBUS_5_TX LSM Function" value="None" />
<!-- listen end-->
+ <!-- split a2dp -->
+ <ctl name="BT SampleRate" value="KHZ_8" />
+ <ctl name="AFE Input Channels" value="Zero" />
+ <ctl name="SLIM7_RX ADM Channels" value="Zero" />
+ <!-- split a2dp end-->
<!-- ADSP testfwk -->
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
@@ -604,17 +624,26 @@
<ctl name="HDMI Mixer MultiMedia1" value="1" />
</path>
+ <path name="deep-buffer-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="1" />
+ </path>
+
<path name="deep-buffer-playback speaker-and-hdmi">
<path name="deep-buffer-playback hdmi" />
<path name="deep-buffer-playback" />
</path>
+ <path name="deep-buffer-playback speaker-and-display-port">
+ <path name="deep-buffer-playback display-port" />
+ <path name="deep-buffer-playback" />
+ </path>
+
<path name="deep-buffer-playback bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
</path>
<path name="deep-buffer-playback bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="deep-buffer-playback bt-sco" />
</path>
@@ -652,12 +681,16 @@
<ctl name="HDMI Mixer MultiMedia5" value="1" />
</path>
+ <path name="low-latency-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="1" />
+ </path>
+
<path name="low-latency-playback bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
</path>
<path name="low-latency-playback bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="low-latency-playback bt-sco" />
</path>
@@ -666,6 +699,11 @@
<path name="low-latency-playback" />
</path>
+ <path name="low-latency-playback speaker-and-display-port">
+ <path name="low-latency-playback display-port" />
+ <path name="low-latency-playback" />
+ </path>
+
<path name="low-latency-playback afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="1" />
</path>
@@ -689,7 +727,7 @@
</path>
<path name="audio-ull-playback">
- <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback speaker-protected">
@@ -697,7 +735,7 @@
</path>
<path name="audio-ull-playback headphones">
- <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback speaker-and-headphones">
@@ -706,15 +744,19 @@
</path>
<path name="audio-ull-playback hdmi">
- <ctl name="HDMI Mixer MultiMedia3" value="1" />
+ <ctl name="HDMI Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="1" />
</path>
<path name="audio-ull-playback bt-sco">
- <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="audio-ull-playback bt-sco" />
</path>
@@ -723,18 +765,27 @@
<path name="audio-ull-playback" />
</path>
+ <path name="audio-ull-playback speaker-and-display-port">
+ <path name="audio-ull-playback display-port" />
+ <path name="audio-ull-playback" />
+ </path>
+
<path name="audio-ull-playback afe-proxy">
- <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback usb-headphones">
- <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="multi-channel-playback hdmi">
<ctl name="HDMI Mixer MultiMedia2" value="1" />
</path>
+ <path name="multi-channel-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+ </path>
+
<path name="multi-channel-playback afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
</path>
@@ -755,12 +806,20 @@
<ctl name="HDMI Mixer MultiMedia9" value="1" />
</path>
+ <path name="compress-offload-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="silence-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="1" />
+ </path>
+
<path name="compress-offload-playback bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
</path>
<path name="compress-offload-playback bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback bt-sco" />
</path>
@@ -769,6 +828,11 @@
<path name="compress-offload-playback" />
</path>
+ <path name="compress-offload-playback speaker-and-display-port">
+ <path name="compress-offload-playback display-port" />
+ <path name="compress-offload-playback" />
+ </path>
+
<path name="compress-offload-playback afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia4" value="1" />
</path>
@@ -803,12 +867,16 @@
<ctl name="HDMI Mixer MultiMedia7" value="1" />
</path>
+ <path name="compress-offload-playback2 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="1" />
+ </path>
+
<path name="compress-offload-playback2 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
</path>
<path name="compress-offload-playback2 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback2 bt-sco" />
</path>
@@ -817,6 +885,11 @@
<path name="compress-offload-playback2" />
</path>
+ <path name="compress-offload-playback2 speaker-and-display-port">
+ <path name="compress-offload-playback2 display-port" />
+ <path name="compress-offload-playback2" />
+ </path>
+
<path name="compress-offload-playback2 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
</path>
@@ -851,12 +924,16 @@
<ctl name="HDMI Mixer MultiMedia10" value="1" />
</path>
+ <path name="compress-offload-playback3 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="1" />
+ </path>
+
<path name="compress-offload-playback3 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
</path>
<path name="compress-offload-playback3 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback3 bt-sco" />
</path>
@@ -865,6 +942,11 @@
<path name="compress-offload-playback3" />
</path>
+ <path name="compress-offload-playback3 speaker-and-display-port">
+ <path name="compress-offload-playback3 display-port" />
+ <path name="compress-offload-playback3" />
+ </path>
+
<path name="compress-offload-playback3 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
</path>
@@ -899,12 +981,16 @@
<ctl name="HDMI Mixer MultiMedia11" value="1" />
</path>
+ <path name="compress-offload-playback4 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="1" />
+ </path>
+
<path name="compress-offload-playback4 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
</path>
<path name="compress-offload-playback4 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback4 bt-sco" />
</path>
@@ -913,6 +999,12 @@
<path name="compress-offload-playback4" />
</path>
+ <path name="compress-offload-playback4 speaker-and-display-port">
+ <path name="compress-offload-playback4 display-port" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+
<path name="compress-offload-playback4 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="1" />
</path>
@@ -947,12 +1039,16 @@
<ctl name="HDMI Mixer MultiMedia12" value="1" />
</path>
+ <path name="compress-offload-playback5 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="1" />
+ </path>
+
<path name="compress-offload-playback5 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
</path>
<path name="compress-offload-playback5 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback5 bt-sco" />
</path>
@@ -961,6 +1057,11 @@
<path name="compress-offload-playback5" />
</path>
+ <path name="compress-offload-playback5 speaker-and-display-port">
+ <path name="compress-offload-playback5 display-port" />
+ <path name="compress-offload-playback5" />
+ </path>
+
<path name="compress-offload-playback5 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia12" value="1" />
</path>
@@ -995,12 +1096,16 @@
<ctl name="HDMI Mixer MultiMedia13" value="1" />
</path>
+ <path name="compress-offload-playback6 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="1" />
+ </path>
+
<path name="compress-offload-playback6 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
</path>
<path name="compress-offload-playback6 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback6 bt-sco" />
</path>
@@ -1009,6 +1114,11 @@
<path name="compress-offload-playback6" />
</path>
+ <path name="compress-offload-playback6 speaker-and-display-port">
+ <path name="compress-offload-playback6 display-port" />
+ <path name="compress-offload-playback6" />
+ </path>
+
<path name="compress-offload-playback6 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia13" value="1" />
</path>
@@ -1043,12 +1153,16 @@
<ctl name="HDMI Mixer MultiMedia14" value="1" />
</path>
+ <path name="compress-offload-playback7 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="1" />
+ </path>
+
<path name="compress-offload-playback7 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
</path>
<path name="compress-offload-playback7 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback7 bt-sco" />
</path>
@@ -1057,6 +1171,11 @@
<path name="compress-offload-playback7" />
</path>
+ <path name="compress-offload-playback7 speaker-and-display-port">
+ <path name="compress-offload-playback7 display-port" />
+ <path name="compress-offload-playback7" />
+ </path>
+
<path name="compress-offload-playback7 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia14" value="1" />
</path>
@@ -1091,12 +1210,16 @@
<ctl name="HDMI Mixer MultiMedia15" value="1" />
</path>
+ <path name="compress-offload-playback8 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="1" />
+ </path>
+
<path name="compress-offload-playback8 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
</path>
<path name="compress-offload-playback8 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback8 bt-sco" />
</path>
@@ -1105,6 +1228,11 @@
<path name="compress-offload-playback8" />
</path>
+ <path name="compress-offload-playback8 speaker-and-display-port">
+ <path name="compress-offload-playback8 display-port" />
+ <path name="compress-offload-playback8" />
+ </path>
+
<path name="compress-offload-playback8 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia15" value="1" />
</path>
@@ -1139,12 +1267,16 @@
<ctl name="HDMI Mixer MultiMedia16" value="1" />
</path>
+ <path name="compress-offload-playback9 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+ </path>
+
<path name="compress-offload-playback9 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
</path>
<path name="compress-offload-playback9 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback9 bt-sco" />
</path>
@@ -1153,6 +1285,11 @@
<path name="compress-offload-playback9" />
</path>
+ <path name="compress-offload-playback9 speaker-and-display-port">
+ <path name="compress-offload-playback9 display-port" />
+ <path name="compress-offload-playback9" />
+ </path>
+
<path name="compress-offload-playback9 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
</path>
@@ -1192,7 +1329,7 @@
</path>
<path name="audio-record bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="audio-record bt-sco" />
</path>
@@ -1209,7 +1346,7 @@
</path>
<path name="audio-record-compress bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="audio-record-compress bt-sco" />
</path>
@@ -1218,24 +1355,24 @@
</path>
<path name="low-latency-record">
- <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
</path>
<path name="low-latency-record bt-sco">
- <ctl name="MultiMedia5 Mixer SLIM_7_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
</path>
<path name="low-latency-record bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="low-latency-record bt-sco" />
</path>
<path name="low-latency-record usb-headset-mic">
- <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
</path>
<path name="low-latency-record capture-fm">
- <ctl name="MultiMedia5 Mixer SLIM_8_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer SLIM_8_TX" value="1" />
</path>
<path name="fm-virtual-record capture-fm">
@@ -1393,12 +1530,12 @@
</path>
<path name="hfp-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="hfp-sco" />
</path>
<path name="hfp-sco-wb headphones">
- <ctl name="AUX PCM SampleRate" value="16000" />
+ <ctl name="AUX PCM SampleRate" value="KHZ_16" />
<path name="hfp-sco headphones" />
</path>
@@ -1419,7 +1556,7 @@
</path>
<path name="compress-voip-call bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-voip-call bt-sco" />
</path>
@@ -1459,7 +1596,7 @@
</path>
<path name="vowlan-call bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="vowlan-call bt-sco" />
</path>
@@ -1499,7 +1636,7 @@
</path>
<path name="voicemmode1-call bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="voicemmode1-call bt-sco" />
</path>
@@ -1539,7 +1676,7 @@
</path>
<path name="voicemmode2-call bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="voicemmode2-call bt-sco" />
</path>
@@ -1636,15 +1773,6 @@
</path>
<!-- For Tasha, DMIC numbered from 0 to 5 -->
- <path name="dmic3">
- <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
- <ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="SLIM TX7 MUX" value="DEC7" />
- <ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC2" />
- <ctl name="IIR0 INP0 MUX" value="DEC7" />
- </path>
-
<path name="dmic1">
<ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
@@ -1663,6 +1791,15 @@
<ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
+ <path name="dmic3">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="SLIM TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
<path name="dmic4">
<ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
@@ -1691,6 +1828,8 @@
</path>
<path name="speaker">
+ <ctl name="COMP7 Switch" value="1" />
+ <ctl name="COMP8 Switch" value="1" />
<ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
<ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
<ctl name="SLIM_0_RX Channels" value="Two" />
@@ -1707,6 +1846,7 @@
</path>
<path name="speaker-fluid">
+ <ctl name="COMP7 Switch" value="1" />
<ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
<ctl name="SLIM_0_RX Channels" value="One" />
<ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
@@ -1717,6 +1857,7 @@
</path>
<path name="speaker-mono">
+ <ctl name="COMP7 Switch" value="1" />
<ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
<ctl name="SLIM_0_RX Channels" value="One" />
<ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
@@ -1763,11 +1904,11 @@
</path>
<path name="speaker-mic">
- <path name="dmic3" />
+ <path name="dmic2" />
</path>
<path name="speaker-mic-liquid">
- <path name="dmic3" />
+ <path name="dmic2" />
<ctl name="DEC7 Volume" value="111" />
</path>
@@ -1809,18 +1950,18 @@
</path>
<path name="handset">
+ <ctl name="COMP7 Switch" value="1" />
<ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
<ctl name="SLIM_0_RX Channels" value="One" />
<ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
<ctl name="SpkrLeft COMP Switch" value="1" />
<ctl name="SpkrLeft BOOST Switch" value="1" />
- <ctl name="SpkrLeft VISENSE Switch" value="1" />
<ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
- <ctl name="RX7 Digital Volume" value="76" />
+ <ctl name="EAR SPKR PA Gain" value="G_6_DB" />
</path>
<path name="handset-mic">
- <path name="dmic1" />
+ <path name="dmic3" />
</path>
<path name="handset-mic-db">
@@ -1847,32 +1988,30 @@
<ctl name="DMIC MUX5" value="DMIC0" />
<ctl name="SLIM TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
- <ctl name="DMIC MUX6" value="DMIC4" />
+ <ctl name="DMIC MUX6" value="DMIC2" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC3" />
+ <ctl name="DMIC MUX8" value="DMIC5" />
</path>
<path name="anc-handset">
<ctl name="ANC Function" value="ON" />
+ <ctl name="ANC Slot" value="6" />
<ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
<ctl name="SLIM_0_RX Channels" value="One" />
- <ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
- <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
- <ctl name="RX0 Digital Volume" value="81" />
- <ctl name="ANC Slot" value="6" />
- <ctl name="ADC MUX10" value="DMIC" />
- <ctl name="DMIC MUX10" value="DMIC3" />
- <ctl name="ANC0 FB MUX" value="ANC_IN_EAR" />
- <ctl name="ANC EAR Enable Switch" value="1" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+ <ctl name="ANC OUT EAR SPKR Enable Switch" value="1" />
+ <ctl name="ANC SPKR PA Enable Switch" value="1" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ <ctl name="SpkrLeft WSA PA Gain" value="G_6_DB" />
</path>
<path name="headphones">
<ctl name="SLIM RX2 MUX" value="AIF4_PB" />
<ctl name="SLIM RX3 MUX" value="AIF4_PB" />
<ctl name="SLIM_6_RX Channels" value="Two" />
- <ctl name= "RX INT1_1 MIX1 INP0" value="RX2" />
- <ctl name= "RX INT2_1 MIX1 INP0" value="RX3" />
+ <ctl name= "RX INT1_2 MUX" value="RX2" />
+ <ctl name= "RX INT2_2 MUX" value="RX3" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
</path>
@@ -1898,8 +2037,16 @@
<ctl name="RX INT2_2 MUX" value="ZERO" />
<ctl name= "RX INT1_1 MIX1 INP0" value="RX2" />
<ctl name= "RX INT2_1 MIX1 INP0" value="RX3" />
- <ctl name= "RX INT1 SPLINE MIX HPHL Native Switch" value="1" />
- <ctl name= "RX INT2 SPLINE MIX HPHR Native Switch" value="1" />
+ <ctl name= "RX INT1_1 NATIVE MUX" value="ON" />
+ <ctl name= "RX INT2_1 NATIVE MUX" value="ON" />
+ </path>
+
+ <path name="hph-highquality-mode">
+ <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
+ </path>
+
+ <path name="hph-lowpower-mode">
+ <ctl name="RX HPH Mode" value="CLS_H_LP" />
</path>
<path name="line">
@@ -2098,6 +2245,9 @@
<path name="hdmi">
</path>
+ <path name="display-port">
+ </path>
+
<path name="speaker-and-usb-headphones">
<path name="speaker" />
<path name="usb-headphones" />
@@ -2108,6 +2258,11 @@
<path name="hdmi" />
</path>
+ <path name="speaker-and-display-port">
+ <path name="speaker" />
+ <path name="display-port" />
+ </path>
+
<path name="voice-rec-mic">
<path name="handset-mic" />
</path>
@@ -2146,14 +2301,17 @@
<ctl name="AANC_SLIM_0_RX MUX" value="SLIMBUS_0_TX" />
<ctl name="SLIM TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
- <ctl name="DMIC MUX6" value="DMIC0" />
+ <ctl name="DMIC MUX6" value="DMIC2" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC3" />
+ <ctl name="DMIC MUX8" value="DMIC4" />
<ctl name="SLIM TX9 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="DMIC MUX7" value="DMIC0" />
<ctl name="IIR0 INP0 MUX" value="DEC6" />
+ <ctl name="ADC MUX10" value="DMIC" />
+ <ctl name="DMIC MUX10" value="DMIC4" />
+ <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
</path>
<!-- Dual MIC devices -->
@@ -2162,10 +2320,10 @@
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
<ctl name="SLIM TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC0" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC3" />
+ <ctl name="DMIC MUX8" value="DMIC4" />
<ctl name="SLIM_0_TX Channels" value="Two" />
</path>
@@ -2174,10 +2332,10 @@
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
<ctl name="SLIM TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="DMIC MUX7" value="DMIC1" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC3" />
+ <ctl name="DMIC MUX8" value="DMIC5" />
<ctl name="SLIM_0_TX Channels" value="Two" />
</path>
@@ -2249,7 +2407,7 @@
<ctl name="SLIM_0_TX Channels" value="Two" />
<ctl name="SLIM TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC0" />
+ <ctl name="DMIC MUX7" value="DMIC1" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
<ctl name="DMIC MUX8" value="DMIC2" />
@@ -2273,16 +2431,16 @@
<ctl name="SLIM_0_TX Channels" value="Four" />
<ctl name="SLIM TX5 MUX" value="DEC5" />
<ctl name="ADC MUX5" value="DMIC" />
- <ctl name="DMIC MUX5" value="DMIC0" />
+ <ctl name="DMIC MUX5" value="DMIC1" />
<ctl name="SLIM TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
- <ctl name="DMIC MUX6" value="DMIC2" />
+ <ctl name="DMIC MUX6" value="DMIC0" />
<ctl name="SLIM TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
<ctl name="SLIM TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC3" />
+ <ctl name="DMIC MUX8" value="DMIC5" />
</path>
<path name="speaker-qmic-liquid">
@@ -2360,7 +2518,7 @@
<path name="listen-handset-mic">
<ctl name="MADONOFF Switch" value="1" />
- <ctl name="MAD Input" value="DMIC0" />
+ <ctl name="MAD Input" value="DMIC2" />
</path>
<path name="unprocessed-handset-mic">
@@ -2376,4 +2534,122 @@
<ctl name="SLIMBUS_DL_HL Switch" value="1" />
</path>
+ <path name="bt-a2dp">
+ <ctl name="BT SampleRate" value="KHZ_48" />
+ <ctl name="AFE Input Channels" value="Two" />
+ <ctl name="SLIM7_RX ADM Channels" value="Two" />
+ </path>
+
+ <path name="speaker-and-bt-a2dp">
+ <path name="speaker" />
+ <path name="bt-a2dp" />
+ </path>
+
+ <path name="deep-buffer-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="low-latency-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="compress-offload-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="audio-ull-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-bt-a2dp">
+ <path name="deep-buffer-playback bt-a2dp" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-bt-a2dp">
+ <path name="compress-offload-playback bt-a2dp" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-bt-a2dp">
+ <path name="low-latency-playback bt-a2dp" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback2 bt-a2dp" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback3 bt-a2dp" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback4 bt-a2dp" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback5 bt-a2dp" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback6 bt-a2dp" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback7 bt-a2dp" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback8 bt-a2dp" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback9 bt-a2dp" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-bt-a2dp">
+ <path name="audio-ull-playback bt-a2dp" />
+ <path name="audio-ull-playback" />
+ </path>
</mixer>
diff --git a/configs/msmcobalt/mixer_paths_tavil.xml b/configs/msmcobalt/mixer_paths_tavil.xml
index ca132c0..29212f9 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -50,6 +50,7 @@
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="0" />
@@ -67,6 +68,9 @@
<ctl name="MultiMedia1 Mixer SLIM_0_TX" value="0" />
<ctl name="MultiMedia1 Mixer SLIM_4_TX" value="0" />
<ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_4_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
<ctl name="HDMI Mixer MultiMedia1" value="0" />
<ctl name="HDMI Mixer MultiMedia2" value="0" />
<ctl name="HDMI Mixer MultiMedia3" value="0" />
@@ -81,6 +85,22 @@
<ctl name="HDMI Mixer MultiMedia14" value="0" />
<ctl name="HDMI Mixer MultiMedia15" value="0" />
<ctl name="HDMI Mixer MultiMedia16" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia6" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia8" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia2" value="0" />
@@ -88,33 +108,46 @@
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia5" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="0" />
@@ -122,6 +155,7 @@
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="0" />
@@ -132,19 +166,19 @@
<ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
- <ctl name="USB_AUDIO_RX Channels" value="One" />
- <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
- <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
- <ctl name="USB_AUDIO_TX Channels" value="One" />
- <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
- <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
<ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
+ <ctl name="SLIM_2_RX Format" value="UNPACKED" />
+ <ctl name="SLIM_2_RX SampleRate" value="KHZ_48" />
+ <ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
<ctl name="SLIM_0_RX Channels" value="One" />
<ctl name="SLIM_5_RX Channels" value="One" />
+ <ctl name="SLIM_6_RX Channels" value="One" />
+ <ctl name="SLIM_2_RX Channels" value="One" />
<ctl name="SLIM_0_TX Channels" value="One" />
<ctl name="SLIM_1_TX Channels" value="One" />
- <ctl name="AIF1_CAP Mixer SLIM TX7" value="0" />
+ <ctl name="AIF1_CAP Mixer SLIM TX9" value="0"/>
<ctl name="AIF1_CAP Mixer SLIM TX8" value="0"/>
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="0" />
<ctl name="AIF1_CAP Mixer SLIM TX6" value="0" />
<ctl name="AIF1_CAP Mixer SLIM TX5" value="0"/>
<ctl name="AIF1_CAP Mixer SLIM TX4" value="0" />
@@ -154,6 +188,13 @@
<ctl name="AIF1_CAP Mixer SLIM TX0" value="0"/>
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="HDMI Mixer MultiMedia4" value="0" />
+ <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="ZERO" />
+ <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="ZERO" />
+ <ctl name="VI_FEED_TX Channels" value="Two" />
+ <ctl name="AIF4_VI Mixer SPKR_VI_1" value="0" />
+ <ctl name="AIF4_VI Mixer SPKR_VI_2" value="0" />
+ <ctl name="SLIM_4_TX Format" value="UNPACKED" />
+ <ctl name="AANC_SLIM_0_RX MUX" value="ZERO" />
<!-- HFP start -->
<ctl name="HFP_PRI_AUX_UL_HL Switch" value="0" />
<ctl name="SLIMBUS_0_RX Port Mixer SLIM_7_TX" value="0" />
@@ -175,7 +216,7 @@
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
<!-- usb headset end -->
<!-- fm -->
- <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+ <ctl name="SLIMBUS_8 LOOPBACK Volume" value="0" />
<ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="0" />
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
<ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="0" />
@@ -262,6 +303,12 @@
<ctl name="MultiMedia8 Mixer AFE_PCM_TX" value="0" />
<!-- audio record compress end-->
+ <!-- split a2dp -->
+ <ctl name="BT SampleRate" value="KHZ_8" />
+ <ctl name="AFE Input Channels" value="Zero" />
+ <ctl name="SLIM7_RX ADM Channels" value="Zero" />
+ <!-- split a2dp end-->
+
<!-- ADSP testfwk -->
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
<ctl name="SLIMBUS6_DL_HL Switch" value="0" />
@@ -270,14 +317,7 @@
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
<!-- Codec controls -->
- <ctl name="SLIM RX0 MUX" value="ZERO" />
- <ctl name="SLIM RX1 MUX" value="ZERO" />
- <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
- <ctl name="CDC_IF RX1 MUX" value="SLIM RX1" />
- <ctl name="RX INT7_1 MIX1 INP0" value="ZERO" />
- <ctl name="RX INT8_1 MIX1 INP0" value="ZERO" />
- <ctl name="COMP7 Switch" value="0" />
- <ctl name="COMP8 Switch" value="0" />
+ <!-- WSA controls -->
<ctl name="SpkrLeft COMP Switch" value="0" />
<ctl name="SpkrRight COMP Switch" value="0" />
<ctl name="SpkrLeft BOOST Switch" value="0" />
@@ -286,22 +326,205 @@
<ctl name="SpkrRight VISENSE Switch" value="0" />
<ctl name="SpkrLeft SWR DAC_Port Switch" value="0" />
<ctl name="SpkrRight SWR DAC_Port Switch" value="0" />
+ <ctl name="SpkrLeft WSA PA Gain" value="G_0_DB" />
- <ctl name="AIF1_CAP Mixer SLIM TX0" value="0" />
- <ctl name="CDC_IF TX0 MUX" value="ZERO" />
- <ctl name="ADC MUX0" value="ZERO" />
+ <!-- Volume controls -->
+ <ctl name="LINEOUT1 Volume" value="13" />
+ <ctl name="LINEOUT2 Volume" value="13" />
+ <ctl name="HPHL Volume" value="20" />
+ <ctl name="HPHR Volume" value="20" />
+ <ctl name="EAR PA Gain" value="G_6_DB" />
+ <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
+
+ <ctl name="RX0 Digital Volume" value="84" />
+ <ctl name="RX1 Digital Volume" value="84" />
+ <ctl name="RX2 Digital Volume" value="84" />
+ <ctl name="RX3 Digital Volume" value="84" />
+ <ctl name="RX4 Digital Volume" value="84" />
+ <ctl name="RX5 Digital Volume" value="84" />
+ <ctl name="RX6 Digital Volume" value="84" />
+ <ctl name="RX7 Digital Volume" value="84" />
+ <ctl name="ADC1 Volume" value="12" />
+ <ctl name="ADC2 Volume" value="12" />
+ <ctl name="ADC3 Volume" value="12" />
+ <ctl name="ADC4 Volume" value="12" />
+ <ctl name="DEC0 Volume" value="84" />
+ <ctl name="DEC1 Volume" value="84" />
+ <ctl name="DEC2 Volume" value="84" />
+ <ctl name="DEC3 Volume" value="84" />
+ <ctl name="DEC4 Volume" value="84" />
+ <ctl name="DEC5 Volume" value="84" />
+ <ctl name="DEC6 Volume" value="84" />
+ <ctl name="DEC7 Volume" value="84" />
+ <ctl name="DEC8 Volume" value="84" />
+
+ <!-- Compander controls -->
+ <ctl name="COMP1 Switch" value="1" />
+ <ctl name="COMP2 Switch" value="1" />
+ <ctl name="COMP7 Switch" value="0" />
+ <ctl name="COMP8 Switch" value="0" />
+
+ <!-- ADC, DMIC and AMIC controls -->
+ <ctl name="ADC MUX0" value="AMIC" />
+ <ctl name="ADC MUX1" value="AMIC" />
+ <ctl name="ADC MUX2" value="AMIC" />
+ <ctl name="ADC MUX3" value="AMIC" />
+ <ctl name="ADC MUX4" value="AMIC" />
+ <ctl name="ADC MUX5" value="AMIC" />
+ <ctl name="ADC MUX6" value="AMIC" />
+ <ctl name="ADC MUX7" value="AMIC" />
+ <ctl name="ADC MUX8" value="AMIC" />
+ <ctl name="ADC MUX10" value="AMIC" />
+ <ctl name="ADC MUX11" value="AMIC" />
+ <ctl name="ADC MUX12" value="AMIC" />
+ <ctl name="ADC MUX13" value="AMIC" />
<ctl name="DMIC MUX0" value="ZERO" />
- <ctl name="DEC0 Volume" value="0" />
+ <ctl name="DMIC MUX1" value="ZERO" />
+ <ctl name="DMIC MUX2" value="ZERO" />
+ <ctl name="DMIC MUX3" value="ZERO" />
+ <ctl name="DMIC MUX4" value="ZERO" />
+ <ctl name="DMIC MUX5" value="ZERO" />
+ <ctl name="DMIC MUX6" value="ZERO" />
+ <ctl name="DMIC MUX7" value="ZERO" />
+ <ctl name="DMIC MUX8" value="ZERO" />
+ <ctl name="DMIC MUX10" value="ZERO" />
+ <ctl name="DMIC MUX11" value="ZERO" />
+ <ctl name="DMIC MUX12" value="ZERO" />
+ <ctl name="DMIC MUX13" value="ZERO" />
+ <ctl name="AMIC MUX0" value="ZERO" />
+ <ctl name="AMIC MUX1" value="ZERO" />
+ <ctl name="AMIC MUX2" value="ZERO" />
+ <ctl name="AMIC MUX3" value="ZERO" />
+ <ctl name="AMIC MUX4" value="ZERO" />
+ <ctl name="AMIC MUX5" value="ZERO" />
+ <ctl name="AMIC MUX6" value="ZERO" />
+ <ctl name="AMIC MUX7" value="ZERO" />
+ <ctl name="AMIC MUX8" value="ZERO" />
+ <ctl name="AMIC MUX10" value="ZERO" />
+ <ctl name="AMIC MUX11" value="ZERO" />
+ <ctl name="AMIC MUX12" value="ZERO" />
+ <ctl name="AMIC MUX13" value="ZERO" />
+
+ <!-- CDC_IF and SLIM controls -->
+ <ctl name="SLIM RX0 MUX" value="ZERO" />
+ <ctl name="SLIM RX1 MUX" value="ZERO" />
+ <ctl name="SLIM RX2 MUX" value="ZERO" />
+ <ctl name="SLIM RX3 MUX" value="ZERO" />
+ <ctl name="SLIM RX4 MUX" value="ZERO" />
+ <ctl name="SLIM RX5 MUX" value="ZERO" />
+ <ctl name="SLIM RX6 MUX" value="ZERO" />
+ <ctl name="SLIM RX7 MUX" value="ZERO" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="CDC_IF RX1 MUX" value="SLIM RX1" />
+ <ctl name="CDC_IF RX2 MUX" value="SLIM RX2" />
+ <ctl name="CDC_IF RX3 MUX" value="SLIM RX3" />
+ <ctl name="CDC_IF RX4 MUX" value="SLIM RX4" />
+ <ctl name="CDC_IF RX5 MUX" value="SLIM RX5" />
+ <ctl name="CDC_IF RX6 MUX" value="SLIM RX6" />
+ <ctl name="CDC_IF RX7 MUX" value="SLIM RX7" />
+ <ctl name="CDC_IF TX0 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX1 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX2 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX3 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX4 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX5 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX6 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX7 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX8 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX9 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX10 MUX" value="ZERO" />
+ <ctl name="CDC_IF TX13 MUX" value="ZERO" />
+
+ <!-- Interpolator chain controls -->
+ <ctl name="RX INT0_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT0_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT0_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT2_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT2_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT2_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT7_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT7_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT8_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT8_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT8_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT0_2 MUX" value="ZERO" />
+ <ctl name="RX INT1_2 MUX" value="ZERO" />
+ <ctl name="RX INT2_2 MUX" value="ZERO" />
+ <ctl name="RX INT7_2 MUX" value="ZERO" />
+ <ctl name="RX INT8_2 MUX" value="ZERO" />
+ <ctl name="RX INT1_1 NATIVE MUX" value="OFF" />
+ <ctl name="RX INT2_1 NATIVE MUX" value="OFF" />
+ <ctl name="RX INT1_2 NATIVE MUX" value="OFF" />
+ <ctl name="RX INT2_2 NATIVE MUX" value="OFF" />
+ <ctl name="ASRC0 MUX" value="ZERO" />
+ <ctl name="ASRC1 MUX" value="ZERO" />
+ <ctl name="RX INT1 SEC MIX HPHL Switch" value="0" />
+ <ctl name="RX INT2 SEC MIX HPHR Switch" value="0" />
+ <ctl name="DSD_L IF MUX" value="ZERO" />
+ <ctl name="DSD_R IF MUX" value="ZERO" />
+ <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="0" />
+ <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="0" />
+ <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
+
+ <!-- Headphone Default mode - uLP -->
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
+ <ctl name="ASRC0 Output Mode" value="INT" />
+ <ctl name="ASRC1 Output Mode" value="INT" />
<!-- IIR/voice anc -->
- <!-- IIR/voice anc end -->
- <!-- anc handset -->
- <!-- anc handset end -->
- <!-- anc headset end -->
- <!-- aanc handset mic -->
- <!-- aanc handset mic end -->
- <!-- ssr qmic -->
- <!-- ssr qmic end-->
+ <ctl name="IIR0 Band1" id ="0" value="268435456" />
+ <ctl name="IIR0 Band1" id ="1" value="0" />
+ <ctl name="IIR0 Band1" id ="2" value="0" />
+ <ctl name="IIR0 Band1" id ="3" value="0" />
+ <ctl name="IIR0 Band1" id ="4" value="0" />
+ <ctl name="IIR0 Band2" id ="0" value="268435456" />
+ <ctl name="IIR0 Band2" id ="1" value="0" />
+ <ctl name="IIR0 Band2" id ="2" value="0" />
+ <ctl name="IIR0 Band2" id ="3" value="0" />
+ <ctl name="IIR0 Band2" id ="4" value="0" />
+ <ctl name="IIR0 Band3" id ="0" value="268435456" />
+ <ctl name="IIR0 Band3" id ="1" value="0" />
+ <ctl name="IIR0 Band3" id ="2" value="0" />
+ <ctl name="IIR0 Band3" id ="3" value="0" />
+ <ctl name="IIR0 Band3" id ="4" value="0" />
+ <ctl name="IIR0 Band4" id ="0" value="268435456" />
+ <ctl name="IIR0 Band4" id ="1" value="0" />
+ <ctl name="IIR0 Band4" id ="2" value="0" />
+ <ctl name="IIR0 Band4" id ="3" value="0" />
+ <ctl name="IIR0 Band4" id ="4" value="0" />
+ <ctl name="IIR0 Band5" id ="0" value="268435456" />
+ <ctl name="IIR0 Band5" id ="1" value="0" />
+ <ctl name="IIR0 Band5" id ="2" value="0" />
+ <ctl name="IIR0 Band5" id ="3" value="0" />
+ <ctl name="IIR0 Band5" id ="4" value="0" />
+ <ctl name="IIR0 Enable Band1" value="0" />
+ <ctl name="IIR0 Enable Band2" value="0" />
+ <ctl name="IIR0 Enable Band3" value="0" />
+ <ctl name="IIR0 Enable Band4" value="0" />
+ <ctl name="IIR0 Enable Band5" value="0" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="IIR0 INP0 MUX" value="ZERO" />
+ <ctl name="IIR0 INP1 MUX" value="ZERO" />
+ <ctl name="IIR0 INP2 MUX" value="ZERO" />
+ <ctl name="IIR1 INP0 MUX" value="ZERO" />
+ <ctl name="IIR1 INP1 MUX" value="ZERO" />
+ <ctl name="IIR1 INP2 MUX" value="ZERO" />
+
+ <!-- anc related -->
+ <ctl name="ANC Slot" value="0" />
+ <ctl name="ANC Function" value="OFF" />
+ <ctl name="ANC0 FB MUX" value="ZERO" />
+ <ctl name="ANC1 FB MUX" value="ZERO" />
+ <ctl name="ANC OUT EAR Enable Switch" value="0" />
+ <ctl name="ANC OUT EAR SPKR Enable Switch" value="0" />
+ <ctl name="ANC SPKR PA Enable Switch" value="0" />
+
<!-- vbat related data -->
<!-- vbat related data end -->
@@ -319,9 +542,11 @@
</path>
<path name="echo-reference">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
</path>
<path name="echo-reference headphones">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX" />
</path>
<path name="echo-reference headphones-44.1">
@@ -339,17 +564,26 @@
<ctl name="HDMI Mixer MultiMedia1" value="1" />
</path>
+ <path name="deep-buffer-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="1" />
+ </path>
+
<path name="deep-buffer-playback speaker-and-hdmi">
<path name="deep-buffer-playback hdmi" />
<path name="deep-buffer-playback" />
</path>
+ <path name="deep-buffer-playback speaker-and-display-port">
+ <path name="deep-buffer-playback display-port" />
+ <path name="deep-buffer-playback" />
+ </path>
+
<path name="deep-buffer-playback bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
</path>
<path name="deep-buffer-playback bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="deep-buffer-playback bt-sco" />
</path>
@@ -387,12 +621,16 @@
<ctl name="HDMI Mixer MultiMedia5" value="1" />
</path>
+ <path name="low-latency-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="1" />
+ </path>
+
<path name="low-latency-playback bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
</path>
<path name="low-latency-playback bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="low-latency-playback bt-sco" />
</path>
@@ -401,6 +639,11 @@
<path name="low-latency-playback" />
</path>
+ <path name="low-latency-playback speaker-and-display-port">
+ <path name="low-latency-playback display-port" />
+ <path name="low-latency-playback" />
+ </path>
+
<path name="low-latency-playback afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="1" />
</path>
@@ -424,7 +667,7 @@
</path>
<path name="audio-ull-playback">
- <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback speaker-protected">
@@ -432,7 +675,7 @@
</path>
<path name="audio-ull-playback headphones">
- <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback speaker-and-headphones">
@@ -441,15 +684,19 @@
</path>
<path name="audio-ull-playback hdmi">
- <ctl name="HDMI Mixer MultiMedia3" value="1" />
+ <ctl name="HDMI Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="1" />
</path>
<path name="audio-ull-playback bt-sco">
- <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="audio-ull-playback bt-sco" />
</path>
@@ -458,18 +705,27 @@
<path name="audio-ull-playback" />
</path>
+ <path name="audio-ull-playback speaker-and-display-port">
+ <path name="audio-ull-playback display-port" />
+ <path name="audio-ull-playback" />
+ </path>
+
<path name="audio-ull-playback afe-proxy">
- <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="audio-ull-playback usb-headphones">
- <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia3" value="1" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
</path>
<path name="multi-channel-playback hdmi">
<ctl name="HDMI Mixer MultiMedia2" value="1" />
</path>
+ <path name="multi-channel-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+ </path>
+
<path name="multi-channel-playback afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
</path>
@@ -490,12 +746,20 @@
<ctl name="HDMI Mixer MultiMedia9" value="1" />
</path>
+ <path name="compress-offload-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="silence-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="1" />
+ </path>
+
<path name="compress-offload-playback bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
</path>
<path name="compress-offload-playback bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback bt-sco" />
</path>
@@ -504,6 +768,11 @@
<path name="compress-offload-playback" />
</path>
+ <path name="compress-offload-playback speaker-and-display-port">
+ <path name="compress-offload-playback display-port" />
+ <path name="compress-offload-playback" />
+ </path>
+
<path name="compress-offload-playback afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia4" value="1" />
</path>
@@ -525,6 +794,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="1" />
</path>
+ <path name="compress-offload-playback headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
<path name="compress-offload-playback speaker-and-headphones">
<path name="compress-offload-playback headphones" />
<path name="compress-offload-playback" />
@@ -538,12 +811,16 @@
<ctl name="HDMI Mixer MultiMedia7" value="1" />
</path>
+ <path name="compress-offload-playback2 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="1" />
+ </path>
+
<path name="compress-offload-playback2 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
</path>
<path name="compress-offload-playback2 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback2 bt-sco" />
</path>
@@ -552,6 +829,11 @@
<path name="compress-offload-playback2" />
</path>
+ <path name="compress-offload-playback2 speaker-and-display-port">
+ <path name="compress-offload-playback2 display-port" />
+ <path name="compress-offload-playback2" />
+ </path>
+
<path name="compress-offload-playback2 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
</path>
@@ -573,6 +855,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="1" />
</path>
+ <path name="compress-offload-playback2 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
<path name="compress-offload-playback2 speaker-and-headphones">
<path name="compress-offload-playback2 headphones" />
<path name="compress-offload-playback2" />
@@ -586,12 +872,16 @@
<ctl name="HDMI Mixer MultiMedia10" value="1" />
</path>
+ <path name="compress-offload-playback3 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="1" />
+ </path>
+
<path name="compress-offload-playback3 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
</path>
<path name="compress-offload-playback3 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback3 bt-sco" />
</path>
@@ -600,6 +890,11 @@
<path name="compress-offload-playback3" />
</path>
+ <path name="compress-offload-playback3 speaker-and-display-port">
+ <path name="compress-offload-playback3 display-port" />
+ <path name="compress-offload-playback3" />
+ </path>
+
<path name="compress-offload-playback3 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
</path>
@@ -621,6 +916,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="1" />
</path>
+ <path name="compress-offload-playback3 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
<path name="compress-offload-playback3 speaker-and-headphones">
<path name="compress-offload-playback3 headphones" />
<path name="compress-offload-playback3" />
@@ -634,12 +933,16 @@
<ctl name="HDMI Mixer MultiMedia11" value="1" />
</path>
+ <path name="compress-offload-playback4 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="1" />
+ </path>
+
<path name="compress-offload-playback4 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
</path>
<path name="compress-offload-playback4 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback4 bt-sco" />
</path>
@@ -648,6 +951,12 @@
<path name="compress-offload-playback4" />
</path>
+ <path name="compress-offload-playback4 speaker-and-display-port">
+ <path name="compress-offload-playback4 display-port" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+
<path name="compress-offload-playback4 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="1" />
</path>
@@ -669,6 +978,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="1" />
</path>
+ <path name="compress-offload-playback4 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
<path name="compress-offload-playback4 speaker-and-headphones">
<path name="compress-offload-playback4 headphones" />
<path name="compress-offload-playback4" />
@@ -682,12 +995,16 @@
<ctl name="HDMI Mixer MultiMedia12" value="1" />
</path>
+ <path name="compress-offload-playback5 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="1" />
+ </path>
+
<path name="compress-offload-playback5 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
</path>
<path name="compress-offload-playback5 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback5 bt-sco" />
</path>
@@ -696,6 +1013,11 @@
<path name="compress-offload-playback5" />
</path>
+ <path name="compress-offload-playback5 speaker-and-display-port">
+ <path name="compress-offload-playback5 display-port" />
+ <path name="compress-offload-playback5" />
+ </path>
+
<path name="compress-offload-playback5 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia12" value="1" />
</path>
@@ -717,6 +1039,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="1" />
</path>
+ <path name="compress-offload-playback5 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
<path name="compress-offload-playback5 speaker-and-headphones">
<path name="compress-offload-playback5 headphones" />
<path name="compress-offload-playback5" />
@@ -730,12 +1056,16 @@
<ctl name="HDMI Mixer MultiMedia13" value="1" />
</path>
+ <path name="compress-offload-playback6 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="1" />
+ </path>
+
<path name="compress-offload-playback6 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
</path>
<path name="compress-offload-playback6 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback6 bt-sco" />
</path>
@@ -744,6 +1074,11 @@
<path name="compress-offload-playback6" />
</path>
+ <path name="compress-offload-playback6 speaker-and-display-port">
+ <path name="compress-offload-playback6 display-port" />
+ <path name="compress-offload-playback6" />
+ </path>
+
<path name="compress-offload-playback6 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia13" value="1" />
</path>
@@ -765,6 +1100,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="1" />
</path>
+ <path name="compress-offload-playback6 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
<path name="compress-offload-playback6 speaker-and-headphones">
<path name="compress-offload-playback6 headphones" />
<path name="compress-offload-playback6" />
@@ -778,12 +1117,16 @@
<ctl name="HDMI Mixer MultiMedia14" value="1" />
</path>
+ <path name="compress-offload-playback7 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="1" />
+ </path>
+
<path name="compress-offload-playback7 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
</path>
<path name="compress-offload-playback7 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback7 bt-sco" />
</path>
@@ -792,6 +1135,11 @@
<path name="compress-offload-playback7" />
</path>
+ <path name="compress-offload-playback7 speaker-and-display-port">
+ <path name="compress-offload-playback7 display-port" />
+ <path name="compress-offload-playback7" />
+ </path>
+
<path name="compress-offload-playback7 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia14" value="1" />
</path>
@@ -813,6 +1161,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="1" />
</path>
+ <path name="compress-offload-playback7 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
<path name="compress-offload-playback7 speaker-and-headphones">
<path name="compress-offload-playback7 headphones" />
<path name="compress-offload-playback7" />
@@ -826,12 +1178,16 @@
<ctl name="HDMI Mixer MultiMedia15" value="1" />
</path>
+ <path name="compress-offload-playback8 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="1" />
+ </path>
+
<path name="compress-offload-playback8 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
</path>
<path name="compress-offload-playback8 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback8 bt-sco" />
</path>
@@ -840,6 +1196,11 @@
<path name="compress-offload-playback8" />
</path>
+ <path name="compress-offload-playback8 speaker-and-display-port">
+ <path name="compress-offload-playback8 display-port" />
+ <path name="compress-offload-playback8" />
+ </path>
+
<path name="compress-offload-playback8 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia15" value="1" />
</path>
@@ -861,6 +1222,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="1" />
</path>
+ <path name="compress-offload-playback8 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
<path name="compress-offload-playback8 speaker-and-headphones">
<path name="compress-offload-playback8 headphones" />
<path name="compress-offload-playback8" />
@@ -874,12 +1239,16 @@
<ctl name="HDMI Mixer MultiMedia16" value="1" />
</path>
+ <path name="compress-offload-playback9 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+ </path>
+
<path name="compress-offload-playback9 bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
</path>
<path name="compress-offload-playback9 bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-offload-playback9 bt-sco" />
</path>
@@ -888,6 +1257,11 @@
<path name="compress-offload-playback9" />
</path>
+ <path name="compress-offload-playback9 speaker-and-display-port">
+ <path name="compress-offload-playback9 display-port" />
+ <path name="compress-offload-playback9" />
+ </path>
+
<path name="compress-offload-playback9 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
</path>
@@ -909,6 +1283,10 @@
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="1" />
</path>
+ <path name="compress-offload-playback9 headphones-dsd">
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
<path name="compress-offload-playback9 speaker-and-headphones">
<path name="compress-offload-playback9 headphones" />
<path name="compress-offload-playback9" />
@@ -927,7 +1305,7 @@
</path>
<path name="audio-record bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="audio-record bt-sco" />
</path>
@@ -944,7 +1322,7 @@
</path>
<path name="audio-record-compress bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="audio-record-compress bt-sco" />
</path>
@@ -953,24 +1331,24 @@
</path>
<path name="low-latency-record">
- <ctl name="MultiMedia5 Mixer SLIM_0_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer SLIM_0_TX" value="1" />
</path>
<path name="low-latency-record bt-sco">
- <ctl name="MultiMedia5 Mixer SLIM_7_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
</path>
<path name="low-latency-record bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="low-latency-record bt-sco" />
</path>
<path name="low-latency-record usb-headset-mic">
- <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
</path>
<path name="low-latency-record capture-fm">
- <ctl name="MultiMedia5 Mixer SLIM_8_TX" value="1" />
+ <ctl name="MultiMedia8 Mixer SLIM_8_TX" value="1" />
</path>
<path name="fm-virtual-record capture-fm">
@@ -1142,7 +1520,7 @@
</path>
<path name="compress-voip-call bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="compress-voip-call bt-sco" />
</path>
@@ -1182,7 +1560,7 @@
</path>
<path name="voicemmode1-call bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="voicemmode1-call bt-sco" />
</path>
@@ -1222,7 +1600,7 @@
</path>
<path name="voicemmode2-call bt-sco-wb">
- <ctl name="BT_SCO SampleRate" value="16000" />
+ <ctl name="BT SampleRate" value="KHZ_16" />
<path name="voicemmode2-call bt-sco" />
</path>
@@ -1250,34 +1628,76 @@
<!-- These are actual sound device specific mixer settings -->
<path name="amic1">
+ <ctl name="AIF1_CAP Mixer SLIM TX6" value="1"/>
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+ <ctl name="ADC MUX6" value="AMIC" />
+ <ctl name="AMIC MUX6" value="ADC1" />
+ <ctl name="IIR0 INP0 MUX" value="DEC6" />
</path>
<path name="amic2">
+ <ctl name="AIF1_CAP Mixer SLIM TX0" value="1"/>
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+ <ctl name="ADC MUX0" value="AMIC" />
+ <ctl name="AMIC MUX0" value="ADC2" />
+ <ctl name="IIR0 INP0 MUX" value="DEC0" />
</path>
<!-- For Tavil, DMIC numbered from 0 to 5 -->
<path name="dmic1">
- <ctl name="AIF1_CAP Mixer SLIM TX0" value="1" />
- <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="ADC MUX0" value="DMIC" />
- <ctl name="DMIC MUX0" value="DMIC0" />
- <ctl name="DEC0 Volume" value="84" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC0" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="dmic2">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1"/>
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="dmic3">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="dmic4">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC3" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="dmic5">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC4" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="dmic6">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC5" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="speaker">
@@ -1300,10 +1720,20 @@
<ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
</path>
- <path name="speaker-fluid">
+ <path name="speaker-mono">
+ <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="SLIM_0_RX Channels" value="One" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+ <ctl name="COMP7 Switch" value="1" />
+ <ctl name="SpkrLeft COMP Switch" value="1" />
+ <ctl name="SpkrLeft BOOST Switch" value="1" />
+ <ctl name="SpkrLeft VISENSE Switch" value="1" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
</path>
- <path name="speaker-mono">
+ <path name="speaker-fluid">
+ <path name="speaker-mono" />
</path>
<path name="speaker-liquid">
@@ -1319,33 +1749,53 @@
</path>
<path name="sidetone-iir">
- </path>
+ <ctl name="IIR0 Enable Band1" value="1" />
+ <ctl name="IIR0 Enable Band2" value="1" />
+ <ctl name="IIR0 Enable Band3" value="1" />
+ <ctl name="IIR0 Enable Band4" value="1" />
+ <ctl name="IIR0 Enable Band5" value="1" />
+ </path>
<path name="sidetone-headphones">
<path name="sidetone-iir" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="RX INT1 MIX2 INP" value="SRC0" />
+ <ctl name="RX INT2 MIX2 INP" value="SRC0" />
</path>
<path name="sidetone-handset">
<path name="sidetone-iir" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="RX INT7 MIX2 INP" value="SRC0" />
</path>
<path name="speaker-mic">
- <path name="dmic3" />
+ <path name="dmic2" />
</path>
<path name="speaker-mic-liquid">
- <path name="dmic3" />
+ <path name="dmic2" />
</path>
<path name="speaker-mic-sbc">
</path>
<path name="speaker-protected">
+ <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+ <ctl name="AIF4_VI Mixer SPKR_VI_2" value="1" />
+ <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
<path name="speaker" />
+ <ctl name="VI_FEED_TX Channels" value="Two" />
+ <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
+ <ctl name="SLIM0_RX_VI_FB_RCH_MUX" value="SLIM4_TX" />
</path>
<path name="voice-speaker-protected">
+ <ctl name="AIF4_VI Mixer SPKR_VI_1" value="1" />
+ <ctl name="SLIM_4_TX Format" value="PACKED_16B" />
<path name="speaker-mono" />
+ <ctl name="VI_FEED_TX Channels" value="One" />
+ <ctl name="SLIM0_RX_VI_FB_LCH_MUX" value="SLIM4_TX" />
</path>
<path name="vi-feedback">
@@ -1360,10 +1810,19 @@
</path>
<path name="handset">
+ <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="SLIM_0_RX Channels" value="One" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+ <ctl name="COMP7 Switch" value="1" />
+ <ctl name="SpkrLeft COMP Switch" value="1" />
+ <ctl name="SpkrLeft BOOST Switch" value="1" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ <ctl name="EAR SPKR PA Gain" value="G_6_DB" />
</path>
<path name="handset-mic">
- <path name="dmic1" />
+ <path name="dmic3" />
</path>
<path name="handset-mic-db">
@@ -1377,15 +1836,80 @@
</path>
<path name="three-mic">
+ <ctl name="AIF1_CAP Mixer SLIM TX5" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="SLIM_0_TX Channels" value="Three" />
+ <ctl name="CDC_IF TX5 MUX" value="DEC5" />
+ <ctl name="ADC MUX5" value="DMIC" />
+ <ctl name="DMIC MUX5" value="DMIC0" />
+ <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+ <ctl name="ADC MUX6" value="DMIC" />
+ <ctl name="DMIC MUX6" value="DMIC2" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC5" />
</path>
<path name="anc-handset">
+ <ctl name="ANC Function" value="ON" />
+ <ctl name="ANC Slot" value="6" />
+ <ctl name="SLIM RX0 MUX" value="AIF1_PB" />
+ <ctl name="CDC_IF RX0 MUX" value="SLIM RX0" />
+ <ctl name="SLIM_0_RX Channels" value="One" />
+ <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ <ctl name="ANC OUT EAR SPKR Enable Switch" value="1" />
+ <ctl name="ANC SPKR PA Enable Switch" value="1" />
+ <ctl name="SpkrLeft WSA PA Gain" value="G_6_DB" />
</path>
<path name="headphones">
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="Two" />
+ <ctl name="RX INT1_2 MUX" value="RX2" />
+ <ctl name="RX INT2_2 MUX" value="RX3" />
</path>
<path name="headphones-44.1">
+ <ctl name="SLIM RX4 MUX" value="AIF3_PB" />
+ <ctl name="SLIM RX5 MUX" value="AIF3_PB" />
+ <ctl name="SLIM_5_RX Channels" value="Two" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX4" />
+ <ctl name="RX INT2_1 MIX1 INP0" value="RX5" />
+ <ctl name="RX INT1_1 NATIVE MUX" value="ON" />
+ <ctl name="RX INT2_1 NATIVE MUX" value="ON" />
+ </path>
+
+ <path name="asrc-mode">
+ <ctl name="ASRC0 Output Mode" value="FRAC" />
+ <ctl name="ASRC1 Output Mode" value="FRAC" />
+ <ctl name="RX INT1_2 NATIVE MUX" value="ON" />
+ <ctl name="RX INT2_2 NATIVE MUX" value="ON" />
+ <ctl name="ASRC0 MUX" value="ASRC_IN_HPHL" />
+ <ctl name="RX INT1 SEC MIX HPHL Switch" value="1" />
+ <ctl name="ASRC1 MUX" value="ASRC_IN_HPHR" />
+ <ctl name="RX INT2 SEC MIX HPHR Switch" value="1" />
+ </path>
+
+ <path name="headphones-dsd">
+ <ctl name="SLIM RX6 MUX" value="AIF2_PB" />
+ <ctl name="SLIM RX7 MUX" value="AIF2_PB" />
+ <ctl name="SLIM_2_RX Channels" value="Two" />
+ <ctl name="DSD_L IF MUX" value="RX6" />
+ <ctl name="DSD_R IF MUX" value="RX7" />
+ <ctl name="RX INT1 MIX3 DSD HPHL Switch" value="1" />
+ <ctl name="RX INT2 MIX3 DSD HPHR Switch" value="1" />
+ <ctl name="SLIM_2_RX Format" value="DSD_DOP" />
+ </path>
+
+ <path name="hph-highquality-mode">
+ <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
+ </path>
+
+ <path name="hph-lowpower-mode">
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
</path>
<path name="true-native-mode">
@@ -1464,6 +1988,7 @@
</path>
<path name="anc-headphones">
+ <path name="headphones" />
</path>
<path name="speaker-and-anc-headphones">
@@ -1481,9 +2006,11 @@
</path>
<path name="voice-anc-headphones">
+ <path name="voice-headphones" />
</path>
<path name="voice-anc-fb-headphones">
+ <path name="voice-headphones" />
</path>
<path name="speaker-and-anc-headphones-liquid">
@@ -1493,6 +2020,9 @@
<path name="hdmi">
</path>
+ <path name="display-port">
+ </path>
+
<path name="speaker-and-usb-headphones">
<path name="speaker" />
<path name="usb-headphones" />
@@ -1503,6 +2033,11 @@
<path name="hdmi" />
</path>
+ <path name="speaker-and-display-port">
+ <path name="speaker" />
+ <path name="display-port" />
+ </path>
+
<path name="voice-rec-mic">
<path name="handset-mic" />
</path>
@@ -1534,21 +2069,59 @@
</path>
<path name="aanc-handset-mic">
+ <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX9" value="1" />
+ <ctl name="SLIM_0_TX Channels" value="Three" />
+ <ctl name="AANC_SLIM_0_RX MUX" value="SLIMBUS_0_TX" />
+ <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+ <ctl name="ADC MUX6" value="DMIC" />
+ <ctl name="DMIC MUX6" value="DMIC2" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC4" />
+ <ctl name="CDC_IF TX9 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC0" />
+ <ctl name="IIR0 INP0 MUX" value="DEC6" />
+ <ctl name="ADC MUX10" value="DMIC" />
+ <ctl name="DMIC MUX10" value="DMIC4" />
+ <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
</path>
<!-- Dual MIC devices -->
<path name="handset-dmic-endfire">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC4" />
+ <ctl name="SLIM_0_TX Channels" value="Two" />
</path>
<path name="speaker-dmic-endfire">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC5" />
+ <ctl name="SLIM_0_TX Channels" value="Two" />
</path>
<path name="dmic-endfire">
<path name="handset-dmic-endfire" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="dmic-endfire-liquid">
<path name="handset-dmic-endfire" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="handset-stereo-dmic-ef">
@@ -1604,10 +2177,20 @@
</path>
<path name="speaker-dmic-broadside">
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="SLIM_0_TX Channels" value="Two" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC2" />
</path>
<path name="dmic-broadside">
<path name="speaker-dmic-broadside" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
</path>
<path name="voice-speaker-dmic-broadside">
@@ -1616,6 +2199,23 @@
<!-- Quad MIC devices -->
<path name="speaker-qmic">
+ <ctl name="AIF1_CAP Mixer SLIM TX5" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX7" value="1" />
+ <ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
+ <ctl name="SLIM_0_TX Channels" value="Four" />
+ <ctl name="CDC_IF TX5 MUX" value="DEC5" />
+ <ctl name="ADC MUX5" value="DMIC" />
+ <ctl name="DMIC MUX5" value="DMIC1" />
+ <ctl name="CDC_IF TX6 MUX" value="DEC6" />
+ <ctl name="ADC MUX6" value="DMIC" />
+ <ctl name="DMIC MUX6" value="DMIC0" />
+ <ctl name="CDC_IF TX7 MUX" value="DEC7" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="CDC_IF TX8 MUX" value="DEC8" />
+ <ctl name="ADC MUX8" value="DMIC" />
+ <ctl name="DMIC MUX8" value="DMIC5" />
</path>
<path name="speaker-qmic-liquid">
@@ -1640,6 +2240,9 @@
<!-- TTY devices -->
<path name="tty-headphones">
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="One" />
+ <ctl name="RX INT1_2 MUX" value="RX2" />
</path>
<path name="voice-tty-full-headphones">
@@ -1682,4 +2285,122 @@
<ctl name="SLIMBUS_DL_HL Switch" value="1" />
</path>
+ <path name="bt-a2dp">
+ <ctl name="BT SampleRate" value="KHZ_48" />
+ <ctl name="AFE Input Channels" value="Two" />
+ <ctl name="SLIM7_RX ADM Channels" value="Two" />
+ </path>
+
+ <path name="speaker-and-bt-a2dp">
+ <path name="speaker" />
+ <path name="bt-a2dp" />
+ </path>
+
+ <path name="deep-buffer-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="low-latency-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="compress-offload-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="audio-ull-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia3" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-bt-a2dp">
+ <path name="deep-buffer-playback bt-a2dp" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-bt-a2dp">
+ <path name="compress-offload-playback bt-a2dp" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-bt-a2dp">
+ <path name="low-latency-playback bt-a2dp" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback2 bt-a2dp" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback3 bt-a2dp" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback4 bt-a2dp" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback5 bt-a2dp" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback6 bt-a2dp" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback7 bt-a2dp" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback8 bt-a2dp" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback9 bt-a2dp" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-bt-a2dp">
+ <path name="audio-ull-playback bt-a2dp" />
+ <path name="audio-ull-playback" />
+ </path>
</mixer>
diff --git a/configs/msmcobalt/msmcobalt.mk b/configs/msmcobalt/msmcobalt.mk
index 4ba276d..9aa0322 100644
--- a/configs/msmcobalt/msmcobalt.mk
+++ b/configs/msmcobalt/msmcobalt.mk
@@ -3,8 +3,8 @@
#AUDIO_FEATURE_FLAGS
BOARD_USES_ALSA_AUDIO := true
USE_CUSTOM_AUDIO_POLICY := 1
-USE_XML_AUDIO_POLICY_CONF := 0
-BOARD_SUPPORTS_SOUND_TRIGGER := true
+USE_XML_AUDIO_POLICY_CONF := 1
+BOARD_SUPPORTS_SOUND_TRIGGER_HAL := true
AUDIO_USE_LL_AS_PRIMARY_OUTPUT := true
AUDIO_FEATURE_ENABLED_VBAT_MONITOR := true
@@ -19,8 +19,9 @@
AUDIO_FEATURE_ENABLED_FLUENCE := true
AUDIO_FEATURE_ENABLED_HDMI_SPK := true
AUDIO_FEATURE_ENABLED_HDMI_EDID := true
-#AUDIO_FEATURE_ENABLED_HDMI_PASSTHROUGH := true
+AUDIO_FEATURE_ENABLED_HDMI_PASSTHROUGH := true
#AUDIO_FEATURE_ENABLED_KEEP_ALIVE := true
+AUDIO_FEATURE_ENABLED_DISPLAY_PORT := true
#AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
#DOLBY_DDP := true
AUDIO_FEATURE_ENABLED_HFP := true
@@ -34,7 +35,7 @@
AUDIO_FEATURE_ENABLED_ALAC_OFFLOAD := true
AUDIO_FEATURE_ENABLED_APE_OFFLOAD := true
AUDIO_FEATURE_ENABLED_AAC_ADTS_OFFLOAD := true
-#AUDIO_FEATURE_ENABLED_PROXY_DEVICE := true
+AUDIO_FEATURE_ENABLED_PROXY_DEVICE := true
AUDIO_FEATURE_ENABLED_KPI_OPTIMIZE := true
AUDIO_FEATURE_ENABLED_SPKR_PROTECTION := true
AUDIO_FEATURE_ENABLED_SSR := true
@@ -43,7 +44,7 @@
AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
BOARD_USES_SRS_TRUEMEDIA := false
DTS_CODEC_M_ := false
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
MM_AUDIO_ENABLED_FTM := true
MM_AUDIO_ENABLED_SAFX := true
TARGET_USES_QCOM_MM_AUDIO := true
@@ -51,7 +52,9 @@
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
+AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
+AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
##AUDIO_FEATURE_FLAGS
#Audio Specific device overlays
@@ -72,13 +75,18 @@
hardware/qcom/audio/configs/msmcobalt/mixer_paths.xml:system/etc/mixer_paths.xml \
hardware/qcom/audio/configs/msmcobalt/mixer_paths_tasha.xml:system/etc/mixer_paths_tasha.xml \
hardware/qcom/audio/configs/msmcobalt/mixer_paths_tavil.xml:system/etc/mixer_paths_tavil.xml \
+ hardware/qcom/audio/configs/msmcobalt/mixer_paths_skuk.xml:system/etc/mixer_paths_skuk.xml \
+ hardware/qcom/audio/configs/msmcobalt/mixer_paths_skuk.xml:system/etc/mixer_paths_qvr.xml \
hardware/qcom/audio/configs/msmcobalt/mixer_paths_dtp.xml:system/etc/mixer_paths_dtp.xml \
hardware/qcom/audio/configs/msmcobalt/mixer_paths_i2s.xml:system/etc/mixer_paths_i2s.xml \
hardware/qcom/audio/configs/msmcobalt/aanc_tuning_mixer.txt:system/etc/aanc_tuning_mixer.txt \
+ hardware/qcom/audio/configs/msmcobalt/aanc_tuning_mixer_tavil.txt:system/etc/aanc_tuning_mixer_tavil.txt \
hardware/qcom/audio/configs/msmcobalt/audio_platform_info_i2s.xml:system/etc/audio_platform_info_i2s.xml \
hardware/qcom/audio/configs/msmcobalt/sound_trigger_mixer_paths.xml:system/etc/sound_trigger_mixer_paths.xml \
hardware/qcom/audio/configs/msmcobalt/sound_trigger_mixer_paths_wcd9330.xml:system/etc/sound_trigger_mixer_paths_wcd9330.xml \
+ hardware/qcom/audio/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml:system/etc/sound_trigger_mixer_paths_wcd9340.xml \
hardware/qcom/audio/configs/msmcobalt/sound_trigger_platform_info.xml:system/etc/sound_trigger_platform_info.xml \
+ hardware/qcom/audio/configs/msmcobalt/graphite_ipc_platform_info.xml:system/etc/graphite_ipc_platform_info.xml \
hardware/qcom/audio/configs/msmcobalt/audio_platform_info.xml:system/etc/audio_platform_info.xml
#XML Audio configuration files
@@ -166,9 +174,9 @@
PRODUCT_PROPERTY_OVERRIDES += \
audio.offload.multiple.enabled=false
-#Disable Compress passthrough playback
+#Enable Compress passthrough playback
PRODUCT_PROPERTY_OVERRIDES += \
-audio.offload.passthrough=false
+audio.offload.passthrough=true
#Disable surround sound recording
PRODUCT_PROPERTY_OVERRIDES += \
@@ -186,3 +194,21 @@
PRODUCT_PROPERTY_OVERRIDES += \
audio.parser.ip.buffer.size=262144
+#flac sw decoder 24 bit decode capability
+PRODUCT_PROPERTY_OVERRIDES += \
+flac.sw.decoder.24bit.support=true
+
+#split a2dp DSP supported encoder list
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.bt.a2dp_offload_cap=sbc-aptx
+
+#enable software decoders for ALAC and APE
+PRODUCT_PROPERTY_OVERRIDES += \
+use.qti.sw.alac.decoder=true
+PRODUCT_PROPERTY_OVERRIDES += \
+use.qti.sw.ape.decoder=true
+
+#enable hw aac encoder by default
+PRODUCT_PROPERTY_OVERRIDES += \
+qcom.hw.aac.encoder=true
+
diff --git a/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
new file mode 100755
index 0000000..be77fee
--- /dev/null
+++ b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
@@ -0,0 +1,117 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!--- Copyright (c) 2014-2016, The Linux Foundation. All rights reserved. -->
+<!--- -->
+<!--- Redistribution and use in source and binary forms, with or without -->
+<!--- modification, are permitted provided that the following conditions are -->
+<!--- met: -->
+<!--- * Redistributions of source code must retain the above copyright -->
+<!--- notice, this list of conditions and the following disclaimer. -->
+<!--- * Redistributions in binary form must reproduce the above -->
+<!--- copyright notice, this list of conditions and the following -->
+<!--- disclaimer in the documentation and/or other materials provided -->
+<!--- with the distribution. -->
+<!--- * Neither the name of The Linux Foundation nor the names of its -->
+<!--- contributors may be used to endorse or promote products derived -->
+<!--- from this software without specific prior written permission. -->
+<!--- -->
+<!--- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED -->
+<!--- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF -->
+<!--- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT -->
+<!--- ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS -->
+<!--- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR -->
+<!--- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF -->
+<!--- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR -->
+<!--- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, -->
+<!--- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE -->
+<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
+<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -->
+
+<mixer>
+ <!-- These are the initial mixer settings -->
+ <ctl name="LSM1 MUX" value="None" />
+ <ctl name="LSM2 MUX" value="None" />
+ <ctl name="LSM3 MUX" value="None" />
+ <ctl name="LSM4 MUX" value="None" />
+ <ctl name="LSM5 MUX" value="None" />
+ <ctl name="LSM6 MUX" value="None" />
+ <ctl name="LSM7 MUX" value="None" />
+ <ctl name="LSM8 MUX" value="None" />
+ <ctl name="SLIMBUS_5_TX LSM Function" value="None" />
+ <ctl name="MADONOFF Switch" value="0" />
+ <ctl name="MAD Input" value="DMIC1" />
+ <ctl name="MAD_BROADCAST Switch" value="0" />
+ <ctl name="TX13 INP MUX" value="CDC_DEC_5" />
+ <ctl name="AIF4_MAD Mixer SLIM TX12" value="0" />
+ <ctl name="AIF4_MAD Mixer SLIM TX13" value="0" />
+ <ctl name="CPE AFE MAD Enable" value="0"/>
+ <ctl name="CLK MODE" value="EXTERNAL" />
+ <ctl name="EC BUF MUX INP" value="ZERO" />
+ <ctl name="ADC MUX1" value="DMIC" />
+ <ctl name="DMIC MUX1" value="ZERO" />
+
+ <path name="listen-voice-wakeup-1">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-2">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-3">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-4">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-5">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-6">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-7">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-8">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+ </path>
+
+ <path name="listen-cpe-handset-mic">
+ <ctl name="MAD Input" value="DMIC0" />
+ <ctl name="MAD_SEL MUX" value="SPE" />
+ <ctl name="MAD_INP MUX" value="MAD" />
+ <ctl name="MAD_CPE1 Switch" value="1" />
+ </path>
+
+ <path name="listen-cpe-handset-mic-ecpp">
+ <ctl name="CLK MODE" value="INTERNAL" />
+ <ctl name="EC BUF MUX INP" value="DEC1" />
+ <ctl name="ADC MUX1" value="DMIC" />
+ <ctl name="DMIC MUX1" value="DMIC0" />
+ </path>
+
+ <!-- path name used for low bandwidth FTRT codec interface -->
+ <path name="listen-cpe-handset-mic low-speed-intf">
+ <ctl name="MADONOFF Switch" value="1" />
+ <ctl name="AIF4_MAD Mixer SLIM TX12" value="1" />
+ <ctl name="MAD Input" value="DMIC0" />
+ <ctl name="CPE AFE MAD Enable" value="1"/>
+ </path>
+
+ <path name="listen-ape-handset-mic">
+ <ctl name="MAD Input" value="DMIC0" />
+ <ctl name="MAD_SEL MUX" value="MSM" />
+ <ctl name="MAD_INP MUX" value="MAD" />
+ <ctl name="MAD_BROADCAST Switch" value="1" />
+ <ctl name="CDC_IF TX13 MUX" value="MAD_BRDCST" />
+ <ctl name="AIF4_MAD Mixer SLIM TX13" value="1" />
+ </path>
+
+</mixer>
diff --git a/configs/msmcobalt/sound_trigger_platform_info.xml b/configs/msmcobalt/sound_trigger_platform_info.xml
index b92ea48..6c9f55e 100644
--- a/configs/msmcobalt/sound_trigger_platform_info.xml
+++ b/configs/msmcobalt/sound_trigger_platform_info.xml
@@ -27,25 +27,23 @@
<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -->
<sound_trigger_platform_info>
<param version="0x0101" /> <!-- this must be the first param -->
-
<common_config>
- <param execution_type="CPE" /> <!-- value: "CPE" "APE" -->
<param max_cpe_sessions="1" />
+ <param max_wdsp_sessions="2" />
<param max_ape_sessions="8" />
<param enable_failure_detection="false" />
</common_config>
-
<acdb_ids>
- <param DEVICE_HANDSET_APE_ACDB_ID="100" />
- <param DEVICE_HANDSET_CPE_ACDB_ID="128" />
- <param DEVICE_HANDSET_CPE_ECPP_ACDB_ID="128" />
+ <param DEVICE_HANDSET_MIC_APE="100" />
+ <param DEVICE_HANDSET_MIC_CPE="128" />
+ <param DEVICE_HANDSET_MIC_ECPP_CPE="128" />
</acdb_ids>
-
- <!-- Multiple sound_model_config tags can be listed, each with unique -->
- <!-- vendor_uuid. The below tag represents QTI SVA engine sound model -->
- <!-- configuration. ISV must use their own unique vendor_uuid. -->
+ <!-- Multiple sound_model_config tags can be listed, each with unique -->
+ <!-- vendor_uuid. The below tag represents QTI SVA engine sound model -->
+ <!-- configuration. ISV must use their own unique vendor_uuid. -->
<sound_model_config>
<param vendor_uuid="68ab2d40-e860-11e3-95ef-0002a5d5c51b" />
+ <param execution_type="WDSP" /> <!-- value: "WDSP" "ADSP" "DYNAMIC" -->
<param app_type="2" /> <!-- app type used in ACDB -->
<param library="libsmwrapper.so" />
<param max_cpe_phrases="6" />
@@ -53,8 +51,25 @@
<param max_ape_phrases="10" />
<param max_ape_users="10" />
<param sample_rate="16000" />
-
- <!-- Module and param ids with which the algorithm is integrated in firmware -->
+ <gcs_usecase>
+ <param uid="0x1" />
+ <!-- module_id, instance_id, param_id -->
+ <param load_sound_model_ids="0x00012C0D, 0x2, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x2, 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x2, 0x00012C29" />
+ <param read_cmd_ids="0x00020013, 0x2, 0x00020015" />
+ <param read_rsp_ids="0x00020013, 0x2, 0x00020016" />
+ </gcs_usecase>
+ <gcs_usecase>
+ <param uid="0x2" />
+ <param load_sound_model_ids="0x00012C0D, 0x3, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x3, 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x3, 0x00012C29" />
+ <param read_cmd_ids="0x00020013, 0x3, 0x00020015" />
+ <param read_rsp_ids="0x00020013, 0x3, 0x00020016" />
+ </gcs_usecase>
+ <!-- Module and param ids with which the algorithm is integrated
+ in non-graphite firmware (note these must come after gcs params) -->
<param load_sound_model_ids="0x00012C0D, 0x00012C14" />
<param unload_sound_model_ids="0x00012C0D, 0x00012C15" />
<param confidence_levels_ids="0x00012C0D, 0x00012C07" />
@@ -62,9 +77,9 @@
<!-- format: "ADPCM_packet" or "PCM_packet" !-->
<!-- transfer_mode: "FTRT" or "RT" -->
- <!-- kw_duration is in milli seconds. It is valid only for FTRT transfer mode -->
+ <!-- kw_duration is in milli seconds. It is valid only for FTRT
+ transfer mode -->
<param capture_keyword="PCM_packet, RT, 2000" />
<param client_capture_read_delay="2000" />
</sound_model_config>
-
</sound_trigger_platform_info>
diff --git a/configs/msmfalcon/audio_policy_configuration.xml b/configs/msmfalcon/audio_policy_configuration.xml
index 56848ad..b1ea1b9 100644
--- a/configs/msmfalcon/audio_policy_configuration.xml
+++ b/configs/msmfalcon/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
@@ -256,13 +256,13 @@
<!-- route declaration, i.e. list all available sources for a given sink -->
<routes>
<route type="mix" sink="Earpiece"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Speaker"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Wired Headset"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Wired Headphones"
- sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,BT SCO Headset Mic,Telephony Rx"/>
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="Line"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx"/>
<route type="mix" sink="HDMI"
@@ -281,8 +281,6 @@
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="record_24"
sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
- <route type="mix" sink="Telephony Tx"
- sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
<route type="mix" sink="voice_rx"
sources="Telephony Rx"/>
</routes>
diff --git a/configs/msmfalcon/msmfalcon.mk b/configs/msmfalcon/msmfalcon.mk
index 2e29d7a..554f32b 100644
--- a/configs/msmfalcon/msmfalcon.mk
+++ b/configs/msmfalcon/msmfalcon.mk
@@ -39,7 +39,7 @@
AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
BOARD_USES_SRS_TRUEMEDIA := false
DTS_CODEC_M_ := false
-AUDIO_FEATURE_ENABLED_DEV_ARBI := true
+AUDIO_FEATURE_ENABLED_DEV_ARBI := false
MM_AUDIO_ENABLED_FTM := true
MM_AUDIO_ENABLED_SAFX := true
TARGET_USES_QCOM_MM_AUDIO := true
diff --git a/hal/Android.mk b/hal/Android.mk
index 83787e3..daf7397 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -243,6 +243,17 @@
LOCAL_SRC_FILES += audio_extn/source_track.c
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_SPLIT_A2DP)),true)
+ LOCAL_CFLAGS += -DSPLIT_A2DP_ENABLED
+ LOCAL_SRC_FILES += audio_extn/a2dp.c
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_QAF)),true)
+ LOCAL_CFLAGS += -DQAF_EXTN_ENABLED
+ LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/qaf/
+ LOCAL_SRC_FILES += audio_extn/qaf.c
+endif
+
LOCAL_SHARED_LIBRARIES := \
liblog \
libcutils \
@@ -251,6 +262,7 @@
libaudioroute \
libdl \
libaudioutils \
+ libhardware \
libexpat
LOCAL_C_INCLUDES += \
@@ -279,6 +291,14 @@
endif
ifeq ($(strip $(BOARD_SUPPORTS_SOUND_TRIGGER)),true)
+ ST_FEATURE_ENABLE := true
+endif
+
+ifeq ($(strip $(BOARD_SUPPORTS_SOUND_TRIGGER_HAL)),true)
+ ST_FEATURE_ENABLE := true
+endif
+
+ifeq ($(ST_FEATURE_ENABLE), true)
LOCAL_CFLAGS += -DSOUND_TRIGGER_ENABLED
LOCAL_CFLAGS += -DSOUND_TRIGGER_PLATFORM_NAME=$(TARGET_BOARD_PLATFORM)
LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/sound_trigger
@@ -296,11 +316,25 @@
LOCAL_SHARED_LIBRARIES += libperipheral_client
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_DISPLAY_PORT)),true)
+ LOCAL_CFLAGS += -DDISPLAY_PORT_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_GEF_SUPPORT)),true)
+ LOCAL_CFLAGS += -DAUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED
+ LOCAL_SRC_FILES += audio_extn/gef.c
+endif
+
LOCAL_CFLAGS += -Wall -Werror
LOCAL_COPY_HEADERS_TO := mm-audio
LOCAL_COPY_HEADERS := audio_extn/audio_defs.h
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_SND_MONITOR)), true)
+ LOCAL_CFLAGS += -DSND_MONITOR_ENABLED
+ LOCAL_SRC_FILES += audio_extn/sndmonitor.c
+endif
+
LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
LOCAL_MODULE_RELATIVE_PATH := hw
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
new file mode 100644
index 0000000..e72cb76
--- /dev/null
+++ b/hal/audio_extn/a2dp.c
@@ -0,0 +1,787 @@
+/*
+* Copyright (c) 2015-16, The Linux Foundation. All rights reserved.
+*
+* Redistribution and use in source and binary forms, with or without
+* modification, are permitted provided that the following conditions are
+* met:
+* * Redistributions of source code must retain the above copyright
+* notice, this list of conditions and the following disclaimer.
+* * Redistributions in binary form must reproduce the above
+* copyright notice, this list of conditions and the following
+* disclaimer in the documentation and/or other materials provided
+* with the distribution.
+* * Neither the name of The Linux Foundation nor the names of its
+* contributors may be used to endorse or promote products derived
+* from this software without specific prior written permission.
+*
+* THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+* ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+* BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+* OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+* IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+#define LOG_TAG "split_a2dp"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+#include <errno.h>
+#include <cutils/log.h>
+#include <dlfcn.h>
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <stdlib.h>
+#include <cutils/str_parms.h>
+#include <hardware/audio.h>
+#include <hardware/hardware.h>
+#include <cutils/properties.h>
+
+#ifdef SPLIT_A2DP_ENABLED
+#define AUDIO_PARAMETER_A2DP_STARTED "A2dpStarted"
+#define BT_IPC_LIB_NAME "libbthost_if.so"
+#define ENC_MEDIA_FMT_NONE 0
+#define ENC_MEDIA_FMT_AAC 0x00010DA6
+#define ENC_MEDIA_FMT_APTX 0x000131ff
+#define ENC_MEDIA_FMT_APTX_HD 0x00013200
+#define ENC_MEDIA_FMT_SBC 0x00010BF2
+#define MEDIA_FMT_AAC_AOT_LC 2
+#define MEDIA_FMT_AAC_AOT_SBR 5
+#define MEDIA_FMT_AAC_AOT_PS 29
+#define PCM_CHANNEL_L 1
+#define PCM_CHANNEL_R 2
+#define PCM_CHANNEL_C 3
+#define MEDIA_FMT_SBC_CHANNEL_MODE_MONO 1
+#define MEDIA_FMT_SBC_CHANNEL_MODE_STEREO 2
+#define MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO 8
+#define MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO 9
+#define MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS 0
+#define MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR 1
+#define MIXER_ENC_CONFIG_BLOCK "SLIM_7_RX Encoder Config"
+#define MIXER_ENC_BIT_FORMAT "AFE Input Bit Format"
+#define MIXER_ENC_FMT_SBC "SBC"
+#define MIXER_ENC_FMT_AAC "AAC"
+#define MIXER_ENC_FMT_APTX "APTX"
+#define MIXER_ENC_FMT_APTXHD "APTXHD"
+#define MIXER_ENC_FMT_NONE "NONE"
+
+
+typedef int (*audio_stream_open_t)(void);
+typedef int (*audio_stream_close_t)(void);
+typedef int (*audio_start_stream_t)(void);
+typedef int (*audio_stop_stream_t)(void);
+typedef int (*audio_suspend_stream_t)(void);
+typedef void (*audio_handoff_triggered_t)(void);
+typedef void (*clear_a2dpsuspend_flag_t)(void);
+typedef void * (*audio_get_codec_config_t)(uint8_t *multicast_status,uint8_t *num_dev,
+ audio_format_t *codec_type);
+
+enum A2DP_STATE {
+ A2DP_STATE_CONNECTED,
+ A2DP_STATE_STARTED,
+ A2DP_STATE_STOPPED,
+ A2DP_STATE_DISCONNECTED,
+};
+
+/* structure used to update a2dp state machine
+ * to communicate IPC library
+ * to store DSP encoder configuration information
+ */
+struct a2dp_data {
+ struct audio_device *adev;
+ void *bt_lib_handle;
+ audio_stream_open_t audio_stream_open;
+ audio_stream_close_t audio_stream_close;
+ audio_start_stream_t audio_start_stream;
+ audio_stop_stream_t audio_stop_stream;
+ audio_suspend_stream_t audio_suspend_stream;
+ audio_handoff_triggered_t audio_handoff_triggered;
+ clear_a2dpsuspend_flag_t clear_a2dpsuspend_flag;
+ audio_get_codec_config_t audio_get_codec_config;
+ enum A2DP_STATE bt_state;
+ audio_format_t bt_encoder_format;
+ uint32_t enc_sampling_rate;
+ bool a2dp_started;
+ bool a2dp_suspended;
+ int a2dp_total_active_session_request;
+ bool is_a2dp_offload_supported;
+ bool is_handoff_in_progress;
+};
+
+struct a2dp_data a2dp;
+
+/* START of DSP configurable structures
+ * These values should match with DSP interface defintion
+ */
+
+/* AAC encoder configuration structure. */
+typedef struct aac_enc_cfg_t aac_enc_cfg_t;
+
+/* supported enc_mode are AAC_LC, AAC_SBR, AAC_PS
+ * supported aac_fmt_flag are ADTS/RAW
+ * supported channel_cfg are Native mode, Mono , Stereo
+ */
+struct aac_enc_cfg_t {
+ uint32_t enc_format;
+ uint32_t bit_rate;
+ uint32_t enc_mode;
+ uint16_t aac_fmt_flag;
+ uint16_t channel_cfg;
+ uint32_t sample_rate;
+} ;
+
+/* SBC encoder configuration structure. */
+typedef struct sbc_enc_cfg_t sbc_enc_cfg_t;
+
+/* supported num_subbands are 4/8
+ * supported blk_len are 4, 8, 12, 16
+ * supported channel_mode are MONO, STEREO, DUAL_MONO, JOINT_STEREO
+ * supported alloc_method are LOUNDNESS/SNR
+ * supported bit_rate for mono channel is max 320kbps
+ * supported bit rate for stereo channel is max 512 kbps
+ */
+struct sbc_enc_cfg_t{
+ uint32_t enc_format;
+ uint32_t num_subbands;
+ uint32_t blk_len;
+ uint32_t channel_mode;
+ uint32_t alloc_method;
+ uint32_t bit_rate;
+ uint32_t sample_rate;
+};
+
+
+/* supported num_channels are Mono/Stereo
+ * supported channel_mapping for mono is CHANNEL_C
+ * supported channel mapping for stereo is CHANNEL_L and CHANNEL_R
+ * custom size and reserved are not used(for future enhancement)
+ */
+struct custom_enc_cfg_aptx_t
+{
+ uint32_t enc_format;
+ uint32_t sample_rate;
+ uint16_t num_channels;
+ uint16_t reserved;
+ uint8_t channel_mapping[8];
+ uint32_t custom_size;
+};
+
+/*********** END of DSP configurable structures ********************/
+
+/* API to identify DSP encoder captabilities */
+static void a2dp_offload_codec_cap_parser(char *value)
+{
+ char *tok = NULL,*saveptr;
+
+ tok = strtok_r(value, "-", &saveptr);
+ while (tok != NULL) {
+ if (strcmp(tok, "sbc") == 0) {
+ ALOGD("%s: SBC offload supported\n",__func__);
+ a2dp.is_a2dp_offload_supported = true;
+ break;
+ } else if (strcmp(tok, "aptx") == 0) {
+ ALOGD("%s: aptx offload supported\n",__func__);
+ a2dp.is_a2dp_offload_supported = true;
+ break;
+ }
+ tok = strtok_r(NULL, "-", &saveptr);
+ };
+}
+
+static void update_offload_codec_capabilities()
+{
+ char value[PROPERTY_VALUE_MAX] = {'\0'};
+
+ property_get("persist.bt.a2dp_offload_cap", value, "false");
+ ALOGD("get_offload_codec_capabilities = %s",value);
+ a2dp.is_a2dp_offload_supported =
+ property_get_bool("persist.bt.a2dp_offload_cap", false);
+ if (strcmp(value, "false") != 0)
+ a2dp_offload_codec_cap_parser(value);
+ ALOGD("%s: codec cap = %s",__func__,value);
+}
+
+/* API to open BT IPC library to start IPC communication */
+static void open_a2dp_output()
+{
+ int ret = 0;
+
+ ALOGD(" Open A2DP output start ");
+ if (a2dp.bt_lib_handle == NULL){
+ ALOGD(" Requesting for BT lib handle");
+ a2dp.bt_lib_handle = dlopen(BT_IPC_LIB_NAME, RTLD_NOW);
+
+ if (a2dp.bt_lib_handle == NULL) {
+ ALOGE("%s: DLOPEN failed for %s", __func__, BT_IPC_LIB_NAME);
+ ret = -ENOSYS;
+ goto init_fail;
+ } else {
+ a2dp.audio_stream_open = (audio_stream_open_t)
+ dlsym(a2dp.bt_lib_handle, "audio_stream_open");
+ a2dp.audio_start_stream = (audio_start_stream_t)
+ dlsym(a2dp.bt_lib_handle, "audio_start_stream");
+ a2dp.audio_get_codec_config = (audio_get_codec_config_t)
+ dlsym(a2dp.bt_lib_handle, "audio_get_codec_config");
+ a2dp.audio_suspend_stream = (audio_suspend_stream_t)
+ dlsym(a2dp.bt_lib_handle, "audio_suspend_stream");
+ a2dp.audio_handoff_triggered = (audio_handoff_triggered_t)
+ dlsym(a2dp.bt_lib_handle, "audio_handoff_triggered");
+ a2dp.clear_a2dpsuspend_flag = (clear_a2dpsuspend_flag_t)
+ dlsym(a2dp.bt_lib_handle, "clear_a2dpsuspend_flag");
+ a2dp.audio_stop_stream = (audio_stop_stream_t)
+ dlsym(a2dp.bt_lib_handle, "audio_stop_stream");
+ a2dp.audio_stream_close = (audio_stream_close_t)
+ dlsym(a2dp.bt_lib_handle, "audio_stream_close");
+ }
+ }
+
+ if (a2dp.bt_lib_handle && a2dp.audio_stream_open) {
+ if (a2dp.bt_state == A2DP_STATE_DISCONNECTED) {
+ ALOGD("calling BT stream open");
+ ret = a2dp.audio_stream_open();
+ if(ret != 0) {
+ ALOGE("Failed to open output stream for a2dp: status %d", ret);
+ goto init_fail;
+ }
+ a2dp.bt_state = A2DP_STATE_CONNECTED;
+ } else {
+ ALOGD("Called a2dp open with improper state, Ignoring request state %d", a2dp.bt_state);
+ }
+ } else {
+ ALOGE("a2dp handle is not identified, Ignoring open request");
+ a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+ goto init_fail;
+ }
+
+init_fail:
+ if(ret != 0 && (a2dp.bt_lib_handle != NULL)) {
+ dlclose(a2dp.bt_lib_handle);
+ a2dp.bt_lib_handle = NULL;
+ }
+}
+
+static int close_a2dp_output()
+{
+ ALOGV("%s\n",__func__);
+ if (!(a2dp.bt_lib_handle && a2dp.audio_stream_close)) {
+ ALOGE("a2dp handle is not identified, Ignoring close request");
+ return -ENOSYS;
+ }
+ if ((a2dp.bt_state == A2DP_STATE_CONNECTED) &&
+ (a2dp.bt_state == A2DP_STATE_STARTED) &&
+ (a2dp.bt_state == A2DP_STATE_STOPPED)) {
+ ALOGD("calling BT stream close");
+ if(a2dp.audio_stream_close() == false)
+ ALOGE("failed close a2dp control path from BT library");
+ a2dp.a2dp_started = false;
+ a2dp.a2dp_total_active_session_request = 0;
+ a2dp.a2dp_suspended = false;
+ a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+ a2dp.enc_sampling_rate = 48000;
+ a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+ } else {
+ ALOGD("close a2dp called in improper state");
+ a2dp.a2dp_started = false;
+ a2dp.a2dp_total_active_session_request = 0;
+ a2dp.a2dp_suspended = false;
+ a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+ a2dp.enc_sampling_rate = 48000;
+ a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+ }
+
+ return 0;
+}
+
+/* API to configure SBC DSP encoder */
+bool configure_sbc_enc_format(audio_sbc_encoder_config *sbc_bt_cfg)
+{
+ struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
+ struct sbc_enc_cfg_t sbc_dsp_cfg;
+ bool is_configured = false;
+ int ret = 0;
+
+ if(sbc_bt_cfg == NULL)
+ return false;
+
+ ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+ if (!ctl_enc_data) {
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identifed");
+ is_configured = false;
+ goto fail;
+ }
+ a2dp.bt_encoder_format = AUDIO_FORMAT_SBC;
+ memset(&sbc_dsp_cfg, 0x0, sizeof(struct sbc_enc_cfg_t));
+ sbc_dsp_cfg.enc_format = ENC_MEDIA_FMT_SBC;
+ sbc_dsp_cfg.num_subbands = sbc_bt_cfg->subband;
+ sbc_dsp_cfg.blk_len = sbc_bt_cfg->blk_len;
+ switch(sbc_bt_cfg->channels) {
+ case 0:
+ sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_MONO;
+ break;
+ case 1:
+ sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO;
+ break;
+ case 3:
+ sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO;
+ break;
+ case 2:
+ default:
+ sbc_dsp_cfg.channel_mode = MEDIA_FMT_SBC_CHANNEL_MODE_STEREO;
+ break;
+ }
+ if (sbc_bt_cfg->alloc)
+ sbc_dsp_cfg.alloc_method = MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS;
+ else
+ sbc_dsp_cfg.alloc_method = MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR;
+ sbc_dsp_cfg.bit_rate = sbc_bt_cfg->bitrate;
+ sbc_dsp_cfg.sample_rate = sbc_bt_cfg->sampling_rate;
+ ret = mixer_ctl_set_array(ctl_enc_data, (void *)&sbc_dsp_cfg,
+ sizeof(struct sbc_enc_cfg_t));
+ if (ret != 0) {
+ ALOGE("%s: failed to set SBC encoder config", __func__);
+ is_configured = false;
+ goto fail;
+ }
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ENC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ ALOGE(" ERROR bit format CONFIG data mixer control not identifed");
+ is_configured = false;
+ goto fail;
+ }
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set bit format to encoder", __func__);
+ is_configured = false;
+ goto fail;
+ }
+ is_configured = true;
+ a2dp.enc_sampling_rate = sbc_bt_cfg->sampling_rate;
+ ALOGV("Successfully updated SBC enc format with samplingrate: %d channelmode:%d",
+ sbc_dsp_cfg.sample_rate, sbc_dsp_cfg.channel_mode);
+fail:
+ return is_configured;
+}
+
+/* API to configure APTX DSP encoder */
+bool configure_aptx_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
+{
+ struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
+ struct custom_enc_cfg_aptx_t aptx_dsp_cfg;
+ bool is_configured = false;
+ int ret = 0;
+
+ if(aptx_bt_cfg == NULL)
+ return false;
+
+ ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+ if (!ctl_enc_data) {
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identifed");
+ is_configured = false;
+ goto fail;
+ }
+ a2dp.bt_encoder_format = AUDIO_FORMAT_APTX;
+ memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_aptx_t));
+ aptx_dsp_cfg.enc_format = ENC_MEDIA_FMT_APTX;
+ aptx_dsp_cfg.sample_rate = aptx_bt_cfg->sampling_rate;
+ aptx_dsp_cfg.num_channels = aptx_bt_cfg->channels;
+ switch(aptx_dsp_cfg.num_channels) {
+ case 1:
+ aptx_dsp_cfg.channel_mapping[0] = PCM_CHANNEL_C;
+ break;
+ case 2:
+ default:
+ aptx_dsp_cfg.channel_mapping[0] = PCM_CHANNEL_L;
+ aptx_dsp_cfg.channel_mapping[1] = PCM_CHANNEL_R;
+ break;
+ }
+ ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
+ sizeof(struct custom_enc_cfg_aptx_t));
+ if (ret != 0) {
+ ALOGE("%s: Failed to set APTX encoder config", __func__);
+ is_configured = false;
+ goto fail;
+ }
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ENC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ ALOGE("ERROR bit format CONFIG data mixer control not identifed");
+ is_configured = false;
+ goto fail;
+ } else {
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set bit format to encoder", __func__);
+ is_configured = false;
+ goto fail;
+ }
+ }
+ is_configured = true;
+ a2dp.enc_sampling_rate = aptx_bt_cfg->sampling_rate;
+ ALOGV("Successfully updated APTX enc format with samplingrate: %d channels:%d",
+ aptx_dsp_cfg.sample_rate, aptx_dsp_cfg.num_channels);
+fail:
+ return is_configured;
+}
+
+/* API to configure APTX HD DSP encoder
+ */
+bool configure_aptx_hd_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
+{
+ struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
+ struct custom_enc_cfg_aptx_t aptx_dsp_cfg;
+ bool is_configured = false;
+ int ret = 0;
+
+ if(aptx_bt_cfg == NULL)
+ return false;
+
+ ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+ if (!ctl_enc_data) {
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identifed");
+ is_configured = false;
+ goto fail;
+ }
+
+ a2dp.bt_encoder_format = AUDIO_FORMAT_APTX_HD;
+ memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_aptx_t));
+ aptx_dsp_cfg.enc_format = ENC_MEDIA_FMT_APTX_HD;
+ aptx_dsp_cfg.sample_rate = aptx_bt_cfg->sampling_rate;
+ aptx_dsp_cfg.num_channels = aptx_bt_cfg->channels;
+ switch(aptx_dsp_cfg.num_channels) {
+ case 1:
+ aptx_dsp_cfg.channel_mapping[0] = PCM_CHANNEL_C;
+ break;
+ case 2:
+ default:
+ aptx_dsp_cfg.channel_mapping[0] = PCM_CHANNEL_L;
+ aptx_dsp_cfg.channel_mapping[1] = PCM_CHANNEL_R;
+ break;
+ }
+ ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
+ sizeof(struct custom_enc_cfg_aptx_t));
+ if (ret != 0) {
+ ALOGE("%s: Failed to set APTX HD encoder config", __func__);
+ is_configured = false;
+ goto fail;
+ }
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ ALOGE(" ERROR bit format CONFIG data mixer control not identifed");
+ is_configured = false;
+ goto fail;
+ }
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S24_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set APTX HD encoder config", __func__);
+ is_configured = false;
+ goto fail;
+ }
+ is_configured = true;
+ a2dp.enc_sampling_rate = aptx_bt_cfg->sampling_rate;
+ ALOGV("Successfully updated APTX HD encformat with samplingrate: %d channels:%d",
+ aptx_dsp_cfg.sample_rate, aptx_dsp_cfg.num_channels);
+fail:
+ return is_configured;
+}
+
+/* API to configure AAC DSP encoder */
+bool configure_aac_enc_format(audio_aac_encoder_config *aac_bt_cfg)
+{
+ struct mixer_ctl *ctl_enc_data = NULL, *ctrl_bit_format = NULL;
+ struct aac_enc_cfg_t aac_dsp_cfg;
+ bool is_configured = false;
+ int ret = 0;
+
+ if(aac_bt_cfg == NULL)
+ return false;
+
+ ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+ if (!ctl_enc_data) {
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identifed");
+ is_configured = false;
+ goto fail;
+ }
+ a2dp.bt_encoder_format = AUDIO_FORMAT_AAC;
+ memset(&aac_dsp_cfg, 0x0, sizeof(struct aac_enc_cfg_t));
+ aac_dsp_cfg.enc_format = ENC_MEDIA_FMT_AAC;
+ aac_dsp_cfg.bit_rate = aac_bt_cfg->bitrate;
+ aac_dsp_cfg.sample_rate = aac_bt_cfg->sampling_rate;
+ switch(aac_bt_cfg->enc_mode) {
+ case 0:
+ aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_LC;
+ break;
+ case 2:
+ aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_PS;
+ break;
+ case 1:
+ default:
+ aac_dsp_cfg.enc_mode = MEDIA_FMT_AAC_AOT_SBR;
+ break;
+ }
+ aac_dsp_cfg.aac_fmt_flag = aac_bt_cfg->format_flag;
+ aac_dsp_cfg.channel_cfg = aac_bt_cfg->channels;
+ ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aac_dsp_cfg,
+ sizeof(struct aac_enc_cfg_t));
+ if (ret != 0) {
+ ALOGE("%s: failed to set SBC encoder config", __func__);
+ is_configured = false;
+ goto fail;
+ }
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ENC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ is_configured = false;
+ ALOGE(" ERROR bit format CONFIG data mixer control not identifed");
+ goto fail;
+ }
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set bit format to encoder", __func__);
+ is_configured = false;
+ goto fail;
+ }
+ is_configured = true;
+ a2dp.enc_sampling_rate = aac_bt_cfg->sampling_rate;
+ ALOGV("Successfully updated AAC enc format with samplingrate: %d channels:%d",
+ aac_dsp_cfg.sample_rate, aac_dsp_cfg.channel_cfg);
+fail:
+ return is_configured;
+}
+
+bool configure_a2dp_encoder_format()
+{
+ void *codec_info = NULL;
+ uint8_t multi_cast = 0, num_dev = 1;
+ audio_format_t codec_type = AUDIO_FORMAT_INVALID;
+ bool is_configured = false;
+
+ if (!a2dp.audio_get_codec_config) {
+ ALOGE(" a2dp handle is not identified, ignoring a2dp encoder config");
+ return false;
+ }
+ ALOGD("configure_a2dp_encoder_format start");
+ codec_info = a2dp.audio_get_codec_config(&multi_cast, &num_dev,
+ &codec_type);
+
+ switch(codec_type) {
+ case AUDIO_FORMAT_SBC:
+ ALOGD(" Received SBC encoder supported BT device");
+ is_configured =
+ configure_sbc_enc_format((audio_sbc_encoder_config *)codec_info);
+ break;
+ case AUDIO_FORMAT_APTX:
+ ALOGD(" Received APTX encoder supported BT device");
+ is_configured =
+ configure_aptx_enc_format((audio_aptx_encoder_config *)codec_info);
+ break;
+ case AUDIO_FORMAT_APTX_HD:
+ ALOGD(" Received APTX HD encoder supported BT device");
+ is_configured =
+ configure_aptx_hd_enc_format((audio_aptx_encoder_config *)codec_info);
+ break;
+ case AUDIO_FORMAT_AAC:
+ ALOGD(" Received AAC encoder supported BT device");
+ is_configured =
+ configure_aac_enc_format((audio_aac_encoder_config *)codec_info);
+ break;
+ default:
+ ALOGD(" Received Unsupported encoder formar");
+ is_configured = false;
+ break;
+ }
+ return is_configured;
+}
+
+int audio_extn_a2dp_start_playback()
+{
+ int ret = 0;
+
+ ALOGD("audio_extn_a2dp_start_playback start");
+
+ if(!(a2dp.bt_lib_handle && a2dp.audio_start_stream
+ && a2dp.audio_get_codec_config)) {
+ ALOGE("a2dp handle is not identified, Ignoring start request");
+ return -ENOSYS;
+ }
+
+ if(a2dp.a2dp_suspended == true) {
+ //session will be restarted after suspend completion
+ ALOGD("a2dp start requested during suspend state");
+ return -ENOSYS;
+ }
+
+ if (!a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
+ ALOGD("calling BT module stream start");
+ /* This call indicates BT IPC lib to start playback */
+ ret = a2dp.audio_start_stream();
+ ALOGE("BT controller start return = %d",ret);
+ if (ret != 0 ) {
+ ALOGE("BT controller start failed");
+ a2dp.a2dp_started = false;
+ ret = -ETIMEDOUT;
+ } else {
+ if(configure_a2dp_encoder_format() == true) {
+ a2dp.a2dp_started = true;
+ ret = 0;
+ ALOGD("Start playback successful to BT library");
+ } else {
+ ALOGD(" unable to configure DSP encoder");
+ a2dp.a2dp_started = false;
+ ret = -ETIMEDOUT;
+ }
+ }
+ }
+
+ if (a2dp.a2dp_started)
+ a2dp.a2dp_total_active_session_request++;
+
+ ALOGD("start A2DP playback total active sessions :%d",
+ a2dp.a2dp_total_active_session_request);
+ return ret;
+}
+
+int audio_extn_a2dp_stop_playback()
+{
+ int ret =0;
+
+ ALOGV("audio_extn_a2dp_stop_playback start");
+ if(!(a2dp.bt_lib_handle && a2dp.audio_stop_stream)) {
+ ALOGE("a2dp handle is not identified, Ignoring start request");
+ return -ENOSYS;
+ }
+
+ if (a2dp.a2dp_started && (a2dp.a2dp_total_active_session_request > 0))
+ a2dp.a2dp_total_active_session_request--;
+
+ if ( a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
+ struct mixer_ctl *ctl_enc_config, *ctrl_bit_format;
+ struct sbc_enc_cfg_t dummy_reset_config;
+
+ ALOGV("calling BT module stream stop");
+ ret = a2dp.audio_stop_stream();
+ if (ret < 0)
+ ALOGE("stop stream to BT IPC lib failed");
+ else
+ ALOGV("stop steam to BT IPC lib successful");
+ memset(&dummy_reset_config, 0x0, sizeof(struct sbc_enc_cfg_t));
+ ctl_enc_config = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ENC_CONFIG_BLOCK);
+ if (!ctl_enc_config) {
+ ALOGE(" ERROR a2dp encoder format mixer control not identifed");
+ } else {
+ ret = mixer_ctl_set_array(ctl_enc_config, (void *)&dummy_reset_config,
+ sizeof(struct sbc_enc_cfg_t));
+ a2dp.bt_encoder_format = ENC_MEDIA_FMT_NONE;
+ }
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ENC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ ALOGE(" ERROR bit format CONFIG data mixer control not identifed");
+ } else {
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set bit format to encoder", __func__);
+ }
+ }
+ }
+ if(!a2dp.a2dp_total_active_session_request)
+ a2dp.a2dp_started = false;
+ ALOGD("Stop A2DP playback total active sessions :%d",
+ a2dp.a2dp_total_active_session_request);
+ return 0;
+}
+
+void audio_extn_a2dp_set_parameters(struct str_parms *parms)
+{
+ int ret, val;
+ char value[32]={0};
+
+ if(a2dp.is_a2dp_offload_supported == false) {
+ ALOGV("no supported encoders identified,ignoring a2dp setparam");
+ return;
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value,
+ sizeof(value));
+ if( ret >= 0) {
+ val = atoi(value);
+ if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ ALOGV("Received device connect request for A2DP");
+ open_a2dp_output();
+ }
+ goto param_handled;
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value,
+ sizeof(value));
+
+ if( ret >= 0) {
+ val = atoi(value);
+ if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ ALOGV("Received device dis- connect request");
+ close_a2dp_output();
+ }
+ goto param_handled;
+ }
+
+ ret = str_parms_get_str(parms, "A2dpSuspended", value, sizeof(value));
+ if (ret >= 0) {
+ if (a2dp.bt_lib_handle && (a2dp.bt_state != A2DP_STATE_DISCONNECTED) ) {
+ if ((!strncmp(value,"true",sizeof(value)))) {
+ ALOGD("Setting a2dp to suspend state");
+ a2dp.a2dp_suspended = true;
+ if(a2dp.audio_suspend_stream)
+ a2dp.audio_suspend_stream();
+ } else if (a2dp.a2dp_suspended == true) {
+ ALOGD("Resetting a2dp suspend state");
+ if(a2dp.clear_a2dpsuspend_flag)
+ a2dp.clear_a2dpsuspend_flag();
+ a2dp.a2dp_suspended = false;
+ }
+ }
+ goto param_handled;
+ }
+param_handled:
+ ALOGV("end of a2dp setparam");
+}
+
+void audio_extn_a2dp_set_handoff_mode(bool is_on)
+{
+ a2dp.is_handoff_in_progress = is_on;
+}
+
+bool audio_extn_a2dp_is_force_device_switch()
+{
+ //During encoder reconfiguration mode, force a2dp device switch
+ return a2dp.is_handoff_in_progress;
+}
+
+void audio_extn_a2dp_get_apptype_params(uint32_t *sample_rate,
+ uint32_t *bit_width)
+{
+ if(a2dp.bt_encoder_format == AUDIO_FORMAT_APTX_HD)
+ *bit_width = 24;
+ else
+ *bit_width = 16;
+ *sample_rate = a2dp.enc_sampling_rate;
+}
+void audio_extn_a2dp_init (void *adev)
+{
+ a2dp.adev = (struct audio_device*)adev;
+ a2dp.bt_lib_handle = NULL;
+ a2dp.a2dp_started = false;
+ a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+ a2dp.a2dp_total_active_session_request = 0;
+ a2dp.a2dp_suspended = false;
+ a2dp.bt_encoder_format = AUDIO_FORMAT_INVALID;
+ a2dp.enc_sampling_rate = 48000;
+ a2dp.is_a2dp_offload_supported = false;
+ a2dp.is_handoff_in_progress = false;
+ update_offload_codec_capabilities();
+}
+#endif // SPLIT_A2DP_ENABLED
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 49e649c..083b925 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -96,7 +96,7 @@
* this is done when device switch happens by setting audioparamter
*/
-#define HDMI_PLUG_STATUS_NOTIFY_ENABLE 0x30
+#define EXT_DISPLAY_PLUG_STATUS_NOTIFY_ENABLE 0x30
static ssize_t update_sysfs_node(const char *path, const char *data, size_t len)
{
@@ -121,56 +121,62 @@
return err;
}
-static int get_hdmi_sysfs_node_index()
+static int get_ext_disp_sysfs_node_index(int ext_disp_type)
{
- static int node_index = -1;
+ int node_index = -1;
char fbvalue[80] = {0};
char fbpath[80] = {0};
int i = 0;
- FILE *hdmi_fp = NULL;
+ FILE *ext_disp_fd = NULL;
- if(node_index >= 0) {
- //hdmi sysfs node will not change so we just need to get the index once.
- ALOGV("HDMI sysfs node is at fb%d", node_index);
- return node_index;
- }
-
- for(i = 0; i < 3; i++) {
+ while (1) {
snprintf(fbpath, sizeof(fbpath),
"/sys/class/graphics/fb%d/msm_fb_type", i);
- hdmi_fp = fopen(fbpath, "r");
- if(hdmi_fp) {
- fread(fbvalue, sizeof(char), 80, hdmi_fp);
- if(strncmp(fbvalue, "dtv panel", strlen("dtv panel")) == 0) {
- node_index = i;
- ALOGV("HDMI is at fb%d",i);
- fclose(hdmi_fp);
- return node_index;
+ ext_disp_fd = fopen(fbpath, "r");
+ if (ext_disp_fd) {
+ if (fread(fbvalue, sizeof(char), 80, ext_disp_fd)) {
+ if(((strncmp(fbvalue, "dtv panel", strlen("dtv panel")) == 0) &&
+ (ext_disp_type == EXT_DISPLAY_TYPE_HDMI)) ||
+ ((strncmp(fbvalue, "dp panel", strlen("dp panel")) == 0) &&
+ (ext_disp_type == EXT_DISPLAY_TYPE_DP))) {
+ node_index = i;
+ ALOGD("%s: Ext Disp:%d is at fb%d", __func__, ext_disp_type, i);
+ fclose(ext_disp_fd);
+ return node_index;
+ }
}
- fclose(hdmi_fp);
+ fclose(ext_disp_fd);
+ i++;
} else {
- ALOGE("Failed to open fb node %d",i);
+ ALOGE("%s: Scanned till end of fbs or Failed to open fb node %d", __func__, i);
+ break;
}
}
return -1;
}
-static int update_hdmi_sysfs_node(int node_value)
+static int update_ext_disp_sysfs_node(const struct audio_device *adev, int node_value)
{
- char hdmi_ack_path[80] = {0};
- char hdmi_ack_value[3] = {0};
+ char ext_disp_ack_path[80] = {0};
+ char ext_disp_ack_value[3] = {0};
int index, ret = -1;
+ int ext_disp_type = platform_get_ext_disp_type(adev->platform);
- index = get_hdmi_sysfs_node_index();
+ if (ext_disp_type < 0) {
+ ALOGE("%s, Unable to get the external display type, err:%d",
+ __func__, ext_disp_type);
+ return -EINVAL;
+ }
+ index = get_ext_disp_sysfs_node_index(ext_disp_type);
if (index >= 0) {
- snprintf(hdmi_ack_value, sizeof(hdmi_ack_value), "%d", node_value);
- snprintf(hdmi_ack_path, sizeof(hdmi_ack_path),
+ snprintf(ext_disp_ack_value, sizeof(ext_disp_ack_value), "%d", node_value);
+ snprintf(ext_disp_ack_path, sizeof(ext_disp_ack_path),
"/sys/class/graphics/fb%d/hdmi_audio_cb", index);
- ret = update_sysfs_node(hdmi_ack_path, hdmi_ack_value,
- sizeof(hdmi_ack_value));
+ ret = update_sysfs_node(ext_disp_ack_path, ext_disp_ack_value,
+ sizeof(ext_disp_ack_value));
ALOGI("update hdmi_audio_cb at fb[%d] to:[%d] %s",
index, node_value, (ret >= 0) ? "success":"fail");
@@ -179,25 +185,27 @@
return ret;
}
-static void check_and_set_hdmi_connection_status(struct str_parms *parms)
+static void check_and_set_ext_disp_connection_status(const struct audio_device *adev,
+ struct str_parms *parms)
{
char value[32] = {0};
static bool is_hdmi_sysfs_node_init = false;
if (str_parms_get_str(parms, "connect", value, sizeof(value)) >= 0
- && (atoi(value) & AUDIO_DEVICE_OUT_HDMI)) {
- //params = "connect=1024" for HDMI connection.
+ && (atoi(value) & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ //params = "connect=1024" for external display connection.
if (is_hdmi_sysfs_node_init == false) {
+ //check if this is different for dp and hdmi
is_hdmi_sysfs_node_init = true;
- update_hdmi_sysfs_node(HDMI_PLUG_STATUS_NOTIFY_ENABLE);
+ update_ext_disp_sysfs_node(adev, EXT_DISPLAY_PLUG_STATUS_NOTIFY_ENABLE);
}
- update_hdmi_sysfs_node(1);
+ update_ext_disp_sysfs_node(adev, 1);
} else if(str_parms_get_str(parms, "disconnect", value, sizeof(value)) >= 0
- && (atoi(value) & AUDIO_DEVICE_OUT_HDMI)){
- //params = "disconnect=1024" for HDMI disconnection.
- update_hdmi_sysfs_node(0);
+ && (atoi(value) & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
+ //params = "disconnect=1024" for external display disconnection.
+ update_ext_disp_sysfs_node(adev, 0);
} else {
- // handle hdmi devices only
+ // handle ext disp devices only
return;
}
}
@@ -755,6 +763,7 @@
audio_extn_ssr_set_parameters(adev, parms);
audio_extn_hfp_set_parameters(adev, parms);
audio_extn_dts_eagle_set_parameters(adev, parms);
+ audio_extn_a2dp_set_parameters(parms);
audio_extn_ddp_set_parameters(adev, parms);
audio_extn_ds2_set_parameters(adev, parms);
audio_extn_customstereo_set_parameters(adev, parms);
@@ -762,7 +771,8 @@
audio_extn_pm_set_parameters(parms);
audio_extn_source_track_set_parameters(adev, parms);
audio_extn_fbsp_set_parameters(parms);
- check_and_set_hdmi_connection_status(parms);
+ audio_extn_keep_alive_set_parameters(adev, parms);
+ check_and_set_ext_disp_connection_status(adev, parms);
if (adev->offload_effects_set_parameters != NULL)
adev->offload_effects_set_parameters(parms);
}
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index fe3fe95..cd9763e 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -155,7 +155,7 @@
#define audio_extn_usb_deinit() (0)
#define audio_extn_usb_add_device(device, card) (0)
#define audio_extn_usb_remove_device(device, card) (0)
-#define audio_extn_usb_is_config_supported(bit_width, sample_rate, ch) (0)
+#define audio_extn_usb_is_config_supported(bit_width, sample_rate, ch, pb) (0)
#define audio_extn_usb_enable_sidetone(device, enable) (0)
#define audio_extn_usb_set_sidetone_gain(parms, value, len) (0)
#else
@@ -165,14 +165,36 @@
void audio_extn_usb_remove_device(audio_devices_t device, int card);
bool audio_extn_usb_is_config_supported(unsigned int *bit_width,
unsigned int *sample_rate,
- unsigned int ch);
+ unsigned int *ch,
+ bool is_playback);
int audio_extn_usb_enable_sidetone(int device, bool enable);
int audio_extn_usb_set_sidetone_gain(struct str_parms *parms,
char *value, int len);
#endif
+#ifndef SPLIT_A2DP_ENABLED
+#define audio_extn_a2dp_init(adev) (0)
+#define audio_extn_a2dp_start_playback() (0)
+#define audio_extn_a2dp_stop_playback() (0)
+#define audio_extn_a2dp_set_parameters(parms) (0)
+#define audio_extn_a2dp_is_force_device_switch() (0)
+#define audio_extn_a2dp_set_handoff_mode(is_on) (0)
+#define audio_extn_a2dp_get_apptype_params(sample_rate,bit_width) (0)
+
+#else
+void audio_extn_a2dp_init(void *adev);
+int audio_extn_a2dp_start_playback();
+void audio_extn_a2dp_stop_playback();
+void audio_extn_a2dp_set_parameters(struct str_parms *parms);
+bool audio_extn_a2dp_is_force_device_switch();
+void audio_extn_a2dp_set_handoff_mode(bool is_on);
+void audio_extn_a2dp_get_apptype_params(uint32_t *sample_rate,
+ uint32_t *bit_width);
+
+#endif
+
#ifndef SSR_ENABLED
-#define audio_extn_ssr_check_and_set_usecase(in) (0)
+#define audio_extn_ssr_check_and_set_usecase(in) (-1)
#define audio_extn_ssr_init(in, num_out_chan) (0)
#define audio_extn_ssr_deinit() (0)
#define audio_extn_ssr_update_enabled() (0)
@@ -387,6 +409,16 @@
#endif
+#ifndef AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
+#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH 0x10000
+#endif
+
+enum {
+ EXT_DISPLAY_TYPE_NONE,
+ EXT_DISPLAY_TYPE_HDMI,
+ EXT_DISPLAY_TYPE_DP
+};
+
#ifndef HDMI_PASSTHROUGH_ENABLED
#define audio_extn_passthru_update_stream_configuration(adev, out) (0)
#define audio_extn_passthru_is_convert_supported(adev, out) (0)
@@ -405,8 +437,6 @@
#define audio_extn_passthru_set_parameters(a, p) (-ENOSYS)
#define audio_extn_passthru_init(a) do {} while(0)
#define audio_extn_passthru_should_standby(o) (1)
-
-#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH 0x1000
#else
bool audio_extn_passthru_is_convert_supported(struct audio_device *adev,
struct stream_out *out);
@@ -434,9 +464,11 @@
#ifndef HFP_ENABLED
#define audio_extn_hfp_is_active(adev) (0)
#define audio_extn_hfp_get_usecase() (-1)
+#define hfp_set_mic_mute(dev, state) (0)
#else
bool audio_extn_hfp_is_active(struct audio_device *adev);
audio_usecase_t audio_extn_hfp_get_usecase();
+int hfp_set_mic_mute(struct audio_device *dev, bool state);
#endif
#ifndef DEV_ARBI_ENABLED
@@ -556,6 +588,29 @@
void audio_utils_set_hdmi_channel_status(struct stream_out *out, char * buffer, size_t bytes);
#endif
+#ifdef QAF_EXTN_ENABLED
+bool audio_extn_qaf_is_enabled();
+void audio_extn_qaf_deinit();
+void audio_extn_qaf_close_output_stream(struct audio_hw_device *dev __unused,
+ struct audio_stream_out *stream);
+int audio_extn_qaf_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address __unused);
+int audio_extn_qaf_init(struct audio_device *adev);
+int audio_extn_qaf_set_parameters(struct audio_device *adev, struct str_parms *parms);
+#else
+#define audio_extn_qaf_is_enabled() (0)
+#define audio_extn_qaf_deinit() (0)
+#define audio_extn_qaf_close_output_stream adev_close_output_stream
+#define audio_extn_qaf_open_output_stream adev_open_output_stream
+#define audio_extn_qaf_init(adev) (0)
+#define audio_extn_qaf_set_parameters(adev, parms) (0)
+#endif
+
#ifndef KEEP_ALIVE_ENABLED
#define audio_extn_keep_alive_init(a) do {} while(0)
#define audio_extn_keep_alive_start() do {} while(0)
@@ -571,5 +626,53 @@
struct str_parms *parms);
#endif
+#ifndef AUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED
+
+#define audio_extn_gef_init(adev) (0)
+#define audio_extn_gef_deinit() (0)
+#define audio_extn_gef_notify_device_config(devices, cmask, acdb_id) (0)
+#define audio_extn_gef_send_audio_cal(dev, acdb_dev_id, acdb_device_type,\
+ app_type, topology_id, sample_rate, module_id, param_id, data, length, persist) (0)
+#define audio_extn_gef_get_audio_cal(adev, acdb_dev_id, acdb_device_type,\
+ app_type, topology_id, sample_rate, module_id, param_id, data, length, persist) (0)
+#define audio_extn_gef_store_audio_cal(adev, acdb_dev_id, acdb_device_type,\
+ app_type, topology_id, sample_rate, module_id, param_id, data, length) (0)
+#define audio_extn_gef_retrieve_audio_cal(adev, acdb_dev_id, acdb_device_type,\
+ app_type, topology_id, sample_rate, module_id, param_id, data, length) (0)
+
+#else
+
+void audio_extn_gef_init(struct audio_device *adev);
+void audio_extn_gef_deinit();
+
+void audio_extn_gef_notify_device_config(audio_devices_t audio_device,
+ audio_channel_mask_t channel_mask, int acdb_id);
+int audio_extn_gef_send_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
+ int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int length, bool persist);
+int audio_extn_gef_get_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
+ int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int* length, bool persist);
+int audio_extn_gef_store_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
+ int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int length);
+int audio_extn_gef_retrieve_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
+ int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int* length);
+
+#endif /* AUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED */
+
+typedef void (* snd_mon_cb)(void * stream, struct str_parms * parms);
+#ifndef SND_MONITOR_ENABLED
+#define audio_extn_snd_mon_init() (0)
+#define audio_extn_snd_mon_deinit() (0)
+#define audio_extn_snd_mon_register_listener(stream, cb) (0)
+#define audio_extn_snd_mon_unregister_listener(stream) (0)
+#else
+int audio_extn_snd_mon_init();
+int audio_extn_snd_mon_deinit();
+int audio_extn_snd_mon_register_listener(void *stream, snd_mon_cb cb);
+int audio_extn_snd_mon_unregister_listener(void *stream);
+#endif
#endif /* AUDIO_EXTN_H */
diff --git a/hal/audio_extn/dev_arbi.c b/hal/audio_extn/dev_arbi.c
index d7ab5ff..69d8568 100644
--- a/hal/audio_extn/dev_arbi.c
+++ b/hal/audio_extn/dev_arbi.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014, 2016 The Linux Foundation. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are
@@ -131,6 +131,7 @@
{SND_DEVICE_OUT_VOICE_HANDSET, AUDIO_DEVICE_OUT_EARPIECE},
{SND_DEVICE_OUT_SPEAKER, AUDIO_DEVICE_OUT_SPEAKER},
{SND_DEVICE_OUT_VOICE_SPEAKER, AUDIO_DEVICE_OUT_SPEAKER},
+ {SND_DEVICE_OUT_VOICE_SPEAKER_2, AUDIO_DEVICE_OUT_SPEAKER},
{SND_DEVICE_OUT_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADPHONE},
{SND_DEVICE_OUT_VOICE_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADPHONE},
{SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
diff --git a/hal/audio_extn/gef.c b/hal/audio_extn/gef.c
new file mode 100644
index 0000000..d0ccd8c
--- /dev/null
+++ b/hal/audio_extn/gef.c
@@ -0,0 +1,290 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "audio_hw_generic_effect"
+//#define LOG_NDEBUG 0
+#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <math.h>
+#include <cutils/log.h>
+#include <fcntl.h>
+#include <dirent.h>
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <sys/stat.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <math.h>
+#include <cutils/properties.h>
+#include "audio_extn.h"
+#include "audio_hw.h"
+
+#ifdef AUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED
+
+#define GEF_LIBRARY "/system/vendor/lib/libqtigef.so"
+
+typedef void* (*gef_init_t)(void*);
+typedef void (*gef_device_config_cb_t)(void*, audio_devices_t,
+ audio_channel_mask_t, int);
+
+typedef struct {
+ void* handle;
+ void* gef_ptr;
+ gef_init_t init;
+ gef_device_config_cb_t device_config_cb;
+} gef_data;
+
+static gef_data gef_hal_handle;
+
+typedef enum {
+ ASM = 0,
+ ADM
+} gef_calibration_type;
+
+typedef enum {
+ AUDIO_DEVICE_CAL_TYPE = 0,
+ AUDIO_STREAM_CAL_TYPE,
+} acdb_device_type;
+
+
+static acdb_device_type make_acdb_device_type_from_gef_cal_type
+ (gef_calibration_type gef_cal_type)
+{
+ int acdb_device_type = 0;
+
+ switch (gef_cal_type) {
+ case ASM:
+ acdb_device_type = AUDIO_STREAM_CAL_TYPE;
+ break;
+ case ADM:
+ acdb_device_type = AUDIO_DEVICE_CAL_TYPE;
+ break;
+ default:
+ acdb_device_type = -1;
+ break;
+ }
+
+ return ((int)acdb_device_type);
+}
+
+void audio_extn_gef_init(struct audio_device *adev)
+{
+ int ret = 0;
+ const char* error = NULL;
+
+ ALOGV("%s: Enter with error", __func__);
+
+ memset(&gef_hal_handle, 0, sizeof(gef_data));
+
+ ret = access(GEF_LIBRARY, R_OK);
+ if (ret == 0) {
+ //: check error for dlopen
+ gef_hal_handle.handle = dlopen(GEF_LIBRARY, RTLD_LAZY);
+ if (gef_hal_handle.handle == NULL) {
+ ALOGE("%s: DLOPEN failed for %s with error %s",
+ __func__, GEF_LIBRARY, dlerror());
+ goto ERROR_RETURN;
+ } else {
+ ALOGV("%s: DLOPEN successful for %s", __func__, GEF_LIBRARY);
+
+ //call dlerror to clear the error
+ dlerror();
+ gef_hal_handle.init =
+ (gef_init_t)dlsym(gef_hal_handle.handle, "gef_init");
+ error = dlerror();
+
+ if(error != NULL) {
+ ALOGE("%s: dlsym of %s failed with error %s",
+ __func__, "gef_init", error);
+ goto ERROR_RETURN;
+ }
+
+ //call dlerror to clear the error
+ error = dlerror();
+ gef_hal_handle.device_config_cb =
+ (gef_device_config_cb_t)dlsym(gef_hal_handle.handle,
+ "gef_device_config_cb");
+ error = dlerror();
+
+ if(error != NULL) {
+ ALOGE("%s: dlsym of %s failed with error %s",
+ __func__, "gef_device_config_cb", error);
+ goto ERROR_RETURN;
+ }
+
+ gef_hal_handle.gef_ptr = gef_hal_handle.init((void*)adev);
+ }
+ } else {
+ ALOGE("%s: %s access failed", __func__, GEF_LIBRARY);
+ }
+
+ERROR_RETURN:
+ ALOGV("%s: Exit with error %d", __func__, ret);
+ return;
+}
+
+
+//this will be called from GEF to exchange calibration using acdb
+int audio_extn_gef_send_audio_cal(void* dev, int acdb_dev_id,
+ int gef_cal_type, int app_type, int topology_id, int sample_rate,
+ uint32_t module_id, uint32_t param_id, void* data, int length, bool persist)
+{
+ int ret = 0;
+ struct audio_device *adev = (struct audio_device*)dev;
+ int acdb_device_type =
+ make_acdb_device_type_from_gef_cal_type(gef_cal_type);
+
+ ALOGV("%s: Enter", __func__);
+
+ //lock adev
+ pthread_mutex_lock(&adev->lock);
+
+ //send cal
+ ret = platform_send_audio_cal(adev->platform, acdb_dev_id,
+ acdb_device_type, app_type, topology_id, sample_rate,
+ module_id, param_id, data, length, persist);
+
+ pthread_mutex_unlock(&adev->lock);
+
+ ALOGV("%s: Exit with error %d", __func__, ret);
+
+ return ret;
+}
+
+//this will be called from GEF to exchange calibration using acdb
+int audio_extn_gef_get_audio_cal(void* dev, int acdb_dev_id,
+ int gef_cal_type, int app_type, int topology_id, int sample_rate,
+ uint32_t module_id, uint32_t param_id, void* data, int* length, bool persist)
+{
+ int ret = 0;
+ struct audio_device *adev = (struct audio_device*)dev;
+ int acdb_device_type =
+ make_acdb_device_type_from_gef_cal_type(gef_cal_type);
+
+ ALOGV("%s: Enter", __func__);
+
+ //lock adev
+ pthread_mutex_lock(&adev->lock);
+
+ ret = platform_get_audio_cal(adev->platform, acdb_dev_id,
+ acdb_device_type, app_type, topology_id, sample_rate,
+ module_id, param_id, data, length, persist);
+
+ pthread_mutex_unlock(&adev->lock);
+
+ ALOGV("%s: Exit with error %d", __func__, ret);
+
+ return ret;
+}
+
+//this will be called from GEF to store into acdb
+int audio_extn_gef_store_audio_cal(void* dev, int acdb_dev_id,
+ int gef_cal_type, int app_type, int topology_id, int sample_rate,
+ uint32_t module_id, uint32_t param_id, void* data, int length)
+{
+ int ret = 0;
+ struct audio_device *adev = (struct audio_device*)dev;
+ int acdb_device_type =
+ make_acdb_device_type_from_gef_cal_type(gef_cal_type);
+
+ ALOGV("%s: Enter", __func__);
+
+ //lock adev
+ pthread_mutex_lock(&adev->lock);
+
+ ret = platform_store_audio_cal(adev->platform, acdb_dev_id,
+ acdb_device_type, app_type, topology_id, sample_rate,
+ module_id, param_id, data, length);
+
+ pthread_mutex_unlock(&adev->lock);
+
+ ALOGV("%s: Exit with error %d", __func__, ret);
+
+ return ret;
+}
+
+//this will be called from GEF to retrieve calibration using acdb
+int audio_extn_gef_retrieve_audio_cal(void* dev, int acdb_dev_id,
+ int gef_cal_type, int app_type, int topology_id, int sample_rate,
+ uint32_t module_id, uint32_t param_id, void* data, int* length)
+{
+ int ret = 0;
+ struct audio_device *adev = (struct audio_device*)dev;
+ int acdb_device_type =
+ make_acdb_device_type_from_gef_cal_type(gef_cal_type);
+
+ ALOGV("%s: Enter", __func__);
+
+ //lock adev
+ pthread_mutex_lock(&adev->lock);
+
+ ret = platform_retrieve_audio_cal(adev->platform, acdb_dev_id,
+ acdb_device_type, app_type, topology_id, sample_rate,
+ module_id, param_id, data, length);
+
+ pthread_mutex_unlock(&adev->lock);
+
+ ALOGV("%s: Exit with error %d", __func__, ret);
+
+ return ret;
+}
+
+//this will be called from HAL to notify GEF of new device configuration
+void audio_extn_gef_notify_device_config(audio_devices_t audio_device,
+ audio_channel_mask_t channel_mask, int acdb_id)
+{
+ ALOGV("%s: Enter", __func__);
+
+ //call into GEF to share channel mask and device info
+ if (gef_hal_handle.handle && gef_hal_handle.device_config_cb) {
+ gef_hal_handle.device_config_cb(gef_hal_handle.gef_ptr, audio_device, channel_mask,
+ acdb_id);
+ }
+
+ ALOGV("%s: Exit", __func__);
+
+ return;
+}
+
+void audio_extn_gef_deinit()
+{
+ ALOGV("%s: Enter", __func__);
+
+ if (gef_hal_handle.handle) {
+ dlclose(gef_hal_handle.handle);
+ }
+
+ memset(&gef_hal_handle, 0, sizeof(gef_data));
+
+ ALOGV("%s: Exit", __func__);
+}
+
+#endif
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 5a45b80..243d48d 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -302,6 +302,26 @@
return false;
}
+int hfp_set_mic_mute(struct audio_device *adev, bool state)
+{
+ struct mixer_ctl *ctl;
+ const char *mixer_ctl_name = "HFP TX Mute";
+ uint32_t set_values[ ] = {0};
+
+ ALOGI("%s: enter, state=%d", __func__, state);
+
+ set_values[0] = state;
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+ mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+ ALOGV("%s: exit", __func__);
+ return 0;
+}
+
audio_usecase_t audio_extn_hfp_get_usecase()
{
return hfpmod.ucid;
diff --git a/hal/audio_extn/keep_alive.c b/hal/audio_extn/keep_alive.c
index 1a4f135..1df6d08 100644
--- a/hal/audio_extn/keep_alive.c
+++ b/hal/audio_extn/keep_alive.c
@@ -36,9 +36,7 @@
#include "platform_api.h"
#include <platform.h>
-#define SILENCE_MIXER_PATH "silence-playback hdmi"
-#define SILENCE_DEV_ID 32 /* index into machine driver */
-#define SILENCE_INTERVAL_US 2000000
+#define SILENCE_INTERVAL 2 /*In secs*/
typedef enum {
STATE_DEINIT = -1,
@@ -52,7 +50,9 @@
typedef struct {
pthread_mutex_t lock;
+ pthread_mutex_t sleep_lock;
pthread_cond_t cond;
+ pthread_cond_t wake_up_cond;
pthread_t thread;
state_t state;
struct listnode cmd_list;
@@ -88,6 +88,8 @@
ka.pcm = NULL;
pthread_mutex_init(&ka.lock, (const pthread_mutexattr_t *) NULL);
pthread_cond_init(&ka.cond, (const pthread_condattr_t *) NULL);
+ pthread_cond_init(&ka.wake_up_cond, (const pthread_condattr_t *) NULL);
+ pthread_mutex_init(&ka.sleep_lock, (const pthread_mutexattr_t *) NULL);
list_init(&ka.cmd_list);
if (pthread_create(&ka.thread, (const pthread_attr_t *) NULL,
keep_alive_loop, NULL) < 0) {
@@ -127,11 +129,14 @@
if (ka.pcm)
return -EEXIST;
- ALOGD("opening silence device %d", SILENCE_DEV_ID);
+ int silence_pcm_dev_id = platform_get_pcm_device_id(USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE,
+ PCM_PLAYBACK);
+
+ ALOGD("opening silence device %d", silence_pcm_dev_id);
struct audio_device * adev = (struct audio_device *)ka.userdata;
- ka.pcm = pcm_open(adev->snd_card, SILENCE_DEV_ID,
+ ka.pcm = pcm_open(adev->snd_card, silence_pcm_dev_id,
flags, &silence_config);
- ALOGD("opened silence device %d", SILENCE_DEV_ID);
+ ALOGD("opened silence device %d", silence_pcm_dev_id);
if (ka.pcm == NULL || !pcm_is_ready(ka.pcm)) {
ALOGE("%s: %s", __func__, pcm_get_error(ka.pcm));
if (ka.pcm != NULL) {
@@ -143,6 +148,27 @@
return 0;
}
+
+static int set_mixer_control(struct mixer *mixer,
+ const char * mixer_ctl_name,
+ const char *mixer_val)
+{
+ struct mixer_ctl *ctl;
+ if ((mixer == NULL) || (mixer_ctl_name == NULL) || (mixer_val == NULL)) {
+ ALOGE("%s: Invalid input", __func__);
+ return -EINVAL;
+ }
+ ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
+ ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ return mixer_ctl_set_enum_by_string(ctl, mixer_val);
+}
+
/* must be called with adev lock held */
void audio_extn_keep_alive_start()
{
@@ -151,18 +177,20 @@
int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT], len = 0, rc;
struct mixer_ctl *ctl;
int acdb_dev_id, snd_device;
+ struct listnode *node;
+ struct audio_usecase *usecase;
int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
pthread_mutex_lock(&ka.lock);
if (ka.state == STATE_DEINIT) {
ALOGE(" %s : Invalid state ",__func__);
- return;
+ goto exit;
}
if (audio_extn_passthru_is_active()) {
ALOGE(" %s : Pass through is already active", __func__);
- return;
+ goto exit;
}
if (ka.state == STATE_ACTIVE) {
@@ -170,11 +198,21 @@
goto exit;
}
+ /* Dont start keep_alive if any other PCM session is routed to HDMI*/
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == PCM_PLAYBACK &&
+ usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+ goto exit;
+ }
+
ka.done = false;
/*configure app type */
+ int silence_pcm_dev_id = platform_get_pcm_device_id(USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE,
+ PCM_PLAYBACK);
snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
- "Audio Stream %d App Type Cfg",SILENCE_DEV_ID);
+ "Audio Stream %d App Type Cfg", silence_pcm_dev_id);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
@@ -184,7 +222,28 @@
goto exit;
}
- snd_device = SND_DEVICE_OUT_HDMI;
+ /* Configure HDMI/DP Backend with default values, this as well
+ * helps reconfigure HDMI/DP backend after passthrough.
+ */
+ int ext_disp_type = platform_get_ext_disp_type(adev->platform);
+ switch(ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ snd_device = SND_DEVICE_OUT_HDMI;
+ set_mixer_control(adev->mixer, "HDMI RX Format", "LPCM");
+ set_mixer_control(adev->mixer, "HDMI_RX SampleRate", "KHZ_48");
+ set_mixer_control(adev->mixer, "HDMI_RX Channels", "Two");
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ snd_device = SND_DEVICE_OUT_DISPLAY_PORT;
+ set_mixer_control(adev->mixer, "Display Port Format", "LPCM");
+ set_mixer_control(adev->mixer, "Display Port RX SampleRate", "KHZ_48");
+ set_mixer_control(adev->mixer, "Display Port RX Channels", "Two");
+ break;
+ default:
+ ALOGE("%s: Invalid external display type:%d", __func__, ext_disp_type);
+ goto exit;
+ }
+
acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
if (acdb_dev_id < 0) {
ALOGE("%s: Couldn't get the acdb dev id", __func__);
@@ -204,15 +263,25 @@
mixer_ctl_set_array(ctl, app_type_cfg, len);
/*send calibration*/
- struct audio_usecase *usecase = calloc(1, sizeof(struct audio_usecase));
+ usecase = calloc(1, sizeof(struct audio_usecase));
usecase->type = PCM_PLAYBACK;
- usecase->out_snd_device = SND_DEVICE_OUT_HDMI;
+ usecase->out_snd_device = snd_device;
platform_send_audio_calibration(adev->platform, usecase,
platform_get_default_app_type(adev->platform), sample_rate);
/*apply audio route */
- audio_route_apply_and_update_path(adev->audio_route, SILENCE_MIXER_PATH);
+ switch(ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ audio_route_apply_and_update_path(adev->audio_route, "silence-playback hdmi");
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ audio_route_apply_and_update_path(adev->audio_route, "silence-playback display-port");
+ break;
+ default:
+ ALOGE("%s: Invalid external display type:%d", __func__, ext_disp_type);
+ goto exit;
+ }
if (open_silence_stream() == 0) {
send_cmd_l(REQUEST_WRITE);
@@ -232,18 +301,30 @@
pthread_mutex_lock(&ka.lock);
- if (ka.state == STATE_DEINIT)
- return;
-
- if (ka.state == STATE_IDLE)
+ if ((ka.state == STATE_DEINIT) || (ka.state == STATE_IDLE))
goto exit;
+ pthread_mutex_lock(&ka.sleep_lock);
ka.done = true;
+ pthread_cond_signal(&ka.wake_up_cond);
+ pthread_mutex_unlock(&ka.sleep_lock);
while (ka.state != STATE_IDLE) {
pthread_cond_wait(&ka.cond, &ka.lock);
}
close_silence_stream();
- audio_route_reset_and_update_path(adev->audio_route, SILENCE_MIXER_PATH);
+
+ /*apply audio route */
+ int ext_disp_type = platform_get_ext_disp_type(adev->platform);
+ switch(ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ audio_route_reset_and_update_path(adev->audio_route, "silence-playback hdmi");
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ audio_route_reset_and_update_path(adev->audio_route, "silence-playback display-port");
+ break;
+ default:
+ ALOGE("%s: Invalid external display type:%d", __func__, ext_disp_type);
+ }
exit:
pthread_mutex_unlock(&ka.lock);
@@ -290,6 +371,7 @@
struct listnode *item;
uint8_t * silence = NULL;
int32_t bytes = 0;
+ struct timespec ts;
while (true) {
pthread_mutex_lock(&ka.lock);
@@ -328,9 +410,17 @@
* Just something to keep the connection alive is sufficient.
* Hence a short burst of silence periodically.
*/
- usleep(SILENCE_INTERVAL_US);
- }
+ pthread_mutex_lock(&ka.sleep_lock);
+ clock_gettime(CLOCK_REALTIME, &ts);
+ ts.tv_sec += SILENCE_INTERVAL;
+ ts.tv_nsec = 0;
+ if (!ka.done)
+ pthread_cond_timedwait(&ka.wake_up_cond,
+ &ka.sleep_lock, &ts);
+
+ pthread_mutex_unlock(&ka.sleep_lock);
+ }
pthread_mutex_lock(&ka.lock);
ka.state = STATE_IDLE;
pthread_cond_signal(&ka.cond);
diff --git a/hal/audio_extn/passthru.c b/hal/audio_extn/passthru.c
index e6ac4dd..eaa8c0a 100644
--- a/hal/audio_extn/passthru.c
+++ b/hal/audio_extn/passthru.c
@@ -82,8 +82,14 @@
*/
bool audio_extn_passthru_should_drop_data(struct stream_out * out)
{
-
- if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+ /*Drop data only
+ *stream is routed to HDMI and
+ *stream has PCM format or
+ *if a compress offload (DSP decode) session
+ */
+ if ((out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) &&
+ (((out->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) ||
+ ((out->compr_config.codec != NULL) && (out->compr_config.codec->compr_passthr == LEGACY_PCM)))) {
if (android_atomic_acquire_load(&compress_passthru_active) > 0) {
ALOGI("drop data as pass thru is active");
return true;
@@ -112,9 +118,6 @@
ALOGV("inc pass thru count to notify other streams");
android_atomic_inc(&compress_passthru_active);
- ALOGV("keep_alive_stop");
- audio_extn_keep_alive_stop();
-
while (true) {
/* find max period time among active playback use cases */
list_for_each(node, &adev->usecase_list) {
diff --git a/hal/audio_extn/qaf.c b/hal/audio_extn/qaf.c
new file mode 100644
index 0000000..d631275
--- /dev/null
+++ b/hal/audio_extn/qaf.c
@@ -0,0 +1,1129 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "audio_hw_qaf"
+/*#define LOG_NDEBUG 0*/
+/*#define VERY_VERY_VERBOSE_LOGGING*/
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+#define QAF_DEFAULT_COMPR_AUDIO_HANDLE 1001
+#define QAF_DEFAULT_COMPR_PASSTHROUGH_HANDLE 1002
+
+#include <stdlib.h>
+#include <pthread.h>
+#include <errno.h>
+#include <dlfcn.h>
+#include <sys/resource.h>
+#include <sys/prctl.h>
+#include <cutils/properties.h>
+#include <cutils/str_parms.h>
+#include <cutils/log.h>
+#include <cutils/atomic.h>
+#include "audio_utils/primitives.h"
+#include "audio_hw.h"
+#include "platform_api.h"
+#include <platform.h>
+#include <system/thread_defs.h>
+#include <cutils/sched_policy.h>
+#include "audio_extn.h"
+#include <qti_audio.h>
+#include "sound/compress_params.h"
+
+#define QAF_OUTPUT_SAMPLING_RATE 48000
+#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
+#define QAF_PLAYBACK_LATENCY 30
+
+#define QAF_LATENCY (COMPRESS_OFFLOAD_PLAYBACK_LATENCY + QAF_PLAYBACK_LATENCY)
+
+#ifdef QAF_DUMP_ENABLED
+FILE *fp_output_writer_hdmi = NULL;
+#endif
+
+typedef enum {
+AUDIO_OUTPUT_FLAG_MAIN = 0x4000, // Flag for Main Input Stream
+AUDIO_OUTPUT_FLAG_ASSOCIATED = 0x8000, // Flag for Assocated Input Stream
+} qaf_audio_output_flags_t;
+
+struct qaf {
+ struct audio_device *adev;
+ audio_session_handle_t session_handle;
+ void *qaf_lib;
+ int (*qaf_audio_session_open)(audio_session_handle_t* session_handle, void *p_data, void* license_data);
+ int (*qaf_audio_session_close)(audio_session_handle_t session_handle);
+ int (*qaf_audio_stream_open)(audio_session_handle_t session_handle, audio_stream_handle_t* stream_handle,
+ audio_stream_config_t input_config, audio_devices_t devices, stream_type_t flags);
+ int (*qaf_audio_stream_close)(audio_stream_handle_t stream_handle);
+ int (*qaf_audio_stream_set_param)(audio_stream_handle_t stream_handle, const char* kv_pairs);
+ int (*qaf_audio_session_set_param)(audio_session_handle_t handle, const char* kv_pairs);
+ char* (*qaf_audio_stream_get_param)(audio_stream_handle_t stream_handle, const char* key);
+ char* (*qaf_audio_session_get_param)(audio_session_handle_t handle, const char* key);
+ int (*qaf_audio_stream_start)(audio_stream_handle_t handle);
+ int (*qaf_audio_stream_stop)(audio_stream_handle_t stream_handle);
+ int (*qaf_audio_stream_pause)(audio_stream_handle_t stream_handle);
+ int (*qaf_audio_stream_flush)(audio_stream_handle_t stream_handle);
+ int (*qaf_audio_stream_write)(audio_stream_handle_t stream_handle, const void* buf, int size);
+ void (*qaf_register_event_callback)(audio_session_handle_t session_handle, void *priv_data,
+ notify_event_callback_t event_callback, audio_event_id_t event_id);
+ pthread_mutex_t lock;
+ struct stream_out *stream_drain_main;
+ struct stream_out *qaf_compr_offload_out;
+ struct stream_out *qaf_compr_passthrough_out;
+ int passthrough_enabled;
+ int hdmi_sink_channels;
+ bool multi_ch_out_enabled;
+ bool main_output_active;
+ bool assoc_output_active;
+};
+
+static struct qaf *qaf_mod = NULL;
+
+static void lock_output_stream(struct stream_out *out)
+{
+ pthread_mutex_lock(&out->pre_lock);
+ pthread_mutex_lock(&out->lock);
+ pthread_mutex_unlock(&out->pre_lock);
+}
+
+static int qaf_send_offload_cmd_l(struct stream_out* out, int command)
+{
+ struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));
+
+ if (!cmd) {
+ ALOGE("failed to allocate mem for command 0x%x", command);
+ return -ENOMEM;
+ }
+
+ ALOGV("%s %d", __func__, command);
+
+ cmd->cmd = command;
+ list_add_tail(&out->qaf_offload_cmd_list, &cmd->node);
+ pthread_cond_signal(&out->qaf_offload_cond);
+ return 0;
+}
+
+static int audio_extn_qaf_stream_stop(struct stream_out *out)
+{
+ ALOGV("%s: %d start", __func__, __LINE__);
+ if (!qaf_mod->qaf_audio_stream_stop)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_stop(out->qaf_stream_handle);
+}
+
+static int qaf_out_standby(struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = 0;
+
+ ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
+ stream, out->usecase, use_case_table[out->usecase]);
+
+ lock_output_stream(out);
+ if (!out->standby) {
+ out->standby = true;
+ status = audio_extn_qaf_stream_stop(out);
+ }
+ pthread_mutex_unlock(&out->lock);
+ out->written = 0;
+ return status;
+}
+
+static int qaf_stream_set_param(struct stream_out *out, const char *kv_pair)
+{
+ ALOGV("%s %d kvpair: %s", __func__, __LINE__, kv_pair);
+ if (!qaf_mod->qaf_audio_stream_set_param)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_set_param(out->qaf_stream_handle, kv_pair);
+}
+
+static int qaf_out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int ret = 0;
+
+ ALOGV("%s: enter: usecase(%d: %s) kvpairs: %s",
+ __func__, out->usecase, use_case_table[out->usecase], kvpairs);
+ lock_output_stream(out);
+ ret = qaf_stream_set_param(out, kvpairs);
+ pthread_mutex_unlock(&out->lock);
+ if ((NULL != qaf_mod->qaf_compr_offload_out)) {
+ qaf_mod->qaf_compr_offload_out->stream.common.set_parameters((struct audio_stream *) qaf_mod->qaf_compr_offload_out, kvpairs);
+ }
+ return ret;
+}
+
+static int qaf_write_input_buffer(struct stream_out *out, const void *buffer, int bytes)
+{
+ int ret = 0;
+ ALOGVV("%s bytes = %d [%p]", __func__, bytes, out->qaf_stream_handle);
+ if (!qaf_mod->qaf_audio_stream_write)
+ return -EINVAL;
+
+ if (out->qaf_stream_handle)
+ ret = qaf_mod->qaf_audio_stream_write(out->qaf_stream_handle, buffer, bytes);
+ return ret;
+}
+
+static int qaf_out_set_volume(struct audio_stream_out *stream __unused, float left,
+ float right)
+{
+ if (qaf_mod->qaf_compr_offload_out != NULL) {
+ return qaf_mod->qaf_compr_offload_out->stream.set_volume(
+ (struct audio_stream_out *)qaf_mod->qaf_compr_offload_out, left, right);
+ }
+ return -ENOSYS;
+}
+
+static int qaf_stream_start(struct stream_out *out)
+{
+ if (!qaf_mod->qaf_audio_stream_start)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_start(out->qaf_stream_handle);
+}
+
+static int qaf_start_output_stream(struct stream_out *out)
+{
+ int ret = 0;
+ struct audio_device *adev = out->dev;
+ int snd_card_status = get_snd_card_state(adev);
+
+ if ((out->usecase < 0) || (out->usecase >= AUDIO_USECASE_MAX)) {
+ ret = -EINVAL;
+ usleep(50000);
+ return ret;
+ }
+
+ ALOGD("%s: enter: stream(%p)usecase(%d: %s) devices(%#x)",
+ __func__, &out->stream, out->usecase, use_case_table[out->usecase],
+ out->devices);
+
+ if (SND_CARD_STATE_OFFLINE == snd_card_status) {
+ ALOGE("%s: sound card is not active/SSR returning error", __func__);
+ ret = -EIO;
+ usleep(50000);
+ return ret;
+ }
+
+ return qaf_stream_start(out);
+}
+
+static ssize_t qaf_out_write(struct audio_stream_out *stream, const void *buffer,
+ size_t bytes)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct audio_device *adev = out->dev;
+ ssize_t ret = 0;
+
+ ALOGV("qaf_out_write bytes = %d, usecase[%d] and flags[%x] for handle[%p]",(int)bytes, out->usecase, out->flags, out);
+ lock_output_stream(out);
+
+ if (out->standby) {
+ out->standby = false;
+ pthread_mutex_lock(&adev->lock);
+ ret = qaf_start_output_stream(out);
+ pthread_mutex_unlock(&adev->lock);
+ /* ToDo: If use case is compress offload should return 0 */
+ if (ret != 0) {
+ out->standby = true;
+ goto exit;
+ }
+ }
+
+ if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
+ audio_utils_set_hdmi_channel_status(out, (char *)buffer, bytes);
+ adev->is_channel_status_set = true;
+ }
+
+ ret = qaf_write_input_buffer(out, buffer, bytes);
+ ALOGV("%s, ret [%d] ", __func__, (int)ret);
+ if (ret < 0) {
+ goto exit;
+ }
+ out->written += bytes / ((popcount(out->channel_mask) * sizeof(short)));
+
+exit:
+
+ pthread_mutex_unlock(&out->lock);
+
+ if (ret < 0) {
+ if (ret == -EAGAIN) {
+ ALOGV("No space available in ms12 driver, post msg to cb thread");
+ lock_output_stream(out);
+ ret = qaf_send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
+ pthread_mutex_unlock(&out->lock);
+ bytes = 0;
+ }
+ if(ret == -ENOMEM || ret == -EPERM){
+ if (out->pcm)
+ ALOGE("%s: error %d, %s", __func__, (int)ret, pcm_get_error(out->pcm));
+ qaf_out_standby(&out->stream.common);
+ usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
+ out->stream.common.get_sample_rate(&out->stream.common));
+ }
+ }
+ return bytes;
+}
+
+static int qaf_get_timestamp(struct stream_out *out, uint64_t *frames, struct timespec *timestamp)
+{
+ int ret = 0;
+ struct str_parms *parms;
+ int value = 0;
+ int signed_frames = 0;
+ const char* kvpairs = NULL;
+
+ ALOGV("%s out->format %d", __func__, out->format);
+ if(out->format & AUDIO_FORMAT_PCM_16_BIT) {
+ *frames = out->written;
+ signed_frames = out->written - (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+ // It would be unusual for this value to be negative, but check just in case ...
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ }
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ } else if (qaf_mod->qaf_audio_stream_get_param) {
+ kvpairs = qaf_mod->qaf_audio_stream_get_param(out->qaf_stream_handle, "position");
+ if (kvpairs) {
+ parms = str_parms_create_str(kvpairs);
+ ret = str_parms_get_int(parms, "position", &value);
+ if (ret >= 0) {
+ *frames = value;
+ signed_frames = value - (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+ // It would be unusual for this value to be negative, but check just in case ...
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ }
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ }
+ str_parms_destroy(parms);
+ }
+ } else {
+ ret = -EINVAL;
+ }
+ return ret;
+}
+
+static int qaf_out_get_presentation_position(const struct audio_stream_out *stream,
+ uint64_t *frames, struct timespec *timestamp)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int ret = -1;
+ lock_output_stream(out);
+ ret = qaf_get_timestamp(out, frames, timestamp);
+ pthread_mutex_unlock(&out->lock);
+
+ return ret;
+}
+
+static int qaf_stream_pause(struct stream_out *out)
+{
+ ALOGV("%s: %d start", __func__, __LINE__);
+ if (!qaf_mod->qaf_audio_stream_pause)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_pause(out->qaf_stream_handle);
+}
+
+static int qaf_out_pause(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = -ENOSYS;
+ ALOGE("%s", __func__);
+ lock_output_stream(out);
+ status = qaf_stream_pause(out);
+ pthread_mutex_unlock(&out->lock);
+ return status;
+}
+
+static int qaf_out_resume(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = -ENOSYS;
+ ALOGD("%s", __func__);
+ lock_output_stream(out);
+ status = qaf_stream_start(out);
+ pthread_mutex_unlock(&out->lock);
+ ALOGD("%s Exit", __func__);
+ return status;
+}
+
+static int qaf_out_drain(struct audio_stream_out* stream, audio_drain_type_t type __unused )
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = 0;
+ ALOGV("%s stream_handle = %p , format = %x", __func__, out->qaf_stream_handle, out->format);
+ lock_output_stream(out);
+ if (out->offload_callback && out->qaf_stream_handle) {
+ /* Stream stop will trigger EOS and on EOS_EVENT received
+ from callback DRAIN_READY command is sent */
+ status = audio_extn_qaf_stream_stop(out);
+ if (out->format != AUDIO_FORMAT_PCM_16_BIT)
+ qaf_mod->stream_drain_main = out;
+ }
+ pthread_mutex_unlock(&out->lock);
+ return status;
+}
+
+static int audio_extn_qaf_stream_flush(struct stream_out *out)
+{
+ ALOGV("%s: %d exit", __func__, __LINE__);
+ if (!qaf_mod->qaf_audio_stream_flush)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_flush(out->qaf_stream_handle);
+}
+
+static int qaf_out_flush(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ ALOGV("%s", __func__);
+ int status = -ENOSYS;
+ lock_output_stream(out);
+ status = audio_extn_qaf_stream_flush(out);
+ pthread_mutex_unlock(&out->lock);
+ ALOGV("%s Exit", __func__);
+ return status;
+}
+
+static uint32_t qaf_out_get_latency(const struct audio_stream_out *stream __unused)
+{
+ uint32_t latency = 0;
+
+ latency = QAF_LATENCY;
+ ALOGV("%s: Latency %d", __func__, latency);
+ return latency;
+}
+
+static void notify_event_callback(audio_session_handle_t session_handle __unused, void *prv_data, void *buf, audio_event_id_t event_id, int size, int device)
+{
+
+/*
+ For SPKR:
+ 1. Open pcm device if device_id passed to it SPKR and write the data to pcm device
+
+ For HDMI
+ 1.Open compress device for HDMI(PCM or AC3) based on current_hdmi_output_format
+ 2.create offload_callback thread to receive async events
+ 3.Write the data to compress device. If not all the data is consumed by the driver,
+ add a command to offload_callback thread.
+*/
+ int ret;
+ audio_output_flags_t flags;
+ struct qaf* qaf_module = (struct qaf* ) prv_data;
+ ALOGV("%s device 0x%X, %d in event = %d", __func__, device, __LINE__, event_id);
+
+ if (event_id == AUDIO_DATA_EVENT) {
+ ALOGVV("Device id %x %s %d, bytes to written %d", device, __func__,__LINE__, size);
+
+ pthread_mutex_lock(&qaf_module->lock);
+ if ((device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DD)) ||
+ (device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DDP))) {
+
+ if (NULL == qaf_mod->qaf_compr_passthrough_out) {
+ struct audio_config config;
+ audio_devices_t devices;
+
+ if (qaf_mod->qaf_compr_offload_out) {
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+ (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+ qaf_mod->qaf_compr_offload_out = NULL;
+ }
+
+ config.sample_rate = config.offload_info.sample_rate = QAF_OUTPUT_SAMPLING_RATE;
+ config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+ config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+
+ if (device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DDP))
+ config.format = config.offload_info.format = AUDIO_FORMAT_E_AC3;
+ else
+ config.format = config.offload_info.format = AUDIO_FORMAT_AC3;
+
+ config.offload_info.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+ flags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+ devices = AUDIO_DEVICE_OUT_AUX_DIGITAL;
+
+ ret = adev_open_output_stream((struct audio_hw_device *) qaf_mod->adev, QAF_DEFAULT_COMPR_PASSTHROUGH_HANDLE, devices,
+ flags, &config, (struct audio_stream_out **) &(qaf_mod->qaf_compr_passthrough_out), NULL);
+ if (ret < 0) {
+ ALOGE("%s: adev_open_output_stream failed with ret = %d!", __func__, ret);
+ pthread_mutex_unlock(&qaf_module->lock);
+ return;
+ }
+ }
+
+ if (!qaf_mod->passthrough_enabled)
+ qaf_mod->passthrough_enabled = 1;
+
+ ret = qaf_mod->qaf_compr_passthrough_out->stream.write((struct audio_stream_out *) qaf_mod->qaf_compr_passthrough_out, buf, size);
+ } else {
+ if (device == AUDIO_DEVICE_OUT_AUX_DIGITAL && !qaf_mod->multi_ch_out_enabled) {
+ if (qaf_mod->qaf_compr_offload_out) {
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+ (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+ qaf_mod->qaf_compr_offload_out = NULL;
+ }
+ qaf_mod->multi_ch_out_enabled = 1;
+ } else if (device == AUDIO_DEVICE_OUT_SPEAKER && qaf_mod->multi_ch_out_enabled) {
+ if (qaf_mod->qaf_compr_offload_out) {
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+ (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+ qaf_mod->qaf_compr_offload_out = NULL;
+ }
+ qaf_mod->multi_ch_out_enabled = 0;
+ }
+
+ if (NULL == qaf_mod->qaf_compr_offload_out) {
+ struct audio_config config;
+ audio_devices_t devices;
+
+ config.sample_rate = config.offload_info.sample_rate = QAF_OUTPUT_SAMPLING_RATE;
+ config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+ config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+ config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT_OFFLOAD;
+ config.offload_info.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ config.format = AUDIO_FORMAT_PCM_16_BIT_OFFLOAD;
+ devices = AUDIO_DEVICE_NONE;
+
+ if (device == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ if (qaf_mod->hdmi_sink_channels == 8) {
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_7POINT1;
+ } else if (qaf_mod->hdmi_sink_channels == 6) {
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+ } else {
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ }
+ devices = AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ } else {
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ qaf_mod->multi_ch_out_enabled = 0;
+ }
+ flags = AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+
+ /* TODO:: Need to Propagate errors to framework */
+ ret = adev_open_output_stream((struct audio_hw_device *) qaf_mod->adev, QAF_DEFAULT_COMPR_AUDIO_HANDLE, devices,
+ flags, &config, (struct audio_stream_out **) &(qaf_mod->qaf_compr_offload_out), NULL);
+ if (ret < 0) {
+ ALOGE("%s: adev_open_output_stream failed with ret = %d!", __func__, ret);
+ pthread_mutex_unlock(&qaf_module->lock);
+ return;
+ }
+ }
+
+ if (qaf_mod->passthrough_enabled) {
+ qaf_mod->passthrough_enabled = 0;
+ if (qaf_mod->qaf_compr_passthrough_out) {
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+ (struct audio_stream_out *) (qaf_mod->qaf_compr_passthrough_out));
+ qaf_mod->qaf_compr_passthrough_out = NULL;
+ }
+ }
+
+ /*
+ * TODO:: Since this is mixed data,
+ * need to identify to which stream the error should be sent
+ */
+ ret = qaf_mod->qaf_compr_offload_out->stream.write((struct audio_stream_out *) qaf_mod->qaf_compr_offload_out, buf, size);
+ }
+
+ ALOGVV("%s:%d stream write ret = %d for out handle[%p]", __func__, __LINE__, ret, qaf_mod->qaf_compr_offload_out);
+ pthread_mutex_unlock(&qaf_module->lock);
+ } else if (event_id == AUDIO_EOS_MAIN_DD_DDP_EVENT || event_id == AUDIO_EOS_MAIN_AAC_EVENT) {
+ /* TODO:: Only MAIN Stream EOS Event is added, need to add ASSOC stream EOS Event */
+ struct stream_out *out = qaf_module->stream_drain_main;
+ if (out != NULL) {
+ lock_output_stream(out);
+ out->offload_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out->offload_cookie);
+ pthread_mutex_unlock(&out->lock);
+ qaf_module->stream_drain_main = NULL;
+ ALOGV("%s %d sent DRAIN_READY", __func__, __LINE__);
+ }
+ }
+ ALOGV("%s %d", __func__, __LINE__);
+}
+
+static int qaf_session_close()
+{
+ ALOGV("%s %d", __func__, __LINE__);
+ if (qaf_mod != NULL) {
+ if (!qaf_mod->qaf_audio_session_close)
+ return -EINVAL;
+
+ qaf_mod->qaf_audio_session_close(qaf_mod->session_handle);
+ qaf_mod->session_handle = NULL;
+ pthread_mutex_destroy(&qaf_mod->lock);
+ }
+ return 0;
+}
+
+static int qaf_stream_close(struct stream_out *out)
+{
+ int ret = 0;
+ ALOGV( "%s %d", __func__, __LINE__);
+ if (!qaf_mod->qaf_audio_stream_close)
+ return -EINVAL;
+ if (out->qaf_stream_handle) {
+ ALOGV( "%s %d output active flag is %x and stream handle %p", __func__, __LINE__, out->flags, out->qaf_stream_handle);
+ if ((out->flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) && (out->flags & AUDIO_OUTPUT_FLAG_MAIN)) { /* Close for Stream with Main and Associated Content*/
+ qaf_mod->main_output_active = false;
+ qaf_mod->assoc_output_active = false;
+ } else if (out->flags & AUDIO_OUTPUT_FLAG_MAIN) {/*Close for Main Stream*/
+ qaf_mod->main_output_active = false;
+ qaf_mod->assoc_output_active = false; /* TODO to remove resetting associated stream active flag when main stream is closed*/
+ } else if (out->flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) { /*Close for Associated Stream*/
+ qaf_mod->assoc_output_active = false;
+ } else { /*Close for Local Playback*/
+ qaf_mod->main_output_active = false;
+ }
+ ret = qaf_mod->qaf_audio_stream_close(out->qaf_stream_handle);
+ out->qaf_stream_handle = NULL;
+ }
+ ALOGV( "%s %d", __func__, __LINE__);
+ return ret;
+}
+
+static int qaf_stream_open(struct stream_out *out, struct audio_config *config, audio_output_flags_t flags, audio_devices_t devices)
+{
+ int status = 0;
+ ALOGV("%s %d", __func__, __LINE__);
+
+ if (!qaf_mod->qaf_audio_stream_open)
+ return -EINVAL;
+
+ audio_stream_config_t input_config;
+ input_config.sample_rate = config->sample_rate;
+ input_config.channel_mask = config->channel_mask;
+ input_config.format = config->format;
+
+ if ((config->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) {
+ input_config.format = AUDIO_FORMAT_AAC;
+ } else if((config->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS) {
+ input_config.format = AUDIO_FORMAT_AAC_ADTS;
+ }
+
+ ALOGV("%s %d audio_stream_open sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x) format(%#x)\
+ ",__func__, __LINE__, input_config.sample_rate, input_config.channel_mask, devices, flags, input_config.format);
+
+ /* TODO to send appropriated flags when support for system tones is added */
+ if (input_config.format == AUDIO_FORMAT_PCM_16_BIT) {
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_SYSTEM_TONE);
+ } else if (input_config.format == AUDIO_FORMAT_AC3 ||
+ input_config.format == AUDIO_FORMAT_E_AC3 ||
+ input_config.format == AUDIO_FORMAT_AAC ||
+ input_config.format == AUDIO_FORMAT_AAC_ADTS) {
+ if (qaf_mod->main_output_active == false) {
+ if ((flags & AUDIO_OUTPUT_FLAG_MAIN) && (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)) {
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_MAIN);
+ if (status == 0) {
+ ALOGV("%s %d Open stream for Input with both Main and Associated stream contents with flag [%x] and stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+ qaf_mod->main_output_active = true;
+ qaf_mod->assoc_output_active = true;
+ }
+ } else if (flags & AUDIO_OUTPUT_FLAG_MAIN) {
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_MAIN);
+ if (status == 0) {
+ ALOGV("%s %d Open stream for Input with only Main flag [%x] stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+ qaf_mod->main_output_active = true;
+ }
+ } else if (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) {
+ ALOGE("%s %d Error main input is not active", __func__, __LINE__);
+ return -EINVAL;
+ } else {
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_MAIN);
+ if (status == 0) {
+ ALOGV("%s %d Open stream for Local playback with flag [%x] stream handle [%p] ", __func__, __LINE__, flags, out->qaf_stream_handle);
+ qaf_mod->main_output_active = true;
+ }
+ }
+ } else {
+ if (flags & AUDIO_OUTPUT_FLAG_MAIN) {
+ ALOGE("%s %d Error main input is already active", __func__, __LINE__);
+ return -EINVAL;
+ } else if (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) {
+ if (qaf_mod->assoc_output_active) {
+ ALOGE("%s %d Error assoc input is already active", __func__, __LINE__);
+ return -EINVAL;
+ } else {
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_ASSOCIATED);
+ if (status == 0) {
+ ALOGV("%s %d Open stream for Input with only Associated flag [%x] stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+ qaf_mod->assoc_output_active = true;
+ }
+ }
+ } else {
+ ALOGE("%s %d Error main input is already active", __func__, __LINE__);
+ return -EINVAL;
+ }
+ }
+ }
+
+ return status;
+}
+
+static int qaf_deinit()
+{
+ ALOGV("%s %d", __func__, __LINE__);
+ if (qaf_mod != NULL) {
+ if (qaf_mod->qaf_compr_offload_out != NULL)
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev, (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+ if (qaf_mod->qaf_compr_passthrough_out != NULL)
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev, (struct audio_stream_out *) (qaf_mod->qaf_compr_passthrough_out));
+
+ if (qaf_mod->qaf_lib != NULL) {
+ dlclose(qaf_mod->qaf_lib);
+ qaf_mod->qaf_lib = NULL;
+ }
+ free(qaf_mod);
+ qaf_mod = NULL;
+ }
+ return 0;
+}
+
+static void *qaf_offload_thread_loop(void *context)
+{
+ struct stream_out *out = (struct stream_out *) context;
+ struct listnode *item;
+ int ret = 0;
+ struct str_parms *parms = NULL;
+ int value = 0;
+ char* kvpairs = NULL;
+
+ setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ set_sched_policy(0, SP_FOREGROUND);
+ prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);
+
+ ALOGE("%s", __func__);
+ lock_output_stream(out);
+ for (;;) {
+ struct offload_cmd *cmd = NULL;
+ stream_callback_event_t event;
+ bool send_callback = false;
+
+ ALOGV("%s qaf_offload_cmd_list %d",
+ __func__, list_empty(&out->qaf_offload_cmd_list));
+ if (list_empty(&out->qaf_offload_cmd_list)) {
+ ALOGV("%s SLEEPING", __func__);
+ pthread_cond_wait(&out->qaf_offload_cond, &out->lock);
+ ALOGV("%s RUNNING", __func__);
+ continue;
+ }
+
+ item = list_head(&out->qaf_offload_cmd_list);
+ cmd = node_to_item(item, struct offload_cmd, node);
+ list_remove(item);
+
+ if (cmd->cmd == OFFLOAD_CMD_EXIT) {
+ free(cmd);
+ break;
+ }
+
+ pthread_mutex_unlock(&out->lock);
+ send_callback = false;
+ switch(cmd->cmd) {
+ case OFFLOAD_CMD_WAIT_FOR_BUFFER:
+ ALOGV("wait for ms12 buffer availability");
+ while (1) {
+ kvpairs = qaf_mod->qaf_audio_stream_get_param(out->qaf_stream_handle, "buf_available");
+ if (kvpairs) {
+ parms = str_parms_create_str(kvpairs);
+ ret = str_parms_get_int(parms, "buf_available", &value);
+ if (ret >= 0) {
+ if (value >= (int)out->compr_config.fragment_size) {
+ ALOGV("%s buffer available", __func__);
+ str_parms_destroy(parms);
+ parms = NULL;
+ break;
+ } else {
+ ALOGV("%s sleep", __func__);
+ str_parms_destroy(parms);
+ parms = NULL;
+ usleep(10000);
+ }
+ }
+ free(kvpairs);
+ kvpairs = NULL;
+ }
+ }
+ send_callback = true;
+ event = STREAM_CBK_EVENT_WRITE_READY;
+ break;
+ default:
+ ALOGV("%s unknown command received: %d", __func__, cmd->cmd);
+ break;
+ }
+ lock_output_stream(out);
+ if (send_callback && out->offload_callback) {
+ out->offload_callback(event, NULL, out->offload_cookie);
+ }
+ free(cmd);
+ }
+
+ while (!list_empty(&out->qaf_offload_cmd_list)) {
+ item = list_head(&out->qaf_offload_cmd_list);
+ list_remove(item);
+ free(node_to_item(item, struct offload_cmd, node));
+ }
+ pthread_mutex_unlock(&out->lock);
+
+ return NULL;
+}
+
+static int qaf_create_offload_callback_thread(struct stream_out *out)
+{
+ ALOGV("%s", __func__);
+ pthread_cond_init(&out->qaf_offload_cond, (const pthread_condattr_t *) NULL);
+ list_init(&out->qaf_offload_cmd_list);
+ pthread_create(&out->qaf_offload_thread, (const pthread_attr_t *) NULL,
+ qaf_offload_thread_loop, out);
+ return 0;
+}
+
+static int qaf_destroy_offload_callback_thread(struct stream_out *out)
+{
+ ALOGV("%s", __func__);
+ lock_output_stream(out);
+ qaf_send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
+ pthread_mutex_unlock(&out->lock);
+
+ pthread_join(out->qaf_offload_thread, (void **) NULL);
+ pthread_cond_destroy(&out->qaf_offload_cond);
+
+ return 0;
+}
+
+int audio_extn_qaf_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address __unused)
+{
+ int ret = 0;
+ struct stream_out *out;
+
+ ret = adev_open_output_stream(dev, handle, devices, flags, config, stream_out, address);
+ if (*stream_out == NULL) {
+ goto error_open;
+ }
+
+ out = (struct stream_out *) *stream_out;
+
+ /* Override function pointers based on qaf definitions */
+ out->stream.set_volume = qaf_out_set_volume;
+ out->stream.pause = qaf_out_pause;
+ out->stream.resume = qaf_out_resume;
+ out->stream.drain = qaf_out_drain;
+ out->stream.flush = qaf_out_flush;
+
+ out->stream.common.standby = qaf_out_standby;
+ out->stream.common.set_parameters = qaf_out_set_parameters;
+ out->stream.get_latency = qaf_out_get_latency;
+ out->stream.write = qaf_out_write;
+ out->stream.get_presentation_position = qaf_out_get_presentation_position;
+
+ ret = qaf_stream_open(out, config, flags, devices);
+ if (ret < 0) {
+ ALOGE("%s, Error opening QAF stream err[%d]!", __func__, ret);
+ adev_close_output_stream(dev, *stream_out);
+ goto error_open;
+ }
+
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_LOW_LATENCY) {
+ out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
+ out->config.period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE;
+ out->config.period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT;
+ out->config.start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+ out->config.avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+ }
+
+ *stream_out = &out->stream;
+ if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ qaf_create_offload_callback_thread(out);
+ }
+ ALOGV("%s: exit", __func__);
+ return 0;
+error_open:
+ *stream_out = NULL;
+ ALOGD("%s: exit: ret %d", __func__, ret);
+ return ret;
+}
+
+void audio_extn_qaf_close_output_stream(struct audio_hw_device *dev,
+ struct audio_stream_out *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+
+ ALOGV("%s: enter:stream_handle(%p) format = %x", __func__, out, out->format);
+ if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ qaf_destroy_offload_callback_thread(out);
+ }
+ qaf_mod->stream_drain_main = NULL;
+ lock_output_stream(out);
+ qaf_stream_close(out);
+ pthread_mutex_unlock(&out->lock);
+
+ adev_close_output_stream(dev, stream);
+ ALOGV("%s: exit", __func__);
+}
+
+bool audio_extn_qaf_is_enabled()
+{
+ bool prop_enabled = false;
+ char value[PROPERTY_VALUE_MAX] = {0};
+ property_get("audio.qaf.enabled", value, NULL);
+ prop_enabled = atoi(value) || !strncmp("true", value, 4);
+ return (prop_enabled);
+}
+
+int audio_extn_qaf_session_open(struct qaf *qaf_mod,
+ device_license_config_t* lic_config)
+{
+ ALOGV("%s %d", __func__, __LINE__);
+ int status = -ENOSYS;
+
+ pthread_mutex_init(&qaf_mod->lock, (const pthread_mutexattr_t *) NULL);
+
+ if (!qaf_mod->qaf_audio_session_open)
+ return -EINVAL;
+
+ status = qaf_mod->qaf_audio_session_open(&qaf_mod->session_handle,
+ (void *)(qaf_mod), (void *)lic_config);
+ if(status < 0)
+ return status;
+
+ if (qaf_mod->session_handle == NULL) {
+ ALOGE("%s %d QAF wrapper session handle is NULL", __func__, __LINE__);
+ return -ENOMEM;
+ }
+ if (qaf_mod->qaf_register_event_callback)
+ qaf_mod->qaf_register_event_callback(qaf_mod->session_handle,
+ qaf_mod, ¬ify_event_callback,
+ AUDIO_DATA_EVENT);
+ return status;
+}
+
+char* audio_extn_qaf_stream_get_param(struct stream_out *out __unused, const char *kv_pair __unused)
+{
+ return NULL;
+}
+
+int audio_extn_qaf_set_parameters(struct audio_device *adev, struct str_parms *parms)
+{
+ int status = 0, val = 0, channels = 0;
+ char *format_params, *kv_parirs;
+ struct str_parms *qaf_params;
+ char value[32];
+ bool passth_support = false;
+
+ ALOGV("%s %d ", __func__, __LINE__);
+ if (!qaf_mod || !qaf_mod->qaf_audio_session_set_param) {
+ return -EINVAL;
+ }
+
+ status = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
+ if (status >= 0) {
+ val = atoi(value);
+ if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ if (property_get_bool("audio.offload.passthrough", false) &&
+ property_get_bool("audio.qaf.reencode", false)) {
+
+ qaf_params = str_parms_create();
+ if (platform_is_edid_supported_format(adev->platform, AUDIO_FORMAT_E_AC3)) {
+ passth_support = true;
+ if (qaf_params) {
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+ AUDIO_QAF_PARAMETER_VALUE_REENCODE_EAC3);
+ }
+ } else if (platform_is_edid_supported_format(adev->platform, AUDIO_FORMAT_AC3)) {
+ passth_support = true;
+ if (qaf_params) {
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+ AUDIO_QAF_PARAMETER_VALUE_REENCODE_AC3);
+ }
+ }
+
+ if (passth_support) {
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+ format_params = str_parms_to_str(qaf_params);
+
+ qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+ }
+ str_parms_destroy(qaf_params);
+ }
+
+ if (!passth_support) {
+ channels = platform_edid_get_max_channels(adev->platform);
+
+ qaf_params = str_parms_create();
+ switch (channels) {
+ case 8:
+ ALOGV("%s: Switching Qaf output to 7.1 channels", __func__);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_CHANNELS,
+ AUDIO_QAF_PARAMETER_VALUE_8_CHANNELS);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+ qaf_mod->hdmi_sink_channels = channels;
+ break;
+ case 6:
+ ALOGV("%s: Switching Qaf output to 5.1 channels", __func__);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_CHANNELS,
+ AUDIO_QAF_PARAMETER_VALUE_6_CHANNELS);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+ qaf_mod->hdmi_sink_channels = channels;
+ break;
+ default:
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_SPEAKER);
+ qaf_mod->hdmi_sink_channels = 2;
+ break;
+ }
+
+ format_params = str_parms_to_str(qaf_params);
+ qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+ str_parms_destroy(qaf_params);
+ }
+ }
+ }
+
+ status = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value));
+ if (status >= 0) {
+ val = atoi(value);
+ if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ qaf_params = str_parms_create();
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_SPEAKER);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+ AUDIO_QAF_PARAMETER_VALUE_PCM);
+ qaf_mod->hdmi_sink_channels = 0;
+
+ format_params = str_parms_to_str(qaf_params);
+ qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+ str_parms_destroy(qaf_params);
+ }
+ }
+
+ kv_parirs = str_parms_to_str(parms);
+ qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, kv_parirs);
+
+ return status;
+}
+
+char* audio_extn_qaf_get_param(struct audio_device *adev __unused, const char *kv_pair __unused)
+{
+ return 0;
+}
+
+int audio_extn_qaf_init(struct audio_device *adev)
+{
+ char value[PROPERTY_VALUE_MAX] = {0};
+ char lib_name[PROPERTY_VALUE_MAX] = {0};
+ unsigned char* license_data = NULL;
+ device_license_config_t* lic_config = NULL;
+ ALOGV("%s %d", __func__, __LINE__);
+ int ret = 0, size = 0;
+
+ qaf_mod = malloc(sizeof(struct qaf));
+ if(qaf_mod == NULL) {
+ ALOGE("%s, out of memory", __func__);
+ ret = -ENOMEM;
+ goto done;
+ }
+ memset(qaf_mod, 0, sizeof(struct qaf));
+ lic_config = (device_license_config_t*) calloc(1, sizeof(device_license_config_t));
+ if(lic_config == NULL) {
+ ALOGE("%s, out of memory", __func__);
+ ret = -ENOMEM;
+ goto done;
+ }
+ qaf_mod->adev = adev;
+ property_get("audio.qaf.library", value, NULL);
+ snprintf(lib_name, PROPERTY_VALUE_MAX, "%s", value);
+
+ license_data = platform_get_license((struct audio_hw_device *)(qaf_mod->adev->platform), &size);
+ if (!license_data) {
+ ALOGE("License is not present");
+ ret = -EINVAL;
+ goto done;
+ }
+ lic_config->p_license = (unsigned char* ) calloc(1, size);
+ if(lic_config->p_license == NULL) {
+ ALOGE("%s, out of memory", __func__);
+ ret = -ENOMEM;
+ goto done;
+ }
+ lic_config->l_size = size;
+ memcpy(lic_config->p_license, license_data, size);
+
+ if (property_get("audio.qaf.manufacturer", value, "") && atoi(value)) {
+ lic_config->manufacturer_id = (unsigned long) atoi (value);
+ } else {
+ ALOGE("audio.qaf.manufacturer id is not set");
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ret = audio_extn_qaf_session_open(qaf_mod, lic_config);
+done:
+ if (license_data != NULL) {
+ free(license_data);
+ license_data = NULL;
+ }
+ if (lic_config->p_license != NULL) {
+ free(lic_config->p_license);
+ lic_config->p_license = NULL;
+ }
+ if (lic_config != NULL) {
+ free(lic_config);
+ lic_config = NULL;
+ }
+ if (ret != 0) {
+ if (qaf_mod != NULL) {
+ free(qaf_mod);
+ qaf_mod = NULL;
+ }
+ }
+ return ret;
+}
+
+void audio_extn_qaf_deinit()
+{
+ qaf_session_close();
+ qaf_deinit();
+}
diff --git a/hal/audio_extn/sndmonitor.c b/hal/audio_extn/sndmonitor.c
new file mode 100644
index 0000000..eecc448
--- /dev/null
+++ b/hal/audio_extn/sndmonitor.c
@@ -0,0 +1,684 @@
+/*
+* Copyright (c) 2016, The Linux Foundation. All rights reserved.
+*
+* Redistribution and use in source and binary forms, with or without
+* modification, are permitted provided that the following conditions are
+* met:
+* * Redistributions of source code must retain the above copyright
+* notice, this list of conditions and the following disclaimer.
+* * Redistributions in binary form must reproduce the above
+* copyright notice, this list of conditions and the following
+* disclaimer in the documentation and/or other materials provided
+* with the distribution.
+* * Neither the name of The Linux Foundation nor the names of its
+* contributors may be used to endorse or promote products derived
+* from this software without specific prior written permission.
+*
+* THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+* ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+* BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+* OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+* IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#define LOG_TAG "audio_hw_sndmonitor"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+/* monitor sound card, cpe state
+
+ audio_dev registers for a callback from this module in adev_open
+ Each stream in audio_hal registers for a callback in
+ adev_open_*_stream.
+
+ A thread is spawned to poll() on sound card state files in /proc.
+ On observing a sound card state change, this thread invokes the
+ callbacks registered.
+
+ Callbacks are deregistered in adev_close_*_stream and adev_close
+*/
+#include <stdlib.h>
+#include <dirent.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/stat.h>
+#include <sys/poll.h>
+#include <cutils/list.h>
+#include <cutils/hashmap.h>
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include <ctype.h>
+
+#include "audio_hw.h"
+#include "audio_extn.h"
+
+//#define MONITOR_DEVICE_EVENTS
+#define CPE_MAGIC_NUM 0x2000
+#define MAX_CPE_SLEEP_RETRY 2
+#define CPE_SLEEP_WAIT 100
+
+#define MAX_SLEEP_RETRY 100
+#define AUDIO_INIT_SLEEP_WAIT 100 /* 100 ms */
+
+#define AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE "ext_audio_device"
+#define INIT_MAP_SIZE 5
+
+typedef enum {
+ audio_event_on,
+ audio_event_off
+} audio_event_status;
+
+typedef struct {
+ int card;
+ int fd;
+ struct listnode node; // membership in sndcards list
+ card_status_t status;
+} sndcard_t;
+
+typedef struct {
+ char *dev;
+ int fd;
+ int status;
+ struct listnode node; // membership in deviceevents list;
+} dev_event_t;
+
+typedef void (*notifyfn)(const void *target, const char *msg);
+
+typedef struct {
+ const void *target;
+ notifyfn notify;
+ struct listnode cards;
+ unsigned int num_cards;
+ struct listnode dev_events;
+ unsigned int num_dev_events;
+ pthread_t monitor_thread;
+ int intpipe[2];
+ Hashmap *listeners; // from stream * -> callback func
+ bool initcheck;
+} sndmonitor_state_t;
+
+static sndmonitor_state_t sndmonitor;
+
+static char *read_state(int fd)
+{
+ struct stat buf;
+ if (fstat(fd, &buf) < 0)
+ return NULL;
+
+ off_t pos = lseek(fd, 0, SEEK_CUR);
+ off_t avail = buf.st_size - pos;
+ if (avail <= 0) {
+ ALOGE("avail %ld", avail);
+ return NULL;
+ }
+
+ char *state = (char *)calloc(avail+1, sizeof(char));
+ if (!state)
+ return NULL;
+
+ ssize_t bytes = read(fd, state, avail);
+ if (bytes <= 0)
+ return NULL;
+
+ // trim trailing whitespace
+ while (bytes && isspace(*(state+bytes-1))) {
+ *(state + bytes - 1) = '\0';
+ --bytes;
+ }
+ lseek(fd, 0, SEEK_SET);
+ return state;
+}
+
+static int add_new_sndcard(int card, int fd)
+{
+ sndcard_t *s = (sndcard_t *)calloc(sizeof(sndcard_t), 1);
+
+ if (!s)
+ return -1;
+
+ s->card = card;
+ s->fd = fd; // dup?
+
+ char *state = read_state(fd);
+
+ if (!state)
+ return -1;
+
+ bool online = state && !strcmp(state, "ONLINE");
+
+ ALOGV("card %d initial state %s %d", card, state, online);
+
+ if (state)
+ free(state);
+
+ s->status = online ? CARD_STATUS_ONLINE : CARD_STATUS_OFFLINE;
+ list_add_tail(&sndmonitor.cards, &s->node);
+ return 0;
+}
+
+static int validate_snd_card(const char *id)
+{
+ return !strncasecmp(id, "msm", 3) ? 0 : -1;
+}
+
+static int enum_sndcards()
+{
+ const char *cards = "/proc/asound/cards";
+ int tries = 10;
+ char *line = NULL;
+ size_t len = 0;
+ ssize_t bytes_read = -1;
+ char path[128] = {0};
+ char *ptr = NULL, *saveptr = NULL, *card_id = NULL;
+ int line_no=0;
+ unsigned int num_cards=0, num_cpe=0;
+ FILE *fp = NULL;
+ int fd = -1, ret = -1;
+
+ while (--tries) {
+ if ((fp = fopen(cards, "r")) == NULL) {
+ ALOGE("Cannot open %s file to get list of sound cards", cards);
+ usleep(100000);
+ continue;
+ }
+ break;
+ }
+
+ if (!tries)
+ return -ENODEV;
+
+ while ((bytes_read = getline(&line, &len, fp) != -1)) {
+ // skip every other line to to match
+ // the output format of /proc/asound/cards
+ if (line_no++ % 2)
+ continue;
+
+ ptr = strtok_r(line, " [", &saveptr);
+ if (!ptr)
+ continue;
+
+ card_id = strtok_r(saveptr+1, "]", &saveptr);
+ if (!card_id)
+ continue;
+
+ // Only consider sound cards associated with ADSP
+ if (validate_snd_card((const char *)card_id) < 0) {
+ ALOGW("Skip over non-ADSP snd card %s", card_id);
+ continue;
+ }
+
+ snprintf(path, sizeof(path), "/proc/asound/card%s/state", ptr);
+ ALOGV("Opening sound card state : %s", path);
+
+ fd = open(path, O_RDONLY);
+ if (fd == -1) {
+ ALOGE("Open %s failed : %s", path, strerror(errno));
+ continue;
+ }
+
+ ret = add_new_sndcard(atoi(ptr), fd);
+ if (ret != 0)
+ continue;
+
+ num_cards++;
+
+ // query cpe state for this card as well
+ tries = MAX_CPE_SLEEP_RETRY;
+ snprintf(path, sizeof(path), "/proc/asound/card%s/cpe0_state", ptr);
+
+ if (access(path, R_OK) < 0) {
+ ALOGW("access %s failed w/ err %s", path, strerror(errno));
+ continue;
+ }
+
+ ALOGV("Open cpe state card state %s", path);
+ while (--tries) {
+ if ((fd = open(path, O_RDONLY)) < 0) {
+ ALOGW("Open cpe state card state failed, retry : %s", path);
+ usleep(CPE_SLEEP_WAIT*1000);
+ continue;
+ }
+ break;
+ }
+
+ if (!tries)
+ continue;
+
+ ret = add_new_sndcard(CPE_MAGIC_NUM+num_cpe, fd);
+ if (ret != 0)
+ continue;
+
+ num_cpe++;
+ num_cards++;
+ }
+ if (line)
+ free(line);
+ fclose(fp);
+ ALOGV("sndmonitor registerer num_cards %d", num_cards);
+ sndmonitor.num_cards = num_cards;
+ return num_cards ? 0 : -1;
+}
+
+static void free_sndcards()
+{
+ while (!list_empty(&sndmonitor.cards)) {
+ struct listnode *n = list_head(&sndmonitor.cards);
+ sndcard_t *s = node_to_item(n, sndcard_t, node);
+ list_remove(n);
+ close(s->fd);
+ free(s);
+ }
+}
+
+#ifdef MONITOR_DEVICE_EVENTS
+static int add_new_dev_event(char *d_name, int fd)
+{
+ dev_event_t *d = (dev_event_t *)calloc(sizeof(dev_event_t), 1);
+
+ if (!d)
+ return -1;
+
+ d->dev = strdup(d_name);
+ d->fd = fd;
+ list_add_tail(&sndmonitor.dev_events, &d->node);
+ return 0;
+}
+
+static int enum_dev_events()
+{
+ const char *events_dir = "/sys/class/switch/";
+ DIR *dp;
+ struct dirent *in_file;
+ int fd;
+ char path[128] = {0};
+ unsigned int num_dev_events = 0;
+
+ if ((dp = opendir(events_dir)) == NULL) {
+ ALOGE("Cannot open switch directory %s err %s",
+ events_dir, strerror(errno));
+ return -1;
+ }
+
+ while ((in_file = readdir(dp)) != NULL) {
+ if (!strstr(in_file->d_name, "qc_"))
+ continue;
+
+ snprintf(path, sizeof(path), "%s/%s/state",
+ events_dir, in_file->d_name);
+
+ ALOGV("Opening audio dev event state : %s ", path);
+ fd = open(path, O_RDONLY);
+ if (fd == -1) {
+ ALOGE("Open %s failed : %s", path, strerror(errno));
+ } else {
+ if (!add_new_dev_event(in_file->d_name, fd))
+ num_dev_events++;
+ }
+ }
+ closedir(dp);
+ sndmonitor.num_dev_events = num_dev_events;
+ return num_dev_events ? 0 : -1;
+}
+#endif
+
+static void free_dev_events()
+{
+ while (!list_empty(&sndmonitor.dev_events)) {
+ struct listnode *n = list_head(&sndmonitor.dev_events);
+ dev_event_t *d = node_to_item(n, dev_event_t, node);
+ list_remove(n);
+ close(d->fd);
+ free(d->dev);
+ free(d);
+ }
+}
+
+static int notify(const struct str_parms *params)
+{
+ if (!params)
+ return -1;
+
+ char *str = str_parms_to_str((struct str_parms *)params);
+
+ if (!str)
+ return -1;
+
+ if (sndmonitor.notify)
+ sndmonitor.notify(sndmonitor.target, str);
+
+ ALOGV("%s", str);
+ free(str);
+ return 0;
+}
+
+int on_dev_event(dev_event_t *dev_event)
+{
+ char state_buf[2];
+ if (read(dev_event->fd, state_buf, 1) <= 0)
+ return -1;
+
+ lseek(dev_event->fd, 0, SEEK_SET);
+ state_buf[1]='\0';
+ if (atoi(state_buf) == dev_event->status)
+ return 0;
+
+ dev_event->status = atoi(state_buf);
+
+ struct str_parms *params = str_parms_create();
+
+ if (!params)
+ return -1;
+
+ char val[32] = {0};
+ snprintf(val, sizeof(val), "%s,%s", dev_event->dev,
+ dev_event->status ? "ON" : "OFF");
+
+ if (str_parms_add_str(params, AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE, val) < 0)
+ return -1;
+
+ int ret = notify(params);
+ str_parms_destroy(params);
+ return ret;
+}
+
+bool on_sndcard_state_update(sndcard_t *s)
+{
+ char rd_buf[9]={0};
+ card_status_t status;
+
+ if (read(s->fd, rd_buf, 8) <= 0)
+ return -1;
+
+ rd_buf[8] = '\0';
+ lseek(s->fd, 0, SEEK_SET);
+
+ ALOGV("card num %d, new state %s", s->card, rd_buf);
+
+ bool is_cpe = (s->card >= CPE_MAGIC_NUM);
+ if (strstr(rd_buf, "OFFLINE"))
+ status = CARD_STATUS_OFFLINE;
+ else if (strstr(rd_buf, "ONLINE"))
+ status = CARD_STATUS_ONLINE;
+ else {
+ ALOGE("unknown state");
+ return 0;
+ }
+
+ if (status == s->status) // no change
+ return 0;
+
+ s->status = status;
+
+ struct str_parms *params = str_parms_create();
+
+ if (!params)
+ return -1;
+
+ char val[32] = {0};
+ // cpe actual card num is (card - MAGIC_NUM). so subtract accordingly
+ snprintf(val, sizeof(val), "%d,%s", s->card - (is_cpe ? CPE_MAGIC_NUM : 0),
+ status == CARD_STATUS_ONLINE ? "ONLINE" : "OFFLINE");
+
+ if (str_parms_add_str(params, is_cpe ? "CPE_STATUS" : "SND_CARD_STATUS",
+ val) < 0)
+ return -1;
+
+ int ret = notify(params);
+ str_parms_destroy(params);
+ return ret;
+}
+
+void *monitor_thread_loop(void *args __unused)
+{
+ ALOGV("Start threadLoop()");
+ unsigned int num_poll_fds = sndmonitor.num_cards +
+ sndmonitor.num_dev_events + 1/*pipe*/;
+ struct pollfd *pfd = (struct pollfd *)calloc(sizeof(struct pollfd),
+ num_poll_fds);
+ if (!pfd)
+ return NULL;
+
+ pfd[0].fd = sndmonitor.intpipe[0];
+ pfd[0].events = POLLPRI|POLLIN;
+
+ int i = 1;
+ struct listnode *node;
+ list_for_each(node, &sndmonitor.cards) {
+ sndcard_t *s = node_to_item(node, sndcard_t, node);
+ pfd[i].fd = s->fd;
+ pfd[i].events = POLLPRI;
+ ++i;
+ }
+
+ list_for_each(node, &sndmonitor.dev_events) {
+ dev_event_t *d = node_to_item(node, dev_event_t, node);
+ pfd[i].fd = d->fd;
+ pfd[i].events = POLLPRI;
+ ++i;
+ }
+
+ while (1) {
+ if (poll(pfd, num_poll_fds, -1) < 0) {
+ int errno_ = errno;
+ ALOGE("poll() failed w/ err %s", strerror(errno_));
+ switch (errno_) {
+ case EINTR:
+ case ENOMEM:
+ sleep(2);
+ continue;
+ default:
+ /* above errors can be caused due to current system
+ state .. any other error is not expected */
+ LOG_ALWAYS_FATAL("unxpected poll() system call failure");
+ break;
+ }
+ }
+ ALOGV("out of poll()");
+
+#define READY_TO_READ(p) ((p)->revents & (POLLIN|POLLPRI))
+#define ERROR_IN_FD(p) ((p)->revents & (POLLERR|POLLHUP|POLLNVAL))
+
+ // check if requested to exit
+ if (READY_TO_READ(&pfd[0])) {
+ char buf[2]={0};
+ read(pfd[0].fd, buf, 1);
+ if (!strcmp(buf, "Q"))
+ break;
+ } else if (ERROR_IN_FD(&pfd[0])) {
+ // do not consider for poll again
+ // POLLERR - can this happen?
+ // POLLHUP - adev must not close pipe
+ // POLLNVAL - fd is valid
+ LOG_ALWAYS_FATAL("unxpected error in pipe poll fd 0x%x",
+ pfd[0].revents);
+ // FIXME: If not fatal, then need some logic to close
+ // these fds on error
+ pfd[0].fd *= -1;
+ }
+
+ i = 1;
+ list_for_each(node, &sndmonitor.cards) {
+ sndcard_t *s = node_to_item(node, sndcard_t, node);
+ if (READY_TO_READ(&pfd[i]))
+ on_sndcard_state_update(s);
+ else if (ERROR_IN_FD(&pfd[i])) {
+ // do not consider for poll again
+ // POLLERR - can this happen as we are reading from a fs?
+ // POLLHUP - not valid for cardN/state
+ // POLLNVAL - fd is valid
+ LOG_ALWAYS_FATAL("unxpected error in card poll fd 0x%x",
+ pfd[i].revents);
+ // FIXME: If not fatal, then need some logic to close
+ // these fds on error
+ pfd[i].fd *= -1;
+ }
+ ++i;
+ }
+
+ list_for_each(node, &sndmonitor.dev_events) {
+ dev_event_t *d = node_to_item(node, dev_event_t, node);
+ if (READY_TO_READ(&pfd[i]))
+ on_dev_event(d);
+ else if (ERROR_IN_FD(&pfd[i])) {
+ // do not consider for poll again
+ // POLLERR - can this happen as we are reading from a fs?
+ // POLLHUP - not valid for switch/state
+ // POLLNVAL - fd is valid
+ LOG_ALWAYS_FATAL("unxpected error in dev poll fd 0x%x",
+ pfd[i].revents);
+ // FIXME: If not fatal, then need some logic to close
+ // these fds on error
+ pfd[i].fd *= -1;
+ }
+ ++i;
+ }
+ }
+
+ return NULL;
+}
+
+// ---- listener static APIs ---- //
+static int hashfn(void *key)
+{
+ return (int)key;
+}
+
+static bool hasheq(void *key1, void *key2)
+{
+ return key1 == key2;
+}
+
+static bool snd_cb(void *key, void *value, void *context)
+{
+ snd_mon_cb cb = (snd_mon_cb)value;
+ cb(key, context);
+ return true;
+}
+
+static void snd_mon_update(const void *target __unused, const char *msg)
+{
+ // target can be used to check if this message is intended for the
+ // recipient or not. (using some statically saved state)
+
+ struct str_parms *parms = str_parms_create_str(msg);
+
+ if (!parms)
+ return;
+
+ hashmapLock(sndmonitor.listeners);
+ hashmapForEach(sndmonitor.listeners, snd_cb, parms);
+ hashmapUnlock(sndmonitor.listeners);
+
+ str_parms_destroy(parms);
+}
+
+static int listeners_init()
+{
+ sndmonitor.listeners = hashmapCreate(INIT_MAP_SIZE, hashfn, hasheq);
+ if (!sndmonitor.listeners)
+ return -1;
+ return 0;
+}
+
+static int listeners_deinit()
+{
+ // XXX TBD
+ return -1;
+}
+
+static int add_listener(void *stream, snd_mon_cb cb)
+{
+ Hashmap *map = sndmonitor.listeners;
+ hashmapLock(map);
+ hashmapPut(map, stream, cb);
+ hashmapUnlock(map);
+ return 0;
+}
+
+static int del_listener(void * stream)
+{
+ Hashmap *map = sndmonitor.listeners;
+ hashmapLock(map);
+ hashmapRemove(map, stream);
+ hashmapUnlock(map);
+ return 0;
+}
+
+// --- public APIs --- //
+
+int audio_extn_snd_mon_deinit()
+{
+ if (!sndmonitor.initcheck)
+ return -1;
+
+ write(sndmonitor.intpipe[1], "Q", 1);
+ pthread_join(sndmonitor.monitor_thread, (void **) NULL);
+ listeners_deinit();
+ free_sndcards();
+ free_dev_events();
+ sndmonitor.initcheck = 0;
+ return 0;
+}
+
+int audio_extn_snd_mon_init()
+{
+ sndmonitor.notify = snd_mon_update;
+ sndmonitor.target = NULL; // unused for now
+ list_init(&sndmonitor.cards);
+ list_init(&sndmonitor.dev_events);
+ sndmonitor.initcheck = false;
+
+ if (pipe(sndmonitor.intpipe) < 0)
+ return -ENODEV;
+
+ if (enum_sndcards() < 0)
+ return -ENODEV;
+
+ if (listeners_init() < 0)
+ return -ENODEV;
+
+#ifdef MONITOR_DEVICE_EVENTS
+ enum_dev_events(); // failure here isn't fatal
+#endif
+
+ int ret = pthread_create(&sndmonitor.monitor_thread,
+ (const pthread_attr_t *) NULL,
+ monitor_thread_loop, NULL);
+
+ if (ret) {
+ free_sndcards();
+ free_dev_events();
+ close(sndmonitor.intpipe[0]);
+ close(sndmonitor.intpipe[1]);
+ return -ENODEV;
+ }
+ sndmonitor.initcheck = true;
+ return 0;
+}
+
+int audio_extn_snd_mon_register_listener(void *stream, snd_mon_cb cb)
+{
+ if (!sndmonitor.initcheck) {
+ ALOGW("sndmonitor initcheck failed, cannot register");
+ return -1;
+ }
+
+ return add_listener(stream, cb);
+}
+
+int audio_extn_snd_mon_unregister_listener(void *stream)
+{
+ if (!sndmonitor.initcheck) {
+ ALOGW("sndmonitor initcheck failed, cannot deregister");
+ return -1;
+ }
+
+ ALOGV("deregister listener for stream %p ", stream);
+ return del_listener(stream);
+}
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index 7e37efc..6142e86 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -98,9 +98,9 @@
status = -ENOMEM;
break;
}
- memcpy(&st_ses_info->st_ses, &config->st_ses, sizeof (config->st_ses));
- ALOGV("%s: add capture_handle %d pcm %p", __func__,
- st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.pcm);
+ memcpy(&st_ses_info->st_ses, &config->st_ses, sizeof (struct sound_trigger_session_info));
+ ALOGV("%s: add capture_handle %d st session opaque ptr %p", __func__,
+ st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.p_ses);
list_add_tail(&st_dev->st_ses_list, &st_ses_info->list);
break;
@@ -112,12 +112,12 @@
}
st_ses_info = get_sound_trigger_info(config->st_ses.capture_handle);
if (!st_ses_info) {
- ALOGE("%s: pcm %p not in the list!", __func__, config->st_ses.pcm);
+ ALOGE("%s: st session opaque ptr %p not in the list!", __func__, config->st_ses.p_ses);
status = -EINVAL;
break;
}
- ALOGV("%s: remove capture_handle %d pcm %p", __func__,
- st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.pcm);
+ ALOGV("%s: remove capture_handle %d st session opaque ptr %p", __func__,
+ st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.p_ses);
list_remove(&st_ses_info->list);
free(st_ses_info);
break;
@@ -181,7 +181,7 @@
pthread_mutex_unlock(&st_dev->lock);
if (st_ses_info) {
event.u.ses_info = st_ses_info->st_ses;
- ALOGV("%s: AUDIO_EVENT_STOP_LAB pcm %p", __func__, st_ses_info->st_ses.pcm);
+ ALOGV("%s: AUDIO_EVENT_STOP_LAB st sess %p", __func__, st_ses_info->st_ses.p_ses);
st_dev->st_callback(AUDIO_EVENT_STOP_LAB, &event);
in->is_st_session_active = false;
}
@@ -201,7 +201,6 @@
list_for_each(node, &st_dev->st_ses_list) {
st_ses_info = node_to_item(node, struct sound_trigger_info , list);
if (st_ses_info->st_ses.capture_handle == in->capture_handle) {
- in->pcm = st_ses_info->st_ses.pcm;
in->config = st_ses_info->st_ses.config;
in->channel_mask = audio_channel_in_mask_from_count(in->config.channels);
in->is_st_session = true;
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 1f88c71..008130f 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -117,6 +117,11 @@
SPKR_PROTECTION_MODE_CALIBRATE = 1,
};
+struct spkr_prot_r0t0 {
+ int r0[SP_V2_NUM_MAX_SPKRS];
+ int t0[SP_V2_NUM_MAX_SPKRS];
+};
+
struct speaker_prot_session {
int spkr_prot_mode;
int spkr_processing_state;
@@ -142,6 +147,7 @@
bool spkr_prot_enable;
bool spkr_in_use;
struct timespec spkr_last_time_used;
+ struct spkr_prot_r0t0 sp_r0t0_cal;
bool wsa_found;
int spkr_1_tzn;
int spkr_2_tzn;
@@ -340,6 +346,7 @@
int ret = 0;
struct audio_cal_fb_spk_prot_cfg cal_data;
char value[PROPERTY_VALUE_MAX];
+ static int cal_done = 0;
if (cal_fd < 0) {
ALOGE("%s: Error: cal_fd = %d", __func__, cal_fd);
@@ -382,6 +389,13 @@
ret = -ENODEV;
goto done;
}
+ if (protCfg->mode == MSM_SPKR_PROT_CALIBRATED && !cal_done) {
+ handle.sp_r0t0_cal.r0[SP_V2_SPKR_1] = protCfg->r0[SP_V2_SPKR_1];
+ handle.sp_r0t0_cal.r0[SP_V2_SPKR_2] = protCfg->r0[SP_V2_SPKR_2];
+ handle.sp_r0t0_cal.t0[SP_V2_SPKR_1] = protCfg->t0[SP_V2_SPKR_1];
+ handle.sp_r0t0_cal.t0[SP_V2_SPKR_2] = protCfg->t0[SP_V2_SPKR_2];
+ cal_done = 1;
+ }
done:
return ret;
}
@@ -1347,12 +1361,48 @@
}
}
+int audio_extn_select_spkr_prot_cal_data(snd_device_t snd_device)
+{
+ struct audio_cal_info_spk_prot_cfg protCfg;
+ int acdb_fd = -1;
+ int ret = 0;
+
+ acdb_fd = open("/dev/msm_audio_cal", O_RDWR | O_NONBLOCK);
+ if (acdb_fd < 0) {
+ ALOGE("%s: open msm_acdb failed", __func__);
+ return -ENODEV;
+ }
+ switch(snd_device) {
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+ protCfg.r0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_2];
+ protCfg.r0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_1];
+ protCfg.t0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_2];
+ protCfg.t0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_1];
+ break;
+ default:
+ protCfg.r0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_1];
+ protCfg.r0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_2];
+ protCfg.t0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_1];
+ protCfg.t0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_2];
+ break;
+ }
+ protCfg.mode = MSM_SPKR_PROT_CALIBRATED;
+ ret = set_spkr_prot_cal(acdb_fd, &protCfg);
+ if (ret)
+ ALOGE("%s: speaker protection cal data swap failed", __func__);
+
+ close(acdb_fd);
+ return ret;
+}
+
int audio_extn_spkr_prot_start_processing(snd_device_t snd_device)
{
struct audio_usecase *uc_info_tx;
struct audio_device *adev = handle.adev_handle;
int32_t pcm_dev_tx_id = -1, ret = 0;
bool disable_tx = false;
+ snd_device_t in_snd_device;
ALOGV("%s: Entry", __func__);
/* cancel speaker calibration */
@@ -1361,6 +1411,15 @@
return -EINVAL;
}
snd_device = platform_get_spkr_prot_snd_device(snd_device);
+ if (handle.spkr_prot_mode == MSM_SPKR_PROT_CALIBRATED) {
+ ret = audio_extn_select_spkr_prot_cal_data(snd_device);
+ if (ret) {
+ ALOGE("%s: Setting speaker protection cal data failed", __func__);
+ return ret;
+ }
+ }
+
+ in_snd_device = platform_get_vi_feedback_snd_device(snd_device);
spkr_prot_set_spkrstatus(true);
uc_info_tx = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
if (!uc_info_tx) {
@@ -1375,12 +1434,12 @@
if (handle.spkr_processing_state == SPKR_PROCESSING_IN_IDLE) {
uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
uc_info_tx->type = PCM_CAPTURE;
- uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ uc_info_tx->in_snd_device = in_snd_device;
uc_info_tx->out_snd_device = SND_DEVICE_NONE;
handle.pcm_tx = NULL;
list_add_tail(&adev->usecase_list, &uc_info_tx->list);
disable_tx = true;
- enable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ enable_snd_device(adev, in_snd_device);
enable_audio_route(adev, uc_info_tx);
pcm_dev_tx_id = platform_get_pcm_device_id(uc_info_tx->id, PCM_CAPTURE);
@@ -1420,9 +1479,9 @@
list_remove(&uc_info_tx->list);
uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
uc_info_tx->type = PCM_CAPTURE;
- uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ uc_info_tx->in_snd_device = in_snd_device;
uc_info_tx->out_snd_device = SND_DEVICE_NONE;
- disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ disable_snd_device(adev, in_snd_device);
disable_audio_route(adev, uc_info_tx);
free(uc_info_tx);
} else
@@ -1436,17 +1495,20 @@
{
struct audio_usecase *uc_info_tx;
struct audio_device *adev = handle.adev_handle;
+ snd_device_t in_snd_device;
ALOGV("%s: Entry", __func__);
snd_device = platform_get_spkr_prot_snd_device(snd_device);
spkr_prot_set_spkrstatus(false);
+ in_snd_device = platform_get_vi_feedback_snd_device(snd_device);
+
pthread_mutex_lock(&handle.mutex_spkr_prot);
if (adev && handle.spkr_processing_state == SPKR_PROCESSING_IN_PROGRESS) {
uc_info_tx = get_usecase_from_list(adev, USECASE_AUDIO_SPKR_CALIB_TX);
if (handle.pcm_tx)
pcm_close(handle.pcm_tx);
handle.pcm_tx = NULL;
- disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ disable_snd_device(adev, in_snd_device);
if (uc_info_tx) {
list_remove(&uc_info_tx->list);
disable_audio_route(adev, uc_info_tx);
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index 9e19eba..f936f99 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -47,7 +47,7 @@
#define SAMPLE_RATE_11025 11025
// Supported sample rates for USB
static uint32_t supported_sample_rates[] =
- {44100, 48000, 64000, 88200, 96000, 176400, 192000};
+ {44100, 48000, 64000, 88200, 96000, 176400, 192000, 384000};
#define MAX_SAMPLE_RATE_SIZE sizeof(supported_sample_rates)/sizeof(supported_sample_rates[0])
@@ -93,7 +93,7 @@
static const char * const usb_sidetone_enable_str[] = {
"Sidetone Playback Switch",
- "Mic Playback Switchs",
+ "Mic Playback Switch",
};
static const char * const usb_sidetone_volume_str[] = {
@@ -190,33 +190,6 @@
}
}
-static int usb_set_channel_mixer_ctl(int channel,
- char *ch_mixer_ctl_name)
-{
- struct mixer_ctl *ctl;
-
- ctl = mixer_get_ctl_by_name(usbmod->adev->mixer, ch_mixer_ctl_name);
- if (!ctl) {
- ALOGE("%s: Could not get ctl for mixer cmd - %s",
- __func__, ch_mixer_ctl_name);
- return -EINVAL;
- }
- switch (channel) {
- case 1:
- mixer_ctl_set_enum_by_string(ctl, "One");
- break;
- case 2:
- mixer_ctl_set_enum_by_string(ctl, "Two");
- break;
- default:
- ALOGV("%s: channel(%d) not supported, set as default 2 channels",
- __func__, channel);
- mixer_ctl_set_enum_by_string(ctl, "Two");
- break;
- }
- return 0;
-}
-
static int usb_set_dev_id_mixer_ctl(unsigned int usb_usecase_type, int card,
char *dev_mixer_ctl_name)
{
@@ -300,25 +273,34 @@
struct usb_card_config *usb_card_info,
int card)
{
- int32_t err = 1;
int32_t size = 0;
int32_t fd=-1;
- int32_t altset_index = 1;
int32_t channels_no;
- char *str_start, *channel_start, *bit_width_start, *rates_str_start,
- *target;
+ char *str_start = NULL;
+ char *str_end = NULL;
+ char *channel_start = NULL;
+ char *bit_width_start = NULL;
+ char *rates_str_start = NULL;
+ char *target = NULL;
char *read_buf = NULL;
char *rates_str = NULL;
- char path[128], altset[9];
+ char path[128];
int ret = 0;
char *bit_width_str = NULL;
struct usb_device_config * usb_device_info;
+ bool check = false;
+ memset(path, 0, sizeof(path));
ALOGV("%s: for %s", __func__, (type == USB_PLAYBACK) ?
PLAYBACK_PROFILE_STR : CAPTURE_PROFILE_STR);
- snprintf(path, sizeof(path), "/proc/asound/card%u/stream0",
+ ret = snprintf(path, sizeof(path), "/proc/asound/card%u/stream0",
card);
+ if(ret < 0) {
+ ALOGE("%s: failed on snprintf (%d) to path %s\n",
+ __func__, ret, path);
+ goto done;
+ }
fd = open(path, O_RDONLY);
if (fd <0) {
@@ -336,7 +318,10 @@
goto done;
}
- err = read(fd, read_buf, USB_BUFF_SIZE);
+ if(read(fd, read_buf, USB_BUFF_SIZE) < 0) {
+ ALOGE("file read error\n");
+ goto done;
+ }
str_start = strstr(read_buf, ((type == USB_PLAYBACK) ?
PLAYBACK_PROFILE_STR : CAPTURE_PROFILE_STR));
if (str_start == NULL) {
@@ -346,21 +331,21 @@
ret = -EINVAL;
goto done;
}
- ALOGV("%s: usb_config = %s\n", __func__, str_start);
+ str_end = strstr(read_buf, ((type == USB_PLAYBACK) ?
+ CAPTURE_PROFILE_STR : PLAYBACK_PROFILE_STR));
+ if (str_end > str_start)
+ check = true;
+
+ ALOGV("%s: usb_config = %s, check %d\n", __func__, str_start, check);
while (str_start != NULL) {
- sprintf(altset, "Altset %d", altset_index);
- ALOGV("%s: altset_index %d\n", __func__, altset_index);
- str_start = strstr(str_start, altset);
- if (str_start == NULL) {
- if (altset_index == 1) {
- ALOGE("%s: error %s section not found in usb config file",
- __func__, (type == USB_PLAYBACK) ?
- PLAYBACK_PROFILE_STR : CAPTURE_PROFILE_STR);
- ret = -EINVAL;
- }
+ str_start = strstr(str_start, "Altset");
+ if ((str_start == NULL) || (check && (str_start >= str_end))) {
+ ALOGV("%s: done parsing %s\n", __func__, str_start);
break;
}
+ ALOGV("%s: remaining string %s\n", __func__, str_start);
+ str_start += sizeof("Altset");
usb_device_info = calloc(1, sizeof(struct usb_device_config));
if (usb_device_info == NULL) {
ALOGE("%s: error unable to allocate memory",
@@ -368,7 +353,6 @@
ret = -ENOMEM;
break;
}
- altset_index++;
/* Bit bit_width parsing */
bit_width_start = strstr(str_start, "Format: ");
if (bit_width_start == NULL) {
@@ -461,8 +445,6 @@
int card)
{
int ret;
- struct listnode *node_d;
- struct usb_device_config *dev_info;
/* get capabilities */
if ((ret = usb_get_capability(USB_PLAYBACK, usb_card_info, card))) {
@@ -470,14 +452,6 @@
__func__);
goto exit;
}
- /* Currently only use the first profile using to configure channel for simplification */
- list_for_each(node_d, &usb_card_info->usb_device_conf_list) {
- dev_info = node_to_item(node_d, struct usb_device_config, list);
- if (dev_info != NULL) {
- usb_set_channel_mixer_ctl(dev_info->channels, "USB_AUDIO_RX Channels");
- break;
- }
- }
usb_set_dev_id_mixer_ctl(USB_PLAYBACK, card, "USB_AUDIO_RX dev_token");
exit:
@@ -489,8 +463,6 @@
int card)
{
int ret;
- struct listnode *node_d;
- struct usb_device_config *dev_info;
/* get capabilities */
if ((ret = usb_get_capability(USB_CAPTURE, usb_card_info, card))) {
@@ -498,14 +470,6 @@
__func__);
goto exit;
}
- /* Currently only use the first profile using to configure channel for simplification */
- list_for_each(node_d, &usb_card_info->usb_device_conf_list) {
- dev_info = node_to_item(node_d, struct usb_device_config, list);
- if (dev_info != NULL) {
- usb_set_channel_mixer_ctl(dev_info->channels, "USB_AUDIO_TX Channels");
- break;
- }
- }
usb_set_dev_id_mixer_ctl(USB_CAPTURE, card, "USB_AUDIO_TX dev_token");
exit:
@@ -739,7 +703,7 @@
"%s: USB ch(%d)bw(%d), stm ch(%d)bw(%d)sr(%d), candidate(%d)",
__func__, dev_info->channels, dev_info->bit_width,
ch, bit_width, stream_sample_rate, candidate);
- if ((dev_info->bit_width != bit_width) && dev_info->channels != ch)
+ if ((dev_info->bit_width != bit_width) || dev_info->channels != ch)
continue;
candidate = 0;
@@ -786,34 +750,32 @@
static bool usb_audio_backend_apply_policy(struct listnode *dev_list,
unsigned int *bit_width,
unsigned int *sample_rate,
- unsigned int ch)
+ unsigned int *ch)
{
- unsigned int channel;
bool is_usb_supported = true;
ALOGV("%s: from stream: bit-width(%d) sample_rate(%d) channels (%d)",
- __func__, *bit_width, *sample_rate, ch);
+ __func__, *bit_width, *sample_rate, *ch);
if (list_empty(dev_list)) {
*sample_rate = 48000;
*bit_width = 16;
- channel = 2;
+ *ch = 2;
ALOGI("%s: list is empty,fall back to default setting", __func__);
goto exit;
}
usb_get_best_match_for_bit_width(dev_list, *bit_width, bit_width);
usb_get_best_match_for_channels(dev_list,
*bit_width,
- ch,
- &channel);
+ *ch,
+ ch);
usb_get_best_match_for_sample_rate(dev_list,
*bit_width,
- channel,
+ *ch,
*sample_rate,
sample_rate);
exit:
ALOGV("%s: Updated sample rate per profile: bit-width(%d) rate(%d) chs(%d)",
- __func__, *bit_width, *sample_rate, channel);
- usb_set_channel_mixer_ctl(channel, "USB_AUDIO_RX Channels");
+ __func__, *bit_width, *sample_rate, *ch);
return is_usb_supported;
}
@@ -885,21 +847,23 @@
bool audio_extn_usb_is_config_supported(unsigned int *bit_width,
unsigned int *sample_rate,
- unsigned int ch)
+ unsigned int *ch,
+ bool is_playback)
{
struct listnode *node_i;
struct usb_card_config *card_info;
bool is_usb_supported = false;
ALOGV("%s: from stream: bit-width(%d) sample_rate(%d) ch(%d)",
- __func__, *bit_width, *sample_rate, ch);
+ __func__, *bit_width, *sample_rate, *ch);
list_for_each(node_i, &usbmod->usb_card_conf_list) {
card_info = node_to_item(node_i, struct usb_card_config, list);
ALOGI_IF(usb_audio_debug_enable,
"%s: card_dev_type (0x%x), card_no(%d)",
__func__, card_info->usb_device_type, card_info->usb_card);
/* Currently only apply the first playback sound card configuration */
- if (card_info->usb_device_type == AUDIO_DEVICE_OUT_USB_DEVICE) {
+ if ((is_playback && card_info->usb_device_type == AUDIO_DEVICE_OUT_USB_DEVICE) ||
+ ((!is_playback) && card_info->usb_device_type == AUDIO_DEVICE_IN_USB_DEVICE)){
is_usb_supported = usb_audio_backend_apply_policy(
&card_info->usb_device_conf_list,
bit_width,
@@ -908,8 +872,8 @@
break;
}
}
- ALOGV("%s: updated: bit-width(%d) sample_rate(%d)",
- __func__, *bit_width, *sample_rate);
+ ALOGV("%s: updated: bit-width(%d) sample_rate(%d) channels (%d)",
+ __func__, *bit_width, *sample_rate, *ch);
return is_usb_supported;
}
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index e3f1b6c..9542fbd 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -90,9 +90,7 @@
#ifdef INCALL_MUSIC_ENABLED
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_INCALL_MUSIC),
#endif
-#ifdef HDMI_PASSTHROUGH_ENABLED
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH),
-#endif
};
const struct string_to_enum s_format_name_to_enum_table[] = {
@@ -133,6 +131,7 @@
STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_LC),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_HE_V1),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_HE_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_DSD),
#endif
};
@@ -496,7 +495,7 @@
struct stream_format *sf_info;
char value[PROPERTY_VALUE_MAX] = {0};
- if ((24 == bit_width) &&
+ if ((bit_width >= 24) &&
(devices & AUDIO_DEVICE_OUT_SPEAKER)) {
int32_t bw = platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
if (-ENOSYS != bw)
@@ -515,6 +514,28 @@
__func__, sample_rate);
}
}
+
+ /* Set sampling rate to 176.4 for DSD64
+ * and 352.8Khz for DSD128.
+ * Set Bit Width to 16. output will be 16 bit
+ * post DoP in ASM.
+ */
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH) &&
+ (format == AUDIO_FORMAT_DSD)) {
+ bit_width = 16;
+ if (sample_rate == INPUT_SAMPLING_RATE_DSD64)
+ sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+ else if (sample_rate == INPUT_SAMPLING_RATE_DSD128)
+ sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
+ }
+
+ if(devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ //TODO: Handle fractional sampling rate configuration for LL
+ audio_extn_a2dp_get_apptype_params(&sample_rate, &bit_width);
+ ALOGI("%s using %d sampling rate %d bit width for A2DP CoPP",
+ __func__, sample_rate, bit_width);
+ }
+
ALOGV("%s: flags: %x, format: %x sample_rate %d",
__func__, flags, format, sample_rate);
list_for_each(node_i, streams_output_cfg_list) {
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index a8ebb6b..3816748 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -36,6 +36,7 @@
*/
#define LOG_TAG "audio_hw_primary"
+#define ATRACE_TAG (ATRACE_TAG_AUDIO|ATRACE_TAG_HAL)
/*#define LOG_NDEBUG 0*/
/*#define VERY_VERY_VERBOSE_LOGGING*/
#ifdef VERY_VERY_VERBOSE_LOGGING
@@ -55,6 +56,7 @@
#include <sys/prctl.h>
#include <cutils/log.h>
+#include <cutils/trace.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <cutils/atomic.h>
@@ -80,6 +82,7 @@
/* ToDo: Check and update a proper value in msec */
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+#define DSD_VOLUME_MIN_DB (-110)
#define PROXY_OPEN_RETRY_COUNT 100
#define PROXY_OPEN_WAIT_TIME 20
@@ -92,6 +95,8 @@
#define PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY pcm_config_deep_buffer
#endif
+#define ULL_PERIOD_SIZE (DEFAULT_OUTPUT_SAMPLING_RATE/1000)
+
static unsigned int configured_low_latency_capture_period_size =
LOW_LATENCY_CAPTURE_PERIOD_SIZE;
@@ -117,6 +122,20 @@
.avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
};
+static int af_period_multiplier = 4;
+struct pcm_config pcm_config_rt = {
+ .channels = 2,
+ .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+ .period_size = ULL_PERIOD_SIZE, //1 ms
+ .period_count = 512, //=> buffer size is 512ms
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = ULL_PERIOD_SIZE*8, //8ms
+ .stop_threshold = INT_MAX,
+ .silence_threshold = 0,
+ .silence_size = 0,
+ .avail_min = ULL_PERIOD_SIZE, //1 ms
+};
+
struct pcm_config pcm_config_hdmi_multi = {
.channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
.rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
@@ -134,6 +153,19 @@
.format = PCM_FORMAT_S16_LE,
};
+struct pcm_config pcm_config_audio_capture_rt = {
+ .channels = 2,
+ .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+ .period_size = ULL_PERIOD_SIZE,
+ .period_count = 512,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = 0,
+ .stop_threshold = INT_MAX,
+ .silence_threshold = 0,
+ .silence_size = 0,
+ .avail_min = ULL_PERIOD_SIZE, //1 ms
+};
+
#define AFE_PROXY_CHANNEL_COUNT 2
#define AFE_PROXY_SAMPLING_RATE 48000
@@ -223,6 +255,7 @@
[USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback",
[USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record",
+ [USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = "silence-playback",
};
static const audio_usecase_t offload_usecases[] = {
@@ -284,6 +317,103 @@
//cache last MBDRC cal step level
static int last_known_cal_step = -1 ;
+static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
+ int flags __unused)
+{
+ int dir = 0;
+ switch (uc_id) {
+ case USECASE_AUDIO_RECORD_LOW_LATENCY:
+ dir = 1;
+ case USECASE_AUDIO_PLAYBACK_ULL:
+ break;
+ default:
+ return false;
+ }
+
+ int dev_id = platform_get_pcm_device_id(uc_id, dir == 0 ?
+ PCM_PLAYBACK : PCM_CAPTURE);
+ if (adev->adm_is_noirq_avail)
+ return adev->adm_is_noirq_avail(adev->adm_data,
+ adev->snd_card, dev_id, dir);
+ return false;
+}
+
+static void register_out_stream(struct stream_out *out)
+{
+ struct audio_device *adev = out->dev;
+ if (is_offload_usecase(out->usecase) ||
+ !adev->adm_register_output_stream)
+ return;
+
+ // register stream first for backward compatibility
+ adev->adm_register_output_stream(adev->adm_data,
+ out->handle,
+ out->flags);
+
+ if (!adev->adm_set_config)
+ return;
+
+ if (out->realtime)
+ adev->adm_set_config(adev->adm_data,
+ out->handle,
+ out->pcm, &out->config);
+}
+
+static void register_in_stream(struct stream_in *in)
+{
+ struct audio_device *adev = in->dev;
+ if (!adev->adm_register_input_stream)
+ return;
+
+ adev->adm_register_input_stream(adev->adm_data,
+ in->capture_handle,
+ in->flags);
+
+ if (!adev->adm_set_config)
+ return;
+
+ if (in->realtime)
+ adev->adm_set_config(adev->adm_data,
+ in->capture_handle,
+ in->pcm,
+ &in->config);
+}
+
+static void request_out_focus(struct stream_out *out, long ns)
+{
+ struct audio_device *adev = out->dev;
+
+ if (adev->adm_request_focus_v2)
+ adev->adm_request_focus_v2(adev->adm_data, out->handle, ns);
+ else if (adev->adm_request_focus)
+ adev->adm_request_focus(adev->adm_data, out->handle);
+}
+
+static void request_in_focus(struct stream_in *in, long ns)
+{
+ struct audio_device *adev = in->dev;
+
+ if (adev->adm_request_focus_v2)
+ adev->adm_request_focus_v2(adev->adm_data, in->capture_handle, ns);
+ else if (adev->adm_request_focus)
+ adev->adm_request_focus(adev->adm_data, in->capture_handle);
+}
+
+static void release_out_focus(struct stream_out *out)
+{
+ struct audio_device *adev = out->dev;
+
+ if (adev->adm_abandon_focus)
+ adev->adm_abandon_focus(adev->adm_data, out->handle);
+}
+
+static void release_in_focus(struct stream_in *in)
+{
+ struct audio_device *adev = in->dev;
+ if (adev->adm_abandon_focus)
+ adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
+}
+
__attribute__ ((visibility ("default")))
bool audio_hw_send_gain_dep_calibration(int level) {
bool ret_val = false;
@@ -360,6 +490,7 @@
format == AUDIO_FORMAT_FLAC ||
format == AUDIO_FORMAT_ALAC ||
format == AUDIO_FORMAT_APE ||
+ format == AUDIO_FORMAT_DSD ||
format == AUDIO_FORMAT_VORBIS ||
format == AUDIO_FORMAT_WMA ||
format == AUDIO_FORMAT_WMA_PRO)
@@ -368,6 +499,12 @@
return false;
}
+static inline bool is_mmap_usecase(audio_usecase_t uc_id)
+{
+ return (uc_id == USECASE_AUDIO_RECORD_AFE_PROXY) ||
+ (uc_id == USECASE_AUDIO_PLAYBACK_AFE_PROXY);
+}
+
static int get_snd_codec_id(audio_format_t format)
{
int id = 0;
@@ -394,6 +531,9 @@
case AUDIO_FORMAT_APE:
id = SND_AUDIOCODEC_APE;
break;
+ case AUDIO_FORMAT_DSD:
+ id = SND_AUDIOCODEC_DSD;
+ break;
case AUDIO_FORMAT_VORBIS:
id = SND_AUDIOCODEC_VORBIS;
break;
@@ -469,6 +609,82 @@
return 0;
}
+static void enable_asrc_mode(struct audio_device *adev)
+{
+ ALOGV("%s", __func__);
+ audio_route_apply_and_update_path(adev->audio_route,
+ "asrc-mode");
+ adev->asrc_mode_enabled = true;
+}
+
+static void disable_asrc_mode(struct audio_device *adev)
+{
+ ALOGV("%s", __func__);
+ audio_route_reset_and_update_path(adev->audio_route,
+ "asrc-mode");
+ adev->asrc_mode_enabled = false;
+}
+
+/*
+ * - Enable ASRC mode for incoming mix path use case(Headphone backend)if Headphone
+ * 44.1 or Native DSD backends are enabled for any of current use case.
+ * e.g. 48-> + (Naitve DSD or Headphone 44.1)
+ * - Disable current mix path use case(Headphone backend) and re-enable it with
+ * ASRC mode for incoming Headphone 44.1 or Native DSD use case.
+ * e.g. Naitve DSD or Headphone 44.1 -> + 48
+ */
+static void check_and_set_asrc_mode(struct audio_device *adev, snd_device_t snd_device)
+{
+ ALOGV("%s snd device %d", __func__, snd_device);
+ int new_backend_idx = platform_get_backend_index(snd_device);
+
+ if (((new_backend_idx == HEADPHONE_BACKEND) ||
+ (new_backend_idx == HEADPHONE_44_1_BACKEND) ||
+ (new_backend_idx == DSD_NATIVE_BACKEND)) &&
+ !adev->asrc_mode_enabled) {
+ struct listnode *node = NULL;
+ struct audio_usecase *uc = NULL;
+ struct stream_out *curr_out = NULL;
+ int usecase_backend_idx = DEFAULT_CODEC_BACKEND;
+
+ list_for_each(node, &adev->usecase_list) {
+ uc = node_to_item(node, struct audio_usecase, list);
+ curr_out = (struct stream_out*) uc->stream.out;
+ if (curr_out && PCM_PLAYBACK == uc->type) {
+ usecase_backend_idx = platform_get_backend_index(uc->out_snd_device);
+
+ if((new_backend_idx == HEADPHONE_BACKEND) &&
+ ((usecase_backend_idx == HEADPHONE_44_1_BACKEND) ||
+ (usecase_backend_idx == DSD_NATIVE_BACKEND))) {
+ ALOGD("%s:DSD or native stream detected enabling asrcmode in hardware",
+ __func__);
+ enable_asrc_mode(adev);
+ break;
+ } else if(((new_backend_idx == HEADPHONE_44_1_BACKEND) ||
+ (new_backend_idx == DSD_NATIVE_BACKEND)) &&
+ (usecase_backend_idx == HEADPHONE_BACKEND)) {
+ ALOGD("%s:48K stream detected, disabling and enabling it with asrcmode in hardware",
+ __func__);
+ disable_audio_route(adev, uc);
+ disable_snd_device(adev, uc->out_snd_device);
+ // Apply true-high-quality-mode if DSD or > 44.1KHz or >=24-bit
+ if (new_backend_idx == DSD_NATIVE_BACKEND)
+ audio_route_apply_and_update_path(adev->audio_route,
+ "hph-true-highquality-mode");
+ else if ((new_backend_idx == HEADPHONE_44_1_BACKEND) &&
+ (curr_out->bit_width >= 24))
+ audio_route_apply_and_update_path(adev->audio_route,
+ "hph-highquality-mode");
+ enable_asrc_mode(adev);
+ enable_snd_device(adev, uc->out_snd_device);
+ enable_audio_route(adev, uc);
+ break;
+ }
+ }
+ }
+ }
+}
+
int pcm_ioctl(struct pcm *pcm, int request, ...)
{
va_list ap;
@@ -568,7 +784,6 @@
if (audio_extn_spkr_prot_is_enabled())
audio_extn_spkr_prot_calib_cancel(adev);
-
if (platform_can_enable_spkr_prot_on_device(snd_device) &&
audio_extn_spkr_prot_is_enabled()) {
if (platform_get_spkr_prot_acdb_id(snd_device) < 0) {
@@ -588,6 +803,13 @@
}
} else {
ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
+
+ if ((SND_DEVICE_OUT_BT_A2DP == snd_device) &&
+ (audio_extn_a2dp_start_playback() < 0)) {
+ ALOGE(" fail to configure A2dp control path ");
+ return -EINVAL;
+ }
+
/* due to the possibility of calibration overwrite between listen
and audio, notify listen hal before audio calibration is sent */
audio_extn_sound_trigger_update_device_status(snd_device,
@@ -644,6 +866,7 @@
if (adev->snd_dev_ref_cnt[snd_device] == 0) {
ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
+
if (platform_can_enable_spkr_prot_on_device(snd_device) &&
audio_extn_spkr_prot_is_enabled()) {
audio_extn_spkr_prot_stop_processing(snd_device);
@@ -656,7 +879,10 @@
audio_route_reset_and_update_path(adev->audio_route, device_name);
}
- if (snd_device == SND_DEVICE_OUT_HDMI)
+ if (SND_DEVICE_OUT_BT_A2DP == snd_device)
+ audio_extn_a2dp_stop_playback();
+
+ if (snd_device == SND_DEVICE_OUT_HDMI || snd_device == SND_DEVICE_OUT_DISPLAY_PORT)
adev->is_channel_status_set = false;
else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
adev->native_playback_enabled) {
@@ -665,6 +891,11 @@
audio_route_reset_and_update_path(adev->audio_route,
"true-native-mode");
adev->native_playback_enabled = false;
+ } else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
+ adev->asrc_mode_enabled) {
+ ALOGD("%s: %d: disabling asrc mode in hardware", __func__, __LINE__);
+ disable_asrc_mode(adev);
+ audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
}
audio_extn_dev_arbi_release(snd_device);
@@ -685,7 +916,7 @@
struct audio_usecase *usecase;
bool switch_device[AUDIO_USECASE_MAX];
int i, num_uc_to_switch = 0;
-
+ bool force_restart_session = false;
/*
* This function is to make sure that all the usecases that are active on
* the hardware codec backend are always routed to any one device that is
@@ -705,7 +936,15 @@
*/
bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info,
snd_device);
-
+ /* For a2dp device reconfigure all active sessions
+ * with new AFE encoder format based on a2dp state
+ */
+ if ((SND_DEVICE_OUT_BT_A2DP == snd_device ||
+ SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) &&
+ audio_extn_a2dp_is_force_device_switch()) {
+ force_routing = true;
+ force_restart_session = true;
+ }
ALOGD("%s:becf: force routing %d", __func__, force_routing);
/* Disable all the usecases on the shared backend other than the
@@ -724,9 +963,13 @@
platform_check_backends_match(snd_device, usecase->out_snd_device));
if (usecase->type != PCM_CAPTURE &&
usecase != uc_info &&
- (usecase->out_snd_device != snd_device || force_routing) &&
- usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND &&
- platform_check_backends_match(snd_device, usecase->out_snd_device)) {
+ (usecase->out_snd_device != snd_device || force_routing) &&
+ ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
+ (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
+ (usecase->devices & AUDIO_DEVICE_OUT_USB_DEVICE) ||
+ (force_restart_session)) &&
+ (platform_check_backends_match(snd_device, usecase->out_snd_device))) {
+
ALOGD("%s:becf: check_usecases (%s) is active on (%s) - disabling ..",
__func__, use_case_table[usecase->id],
platform_get_snd_device_name(usecase->out_snd_device));
@@ -874,6 +1117,12 @@
reset_hdmi_sink_caps(out);
+ /* Cache ext disp type */
+ if (platform_get_ext_disp_type(adev->platform) <= 0) {
+ ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret);
+ return -EINVAL;
+ }
+
switch (channels) {
case 8:
ALOGV("%s: HDMI supports 7.1 channels", __func__);
@@ -997,6 +1246,29 @@
return active;
}
+/*
+ * if native DSD playback active
+ */
+bool audio_is_dsd_native_stream_active(struct audio_device *adev)
+{
+ bool active = false;
+ struct listnode *node = NULL;
+ struct audio_usecase *uc = NULL;
+ struct stream_out *curr_out = NULL;
+
+ list_for_each(node, &adev->usecase_list) {
+ uc = node_to_item(node, struct audio_usecase, list);
+ curr_out = (struct stream_out*) uc->stream.out;
+
+ if (curr_out && PCM_PLAYBACK == uc->type &&
+ (DSD_NATIVE_BACKEND == platform_get_backend_index(uc->out_snd_device))) {
+ active = true;
+ ALOGV("%s:DSD playback is active", __func__);
+ break;
+ }
+ }
+ return active;
+}
static bool force_device_switch(struct audio_usecase *usecase)
{
@@ -1016,6 +1288,14 @@
}
}
+ // Force all a2dp output devices to reconfigure for proper AFE encode format
+ if((usecase->stream.out) &&
+ (usecase->stream.out->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) &&
+ audio_extn_a2dp_is_force_device_switch()) {
+ ALOGD("Force a2dp device switch to update new encoder config");
+ ret = true;
+ }
+
return ret;
}
@@ -1060,6 +1340,8 @@
get_usecase_id_from_usecase_type(adev, VOICE_CALL));
if ((vc_usecase) && (((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) ||
+ ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ (usecase->devices & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) ||
(usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
in_snd_device = vc_usecase->in_snd_device;
out_snd_device = vc_usecase->out_snd_device;
@@ -1067,7 +1349,8 @@
} else if (voice_extn_compress_voip_is_active(adev)) {
voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
- (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
+ ((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) &&
(voip_usecase->stream.out != adev->primary_output))) {
in_snd_device = voip_usecase->in_snd_device;
out_snd_device = voip_usecase->out_snd_device;
@@ -1169,6 +1452,8 @@
/* Enable new sound devices */
if (out_snd_device != SND_DEVICE_NONE) {
check_usecases_codec_backend(adev, usecase, out_snd_device);
+ if (platform_check_codec_asrc_support(adev->platform))
+ check_and_set_asrc_mode(adev, out_snd_device);
enable_snd_device(adev, out_snd_device);
}
@@ -1182,10 +1467,6 @@
out_snd_device,
in_snd_device);
enable_audio_route_for_voice_usecases(adev, usecase);
- /* Enable sidetone only if voice/voip call already exists */
- if (voice_is_call_state_active(adev) ||
- voice_extn_compress_voip_is_started(adev))
- voice_set_sidetone(adev, out_snd_device, true);
}
usecase->in_snd_device = in_snd_device;
@@ -1202,10 +1483,23 @@
usecase->stream.out->channel_mask,
&usecase->stream.out->app_type_cfg);
ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
+
+ /* Notify device change info to effect clients registered */
+ audio_extn_gef_notify_device_config(
+ usecase->stream.out->devices,
+ usecase->stream.out->channel_mask,
+ platform_get_snd_device_acdb_id(usecase->out_snd_device));
}
enable_audio_route(adev, usecase);
+ if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
+ /* Enable sidetone only if other voice/voip call already exists */
+ if (voice_is_call_state_active(adev) ||
+ voice_extn_compress_voip_is_started(adev))
+ voice_set_sidetone(adev, out_snd_device, true);
+ }
+
/* Applicable only on the targets that has external modem.
* Enable device command should be sent to modem only after
* enabling voice call mixer controls
@@ -1324,6 +1618,8 @@
if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
flags |= PCM_MMAP | PCM_NOIRQ;
pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+ } else if (in->realtime) {
+ flags |= PCM_MMAP | PCM_NOIRQ;
}
while (1) {
@@ -1354,6 +1650,13 @@
goto error_open;
}
+ register_in_stream(in);
+ if (in->realtime) {
+ ret = pcm_start(in->pcm);
+ if (ret < 0)
+ goto error_open;
+ }
+
audio_extn_perf_lock_release(&adev->perf_lock_handle);
ALOGD("%s: exit", __func__);
@@ -1614,193 +1917,6 @@
return 0;
}
-static bool allow_hdmi_channel_config(struct audio_device *adev,
- bool enable_passthru)
-{
- struct listnode *node;
- struct audio_usecase *usecase;
- bool ret = true;
-
- if (enable_passthru && !audio_extn_passthru_is_enabled()) {
- ret = false;
- goto exit;
- }
-
- if (audio_extn_passthru_is_active()) {
- ALOGI("%s: Compress audio passthrough is active,"
- "no HDMI config change allowed", __func__);
- ret = false;
- goto exit;
- }
-
- list_for_each(node, &adev->usecase_list) {
- usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
- /*
- * If voice call is already existing, do not proceed further to avoid
- * disabling/enabling both RX and TX devices, CSD calls, etc.
- * Once the voice call done, the HDMI channels can be configured to
- * max channels of remaining use cases.
- */
- if (usecase->id == USECASE_VOICE_CALL) {
- ALOGV("%s: voice call is active, no change in HDMI channels",
- __func__);
- ret = false;
- break;
- } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
- if (!enable_passthru) {
- ALOGV("%s: multi channel playback is active, "
- "no change in HDMI channels", __func__);
- ret = false;
- break;
- }
- } else if (is_offload_usecase(usecase->id) &&
- audio_channel_count_from_out_mask(usecase->stream.out->channel_mask) > 2) {
- if (!enable_passthru) {
- ALOGD("%s:multi-channel(%x) compress offload playback is active"
- ", no change in HDMI channels", __func__,
- usecase->stream.out->channel_mask);
- ret = false;
- break;
- }
- }
- }
- }
- ALOGV("allow hdmi config %d", ret);
-exit:
- return ret;
-}
-
-static int check_and_set_hdmi_config(struct audio_device *adev,
- uint32_t channels,
- uint32_t sample_rate,
- audio_format_t format,
- bool enable_passthru)
-{
- struct listnode *node;
- struct audio_usecase *usecase;
- int32_t factor = 1;
- bool config = false;
-
- ALOGV("%s channels %d sample_rate %d format:%x enable_passthru:%d",
- __func__, channels, sample_rate, format, enable_passthru);
-
- if (channels != adev->cur_hdmi_channels) {
- ALOGV("channel does not match current hdmi channels");
- config = true;
- }
-
- if (sample_rate != adev->cur_hdmi_sample_rate) {
- ALOGV("sample rate does not match current hdmi sample rate");
- config = true;
- }
-
- if (format != adev->cur_hdmi_format) {
- ALOGV("format does not match current hdmi format");
- config = true;
- }
-
- /* TBD - add check for bit width */
- if (!config) {
- ALOGV("No need to config hdmi");
- return 0;
- }
-
- if (enable_passthru &&
- (format == AUDIO_FORMAT_E_AC3)) {
- ALOGV("factor 4 for E_AC3 passthru");
- factor = 4;
- }
-
- platform_set_hdmi_config(adev->platform, channels, factor * sample_rate,
- enable_passthru);
- adev->cur_hdmi_channels = channels;
- adev->cur_hdmi_format = format;
- adev->cur_hdmi_sample_rate = sample_rate;
-
- /*
- * Deroute all the playback streams routed to HDMI so that
- * the back end is deactivated. Note that backend will not
- * be deactivated if any one stream is connected to it.
- */
- list_for_each(node, &adev->usecase_list) {
- usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->type == PCM_PLAYBACK &&
- usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
- disable_audio_route(adev, usecase);
- }
- }
-
- bool was_active = audio_extn_keep_alive_is_active();
- if (was_active)
- audio_extn_keep_alive_stop();
-
- /*
- * Enable all the streams disabled above. Now the HDMI backend
- * will be activated with new channel configuration
- */
- list_for_each(node, &adev->usecase_list) {
- usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->type == PCM_PLAYBACK &&
- usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
- enable_audio_route(adev, usecase);
- }
- }
-
- if (was_active)
- audio_extn_keep_alive_start();
-
- return 0;
-}
-
-/* called with out lock taken */
-static int check_and_set_hdmi_backend(struct stream_out *out)
-{
- struct audio_device *adev = out->dev;
- int ret;
- bool enable_passthru = false;
-
- if (!(out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL))
- return -1;
-
- ALOGV("%s usecase %s out->format:%x out->bit_width:%d", __func__, use_case_table[out->usecase],out->format,out->bit_width);
-
- if (is_offload_usecase(out->usecase) &&
- audio_extn_passthru_is_passthrough_stream(out)) {
- enable_passthru = true;
- ALOGV("%s : enable_passthru is set to true", __func__);
- }
-
- /* Check if change in HDMI channel config is allowed */
- if (!allow_hdmi_channel_config(adev, enable_passthru)) {
- return -EPERM;
- }
-
- if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
- uint32_t channels;
- ALOGV("Offload usecase, enable passthru %d", enable_passthru);
-
- if (enable_passthru) {
- audio_extn_passthru_on_start(out);
- audio_extn_passthru_update_stream_configuration(adev, out);
- }
-
- /* For pass through case, the backend should be configured as stereo */
- channels = enable_passthru ? DEFAULT_HDMI_OUT_CHANNELS :
- out->compr_config.codec->ch_in;
-
- ret = check_and_set_hdmi_config(adev, channels,
- out->sample_rate, out->format,
- enable_passthru);
- } else
- ret = check_and_set_hdmi_config(adev, out->config.channels,
- out->config.rate,
- out->format,
- false);
- return ret;
-}
-
-
static int stop_output_stream(struct stream_out *out)
{
int ret = 0;
@@ -1841,17 +1957,14 @@
ALOGV("Disable passthrough , reset mixer to pcm");
/* NO_PASSTHROUGH */
out->compr_config.codec->compr_passthr = 0;
-
audio_extn_passthru_on_stop(out);
audio_extn_dolby_set_dap_bypass(adev, DAP_STATE_ON);
}
/* Must be called after removing the usecase from list */
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
- check_and_set_hdmi_config(adev, DEFAULT_HDMI_OUT_CHANNELS,
- DEFAULT_HDMI_OUT_SAMPLE_RATE,
- DEFAULT_HDMI_OUT_FORMAT,
- false);
+ audio_extn_keep_alive_start();
+
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
}
@@ -1893,12 +2006,6 @@
goto error_config;
}
- /* This must be called before adding this usecase to the list */
- if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
- /* This call can fail if compress pass thru is already active */
- check_and_set_hdmi_backend(out);
- }
-
uc_info->id = out->usecase;
uc_info->type = PCM_PLAYBACK;
uc_info->stream.out = out;
@@ -1910,6 +2017,16 @@
audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
adev->perf_lock_opts,
adev->perf_lock_opts_size);
+
+ if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ audio_extn_keep_alive_stop();
+ if (audio_extn_passthru_is_enabled() &&
+ audio_extn_passthru_is_passthrough_stream(out)) {
+ audio_extn_passthru_on_start(out);
+ audio_extn_passthru_update_stream_configuration(adev, out);
+ }
+ }
+
select_devices(adev, out->usecase);
ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
@@ -1920,6 +2037,8 @@
if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
flags |= PCM_MMAP | PCM_NOIRQ;
pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+ } else if (out->realtime) {
+ flags |= PCM_MMAP | PCM_NOIRQ;
} else
flags |= PCM_MONOTONIC;
@@ -2000,10 +2119,20 @@
audio_extn_check_and_set_dts_hpx_state(adev);
}
}
+
+ if (ret == 0) {
+ register_out_stream(out);
+ if (out->realtime) {
+ ret = pcm_start(out->pcm);
+ if (ret < 0)
+ goto error_open;
+ }
+ }
+
audio_extn_perf_lock_release(&adev->perf_lock_handle);
ALOGD("%s: exit", __func__);
- return 0;
+ return ret;
error_open:
audio_extn_perf_lock_release(&adev->perf_lock_handle);
stop_output_stream(out);
@@ -2086,6 +2215,35 @@
return size;
}
+static size_t get_output_period_size(uint32_t sample_rate,
+ audio_format_t format,
+ int channel_count,
+ int duration /*in millisecs*/)
+{
+ size_t size = 0;
+ uint32_t bytes_per_sample = audio_bytes_per_sample(format);
+
+ if ((duration == 0) || (sample_rate == 0) ||
+ (bytes_per_sample == 0) || (channel_count == 0)) {
+ ALOGW("Invalid config duration %d sr %d bps %d ch %d", duration, sample_rate,
+ bytes_per_sample, channel_count);
+ return -EINVAL;
+ }
+
+ size = (sample_rate *
+ duration *
+ bytes_per_sample *
+ channel_count) / 1000;
+ /*
+ * To have same PCM samples for all channels, the buffer size requires to
+ * be multiple of (number of channels * bytes per sample)
+ * For writes to succeed, the buffer must be written at address which is multiple of 32
+ */
+ size = ALIGN(size, (bytes_per_sample * channel_count * 32));
+
+ return (size/(channel_count * bytes_per_sample));
+}
+
static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out)
{
uint64_t actual_frames_rendered = 0;
@@ -2141,7 +2299,7 @@
else if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)
return out->hal_fragment_size;
- return out->config.period_size *
+ return out->config.period_size * out->af_period_multiplier *
audio_stream_out_frame_size((const struct audio_stream_out *)stream);
}
@@ -2172,13 +2330,6 @@
ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
stream, out->usecase, use_case_table[out->usecase]);
- if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
- /* Ignore standby in case of voip call because the voip output
- * stream is closed in adev_close_output_stream()
- */
- ALOGD("%s: Ignore Standby in VOIP call", __func__);
- return 0;
- }
lock_output_stream(out);
if (!out->standby) {
@@ -2190,7 +2341,13 @@
pthread_mutex_lock(&adev->lock);
out->standby = true;
- if (!is_offload_usecase(out->usecase)) {
+ if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
+ voice_extn_compress_voip_close_output_stream(stream);
+ pthread_mutex_unlock(&adev->lock);
+ pthread_mutex_unlock(&out->lock);
+ ALOGD("VOIP output entered standby");
+ return 0;
+ } else if (!is_offload_usecase(out->usecase)) {
if (out->pcm) {
pcm_close(out->pcm);
out->pcm = NULL;
@@ -2284,6 +2441,17 @@
(platform_get_edid_info(adev->platform) != 0) /* HDMI disconnected */) {
val = AUDIO_DEVICE_OUT_SPEAKER;
}
+ /*
+ * When A2DP is disconnected the
+ * music playback is paused and the policy manager sends routing=0
+ * But the audioflingercontinues to write data until standby time
+ * (3sec). As BT is turned off, the write gets blocked.
+ * Avoid this by routing audio to speaker until standby.
+ */
+ if ((out->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) &&
+ (val == AUDIO_DEVICE_NONE)) {
+ val = AUDIO_DEVICE_OUT_SPEAKER;
+ }
/*
* select_devices() call below switches all the usecases on the same
@@ -2304,15 +2472,9 @@
* playback to headset.
*/
if (val != 0) {
- out->devices = val;
-
- if (!out->standby) {
- audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
- adev->perf_lock_opts,
- adev->perf_lock_opts_size);
- select_devices(adev, out->usecase);
- audio_extn_perf_lock_release(&adev->perf_lock_handle);
- }
+ audio_devices_t new_dev = val;
+ bool same_dev = out->devices == new_dev;
+ out->devices = new_dev;
if (output_drives_call(adev, out)) {
if(!voice_is_in_call(adev)) {
@@ -2325,6 +2487,21 @@
voice_update_devices_for_all_voice_usecases(adev);
}
}
+
+ if (!out->standby) {
+ if (!same_dev) {
+ ALOGV("update routing change");
+ audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
+ adev->perf_lock_opts,
+ adev->perf_lock_opts_size);
+ if (adev->adm_on_routing_change)
+ adev->adm_on_routing_change(adev->adm_data,
+ out->handle);
+ }
+ select_devices(adev, out->usecase);
+ if (!same_dev)
+ audio_extn_perf_lock_release(&adev->perf_lock_handle);
+ }
}
pthread_mutex_unlock(&adev->lock);
@@ -2478,11 +2655,21 @@
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
+ uint32_t period_ms;
struct stream_out *out = (struct stream_out *)stream;
uint32_t latency = 0;
if (is_offload_usecase(out->usecase)) {
latency = COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
+ } else if (out->realtime) {
+ // since the buffer won't be filled up faster than realtime,
+ // return a smaller number
+ if (out->config.rate)
+ period_ms = (out->af_period_multiplier * out->config.period_size *
+ 1000) / (out->config.rate);
+ else
+ period_ms = 0;
+ latency = period_ms + platform_render_latency(out->usecase)/1000;
} else {
latency = (out->config.period_count * out->config.period_size * 1000) /
(out->config.rate);
@@ -2492,6 +2679,17 @@
return latency;
}
+static float AmpToDb(float amplification)
+{
+ float db = DSD_VOLUME_MIN_DB;
+ if (amplification > 0) {
+ db = 20 * log10(amplification);
+ if(db < DSD_VOLUME_MIN_DB)
+ return DSD_VOLUME_MIN_DB;
+ }
+ return db;
+}
+
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
@@ -2510,6 +2708,20 @@
* Mute is 0 and unmute 1
*/
audio_extn_passthru_set_volume(out, (left == 0.0f));
+ } else if (out->format == AUDIO_FORMAT_DSD){
+ char mixer_ctl_name[128] = "DSD Volume";
+ struct audio_device *adev = out->dev;
+ struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+ volume[0] = (int)(AmpToDb(left));
+ volume[1] = (int)(AmpToDb(right));
+ mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
+ return 0;
} else {
char mixer_ctl_name[128];
struct audio_device *adev = out->dev;
@@ -2556,8 +2768,11 @@
/* increase written size during SSR to avoid mismatch
* with the written frames count in AF
*/
- if (audio_bytes_per_sample(out->format) != 0)
- out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
+ // bytes per frame
+ size_t bpf = audio_bytes_per_sample(out->format) *
+ audio_channel_count_from_out_mask(out->channel_mask);
+ if (bpf != 0)
+ out->written += bytes / bpf;
ALOGD(" %s: sound card is not active/SSR state", __func__);
ret= -EIO;
goto exit;
@@ -2565,9 +2780,10 @@
}
if (audio_extn_passthru_should_drop_data(out)) {
- ALOGD(" %s : Drop data as compress passthrough session is going on", __func__);
- usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
- out_get_sample_rate(&out->stream.common));
+ ALOGV(" %s : Drop data as compress passthrough session is going on", __func__);
+ if (audio_bytes_per_sample(out->format) != 0)
+ out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
+ ret = -EIO;
goto exit;
}
@@ -2589,9 +2805,6 @@
ALOGD("%s: retry previous failed cal level set", __func__);
audio_hw_send_gain_dep_calibration(last_known_cal_step);
}
-
- if (!is_offload_usecase(out->usecase) && adev->adm_register_output_stream)
- adev->adm_register_output_stream(adev->adm_data, out->handle, out->flags);
}
if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
@@ -2647,7 +2860,8 @@
if (ret < 0)
ret = -errno;
ALOGVV("%s: writing buffer (%zu bytes) to compress device returned %zd", __func__, bytes, ret);
- if (ret >= 0 && ret < (ssize_t)bytes) {
+ /*msg to cb thread only if non blocking write is enabled*/
+ if (ret >= 0 && ret < (ssize_t)bytes && out->non_blocking) {
ALOGD("No space available in compress driver, post msg to cb thread");
send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
} else if (-ENETRESET == ret) {
@@ -2660,8 +2874,14 @@
if ( ret == (ssize_t)bytes && !out->non_blocking)
out->written += bytes;
- if (!out->playback_started && ret >= 0) {
- compress_start(out->compr);
+ /* Call compr start only when non-zero bytes of data is there to be rendered */
+ if (!out->playback_started && ret > 0) {
+ int status = compress_start(out->compr);
+ if (status < 0) {
+ ret = status;
+ ALOGE("%s: compr start failed with err %d", __func__, errno);
+ goto exit;
+ }
audio_extn_dts_eagle_fade(adev, true, out);
out->playback_started = 1;
out->offload_state = OFFLOAD_STATE_PLAYING;
@@ -2679,12 +2899,19 @@
ALOGVV("%s: writing buffer (%zu bytes) to pcm device", __func__, bytes);
- if (adev->adm_request_focus)
- adev->adm_request_focus(adev->adm_data, out->handle);
+ long ns = 0;
- if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
+ if (out->config.rate)
+ ns = pcm_bytes_to_frames(out->pcm, bytes)*1000000000LL/
+ out->config.rate;
+
+ bool use_mmap = is_mmap_usecase(out->usecase) || out->realtime;
+
+ request_out_focus(out, ns);
+
+ if (use_mmap)
ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
- } else if (out->hal_op_format != out->hal_ip_format &&
+ else if (out->hal_op_format != out->hal_ip_format &&
out->convert_buffer != NULL) {
memcpy_by_audio_format(out->convert_buffer,
@@ -2701,15 +2928,14 @@
ret = pcm_write(out->pcm, (void *)buffer, bytes);
}
+ release_out_focus(out);
+
if (ret < 0)
ret = -errno;
else if (ret == 0 && (audio_bytes_per_sample(out->format) != 0))
out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
else
ret = -EINVAL;
-
- if (adev->adm_abandon_focus)
- adev->adm_abandon_focus(adev->adm_data, out->handle);
}
}
@@ -2732,8 +2958,9 @@
out->standby = true;
}
out_standby(&out->stream.common);
- usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
- out_get_sample_rate(&out->stream.common));
+ if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
+ usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
+ out_get_sample_rate(&out->stream.common));
}
return bytes;
}
@@ -2939,7 +3166,6 @@
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
pthread_mutex_lock(&out->dev->lock);
ALOGV("offload resume, check and set hdmi backend again");
- check_and_set_hdmi_backend(out);
pthread_mutex_unlock(&out->dev->lock);
}
status = compress_resume(out->compr);
@@ -2979,8 +3205,12 @@
if (is_offload_usecase(out->usecase)) {
ALOGD("copl(%p):calling compress flush", out);
lock_output_stream(out);
- stop_compressed_output_l(out);
- out->written = 0;
+ if (out->offload_state == OFFLOAD_STATE_PAUSED) {
+ stop_compressed_output_l(out);
+ out->written = 0;
+ } else {
+ ALOGW("%s called in invalid state %d", __func__, out->offload_state);
+ }
pthread_mutex_unlock(&out->lock);
ALOGD("copl(%p):out of compress flush", out);
return 0;
@@ -3011,8 +3241,8 @@
else if(audio_extn_compr_cap_usecase_supported(in->usecase))
return audio_extn_compr_cap_get_buffer_size(in->config.format);
- return in->config.period_size *
- audio_stream_in_frame_size((const struct audio_stream_in *)stream);
+ return in->config.period_size * in->af_period_multiplier *
+ audio_stream_in_frame_size((const struct audio_stream_in *)stream);
}
static uint32_t in_get_channels(const struct audio_stream *stream)
@@ -3043,14 +3273,6 @@
ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
stream, in->usecase, use_case_table[in->usecase]);
- if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
- /* Ignore standby in case of voip call because the voip input
- * stream is closed in adev_close_input_stream()
- */
- ALOGV("%s: Ignore Standby in VOIP call", __func__);
- return status;
- }
-
lock_input_stream(in);
if (!in->standby && in->is_st_session) {
ALOGD("%s: sound trigger pcm stop lab", __func__);
@@ -3064,11 +3286,16 @@
pthread_mutex_lock(&adev->lock);
in->standby = true;
- if (in->pcm) {
- pcm_close(in->pcm);
- in->pcm = NULL;
+ if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
+ voice_extn_compress_voip_close_input_stream(stream);
+ ALOGD("VOIP input entered standby");
+ } else {
+ if (in->pcm) {
+ pcm_close(in->pcm);
+ in->pcm = NULL;
+ }
+ status = stop_input_stream(in);
}
- status = stop_input_stream(in);
pthread_mutex_unlock(&adev->lock);
}
pthread_mutex_unlock(&in->lock);
@@ -3125,8 +3352,13 @@
if (((int)in->device != val) && (val != 0)) {
in->device = val;
/* If recording is in progress, change the tx device to new device */
- if (!in->standby && !in->is_st_session)
+ if (!in->standby && !in->is_st_session) {
+ ALOGV("update input routing change");
+ if (adev->adm_on_routing_change)
+ adev->adm_on_routing_change(adev->adm_data,
+ in->capture_handle);
ret = select_devices(adev, in->usecase);
+ }
}
}
@@ -3212,19 +3444,24 @@
goto exit;
}
in->standby = 0;
- if (adev->adm_register_input_stream)
- adev->adm_register_input_stream(adev->adm_data, in->capture_handle, in->flags);
}
- if (adev->adm_request_focus)
- adev->adm_request_focus(adev->adm_data, in->capture_handle);
+ // what's the duration requested by the client?
+ long ns = 0;
+
+ if (in->config.rate)
+ ns = pcm_bytes_to_frames(in->pcm, bytes)*1000000000LL/
+ in->config.rate;
+
+ request_in_focus(in, ns);
+ bool use_mmap = is_mmap_usecase(in->usecase) || in->realtime;
if (in->pcm) {
if (audio_extn_ssr_get_stream() == in) {
ret = audio_extn_ssr_read(stream, buffer, bytes);
} else if (audio_extn_compr_cap_usecase_supported(in->usecase)) {
ret = audio_extn_compr_cap_read(in, buffer, bytes);
- } else if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
+ } else if (use_mmap) {
ret = pcm_mmap_read(in->pcm, buffer, bytes);
} else {
ret = pcm_read(in->pcm, buffer, bytes);
@@ -3246,8 +3483,7 @@
}
}
- if (adev->adm_abandon_focus)
- adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
+ release_in_focus(in);
/*
* Instead of writing zeroes here, we could trust the hardware
@@ -3333,7 +3569,7 @@
return add_remove_audio_effect(stream, effect, false);
}
-static int adev_open_output_stream(struct audio_hw_device *dev,
+int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
@@ -3514,7 +3750,12 @@
else if (config->channel_mask) {
out->channel_mask = config->channel_mask;
config->offload_info.channel_mask = config->channel_mask;
+ } else {
+ ALOGE("out->channel_mask not set for OFFLOAD/DIRECT_PCM");
+ ret = -EINVAL;
+ goto error_open;
}
+
format = out->format = config->offload_info.format;
out->sample_rate = config->offload_info.sample_rate;
@@ -3531,7 +3772,7 @@
out->compr_config.codec->bit_rate =
config->offload_info.bit_rate;
out->compr_config.codec->ch_in =
- audio_channel_count_from_out_mask(config->channel_mask);
+ audio_channel_count_from_out_mask(out->channel_mask);
out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
out->bit_width = AUDIO_OUTPUT_BIT_WIDTH;
/*TODO: Do we need to change it for passthrough */
@@ -3609,19 +3850,30 @@
audio_extn_dts_create_state_notifier_node(out->usecase);
- create_offload_callback_thread(out);
ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
__func__, config->offload_info.version,
config->offload_info.bit_rate);
+ /* Check if DSD audio format is supported in codec
+ * and there is no active native DSD use case
+ */
+
+ if ((config->format == AUDIO_FORMAT_DSD) &&
+ (!platform_check_codec_dsd_support(adev->platform) ||
+ audio_is_dsd_native_stream_active(adev))) {
+ ret = -EINVAL;
+ goto error_open;
+ }
+
/* Disable gapless if any of the following is true
* passthrough playback
* AV playback
* Direct PCM playback
*/
if (audio_extn_passthru_is_passthrough_stream(out) ||
- config->offload_info.has_video ||
- out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ (config->format == AUDIO_FORMAT_DSD) ||
+ config->offload_info.has_video ||
+ out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
check_and_set_gapless_mode(adev, false);
} else
check_and_set_gapless_mode(adev, true);
@@ -3629,6 +3881,13 @@
if (audio_extn_passthru_is_passthrough_stream(out)) {
out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
}
+ if (config->format == AUDIO_FORMAT_DSD) {
+ out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
+ out->compr_config.codec->compr_passthr = PASSTHROUGH_DSD;
+ }
+
+ create_offload_callback_thread(out);
+
} else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
if (ret != 0) {
@@ -3659,15 +3918,35 @@
out->config = pcm_config_afe_proxy_playback;
adev->voice_tx_output = out;
} else {
+ unsigned int channels = 0;
+ /*Update config params to default if not set by the caller*/
+ if (config->sample_rate == 0)
+ config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ if (config->channel_mask == AUDIO_CHANNEL_NONE)
+ config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ if (config->format == AUDIO_FORMAT_DEFAULT)
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+
+ channels = audio_channel_count_from_out_mask(out->channel_mask);
+
if (out->flags & AUDIO_OUTPUT_FLAG_RAW) {
out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
- out->config = pcm_config_low_latency;
+ out->realtime = may_use_noirq_mode(adev, USECASE_AUDIO_PLAYBACK_ULL,
+ out->flags);
+ out->config = out->realtime ? pcm_config_rt : pcm_config_low_latency;
} else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
out->config = pcm_config_low_latency;
} else if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
out->config = pcm_config_deep_buffer;
+ out->config.period_size = get_output_period_size(config->sample_rate, out->format,
+ channels, DEEP_BUFFER_OUTPUT_PERIOD_DURATION);
+ if (out->config.period_size <= 0) {
+ ALOGE("Invalid configuration period size is not valid");
+ ret = -EINVAL;
+ goto error_open;
+ }
} else {
/* primary path is the default path selected if no other outputs are available/suitable */
out->usecase = USECASE_AUDIO_PLAYBACK_PRIMARY;
@@ -3679,7 +3958,7 @@
out->bit_width = format_to_bitwidth_table[out->hal_op_format] << 3;
out->config.rate = config->sample_rate;
out->sample_rate = out->config.rate;
- out->config.channels = audio_channel_count_from_out_mask(out->channel_mask);
+ out->config.channels = channels;
if (out->hal_ip_format != out->hal_op_format) {
uint32_t buffer_size = out->config.period_size *
format_to_bitwidth_table[out->hal_op_format] *
@@ -3748,6 +4027,7 @@
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->stream.get_presentation_position = out_get_presentation_position;
+ out->af_period_multiplier = out->realtime ? af_period_multiplier : 1;
out->standby = 1;
/* out->muted = false; by calloc() */
/* out->written = 0; by calloc() */
@@ -3776,7 +4056,7 @@
return ret;
}
-static void adev_close_output_stream(struct audio_hw_device *dev __unused,
+void adev_close_output_stream(struct audio_hw_device *dev __unused,
struct audio_stream_out *stream)
{
struct stream_out *out = (struct stream_out *)stream;
@@ -3942,10 +4222,16 @@
if (ret >= 0) {
val = atoi(value);
if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
- ALOGV("cache new edid");
+ ALOGV("cache new ext disp type and edid");
+ ret = platform_get_ext_disp_type(adev->platform);
+ if (ret < 0) {
+ ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret);
+ status = ret;
+ goto done;
+ }
platform_cache_edid(adev->platform);
} else if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
- (val & AUDIO_DEVICE_IN_USB_DEVICE)) {
+ !(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) {
/*
* Do not allow AFE proxy port usage by WFD source when USB headset is connected.
* Per AudioPolicyManager, USB device is higher priority than WFD.
@@ -3969,7 +4255,7 @@
ALOGV("invalidate cached edid");
platform_invalidate_hdmi_config(adev->platform);
} else if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
- (val & AUDIO_DEVICE_IN_USB_DEVICE)) {
+ !(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) {
ret = str_parms_get_str(parms, "card", value, sizeof(value));
if (ret >= 0) {
audio_extn_usb_remove_device(val, atoi(value));
@@ -3979,8 +4265,26 @@
}
}
+ ret = str_parms_get_str(parms,"reconfigA2dp", value, sizeof(value));
+ if (ret >= 0) {
+ struct audio_usecase *usecase;
+ struct listnode *node;
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if ((usecase->type == PCM_PLAYBACK) &&
+ (usecase->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP)){
+ ALOGD("reconfigure a2dp... forcing device switch");
+ lock_output_stream(usecase->stream.out);
+ audio_extn_a2dp_set_handoff_mode(true);
+ //force device switch to re configure encoder
+ select_devices(adev, usecase->id);
+ audio_extn_a2dp_set_handoff_mode(false);
+ pthread_mutex_unlock(&usecase->stream.out->lock);
+ break;
+ }
+ }
+ }
audio_extn_set_parameters(adev, parms);
-
done:
str_parms_destroy(parms);
pthread_mutex_unlock(&adev->lock);
@@ -4188,12 +4492,21 @@
#if LOW_LATENCY_CAPTURE_USE_CASE
in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
#endif
+ in->realtime = may_use_noirq_mode(adev, in->usecase, in->flags);
}
- in->config = pcm_config_audio_capture;
- in->config.rate = config->sample_rate;
+
in->format = config->format;
+ if (in->realtime) {
+ in->config = pcm_config_audio_capture_rt;
+ in->sample_rate = in->config.rate;
+ in->af_period_multiplier = af_period_multiplier;
+ } else {
+ in->config = pcm_config_audio_capture;
+ in->config.rate = config->sample_rate;
+ in->sample_rate = config->sample_rate;
+ in->af_period_multiplier = 1;
+ }
in->bit_width = 16;
- in->sample_rate = config->sample_rate;
if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
if (adev->mode != AUDIO_MODE_IN_CALL) {
@@ -4269,14 +4582,17 @@
}
}
- in->format = config->format;
in->config.channels = channel_count;
- frame_size = audio_stream_in_frame_size(&in->stream);
- buffer_size = get_input_buffer_size(config->sample_rate,
- config->format,
- channel_count,
- is_low_latency);
- in->config.period_size = buffer_size / frame_size;
+ if (!in->realtime) {
+ in->format = config->format;
+ frame_size = audio_stream_in_frame_size(&in->stream);
+ buffer_size = get_input_buffer_size(config->sample_rate,
+ config->format,
+ channel_count,
+ is_low_latency);
+ in->config.period_size = buffer_size / frame_size;
+ }
+
if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
(in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
(voice_extn_compress_voip_is_format_supported(in->format)) &&
@@ -4357,8 +4673,11 @@
if ((--audio_device_ref_count) == 0) {
audio_extn_sound_trigger_deinit(adev);
audio_extn_listen_deinit(adev);
+ if (audio_extn_qaf_is_enabled())
+ audio_extn_qaf_deinit();
audio_extn_utils_release_streams_output_cfg_list(&adev->streams_output_cfg_list);
audio_route_free(adev->audio_route);
+ audio_extn_gef_deinit();
free(adev->snd_dev_ref_cnt);
platform_deinit(adev->platform);
if (adev->adm_deinit)
@@ -4392,6 +4711,8 @@
static int adev_open(const hw_module_t *module, const char *name,
hw_device_t **device)
{
+ int ret;
+
ALOGD("%s: enter", __func__);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
@@ -4466,9 +4787,26 @@
ALOGE("%s: Failed to init platform data, aborting.", __func__);
*device = NULL;
pthread_mutex_unlock(&adev_init_lock);
+ pthread_mutex_destroy(&adev->lock);
+ pthread_mutex_destroy(&adev->snd_card_status.lock);
return -EINVAL;
}
+ if (audio_extn_qaf_is_enabled()) {
+ ret = audio_extn_qaf_init(adev);
+ if (ret < 0) {
+ free(adev);
+ ALOGE("%s: Failed to init platform data, aborting.", __func__);
+ *device = NULL;
+ pthread_mutex_unlock(&adev_init_lock);
+ pthread_mutex_destroy(&adev->lock);
+ return ret;
+ }
+
+ adev->device.open_output_stream = audio_extn_qaf_open_output_stream;
+ adev->device.close_output_stream = audio_extn_qaf_close_output_stream;
+ }
+
adev->snd_card_status.state = SND_CARD_STATE_ONLINE;
if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) {
@@ -4487,6 +4825,7 @@
}
audio_extn_listen_init(adev, adev->snd_card);
audio_extn_sound_trigger_init(adev);
+ audio_extn_gef_init(adev);
if (access(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, R_OK) == 0) {
adev->offload_effects_lib = dlopen(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, RTLD_NOW);
@@ -4535,6 +4874,14 @@
dlsym(adev->adm_lib, "adm_request_focus");
adev->adm_abandon_focus = (adm_abandon_focus_t)
dlsym(adev->adm_lib, "adm_abandon_focus");
+ adev->adm_set_config = (adm_set_config_t)
+ dlsym(adev->adm_lib, "adm_set_config");
+ adev->adm_request_focus_v2 = (adm_request_focus_v2_t)
+ dlsym(adev->adm_lib, "adm_request_focus_v2");
+ adev->adm_is_noirq_avail = (adm_is_noirq_avail_t)
+ dlsym(adev->adm_lib, "adm_is_noirq_avail");
+ adev->adm_on_routing_change = (adm_on_routing_change_t)
+ dlsym(adev->adm_lib, "adm_on_routing_change");
}
}
@@ -4566,6 +4913,16 @@
}
}
+ if (property_get("audio_hal.period_multiplier", value, NULL) > 0) {
+ af_period_multiplier = atoi(value);
+ if (af_period_multiplier < 0)
+ af_period_multiplier = 2;
+ else if (af_period_multiplier > 4)
+ af_period_multiplier = 4;
+
+ ALOGV("new period_multiplier = %d", af_period_multiplier);
+ }
+
adev->multi_offload_enable = property_get_bool("audio.offload.multiple.enabled", false);
pthread_mutex_unlock(&adev_init_lock);
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 8197fec..83ad54d 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -79,6 +79,11 @@
#define MAX_PERF_LOCK_OPTS 20
+typedef enum card_status_t {
+ CARD_STATUS_OFFLINE,
+ CARD_STATUS_ONLINE
+} card_status_t;
+
/* These are the supported use cases by the hardware.
* Each usecase is mapped to a specific PCM device.
* Refer to pcm_device_table[].
@@ -141,6 +146,8 @@
USECASE_AUDIO_PLAYBACK_AFE_PROXY,
USECASE_AUDIO_RECORD_AFE_PROXY,
+ USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE,
+
AUDIO_USECASE_MAX
};
@@ -162,6 +169,7 @@
OFFLOAD_CMD_DRAIN, /* send a full drain request to DSP */
OFFLOAD_CMD_PARTIAL_DRAIN, /* send a partial drain request to DSP */
OFFLOAD_CMD_WAIT_FOR_BUFFER, /* wait for buffer released by DSP */
+ OFFLOAD_CMD_ERROR, /* offload playback hit some error */
};
enum {
@@ -228,7 +236,13 @@
audio_format_t hal_op_format;
void *convert_buffer;
+ bool realtime;
+ int af_period_multiplier;
struct audio_device *dev;
+ void* qaf_stream_handle;
+ pthread_cond_t qaf_offload_cond;
+ pthread_t qaf_offload_thread;
+ struct listnode qaf_offload_cmd_list;
};
struct stream_in {
@@ -252,6 +266,8 @@
bool is_st_session_active;
int sample_rate;
int bit_width;
+ bool realtime;
+ int af_period_multiplier;
struct audio_device *dev;
};
@@ -308,6 +324,12 @@
typedef void (*adm_deregister_stream_t)(void *, audio_io_handle_t);
typedef void (*adm_request_focus_t)(void *, audio_io_handle_t);
typedef void (*adm_abandon_focus_t)(void *, audio_io_handle_t);
+typedef void (*adm_set_config_t)(void *, audio_io_handle_t,
+ struct pcm *,
+ struct pcm_config *);
+typedef void (*adm_request_focus_v2_t)(void *, audio_io_handle_t, long);
+typedef bool (*adm_is_noirq_avail_t)(void *, int, int, int);
+typedef void (*adm_on_routing_change_t)(void *, audio_io_handle_t);
struct audio_device {
struct audio_hw_device device;
@@ -361,6 +383,10 @@
adm_deregister_stream_t adm_deregister_stream;
adm_request_focus_t adm_request_focus;
adm_abandon_focus_t adm_abandon_focus;
+ adm_set_config_t adm_set_config;
+ adm_request_focus_v2_t adm_request_focus_v2;
+ adm_is_noirq_avail_t adm_is_noirq_avail;
+ adm_on_routing_change_t adm_on_routing_change;
void (*offload_effects_get_parameters)(struct str_parms *,
struct str_parms *);
@@ -371,6 +397,7 @@
int perf_lock_opts[MAX_PERF_LOCK_OPTS];
int perf_lock_opts_size;
bool native_playback_enabled;
+ bool asrc_mode_enabled;
};
int select_devices(struct audio_device *adev,
@@ -392,12 +419,24 @@
bool audio_is_true_native_stream_active(struct audio_device *adev);
+bool audio_is_dsd_native_stream_active(struct audio_device *adev);
+
int pcm_ioctl(struct pcm *pcm, int request, ...);
int get_snd_card_state(struct audio_device *adev);
audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
usecase_type_t type);
+int adev_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address __unused);
+void adev_close_output_stream(struct audio_hw_device *dev __unused,
+ struct audio_stream_out *stream);
+
#define LITERAL_TO_STRING(x) #x
#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
__FILE__ ":" LITERAL_TO_STRING(__LINE__)\
diff --git a/hal/msm8916/hw_info.c b/hal/msm8916/hw_info.c
index d9add29..28b2397 100644
--- a/hal/msm8916/hw_info.c
+++ b/hal/msm8916/hw_info.c
@@ -324,7 +324,8 @@
strstr(snd_card_name, "msm8909") || strstr(snd_card_name, "msm8952") ||
strstr(snd_card_name, "msm8976") || strstr(snd_card_name, "msm8953") ||
strstr(snd_card_name, "msm8937") || strstr(snd_card_name, "msm8917") ||
- strstr(snd_card_name, "msm8940") || strstr(snd_card_name, "msmfalcon")) {
+ strstr(snd_card_name, "msm8940") || strstr(snd_card_name, "msm8920") ||
+ strstr(snd_card_name, "msmfalcon")) {
ALOGV("8x16 - variant soundcard");
update_hardware_info_8x16(hw_info, snd_card_name);
} else {
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 500a28d..a5cc804 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -68,7 +68,6 @@
#define PLATFORM_INFO_XML_PATH_EXTCODEC "/system/etc/audio_platform_info_extcodec.xml"
#define LIB_ACDB_LOADER "libacdbloader.so"
-#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
#define CVD_VERSION_MIXER_CTL "CVD Version"
#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
@@ -119,6 +118,8 @@
#define AUDIO_PARAMETER_KEY_AUD_CALDATA "cal_data"
#define AUDIO_PARAMETER_KEY_AUD_CALRESULT "cal_result"
+#define AUDIO_PARAMETER_KEY_MONO_SPEAKER "mono_speaker"
+
/* Reload ACDB files from specified path */
#define AUDIO_PARAMETER_KEY_RELOAD_ACDB "reload_acdb"
@@ -192,8 +193,10 @@
typedef struct codec_backend_cfg {
uint32_t sample_rate;
uint32_t bit_width;
+ uint32_t channels;
char *bitwidth_mixer_ctl;
char *samplerate_mixer_ctl;
+ char *channels_mixer_ctl;
} codec_backend_cfg_t;
static native_audio_prop na_props = {0, 0, 0};
@@ -220,6 +223,7 @@
/* Vbat monitor related flags */
bool is_vbat_speaker;
bool gsm_mode_enabled;
+ int mono_speaker;
/* Audio calibration related functions */
void *acdb_handle;
int voice_feature_set;
@@ -241,8 +245,8 @@
struct csd_data *csd;
void *edid_info;
bool edid_valid;
+ int ext_disp_type;
codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
- codec_backend_cfg_t current_tx_backend_cfg[MAX_CODEC_TX_BACKENDS];
char ec_ref_mixer_path[64];
char codec_version[CODEC_VERSION_MAX_LENGTH];
int hw_dep_fd;
@@ -316,6 +320,7 @@
AFE_PROXY_RECORD_PCM_DEVICE},
[USECASE_AUDIO_RECORD_AFE_PROXY] = {AFE_PROXY_PLAYBACK_PCM_DEVICE,
AFE_PROXY_RECORD_PCM_DEVICE},
+ [USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = {MULTIMEDIA9_PCM_DEVICE, -1},
};
/* Array to store sound devices */
@@ -340,12 +345,19 @@
[SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
[SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = "wsa-voice-speaker",
[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = "vbat-voice-speaker",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2] = "voice-speaker-2",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA] = "wsa-voice-speaker-2",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = "vbat-voice-speaker-2",
[SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
[SND_DEVICE_OUT_VOICE_LINE] = "voice-line",
[SND_DEVICE_OUT_HDMI] = "hdmi",
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
+ [SND_DEVICE_OUT_DISPLAY_PORT] = "display-port",
+ [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = "speaker-and-display-port",
[SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
[SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+ [SND_DEVICE_OUT_BT_A2DP] = "bt-a2dp",
+ [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = "speaker-and-bt-a2dp",
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
@@ -362,8 +374,10 @@
[SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = "voice-speaker-protected",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = "voice-speaker-2-protected",
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = "speaker-protected-vbat",
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = "voice-speaker-protected-vbat",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = "voice-speaker-2-protected-vbat",
#ifdef RECORD_PLAY_CONCURRENCY
[SND_DEVICE_OUT_VOIP_HANDSET] = "voip-handset",
[SND_DEVICE_OUT_VOIP_SPEAKER] = "voip-speaker",
@@ -416,6 +430,8 @@
[SND_DEVICE_IN_HANDSET_STEREO_DMIC] = "handset-stereo-dmic-ef",
[SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = "speaker-stereo-dmic-ef",
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = "vi-feedback",
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = "vi-feedback-mono-1",
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = "vi-feedback-mono-2",
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = "voice-speaker-dmic-broadside",
[SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = "speaker-dmic-broadside",
[SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = "speaker-dmic-broadside",
@@ -458,13 +474,20 @@
[SND_DEVICE_OUT_VOICE_HANDSET] = 7,
[SND_DEVICE_OUT_VOICE_LINE] = 10,
[SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2] = 14,
[SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = 135,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA] = 135,
[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = 135,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = 135,
[SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
[SND_DEVICE_OUT_HDMI] = 18,
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
+ [SND_DEVICE_OUT_DISPLAY_PORT] = 18,
+ [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = 14,
[SND_DEVICE_OUT_BT_SCO] = 22,
[SND_DEVICE_OUT_BT_SCO_WB] = 39,
+ [SND_DEVICE_OUT_BT_A2DP] = 20,
+ [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = 14,
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
@@ -481,8 +504,10 @@
[SND_DEVICE_OUT_ANC_HANDSET] = 103,
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = 101,
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = 101,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = 101,
#ifdef RECORD_PLAY_CONCURRENCY
[SND_DEVICE_OUT_VOIP_HANDSET] = 133,
[SND_DEVICE_OUT_VOIP_SPEAKER] = 132,
@@ -534,6 +559,8 @@
[SND_DEVICE_IN_HANDSET_STEREO_DMIC] = 34,
[SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = 35,
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = 102,
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = 102,
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = 12,
[SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = 12,
[SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = 119,
@@ -580,12 +607,19 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_WSA)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_LINE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_DISPLAY_PORT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
{TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BT_A2DP)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
@@ -602,8 +636,10 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT)},
#ifdef RECORD_PLAY_CONCURRENCY
{TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_SPEAKER)},
@@ -654,6 +690,8 @@
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_FLUENCE_DMIC_AANC)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE)},
@@ -704,6 +742,7 @@
{TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK_AND_DOWNLINK)},
{TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO)},
{TO_NAME_INDEX(USECASE_AUDIO_SPKR_CALIB_TX)},
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE)},
};
#define NO_COLS 2
@@ -767,6 +806,7 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+#define ULL_PLATFORM_DELAY (6*1000LL)
static void update_codec_type(const char *snd_card_name) {
@@ -1194,6 +1234,8 @@
backend_tag_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("bt-sco-wb");
backend_tag_table[SND_DEVICE_OUT_HDMI] = strdup("hdmi");
backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("speaker-and-hdmi");
+ backend_tag_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("display-port");
+ backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("speaker-and-display-port");
backend_tag_table[SND_DEVICE_OUT_VOICE_TX] = strdup("afe-proxy");
backend_tag_table[SND_DEVICE_IN_VOICE_RX] = strdup("afe-proxy");
backend_tag_table[SND_DEVICE_OUT_AFE_PROXY] = strdup("afe-proxy");
@@ -1205,9 +1247,16 @@
backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("vbat-voice-speaker");
+ backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("vbat-voice-speaker-2");
+ backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
+ backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("SLIMBUS_0_RX-and-HDMI_RX");
+ hw_interface_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("DISPLAY_PORT_RX");
+ hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("SLIMBUS_0_RX-and-DISPLAY_PORT_RX");
+ hw_interface_table[SND_DEVICE_OUT_USB_HEADSET] = strdup("USB_AUDIO_RX");
+ hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = strdup("SLIMBUS_0_RX-and-USB_AUDIO_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_TX] = strdup("AFE_PCM_RX");
my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
@@ -1661,8 +1710,10 @@
my_data->slowtalk = false;
my_data->hd_voice = false;
my_data->edid_info = NULL;
+ my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
my_data->is_wsa_speaker = false;
my_data->hw_dep_fd = -1;
+ my_data->mono_speaker = SPKR_1;
property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
@@ -1847,6 +1898,9 @@
/* init usb */
audio_extn_usb_init(adev);
+ /*init a2dp*/
+ audio_extn_a2dp_init(adev);
+
/* Read one time ssr property */
audio_extn_ssr_update_enabled();
audio_extn_spkr_prot_init(adev);
@@ -1867,13 +1921,14 @@
if (idx == HEADPHONE_44_1_BACKEND)
my_data->current_backend_cfg[idx].sample_rate = OUTPUT_SAMPLING_RATE_44100;
my_data->current_backend_cfg[idx].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_CHANNELS;
+ if (idx > MAX_RX_CODEC_BACKENDS)
+ my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
+ my_data->current_backend_cfg[idx].bitwidth_mixer_ctl = NULL;
+ my_data->current_backend_cfg[idx].samplerate_mixer_ctl = NULL;
+ my_data->current_backend_cfg[idx].channels_mixer_ctl = NULL;
}
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].sample_rate =
- CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bit_width =
- CODEC_BACKEND_DEFAULT_BIT_WIDTH;
-
if (is_external_codec) {
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
strdup("SLIM_0_RX Format");
@@ -1890,9 +1945,9 @@
my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_6_RX SampleRate");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ my_data->current_backend_cfg[SLIMBUS_0_TX].bitwidth_mixer_ctl =
strdup("SLIM_0_TX Format");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ my_data->current_backend_cfg[SLIMBUS_0_TX].samplerate_mixer_ctl =
strdup("SLIM_0_TX SampleRate");
} else {
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
@@ -1900,21 +1955,38 @@
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
strdup("MI2S_RX SampleRate");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
strdup("MI2S_TX Format");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
strdup("MI2S_TX SampleRate");
}
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
+ strdup("USB_AUDIO_TX Format");
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].samplerate_mixer_ctl =
+ strdup("USB_AUDIO_TX SampleRate");
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].channels_mixer_ctl =
+ strdup("USB_AUDIO_TX Channels");
my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].bitwidth_mixer_ctl =
strdup("USB_AUDIO_RX Format");
my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].samplerate_mixer_ctl =
strdup("USB_AUDIO_RX SampleRate");
+ my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].channels_mixer_ctl =
+ strdup("USB_AUDIO_RX Channels");
my_data->current_backend_cfg[HDMI_RX_BACKEND].bitwidth_mixer_ctl =
strdup("HDMI_RX Bit Format");
my_data->current_backend_cfg[HDMI_RX_BACKEND].samplerate_mixer_ctl =
strdup("HDMI_RX SampleRate");
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].channels_mixer_ctl =
+ strdup("HDMI_RX Channels");
+
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].bitwidth_mixer_ctl =
+ strdup("Display Port RX Bit Format");
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].samplerate_mixer_ctl =
+ strdup("Display Port RX SampleRate");
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].channels_mixer_ctl =
+ strdup("Display Port RX Channels");
ret = audio_extn_utils_get_codec_version(snd_card_name,
my_data->adev->snd_card,
@@ -2032,7 +2104,8 @@
return;
}
- if((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
!(usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)) {
ALOGI("%s: Not adding vbat speaker device to non voice use cases", __func__);
return;
@@ -2420,12 +2493,23 @@
return ret;
}
+int codec_device_supports_native_playback(audio_devices_t out_device)
+{
+ int ret = false;
-static int platform_get_backend_index(snd_device_t snd_device)
+ if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+ out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+ out_device & AUDIO_DEVICE_OUT_LINE)
+ ret = true;
+
+ return ret;
+}
+
+int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
- if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+ if (snd_device >= SND_DEVICE_OUT_BEGIN && snd_device < SND_DEVICE_OUT_END) {
if (backend_tag_table[snd_device] != NULL) {
if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
sizeof("headphones-44.1")) == 0)
@@ -2435,15 +2519,22 @@
port = HEADPHONE_BACKEND;
else if (strcmp(backend_tag_table[snd_device], "hdmi") == 0)
port = HDMI_RX_BACKEND;
+ else if (strcmp(backend_tag_table[snd_device], "display-port") == 0)
+ port = DISP_PORT_RX_BACKEND;
else if (strcmp(backend_tag_table[snd_device], "usb-headphones") == 0)
port = USB_AUDIO_RX_BACKEND;
}
+ } else if (snd_device >= SND_DEVICE_IN_BEGIN && snd_device < SND_DEVICE_IN_END) {
+ port = DEFAULT_CODEC_TX_BACKEND;
+ if (backend_tag_table[snd_device] != NULL) {
+ if (strcmp(backend_tag_table[snd_device], "usb-headset-mic") == 0)
+ port = USB_AUDIO_TX_BACKEND;
+ }
} else {
- ALOGV("%s:napb: Invalid device - %d ", __func__, snd_device);
+ ALOGW("%s:napb: Invalid device - %d ", __func__, snd_device);
}
- ALOGV("%s:napb: backend port - %d device - %d ", __func__, port,
- snd_device);
+ ALOGV("%s:napb: backend port - %d device - %d ", __func__, port, snd_device);
return port;
}
@@ -2526,7 +2617,9 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
audio_extn_spkr_prot_is_enabled()) {
if (my_data->is_vbat_speaker)
acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2561,9 +2654,18 @@
if (my_data->acdb_send_voice_cal == NULL) {
ALOGE("%s: dlsym error for acdb_send_voice_call", __func__);
} else {
- if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
- audio_extn_spkr_prot_is_enabled())
- out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ if (audio_extn_spkr_prot_is_enabled()) {
+ if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
+ }
acdb_rx_id = acdb_device_table[out_snd_device];
acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2590,7 +2692,9 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
audio_extn_spkr_prot_is_enabled()) {
if (my_data->is_vbat_speaker)
acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2797,12 +2901,29 @@
new_snd_devices[1] = SND_DEVICE_OUT_HDMI;
status = true;
+ } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT &&
+ !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_DISPLAY_PORT)) {
+ *num_devices = 2;
+
+ if (my_data->is_vbat_speaker)
+ new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_VBAT;
+ else if (my_data->is_wsa_speaker)
+ new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_WSA;
+ else
+ new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+
+ new_snd_devices[1] = SND_DEVICE_OUT_DISPLAY_PORT;
+ status = true;
} else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET &&
!platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_USB_HEADSET)) {
*num_devices = 2;
new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
new_snd_devices[1] = SND_DEVICE_OUT_USB_HEADSET;
status = true;
+ } else if (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+ new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
}
ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
@@ -2811,6 +2932,42 @@
return status;
}
+int platform_get_ext_disp_type(void *platform)
+{
+ int disp_type;
+ struct platform_data *my_data = (struct platform_data *)platform;
+
+ if (my_data->ext_disp_type != EXT_DISPLAY_TYPE_NONE) {
+ ALOGD("%s: Returning cached ext disp type:%s",
+ __func__, (my_data->ext_disp_type == EXT_DISPLAY_TYPE_DP) ? "DisplayPort" : "HDMI");
+ return my_data->ext_disp_type;
+ }
+
+#ifdef DISPLAY_PORT_ENABLED
+ struct audio_device *adev = my_data->adev;
+ struct mixer_ctl *ctl;
+ char *mixer_ctl_name = "External Display Type";
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ disp_type = mixer_ctl_get_value(ctl, 0);
+ if (disp_type == EXT_DISPLAY_TYPE_NONE) {
+ ALOGE("%s: Invalid external display type: %d", __func__, disp_type);
+ return -EINVAL;
+ }
+#else
+ disp_type = EXT_DISPLAY_TYPE_HDMI;
+#endif
+ my_data->ext_disp_type = disp_type;
+ ALOGD("%s: ext disp type:%s", __func__, (disp_type == EXT_DISPLAY_TYPE_DP) ? "DisplayPort" : "HDMI");
+ return disp_type;
+}
+
snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -2868,13 +3025,26 @@
snd_device = SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES;
} else if (devices == (AUDIO_DEVICE_OUT_AUX_DIGITAL |
AUDIO_DEVICE_OUT_SPEAKER)) {
- snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ snd_device = SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT;
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ goto exit;
+ }
} else if (devices == (AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
AUDIO_DEVICE_OUT_SPEAKER)) {
snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
} else if (devices == (AUDIO_DEVICE_OUT_USB_DEVICE |
AUDIO_DEVICE_OUT_SPEAKER)) {
snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
+ } else if ((devices & AUDIO_DEVICE_OUT_SPEAKER) &&
+ (devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
+ snd_device = SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP;
} else {
ALOGE("%s: Invalid combo device(%#x)", __func__, devices);
goto exit;
@@ -2925,13 +3095,25 @@
snd_device = SND_DEVICE_OUT_BT_SCO_WB;
else
snd_device = SND_DEVICE_OUT_BT_SCO;
+ } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ snd_device = SND_DEVICE_OUT_BT_A2DP;
} else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
- if (my_data->is_vbat_speaker)
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
- else if (my_data->is_wsa_speaker)
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_WSA;
- else
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ if (my_data->is_vbat_speaker) {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT;
+ } else if (my_data->is_wsa_speaker) {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_WSA;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA;
+ } else {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2;
+ }
} else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
snd_device = SND_DEVICE_OUT_USB_HEADSET;
@@ -3007,7 +3189,19 @@
else
snd_device = SND_DEVICE_OUT_BT_SCO;
} else if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
- snd_device = SND_DEVICE_OUT_HDMI ;
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ snd_device = SND_DEVICE_OUT_HDMI;
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ snd_device = SND_DEVICE_OUT_DISPLAY_PORT;
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ goto exit;
+ }
+ } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ snd_device = SND_DEVICE_OUT_BT_A2DP;
} else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
ALOGD("%s: setting USB hadset channel capability(2) for Proxy", __func__);
@@ -3430,7 +3624,7 @@
struct audio_device *adev = my_data->adev;
struct mixer_ctl *ctl;
const char *channel_cnt_str = NULL;
- const char *mixer_ctl_name = "HDMI_RX Channels";
+ char *mixer_ctl_name;
switch (channel_count) {
case 8:
channel_cnt_str = "Eight"; break;
@@ -3447,13 +3641,25 @@
default:
channel_cnt_str = "Two"; break;
}
+
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ mixer_ctl_name = "HDMI_RX Channels";
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ mixer_ctl_name = "Display Port RX Channels";
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ return -EINVAL;
+ }
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
- ALOGV("HDMI channel count: %s", channel_cnt_str);
+ ALOGV("Ext disp channel count: %s", channel_cnt_str);
mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
return 0;
}
@@ -3549,6 +3755,8 @@
char *kv_pairs = NULL;
kv_pairs = str_parms_to_str(parms);
+ if(!kv_pairs)
+ return ret;
len = strlen(kv_pairs);
ALOGV("%s: enter: - %s", __func__, kv_pairs);
free(kv_pairs);
@@ -3611,6 +3819,16 @@
}
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER, value, len);
+ if (err >= 0) {
+ if (!strncmp("left", value, sizeof("left")))
+ my_data->mono_speaker = SPKR_1;
+ else if (!strncmp("right", value, sizeof("right")))
+ my_data->mono_speaker = SPKR_2;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER);
+ }
+
#ifdef RECORD_PLAY_CONCURRENCY
err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_REC_PLAY_CONC, value, sizeof(value));
if (err >= 0) {
@@ -3787,8 +4005,8 @@
!strncmp("true", propValue, 4);
}
- if (prop_playback_enabled && (voice_is_in_call(my_data->adev) ||
- (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev)))) {
+ if ((prop_playback_enabled && (voice_is_in_call(my_data->adev))) ||
+ (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev))) {
char *decoder_mime_type = value;
//check if unsupported mime type or not
@@ -3814,6 +4032,13 @@
free(kv_pairs);
}
+unsigned char* platform_get_license(void *platform __unused, int *size __unused)
+{
+ ALOGE("%s: Not implemented", __func__);
+ return NULL;
+}
+
+
/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
@@ -3825,6 +4050,8 @@
case USECASE_AUDIO_PLAYBACK_OFFLOAD:
case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
return PCM_OFFLOAD_PLATFORM_DELAY;
+ case USECASE_AUDIO_PLAYBACK_ULL:
+ return ULL_PLATFORM_DELAY;
default:
return 0;
}
@@ -3845,7 +4072,9 @@
if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
(snd_device < SND_DEVICE_IN_END) &&
(snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
- (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
needs_event = true;
return needs_event;
@@ -3908,7 +4137,9 @@
if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
(snd_device < SND_DEVICE_IN_END) &&
(snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
- (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
needs_event = true;
return needs_event;
@@ -4009,27 +4240,55 @@
return fragment_size;
}
+/*
+ * return backend_idx on which voice call is active
+ */
+static int platform_get_voice_call_backend(struct audio_device* adev)
+{
+ struct audio_usecase *uc = NULL;
+ struct listnode *node;
+ snd_device_t out_snd_device = SND_DEVICE_NONE;
+
+ int backend_idx = -1;
+
+ if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
+ list_for_each(node, &adev->usecase_list) {
+ uc = node_to_item(node, struct audio_usecase, list);
+ if (uc && (uc->type == VOICE_CALL || uc->type == VOIP_CALL) && uc->stream.out) {
+ out_snd_device = platform_get_output_snd_device(adev->platform, uc->stream.out);
+ backend_idx = platform_get_backend_index(out_snd_device);
+ break;
+ }
+ }
+ }
+ return backend_idx;
+}
/*
* configures afe with bit width and Sample Rate
*/
static int platform_set_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device, unsigned int bit_width,
- unsigned int sample_rate, audio_format_t format)
+ snd_device_t snd_device, struct audio_backend_cfg backend_cfg)
{
int ret = 0;
int backend_idx = DEFAULT_CODEC_BACKEND;
struct platform_data *my_data = (struct platform_data *)adev->platform;
+ unsigned int bit_width = backend_cfg.bit_width;
+ unsigned int sample_rate = backend_cfg.sample_rate;
+ unsigned int channels = backend_cfg.channels;
+ audio_format_t format = backend_cfg.format;
+ bool passthrough_enabled = backend_cfg.passthrough_enabled;
backend_idx = platform_get_backend_index(snd_device);
- ALOGI("%s:becf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
- __func__, bit_width, sample_rate, backend_idx,
+
+ ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d, backend_idx %d device (%s)",
+ __func__, bit_width, sample_rate, channels,backend_idx,
platform_get_snd_device_name(snd_device));
if (bit_width !=
my_data->current_backend_cfg[backend_idx].bit_width) {
- struct mixer_ctl *ctl;
+ struct mixer_ctl *ctl = NULL;
ctl = mixer_get_ctl_by_name(adev->mixer,
my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
if (!ctl) {
@@ -4045,7 +4304,7 @@
else
mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else if (bit_width == 32) {
- mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ mixer_ctl_set_enum_by_string(ctl, "S32_LE");
} else {
mixer_ctl_set_enum_by_string(ctl, "S16_LE");
}
@@ -4086,14 +4345,24 @@
rate_str = "KHZ_44P1";
break;
case 64000:
- case 88200:
case 96000:
rate_str = "KHZ_96";
break;
+ case 88200:
+ rate_str = "KHZ_88P2";
+ break;
case 176400:
+ rate_str = "KHZ_176P4";
+ break;
case 192000:
rate_str = "KHZ_192";
break;
+ case 352800:
+ rate_str = "KHZ_352P8";
+ break;
+ case 384000:
+ rate_str = "KHZ_384";
+ break;
default:
rate_str = "KHZ_48";
break;
@@ -4115,19 +4384,163 @@
mixer_ctl_set_enum_by_string(ctl, rate_str);
my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
}
+ if ((my_data->current_backend_cfg[backend_idx].channels_mixer_ctl) &&
+ (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+ struct mixer_ctl *ctl;
+ char *channel_cnt_str = NULL;
+
+ switch (channels) {
+ case 8:
+ channel_cnt_str = "Eight"; break;
+ case 7:
+ channel_cnt_str = "Seven"; break;
+ case 6:
+ channel_cnt_str = "Six"; break;
+ case 5:
+ channel_cnt_str = "Five"; break;
+ case 4:
+ channel_cnt_str = "Four"; break;
+ case 3:
+ channel_cnt_str = "Three"; break;
+ case 1:
+ channel_cnt_str = "One"; break;
+ case 2:
+ default:
+ channel_cnt_str = "Two"; break;
+ }
+
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+ if (!ctl) {
+ ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+ __func__,
+ my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+ return -EINVAL;
+ }
+ mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
+ my_data->current_backend_cfg[backend_idx].channels = channels;
+
+ if (backend_idx == HDMI_RX_BACKEND)
+ platform_set_edid_channels_configuration(adev->platform, channels);
+
+ ALOGD("%s:becf: afe: %s set to %s", __func__,
+ my_data->current_backend_cfg[backend_idx].channels_mixer_ctl, channel_cnt_str);
+ }
+
+ bool set_ext_disp_format = false;
+ char *ext_disp_format = NULL;
+
+ if (backend_idx == HDMI_RX_BACKEND) {
+ ext_disp_format = "HDMI RX Format";
+ set_ext_disp_format = true;
+ } else if (backend_idx == DISP_PORT_RX_BACKEND) {
+ ext_disp_format = "Display Port Rx Format";
+ set_ext_disp_format = true;
+ } else {
+ ALOGV("%s: Format doesnt have to be set", __func__);
+ }
+
+ if (set_ext_disp_format) {
+ struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, ext_disp_format);
+ if (!ctl) {
+ ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+ __func__, ext_disp_format);
+ return -EINVAL;
+ }
+
+ if (passthrough_enabled) {
+ ALOGD("%s:Ext display compress format", __func__);
+ mixer_ctl_set_enum_by_string(ctl, "Compr");
+ } else {
+ ALOGD("%s: Ext display PCM format", __func__);
+ mixer_ctl_set_enum_by_string(ctl, "LPCM");
+ }
+ }
return ret;
}
/*
+ *Validate the selected bit_width, sample_rate and channels using the edid
+ *of the connected sink device.
+ */
+static void platform_check_hdmi_backend_cfg(struct audio_device* adev,
+ struct audio_usecase* usecase,
+ int backend_idx,
+ struct audio_backend_cfg *hdmi_backend_cfg)
+{
+ unsigned int bit_width;
+ unsigned int sample_rate;
+ unsigned int channels, max_supported_channels = 0;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+ edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+ bool passthrough_enabled = false;
+
+ bit_width = hdmi_backend_cfg->bit_width;
+ sample_rate = hdmi_backend_cfg->sample_rate;
+ channels = hdmi_backend_cfg->channels;
+
+
+ ALOGI("%s:becf: HDMI: bitwidth %d, samplerate %d, channels %d"
+ ", usecase = %d", __func__, bit_width,
+ sample_rate, channels, usecase->id);
+
+ if (audio_extn_passthru_is_enabled() && audio_extn_passthru_is_active()
+ && (usecase->stream.out->compr_config.codec->compr_passthr != 0)) {
+ passthrough_enabled = true;
+ ALOGI("passthrough is enabled for this stream");
+ }
+
+ // For voice calls use default configuration i.e. 16b/48K, only applicable to
+ // default backend
+ if (!passthrough_enabled) {
+
+ max_supported_channels = platform_edid_get_max_channels(my_data);
+
+ //Check EDID info for supported samplerate
+ if (!edid_is_supported_sr(edid_info,sample_rate)) {
+ //reset to current sample rate
+ sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
+ }
+
+ //Check EDID info for supported bit width
+ if (!edid_is_supported_bps(edid_info,bit_width)) {
+ //reset to current sample rate
+ bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
+ }
+
+ if (channels > max_supported_channels)
+ channels = max_supported_channels;
+
+ } else {
+ /*During pass through set default bit width and channels*/
+ channels = DEFAULT_HDMI_OUT_CHANNELS;
+ if ((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
+ (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC))
+ sample_rate = sample_rate * 4 ;
+
+ bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ /* We force route so that the BE format can be set to Compr */
+ }
+
+ ALOGI("%s:becf: afe: HDMI backend: passthrough %d updated bit width: %d and sample rate: %d"
+ "channels %d", __func__, passthrough_enabled , bit_width,
+ sample_rate, channels);
+
+ hdmi_backend_cfg->bit_width = bit_width;
+ hdmi_backend_cfg->sample_rate = sample_rate;
+ hdmi_backend_cfg->channels = channels;
+ hdmi_backend_cfg->passthrough_enabled = passthrough_enabled;
+}
+
+/*
* goes through all the current usecases and picks the highest
* bitwidth & samplerate
*/
static bool platform_check_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase* usecase,
snd_device_t snd_device,
- unsigned int* new_bit_width,
- unsigned int* new_sample_rate)
+ struct audio_backend_cfg *backend_cfg)
{
bool backend_change = false;
struct listnode *node;
@@ -4135,24 +4548,25 @@
char value[PROPERTY_VALUE_MAX] = {0};
unsigned int bit_width;
unsigned int sample_rate;
+ unsigned int channels;
+ bool passthrough_enabled = false;
int backend_idx = DEFAULT_CODEC_BACKEND;
struct platform_data *my_data = (struct platform_data *)adev->platform;
- int na_mode = platform_get_native_support();
- edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+ bool channels_updated = false;
backend_idx = platform_get_backend_index(snd_device);
- bit_width = *new_bit_width;
- sample_rate = *new_sample_rate;
+ bit_width = backend_cfg->bit_width;
+ sample_rate = backend_cfg->sample_rate;
+ channels = backend_cfg->channels;
- ALOGI("%s:becf: afe: Codec selected backend: %d current bit width: %d and sample rate: %d",
- __func__, backend_idx, bit_width, sample_rate);
+ ALOGI("%s:becf: afe: Codec selected backend: %d current bit width: %d sample rate: %d channels: %d",
+ __func__, backend_idx, bit_width, sample_rate, channels);
// For voice calls use default configuration i.e. 16b/48K, only applicable to
// default backend
// force routing is not required here, caller will do it anyway
- if ((voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
- backend_idx == DEFAULT_CODEC_BACKEND) {
+ if (backend_idx == platform_get_voice_call_backend(adev)) {
ALOGW("%s:becf: afe:Use default bw and sr for voice/voip calls ",
__func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
@@ -4173,11 +4587,12 @@
uc = node_to_item(node, struct audio_usecase, list);
struct stream_out *out = (struct stream_out*) uc->stream.out;
if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+ unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
- "(%d) device %s", __func__, i++, use_case_table[uc->id],
+ "(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
uc->id, out->sample_rate,
- out->bit_width,
+ out->bit_width, out_channels,
platform_get_snd_device_name(uc->out_snd_device));
if (platform_check_backends_match(snd_device, uc->out_snd_device)) {
@@ -4187,75 +4602,87 @@
sample_rate = out->sample_rate;
if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ if (channels < out_channels)
+ channels = out_channels;
}
}
}
}
- if (audio_is_true_native_stream_active(adev)) {
- if (check_hdset_combo_device(snd_device)) {
- /*
- * In true native mode Tasha has a limitation that one port at 44.1 khz
- * cannot drive both spkr and hdset, to simiplify the solution lets
- * move the AFE to 48khzwhen a ring tone selects combo device.
- */
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
- ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
- __func__);
- } else {
- /*
- * in single BE mode, if native audio playback
- * is active then it will take priority
- */
- sample_rate = OUTPUT_SAMPLING_RATE_44100;
- ALOGD("%s:becf: afe: napb active set rate to 44.1 khz",
- __func__);
+ /* Native playback is preferred for Headphone/HS device over 192Khz */
+ if (codec_device_supports_native_playback(usecase->devices)) {
+ if (audio_is_true_native_stream_active(adev)) {
+ if (check_hdset_combo_device(snd_device)) {
+ /*
+ * In true native mode Tasha has a limitation that one port at 44.1 khz
+ * cannot drive both spkr and hdset, to simiplify the solution lets
+ * move the AFE to 48khzwhen a ring tone selects combo device.
+ * or if NATIVE playback is not enabled.
+ */
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
+ __func__);
+ } else {
+ /*
+ * in single BE mode, if native audio playback
+ * is active then it will take priority
+ */
+ sample_rate = OUTPUT_SAMPLING_RATE_44100;
+ ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
+ __func__);
+ }
+ } else if (OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
+ __func__);
}
- }
-
- /*
- * hifi playback not supported on spkr devices, limit the Sample Rate
- * to 48 khz.
- */
- if (SND_DEVICE_OUT_SPEAKER == snd_device ||
- SND_DEVICE_OUT_SPEAKER_WSA == snd_device ||
- SND_DEVICE_OUT_SPEAKER_VBAT == snd_device) {
+ } else if ((usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) ||
+ (usecase->devices & AUDIO_DEVICE_OUT_EARPIECE) ) {
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: playback on speaker device Configure afe to "
+
+ if (bit_width >= 24) {
+ bit_width = platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
+ ALOGD("%s:becf: afe: reset bitwidth to %d (based on supported"
+ " value for this platform)", __func__, bit_width);
+ }
+
+ ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
"default Sample Rate(48k)", __func__);
}
/*
- * native playback is not enabled.Configure afe to default Sample Rate(48k)
- */
- if (NATIVE_AUDIO_MODE_INVALID == na_mode &&
- OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
- __func__);
- }
-
- /*
* reset the sample rate to default value(48K), if hifi audio is not supported
*/
- if (!my_data->hifi_audio) {
+ if (!my_data->hifi_audio && (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) {
ALOGD("%s:becf: afe: only 48KHZ sample rate is supported "
"Configure afe to default Sample Rate(48k)", __func__);
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
if (backend_idx == HDMI_RX_BACKEND) {
- //Check EDID info for supported samplerate
- if (!edid_is_supported_sr(edid_info,sample_rate)) {
- //reset to current sample rate
- sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
- }
- //Check EDID info for supported bit widhth
- if (!edid_is_supported_bps(edid_info,bit_width)) {
- //reset to current sample rate
- bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
- }
+ struct audio_backend_cfg hdmi_backend_cfg;
+ hdmi_backend_cfg.bit_width = bit_width;
+ hdmi_backend_cfg.sample_rate = sample_rate;
+ hdmi_backend_cfg.channels = channels;
+ hdmi_backend_cfg.passthrough_enabled = false;
+
+ /*HDMI does not support 384Khz/32bit playback hence configure BE to 24b/192Khz*/
+ /* TODO: Instead have the validation against edid return the next best match*/
+ if (bit_width > 24)
+ hdmi_backend_cfg.bit_width = 24;
+ if (sample_rate > 192000)
+ hdmi_backend_cfg.sample_rate = 192000;
+
+ platform_check_hdmi_backend_cfg(adev, usecase, backend_idx, &hdmi_backend_cfg);
+
+ bit_width = hdmi_backend_cfg.bit_width;
+ sample_rate = hdmi_backend_cfg.sample_rate;
+ channels = hdmi_backend_cfg.channels;
+ passthrough_enabled = hdmi_backend_cfg.passthrough_enabled;
+
+ if (channels != my_data->current_backend_cfg[backend_idx].channels)
+ channels_updated = true;
}
//check if mulitchannel clip needs to be down sampled to 48k
@@ -4274,10 +4701,11 @@
}
if (backend_idx == USB_AUDIO_RX_BACKEND) {
- unsigned int channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
- audio_extn_usb_is_config_supported(&bit_width, &sample_rate, channels);
+ audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, true);
ALOGV("%s: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
__func__, bit_width, sample_rate, channels);
+ if (channels != my_data->current_backend_cfg[backend_idx].channels)
+ channels_updated = true;
}
ALOGI("%s:becf: afe: Codec selected backend: %d updated bit width: %d and sample rate: %d",
@@ -4286,13 +4714,16 @@
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
- (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate)) {
- *new_bit_width = bit_width;
- *new_sample_rate = sample_rate;
+ (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+ passthrough_enabled || channels_updated) {
+ backend_cfg->bit_width = bit_width;
+ backend_cfg->sample_rate = sample_rate;
+ backend_cfg->channels = channels;
+ backend_cfg->passthrough_enabled = passthrough_enabled;
backend_change = true;
- ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d new sample rate: %d",
- __func__,
- *new_bit_width, *new_sample_rate);
+ ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d"
+ " new sample rate: %d new channels %d",__func__,
+ backend_cfg->bit_width, backend_cfg->sample_rate, backend_cfg->channels);
}
return backend_change;
@@ -4301,23 +4732,24 @@
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase *usecase, snd_device_t snd_device)
{
- unsigned int new_bit_width;
- unsigned int new_sample_rate;
int backend_idx = DEFAULT_CODEC_BACKEND;
int new_snd_devices[SND_DEVICE_OUT_END];
int i, num_devices = 1;
+ struct audio_backend_cfg backend_cfg;
bool ret = false;
- audio_format_t format;
backend_idx = platform_get_backend_index(snd_device);
- new_bit_width = usecase->stream.out->bit_width;
- new_sample_rate = usecase->stream.out->sample_rate;
- format = usecase->stream.out->format;
+ backend_cfg.bit_width = usecase->stream.out->bit_width;
+ backend_cfg.sample_rate = usecase->stream.out->sample_rate;
+ backend_cfg.format = usecase->stream.out->format;
+ backend_cfg.channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
+ /*this is populated by check_codec_backend_cfg hence set default value to false*/
+ backend_cfg.passthrough_enabled = false;
- ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
- ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
- new_sample_rate, backend_idx, usecase->id,
+ ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+ ", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
+ backend_cfg.sample_rate, backend_cfg.channels, backend_idx, usecase->id,
platform_get_snd_device_name(snd_device));
if (!platform_can_split_snd_device(adev->platform, snd_device,
@@ -4328,9 +4760,9 @@
ALOGI("%s: becf: new_snd_devices[%d] is %s", __func__, i,
platform_get_snd_device_name(new_snd_devices[i]));
if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
- &new_bit_width, &new_sample_rate)) {
+ &backend_cfg)) {
platform_set_codec_backend_cfg(adev, new_snd_devices[i],
- new_bit_width, new_sample_rate, format);
+ backend_cfg);
ret = true;
}
}
@@ -4339,143 +4771,26 @@
}
/*
- * configures afe with bit width and Sample Rate
- */
-
-int platform_set_capture_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- unsigned int bit_width, unsigned int sample_rate,
- audio_format_t format)
-{
- int ret = 0;
- int backend_idx = DEFAULT_CODEC_BACKEND;
- struct platform_data *my_data = (struct platform_data *)adev->platform;
-
- ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
- __func__, bit_width, sample_rate, backend_idx,
- platform_get_snd_device_name(snd_device));
-
- if (bit_width !=
- my_data->current_tx_backend_cfg[backend_idx].bit_width) {
-
- struct mixer_ctl *ctl = NULL;
- ctl = mixer_get_ctl_by_name(adev->mixer,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- if (!ctl) {
- ALOGE("%s:txbecf: afe: Could not get ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- return -EINVAL;
- }
-
- if (bit_width == 24) {
- if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
- ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
- else
- ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
- } else {
- ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
- }
-
- if (ret < 0) {
- ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- return -EINVAL;
- }
-
- my_data->current_tx_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:txbecf: afe: %s mixer set to %d bit", __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
- }
-
- /*
- * Backend sample rate configuration follows:
- * 16 bit record - 48khz for streams at any valid sample rate
- * 24 bit record - 48khz for stream sample rate less than 48khz
- * 24 bit record - 96khz for sample rate range of 48khz to 96khz
- * 24 bit record - 192khz for sample rate range of 96khz to 192 khz
- * Upper limit is inclusive in the sample rate range.
- */
- // TODO: This has to be more dynamic based on policy file
-
- if (sample_rate != my_data->current_tx_backend_cfg[(int)backend_idx].sample_rate) {
- /*
- * sample rate update is needed only for hifi audio enabled platforms
- */
- char *rate_str = NULL;
- struct mixer_ctl *ctl = NULL;
-
- switch (sample_rate) {
- case 8000:
- case 11025:
- case 16000:
- case 22050:
- case 32000:
- case 44100:
- case 48000:
- rate_str = "KHZ_48";
- break;
- case 64000:
- case 88200:
- case 96000:
- rate_str = "KHZ_96";
- break;
- case 176400:
- case 192000:
- rate_str = "KHZ_192";
- break;
- default:
- rate_str = "KHZ_48";
- break;
- }
-
- ctl = mixer_get_ctl_by_name(adev->mixer,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-
- if (ctl < 0) {
- ALOGE("%s:txbecf: afe: Could not get ctl to set the Sample Rate for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
- return -EINVAL;
- }
-
- ALOGD("%s:txbecf: afe: %s set to %s", __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl,
- rate_str);
- ret = mixer_ctl_set_enum_by_string(ctl, rate_str);
- if (ret < 0) {
- ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
- return -EINVAL;
- }
-
- my_data->current_tx_backend_cfg[backend_idx].sample_rate = sample_rate;
- }
-
- return ret;
-}
-
-/*
* goes through all the current usecases and picks the highest
* bitwidth & samplerate
*/
-bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
- unsigned int* new_bit_width,
- unsigned int* new_sample_rate)
+static bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
+ int backend_idx,
+ struct audio_backend_cfg *backend_cfg)
{
bool backend_change = false;
unsigned int bit_width;
unsigned int sample_rate;
- int backend_idx = DEFAULT_CODEC_BACKEND;
+ unsigned int channels;
struct platform_data *my_data = (struct platform_data *)adev->platform;
- bit_width = *new_bit_width;
- sample_rate = *new_sample_rate;
+ bit_width = backend_cfg->bit_width;
+ sample_rate = backend_cfg->sample_rate;
+ channels = backend_cfg->channels;
ALOGI("%s:txbecf: afe: Codec selected backend: %d current bit width: %d and "
- "sample rate: %d",__func__,backend_idx, bit_width, sample_rate);
+ "sample rate: %d, channels %d",__func__,backend_idx, bit_width,
+ sample_rate, channels);
// For voice calls use default configuration i.e. 16b/48K, only applicable to
// default backend
@@ -4487,18 +4802,27 @@
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
+ if (backend_idx == USB_AUDIO_TX_BACKEND) {
+ audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, false);
+ ALOGV("%s: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
+ __func__, bit_width, sample_rate, channels);
+ }
ALOGI("%s:txbecf: afe: Codec selected backend: %d updated bit width: %d and "
"sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
- if ((bit_width != my_data->current_tx_backend_cfg[backend_idx].bit_width) ||
- (sample_rate != my_data->current_tx_backend_cfg[backend_idx].sample_rate)) {
- *new_bit_width = bit_width;
- *new_sample_rate = sample_rate;
+ if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
+ (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+ (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+ backend_cfg->bit_width = bit_width;
+ backend_cfg->sample_rate= sample_rate;
+ backend_cfg->channels = channels;
backend_change = true;
ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
- "new sample rate: %d", __func__, *new_bit_width, *new_sample_rate);
+ "new sample rate: %d new channel: %d",
+ __func__, backend_cfg->bit_width,
+ backend_cfg->sample_rate, backend_cfg->channels);
}
return backend_change;
@@ -4507,29 +4831,34 @@
bool platform_check_and_set_capture_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase *usecase, snd_device_t snd_device)
{
- unsigned int new_bit_width;
- unsigned int new_sample_rate;
- audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
- int backend_idx = DEFAULT_CODEC_BACKEND;
+ int backend_idx = platform_get_backend_index(snd_device);
int ret = 0;
+ struct audio_backend_cfg backend_cfg;
+ backend_cfg.passthrough_enabled = false;
if(usecase->type == PCM_CAPTURE) {
- new_sample_rate = usecase->stream.in->sample_rate;
- new_bit_width = usecase->stream.in->bit_width;
- format = usecase->stream.in->format;
+ backend_cfg.sample_rate= usecase->stream.in->sample_rate;
+ backend_cfg.bit_width= usecase->stream.in->bit_width;
+ backend_cfg.format= usecase->stream.in->format;
+ backend_cfg.channels = audio_channel_count_from_in_mask(usecase->stream.in->channel_mask);
} else {
- new_bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
- new_sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ backend_cfg.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ backend_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ backend_cfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ backend_cfg.channels = 1;
}
- ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d"
- ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
- new_sample_rate, backend_idx, usecase->id,
+ ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, channel %d"
+ ", backend_idx %d usecase = %d device (%s)", __func__,
+ backend_cfg.bit_width,
+ backend_cfg.sample_rate,
+ backend_cfg.channels,
+ backend_idx, usecase->id,
platform_get_snd_device_name(snd_device));
- if (platform_check_capture_codec_backend_cfg(adev, &new_bit_width,
- &new_sample_rate)) {
- ret = platform_set_capture_codec_backend_cfg(adev, snd_device,
- new_bit_width, new_sample_rate, format);
+ if (platform_check_capture_codec_backend_cfg(adev, backend_idx,
+ &backend_cfg)) {
+ ret = platform_set_codec_backend_cfg(adev, snd_device,
+ backend_cfg);
if(!ret)
return true;
}
@@ -4692,7 +5021,7 @@
struct audio_device *adev = my_data->adev;
char block[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE];
int ret, count;
-
+ char *mix_ctl_name;
struct mixer_ctl *ctl;
char edid_data[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE + 1] = {0};
edid_audio_info *info;
@@ -4702,17 +5031,28 @@
return 0;
}
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ mix_ctl_name = "HDMI EDID";
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ mix_ctl_name = "Display Port EDID";
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ return -EINVAL;
+ }
+
if (my_data->edid_info == NULL) {
my_data->edid_info =
(struct edid_audio_info *)calloc(1, sizeof(struct edid_audio_info));
}
info = my_data->edid_info;
-
- ctl = mixer_get_ctl_by_name(adev->mixer, AUDIO_DATA_BLOCK_MIXER_CTL);
+ ctl = mixer_get_ctl_by_name(adev->mixer, mix_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
- __func__, AUDIO_DATA_BLOCK_MIXER_CTL);
+ __func__, mix_ctl_name);
goto fail;
}
@@ -4731,8 +5071,9 @@
}
edid_data[0] = count;
memcpy(&edid_data[1], block, count);
+
if (!edid_get_sink_caps(info, edid_data)) {
- ALOGE("%s: Failed to get HDMI sink capabilities", __func__);
+ ALOGE("%s: Failed to get extn disp sink capabilities", __func__);
goto fail;
}
my_data->edid_valid = true;
@@ -4751,16 +5092,28 @@
int platform_set_channel_allocation(void *platform, int channel_alloc)
{
struct mixer_ctl *ctl;
- const char *mixer_ctl_name = "HDMI RX CA";
+ char *mixer_ctl_name;
int ret;
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ mixer_ctl_name = "HDMI RX CA";
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ mixer_ctl_name = "Display Port RX CA";
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ return -EINVAL;
+ }
+
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
- ret = EINVAL;
+ return -EINVAL;
}
ALOGD(":%s channel allocation = 0x%x", __func__, channel_alloc);
ret = mixer_ctl_set_value(ctl, 0, channel_alloc);
@@ -4775,7 +5128,7 @@
int platform_set_channel_map(void *platform, int ch_count, char *ch_map, int snd_id)
{
struct mixer_ctl *ctl;
- char mixer_ctl_name[44]; // max length of name is 44 as defined
+ char mixer_ctl_name[44] = {0}; // max length of name is 44 as defined
int ret;
unsigned int i;
int set_values[8] = {0};
@@ -4991,15 +5344,25 @@
void platform_invalidate_hdmi_config(void * platform)
{
+ //reset ext display EDID info
struct platform_data *my_data = (struct platform_data *)platform;
my_data->edid_valid = false;
if (my_data->edid_info) {
memset(my_data->edid_info, 0, sizeof(struct edid_audio_info));
}
- //reset HDMI_RX_BACKEND to default values
- my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ if (my_data->ext_disp_type == EXT_DISPLAY_TYPE_HDMI) {
+ //reset HDMI_RX_BACKEND to default values
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ } else {
+ //reset Display port BACKEND to default values
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ }
+ my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
}
int platform_set_mixer_control(struct stream_out *out, const char * mixer_ctl_name,
@@ -5018,91 +5381,6 @@
return mixer_ctl_set_enum_by_string(ctl, mixer_val);
}
-static int set_mixer_control(struct mixer *mixer,
- const char * mixer_ctl_name,
- const char *mixer_val)
-{
- struct mixer_ctl *ctl;
- ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
- ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
- if (!ctl) {
- ALOGE("%s: could not get ctl for mixer cmd - %s",
- __func__, mixer_ctl_name);
- return -EINVAL;
- }
-
- return mixer_ctl_set_enum_by_string(ctl, mixer_val);
-}
-
-int platform_set_hdmi_config(void *platform, uint32_t channel_count,
- uint32_t sample_rate, bool enable_passthrough)
-{
- struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
- const char *hdmi_format_ctrl = "HDMI RX Format";
- const char *hdmi_rate_ctrl = "HDMI_RX SampleRate";
- const char *hdmi_chans_ctrl = "HDMI_RX Channels";
- const char *channel_cnt_str = NULL;
-
- ALOGI("%s ch[%d] sr[%d], pthru[%d]", __func__,
- channel_count, sample_rate, enable_passthrough);
-
- switch (channel_count) {
- case 8:
- channel_cnt_str = "Eight"; break;
- case 7:
- channel_cnt_str = "Seven"; break;
- case 6:
- channel_cnt_str = "Six"; break;
- case 5:
- channel_cnt_str = "Five"; break;
- case 4:
- channel_cnt_str = "Four"; break;
- case 3:
- channel_cnt_str = "Three"; break;
- default:
- channel_cnt_str = "Two"; break;
- }
- ALOGV("%s: HDMI channel count: %s", __func__, channel_cnt_str);
- set_mixer_control(adev->mixer, hdmi_chans_ctrl, channel_cnt_str);
-
- if (enable_passthrough) {
- ALOGD("%s:HDMI compress format", __func__);
- set_mixer_control(adev->mixer, hdmi_format_ctrl, "Compr");
- } else {
- ALOGD("%s: HDMI PCM format", __func__);
- set_mixer_control(adev->mixer, hdmi_format_ctrl, "LPCM");
- }
-
- switch (sample_rate) {
- case 32000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_32");
- break;
- case 44100:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_44P1");
- break;
- case 96000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_96");
- break;
- case 128000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_128");
- break;
- case 176400:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_176_4");
- break;
- case 192000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_192");
- break;
- default:
- case 48000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_48");
- break;
- }
-
- return 0;
-}
-
-
int platform_set_device_params(struct stream_out *out, int param, int value)
{
struct audio_device *adev = out->dev;
@@ -5234,8 +5512,11 @@
snd_device == SND_DEVICE_OUT_SPEAKER_WSA ||
snd_device == SND_DEVICE_OUT_SPEAKER_VBAT ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA) {
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA) {
ret = true;
}
@@ -5255,12 +5536,19 @@
case SND_DEVICE_OUT_VOICE_SPEAKER_WSA:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED);
+ break;
case SND_DEVICE_OUT_SPEAKER_VBAT:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
break;
case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT);
break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT);
+ break;
default:
acdb_id = -EINVAL;
break;
@@ -5280,15 +5568,37 @@
case SND_DEVICE_OUT_VOICE_SPEAKER:
case SND_DEVICE_OUT_VOICE_SPEAKER_WSA:
return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
case SND_DEVICE_OUT_SPEAKER_VBAT:
return SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
default:
return snd_device;
}
}
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device)
+{
+ switch(snd_device) {
+ case SND_DEVICE_OUT_SPEAKER_PROTECTED:
+ case SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2;
+ default:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ }
+}
+
int platform_set_sidetone(struct audio_device *adev,
snd_device_t out_snd_device,
bool enable,
@@ -5311,3 +5621,53 @@
}
return 0;
}
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+ return false;
+}
+
+bool platform_check_codec_asrc_support(void *platform __unused)
+{
+ return false;
+}
+
+int platform_send_audio_cal(void* platform __unused,
+ int acdb_dev_id __unused, int acdb_device_type __unused,
+ int app_type __unused, int topology_id __unused,
+ int sample_rate __unused, uint32_t module_id __unused,
+ uint32_t param_id __unused, void* data __unused,
+ int length __unused, bool persist __unused)
+{
+ return -ENOSYS;
+}
+
+int platform_get_audio_cal(void* platform __unused,
+ int acdb_dev_id __unused, int acdb_device_type __unused,
+ int app_type __unused, int topology_id __unused,
+ int sample_rate __unused, uint32_t module_id __unused,
+ uint32_t param_id __unused, void* data __unused,
+ int* length __unused, bool persist __unused)
+{
+ return -ENOSYS;
+}
+
+int platform_store_audio_cal(void* platform __unused,
+ int acdb_dev_id __unused, int acdb_device_type __unused,
+ int app_type __unused, int topology_id __unused,
+ int sample_rate __unused, uint32_t module_id __unused,
+ uint32_t param_id __unused, void* data __unused,
+ int length __unused)
+{
+ return -ENOSYS;
+}
+
+int platform_retrieve_audio_cal(void* platform __unused,
+ int acdb_dev_id __unused, int acdb_device_type __unused,
+ int app_type __unused, int topology_id __unused,
+ int sample_rate __unused, uint32_t module_id __unused,
+ uint32_t param_id __unused, void* data __unused,
+ int* length __unused)
+{
+ return -ENOSYS;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 756c749..cba9068 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -39,6 +39,11 @@
SOURCE_QUAD_MIC = 0x8, /* Target contains 4 mics */
};
+enum {
+ SPKR_1,
+ SPKR_2
+};
+
#define PLATFORM_IMAGE_NAME "modem"
/*
@@ -92,12 +97,19 @@
SND_DEVICE_OUT_VOICE_SPEAKER,
SND_DEVICE_OUT_VOICE_SPEAKER_WSA,
SND_DEVICE_OUT_VOICE_SPEAKER_VBAT,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT,
SND_DEVICE_OUT_VOICE_HEADPHONES,
SND_DEVICE_OUT_VOICE_LINE,
SND_DEVICE_OUT_HDMI,
SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+ SND_DEVICE_OUT_DISPLAY_PORT,
+ SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT,
SND_DEVICE_OUT_BT_SCO,
SND_DEVICE_OUT_BT_SCO_WB,
+ SND_DEVICE_OUT_BT_A2DP,
+ SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
@@ -114,8 +126,10 @@
SND_DEVICE_OUT_ANC_HANDSET,
SND_DEVICE_OUT_SPEAKER_PROTECTED,
SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED,
SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT,
SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT,
#ifdef RECORD_PLAY_CONCURRENCY
SND_DEVICE_OUT_VOIP_HANDSET,
SND_DEVICE_OUT_VOIP_SPEAKER,
@@ -174,6 +188,8 @@
SND_DEVICE_IN_HANDSET_STEREO_DMIC,
SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2,
SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE,
SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE,
SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE,
@@ -196,24 +212,35 @@
SND_DEVICE_MAX = SND_DEVICE_IN_END,
};
-
+#define INPUT_SAMPLING_RATE_DSD64 2822400
+#define INPUT_SAMPLING_RATE_DSD128 5644800
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
#define OUTPUT_SAMPLING_RATE_44100 44100
+#define OUTPUT_SAMPLING_RATE_DSD64 176400
+#define OUTPUT_SAMPLING_RATE_DSD128 352800
#define MAX_PORT 6
#define ALL_CODEC_BACKEND_PORT 0
#define HEADPHONE_44_1_BACKEND_PORT 5
-#define MAX_CODEC_TX_BACKENDS 1
enum {
DEFAULT_CODEC_BACKEND,
SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,
+ DSD_NATIVE_BACKEND,
+ SLIMBUS_2_RX = DSD_NATIVE_BACKEND,
HEADPHONE_44_1_BACKEND,
SLIMBUS_5_RX = HEADPHONE_44_1_BACKEND,
HEADPHONE_BACKEND,
SLIMBUS_6_RX = HEADPHONE_BACKEND,
HDMI_RX_BACKEND,
+ DISP_PORT_RX_BACKEND,
USB_AUDIO_RX_BACKEND,
+ MAX_RX_CODEC_BACKENDS = USB_AUDIO_RX_BACKEND,
+ /* TX BE follows RX BE */
+ SLIMBUS_0_TX,
+ DEFAULT_CODEC_TX_BACKEND = SLIMBUS_0_TX,
+ USB_AUDIO_TX_BACKEND,
MAX_CODEC_BACKENDS
};
+
#define AUDIO_PARAMETER_KEY_NATIVE_AUDIO "audio.nat.codec.enabled"
#define AUDIO_PARAMETER_KEY_NATIVE_AUDIO_MODE "native_audio_mode"
@@ -240,6 +267,7 @@
* the buffer size of an input/output stream
*/
#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_DURATION 40 /* 40 millisecs */
#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
@@ -274,6 +302,7 @@
#define AUDIO_RECORD_PCM_DEVICE 0
#define MULTIMEDIA2_PCM_DEVICE 1
#define MULTIMEDIA3_PCM_DEVICE 4
+#define MULTIMEDIA9_PCM_DEVICE 32
#define FM_PLAYBACK_PCM_DEVICE 5
#define FM_CAPTURE_PCM_DEVICE 6
#define HFP_PCM_RX 5
@@ -354,7 +383,8 @@
enum {
LEGACY_PCM = 0,
PASSTHROUGH,
- PASSTHROUGH_CONVERT
+ PASSTHROUGH_CONVERT,
+ PASSTHROUGH_DSD
};
/*
* ID for setting mute and lateny on the device side
@@ -370,4 +400,13 @@
char device_name[100];
char interface_name[100];
};
+
+struct audio_backend_cfg {
+ unsigned int sample_rate;
+ unsigned int channels;
+ unsigned int bit_width;
+ bool passthrough_enabled;
+ audio_format_t format;
+};
+
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 2b6a1d7..b5a4f11 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -667,6 +667,11 @@
return -ENOSYS;
}
+int platform_get_ext_disp_type(void *platform)
+{
+ return EXT_DISPLAY_TYPE_HDMI;
+}
+
snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -1034,6 +1039,12 @@
return -ENOSYS;
}
+unsigned char* platform_get_license(void *platform, int *size)
+{
+ ALOGE("%s: Not implemented", __func__);
+ return NULL;
+}
+
/* Delay in Us */
int64_t platform_render_latency(audio_usecase_t usecase)
{
@@ -1218,6 +1229,11 @@
return -ENOSYS;
}
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device __unused)
+{
+ return -ENOSYS;
+}
+
int platform_spkr_prot_is_wsa_analog_mode(void *adev __unused)
{
return 0;
@@ -1232,7 +1248,48 @@
bool platform_check_backends_match(snd_device_t snd_device1 __unused,
snd_device_t snd_device2 __unused)
{
- return true;
+ return -ENOSYS;
+}
+
+int platform_send_audio_cal(void* platform __unused,
+ int acdb_dev_id __unused, int acdb_device_type __unused,
+ int app_type __unused, int topology_id __unused,
+ int sample_rate __unused, uint32_t module_id,
+ uint32_t param_id, void* data __unused,
+ int length __unused, bool persist __unused)
+{
+ return -ENOSYS;
+}
+
+int platform_get_audio_cal(void* platform __unused,
+ int acdb_dev_id __unused, int acdb_device_type __unused,
+ int app_type __unused, int topology_id __unused,
+ int sample_rate __unused, uint32_t module_id,
+ uint32_t param_id, void* data __unused,
+ int* length __unused, bool persist __unused)
+{
+ return -ENOSYS;
+}
+
+int platform_store_audio_cal(void* platform __unused,
+ int acdb_dev_id __unused, int acdb_device_type __unused,
+ int app_type __unused, int topology_id __unused,
+ int sample_rate __unused, uint32_t module_id,
+ uint32_t param_id, void* data __unused,
+ int length __unused)
+{
+ return -ENOSYS;
+}
+
+
+int platform_retrieve_audio_cal(void* platform __unused,
+ int acdb_dev_id __unused, int acdb_device_type __unused,
+ int app_type __unused, int topology_id __unused,
+ int sample_rate __unused, uint32_t module_id,
+ uint32_t param_id, void* data __unused,
+ int* length __unused)
+{
+ return -ENOSYS;
}
int platform_set_sidetone(struct audio_device *adev,
@@ -1257,3 +1314,19 @@
}
return 0;
}
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+ return false;
+}
+
+int platform_get_backend_index(snd_device_t snd_device __unused);
+{
+ return 0;
+}
+
+bool platform_check_codec_asrc_support(void *platform __unused)
+{
+ return false;
+}
+
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index aab5436..c9ac74a 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, 2015 The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013, 2016 The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -63,7 +63,11 @@
SND_DEVICE_OUT_VOICE_HEADPHONES,
SND_DEVICE_OUT_HDMI,
SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+ SND_DEVICE_OUT_DISPLAY_PORT,
+ SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT,
SND_DEVICE_OUT_BT_SCO,
+ SND_DEVICE_OUT_BT_A2DP,
+ SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
SND_DEVICE_OUT_BT_SCO_WB,
SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
@@ -110,6 +114,12 @@
#define SOUND_CARD 0
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+#define INPUT_SAMPLING_RATE_DSD64 2822400
+#define INPUT_SAMPLING_RATE_DSD128 5644800
+#define OUTPUT_SAMPLING_RATE_DSD64 176400
+#define OUTPUT_SAMPLING_RATE_DSD128 352800
+#define DSD_NATIVE_BACKEND 1
+#define PASSTHROUGH_DSD 3
#define ALL_SESSION_VSID 0xFFFFFFFF
#define DEFAULT_MUTE_RAMP_DURATION_MS 20
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 9143d35..dae7ff3 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -135,6 +135,10 @@
SND_DEVICE_OUT_SPEAKER
};
+static const snd_device_t tasha_sbc_variant_devices[] = {
+ SND_DEVICE_IN_HANDSET_MIC
+};
+
static const snd_device_t taiko_apq8084_sbc_variant_devices[] = {
SND_DEVICE_IN_HANDSET_MIC,
SND_DEVICE_IN_SPEAKER_MIC,
@@ -301,6 +305,12 @@
hw_info->snd_devices = (snd_device_t *)tasha_DB_variant_devices;
hw_info->num_snd_devices = ARRAY_SIZE(tasha_DB_variant_devices);
strlcpy(hw_info->dev_extn, "-db", sizeof(hw_info->dev_extn));
+ } else if (!strcmp(snd_card_name, "msm8996-tasha-sbc-snd-card")) {
+ strlcpy(hw_info->type, " sbc", sizeof(hw_info->type));
+ strlcpy(hw_info->name, "msm8996", sizeof(hw_info->name));
+ hw_info->snd_devices = (snd_device_t *)tasha_sbc_variant_devices;
+ hw_info->num_snd_devices = ARRAY_SIZE(tasha_sbc_variant_devices);
+ strlcpy(hw_info->dev_extn, "-sbc", sizeof(hw_info->dev_extn));
} else {
ALOGW("%s: Not a 8996 device", __func__);
}
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index b98cc73..fa67342 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -42,6 +42,7 @@
#include "edid.h"
#include "sound/compress_params.h"
#include "sound/msmcal-hwdep.h"
+#include <linux/msm_audio_calibration.h>
#define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
#define MIXER_XML_DEFAULT_PATH "/system/etc/mixer_paths.xml"
@@ -57,10 +58,10 @@
#include <linux/msm_audio.h>
#define LIB_ACDB_LOADER "libacdbloader.so"
-#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
#define CVD_VERSION_MIXER_CTL "CVD Version"
-#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define FLAC_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024 * 1024)
#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024)
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
@@ -108,6 +109,8 @@
#define AUDIO_PARAMETER_KEY_AUD_CALDATA "cal_data"
#define AUDIO_PARAMETER_KEY_AUD_CALRESULT "cal_result"
+#define AUDIO_PARAMETER_KEY_MONO_SPEAKER "mono_speaker"
+
#define AUDIO_PARAMETER_KEY_PERF_LOCK_OPTS "perf_lock_opts"
/* Reload ACDB files from specified path */
@@ -188,8 +191,10 @@
typedef struct codec_backend_cfg {
uint32_t sample_rate;
uint32_t bit_width;
+ uint32_t channels;
char *bitwidth_mixer_ctl;
char *samplerate_mixer_ctl;
+ char *channels_mixer_ctl;
} codec_backend_cfg_t;
static native_audio_prop na_props = {0, 0, 0};
@@ -216,6 +221,7 @@
/* Vbat monitor related flags */
bool is_vbat_speaker;
bool gsm_mode_enabled;
+ int mono_speaker;
/* Audio calibration related functions */
void *acdb_handle;
int voice_feature_set;
@@ -235,9 +241,9 @@
struct csd_data *csd;
void *edid_info;
bool edid_valid;
+ int ext_disp_type;
char ec_ref_mixer_path[64];
codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
- codec_backend_cfg_t current_tx_backend_cfg[MAX_CODEC_TX_BACKENDS];
char codec_version[CODEC_VERSION_MAX_LENGTH];
int hw_dep_fd;
char cvd_version[MAX_CVD_VERSION_STRING_SIZE];
@@ -245,6 +251,8 @@
int metainfo_key;
int source_mic_type;
int max_mic_count;
+ bool is_dsd_supported;
+ bool is_asrc_supported;
};
static int pcm_device_table[AUDIO_USECASE_MAX][2] = {
@@ -318,6 +326,7 @@
AFE_PROXY_RECORD_PCM_DEVICE},
[USECASE_AUDIO_RECORD_AFE_PROXY] = {AFE_PROXY_PLAYBACK_PCM_DEVICE,
AFE_PROXY_RECORD_PCM_DEVICE},
+ [USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = {MULTIMEDIA9_PCM_DEVICE, -1},
};
@@ -332,6 +341,7 @@
[SND_DEVICE_OUT_SPEAKER_VBAT] = "speaker-vbat",
[SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
[SND_DEVICE_OUT_HEADPHONES] = "headphones",
+ [SND_DEVICE_OUT_HEADPHONES_DSD] = "headphones-dsd",
[SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
[SND_DEVICE_OUT_LINE] = "line",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
@@ -341,12 +351,18 @@
[SND_DEVICE_OUT_VOICE_HANDSET] = "voice-handset",
[SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = "voice-speaker-vbat",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2] = "voice-speaker-2",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = "voice-speaker-2-vbat",
[SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
[SND_DEVICE_OUT_VOICE_LINE] = "voice-line",
[SND_DEVICE_OUT_HDMI] = "hdmi",
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
+ [SND_DEVICE_OUT_DISPLAY_PORT] = "display-port",
+ [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = "speaker-and-display-port",
[SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
[SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+ [SND_DEVICE_OUT_BT_A2DP] = "bt-a2dp",
+ [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = "speaker-and-bt-a2dp",
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
@@ -363,8 +379,10 @@
[SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = "voice-speaker-protected",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = "voice-speaker-2-protected",
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = "speaker-protected-vbat",
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = "voice-speaker-protected-vbat",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = "voice-speaker-2-protected-vbat",
/* Capture sound devices */
[SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
@@ -414,6 +432,8 @@
[SND_DEVICE_IN_HANDSET_STEREO_DMIC] = "handset-stereo-dmic-ef",
[SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = "speaker-stereo-dmic-ef",
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = "vi-feedback",
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = "vi-feedback-mono-1",
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = "vi-feedback-mono-2",
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = "voice-speaker-dmic-broadside",
[SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = "speaker-dmic-broadside",
[SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = "speaker-dmic-broadside",
@@ -423,6 +443,7 @@
[SND_DEVICE_IN_SPEAKER_QMIC_AEC] = "quad-mic",
[SND_DEVICE_IN_SPEAKER_QMIC_NS] = "quad-mic",
[SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = "quad-mic",
+ [SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = "quad-mic",
[SND_DEVICE_IN_THREE_MIC] = "three-mic",
[SND_DEVICE_IN_HANDSET_TMIC] = "three-mic",
[SND_DEVICE_IN_UNPROCESSED_MIC] = "unprocessed-mic",
@@ -446,6 +467,7 @@
[SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
[SND_DEVICE_OUT_LINE] = 10,
[SND_DEVICE_OUT_HEADPHONES] = 10,
+ [SND_DEVICE_OUT_HEADPHONES_DSD] = 10,
[SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_LINE] = 10,
@@ -454,12 +476,18 @@
[SND_DEVICE_OUT_VOICE_HANDSET] = 7,
[SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = 14,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2] = 14,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = 14,
[SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
[SND_DEVICE_OUT_VOICE_LINE] = 10,
[SND_DEVICE_OUT_HDMI] = 18,
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
+ [SND_DEVICE_OUT_DISPLAY_PORT] = 18,
+ [SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = 14,
[SND_DEVICE_OUT_BT_SCO] = 22,
[SND_DEVICE_OUT_BT_SCO_WB] = 39,
+ [SND_DEVICE_OUT_BT_A2DP] = 20,
+ [SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = 14,
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
@@ -476,8 +504,10 @@
[SND_DEVICE_OUT_ANC_HANDSET] = 103,
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = 101,
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = 101,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = 101,
[SND_DEVICE_IN_HANDSET_MIC] = 4,
[SND_DEVICE_IN_HANDSET_MIC_EXTERNAL] = 4,
@@ -526,6 +556,8 @@
[SND_DEVICE_IN_HANDSET_STEREO_DMIC] = 34,
[SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = 35,
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = 102,
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = 102,
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = 12,
[SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = 12,
[SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = 119,
@@ -535,6 +567,7 @@
[SND_DEVICE_IN_SPEAKER_QMIC_AEC] = 126,
[SND_DEVICE_IN_SPEAKER_QMIC_NS] = 127,
[SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = 129,
+ [SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = 125,
[SND_DEVICE_IN_THREE_MIC] = 46, /* for APSS Surround Sound Recording */
[SND_DEVICE_IN_HANDSET_TMIC] = 125, /* for 3mic recording with fluence */
[SND_DEVICE_IN_UNPROCESSED_MIC] = 143,
@@ -560,6 +593,7 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_DSD)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_44_1)},
{TO_NAME_INDEX(SND_DEVICE_OUT_LINE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
@@ -569,12 +603,18 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_LINE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_DISPLAY_PORT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
{TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BT_A2DP)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
@@ -590,8 +630,10 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_EXTERNAL)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC)},
@@ -637,6 +679,8 @@
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE)},
@@ -646,6 +690,7 @@
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_QMIC_AEC)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_QMIC_NS)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE)},
{TO_NAME_INDEX(SND_DEVICE_IN_THREE_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_MIC)},
@@ -691,6 +736,7 @@
{TO_NAME_INDEX(USECASE_AUDIO_SPKR_CALIB_TX)},
{TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_AFE_PROXY)},
{TO_NAME_INDEX(USECASE_AUDIO_RECORD_AFE_PROXY)},
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE)},
};
#define NO_COLS 2
@@ -783,6 +829,7 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+#define ULL_PLATFORM_DELAY (6*1000LL)
bool platform_send_gain_dep_cal(void *platform, int level) {
bool ret_val = false;
@@ -1051,7 +1098,8 @@
sizeof("apq8084-taiko-i2s-cdp-snd-card"))) {
plat_data->is_i2s_ext_modem = true;
}
- ALOGV("%s, is_i2s_ext_modem:%d",__func__, plat_data->is_i2s_ext_modem);
+ ALOGV("%s, is_i2s_ext_modem:%d soundcard name is %s",__func__,
+ plat_data->is_i2s_ext_modem, snd_card_name);
return plat_data->is_i2s_ext_modem;
}
@@ -1080,6 +1128,8 @@
backend_tag_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("bt-sco-wb");
backend_tag_table[SND_DEVICE_OUT_HDMI] = strdup("hdmi");
backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("speaker-and-hdmi");
+ backend_tag_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("display-port");
+ backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("speaker-and-display-port");
backend_tag_table[SND_DEVICE_OUT_VOICE_TX] = strdup("afe-proxy");
backend_tag_table[SND_DEVICE_IN_VOICE_RX] = strdup("afe-proxy");
backend_tag_table[SND_DEVICE_OUT_AFE_PROXY] = strdup("afe-proxy");
@@ -1089,12 +1139,21 @@
backend_tag_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
backend_tag_table[SND_DEVICE_IN_CAPTURE_FM] = strdup("capture-fm");
backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
+ backend_tag_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("headphones-dsd");
backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
+ backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("voice-speaker-2-vbat");
+ backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
+ backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
+ hw_interface_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("SLIMBUS_2_RX");
hw_interface_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("SLIMBUS_5_RX");
hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("SLIMBUS_0_RX-and-HDMI_RX");
+ hw_interface_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("DISPLAY_PORT_RX");
+ hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("SLIMBUS_0_RX-and-DISPLAY_PORT_RX");
+ hw_interface_table[SND_DEVICE_OUT_USB_HEADSET] = strdup("USB_AUDIO_RX");
+ hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = strdup("SLIMBUS_0_RX-and-USB_AUDIO_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_TX] = strdup("AFE_PCM_RX");
my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
@@ -1505,7 +1564,9 @@
my_data->slowtalk = false;
my_data->hd_voice = false;
my_data->edid_info = NULL;
+ my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
my_data->hw_dep_fd = -1;
+ my_data->mono_speaker = SPKR_1;
property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
@@ -1672,6 +1733,9 @@
/* init usb */
audio_extn_usb_init(adev);
+ /*init a2dp*/
+ audio_extn_a2dp_init(adev);
+
/* init dap hal */
audio_extn_dap_hal_init(adev->snd_card);
@@ -1690,6 +1754,12 @@
if (idx == HEADPHONE_44_1_BACKEND)
my_data->current_backend_cfg[idx].sample_rate = OUTPUT_SAMPLING_RATE_44100;
my_data->current_backend_cfg[idx].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_CHANNELS;
+ if (idx > MAX_RX_CODEC_BACKENDS)
+ my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
+ my_data->current_backend_cfg[idx].bitwidth_mixer_ctl = NULL;
+ my_data->current_backend_cfg[idx].samplerate_mixer_ctl = NULL;
+ my_data->current_backend_cfg[idx].channels_mixer_ctl = NULL;
}
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
@@ -1697,16 +1767,28 @@
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_0_RX SampleRate");
+ my_data->current_backend_cfg[DSD_NATIVE_BACKEND].bitwidth_mixer_ctl =
+ strdup("SLIM_2_RX Format");
+ my_data->current_backend_cfg[DSD_NATIVE_BACKEND].samplerate_mixer_ctl =
+ strdup("SLIM_2_RX SampleRate");
+
my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].bitwidth_mixer_ctl =
strdup("SLIM_5_RX Format");
my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_5_RX SampleRate");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
strdup("SLIM_0_TX Format");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_0_TX SampleRate");
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
+ strdup("USB_AUDIO_TX Format");
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].samplerate_mixer_ctl =
+ strdup("USB_AUDIO_TX SampleRate");
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].channels_mixer_ctl =
+ strdup("USB_AUDIO_TX Channels");
+
ret = audio_extn_utils_get_codec_version(snd_card_name,
my_data->adev->snd_card,
my_data->codec_version);
@@ -1727,6 +1809,13 @@
}
}
+ if(strstr(snd_card_name, "tavil")) {
+ ALOGD("%s:DSD playback is supported", __func__);
+ my_data->is_dsd_supported = true;
+ my_data->is_asrc_supported = true;
+ platform_set_native_support(NATIVE_AUDIO_MODE_MULTIPLE_44_1);
+ }
+
my_data->current_backend_cfg[HEADPHONE_BACKEND].bitwidth_mixer_ctl =
strdup("SLIM_6_RX Format");
my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
@@ -1735,11 +1824,21 @@
strdup("HDMI_RX Bit Format");
my_data->current_backend_cfg[HDMI_RX_BACKEND].samplerate_mixer_ctl =
strdup("HDMI_RX SampleRate");
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].channels_mixer_ctl =
+ strdup("HDMI_RX Channels");
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].bitwidth_mixer_ctl =
+ strdup("Display Port RX Bit Format");
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].samplerate_mixer_ctl =
+ strdup("Display Port RX SampleRate");
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].channels_mixer_ctl =
+ strdup("Display Port RX Channels");
my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].bitwidth_mixer_ctl =
strdup("USB_AUDIO_RX Format");
my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].samplerate_mixer_ctl =
strdup("USB_AUDIO_RX SampleRate");
+ my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].channels_mixer_ctl =
+ strdup("USB_AUDIO_RX Channels");
my_data->edid_info = NULL;
free(snd_card_name);
@@ -1839,7 +1938,8 @@
return;
}
- if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
!(usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)) {
ALOGI("%s: Not adding vbat speaker device to non voice use cases", __func__);
return;
@@ -2065,14 +2165,15 @@
{
if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
ALOGE("%s: Invalid snd_device = %d", __func__, snd_device);
- return DEFAULT_OUTPUT_SAMPLING_RATE;
+ return CODEC_BACKEND_DEFAULT_BIT_WIDTH;
}
return backend_bit_width_table[snd_device];
}
int platform_set_native_support(int na_mode)
{
- if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode) {
+ if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode
+ || NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode) {
na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled = true;
na_props.na_mode = na_mode;
ALOGD("%s:napb: native audio playback enabled in (%s) mode v2.0", __func__,
@@ -2087,6 +2188,18 @@
return 0;
}
+bool platform_check_codec_dsd_support(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->is_dsd_supported;
+}
+
+bool platform_check_codec_asrc_support(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->is_asrc_supported;
+}
+
int platform_get_native_support()
{
int ret = NATIVE_AUDIO_MODE_INVALID;
@@ -2139,6 +2252,8 @@
mode = NATIVE_AUDIO_MODE_SRC;
else if (value && !strncmp(value, "true", sizeof("true")))
mode = NATIVE_AUDIO_MODE_TRUE_44_1;
+ else if (value && !strncmp(value, "multiple", sizeof("multiple")))
+ mode = NATIVE_AUDIO_MODE_MULTIPLE_44_1;
else {
mode = NATIVE_AUDIO_MODE_INVALID;
ALOGE("%s:napb:native_audio_mode in platform info xml,invalid mode string",
@@ -2205,28 +2320,52 @@
return ret;
}
-static int platform_get_backend_index(snd_device_t snd_device)
+
+int codec_device_supports_native_playback(audio_devices_t out_device)
+{
+ int ret = false;
+
+ if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+ out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+ out_device & AUDIO_DEVICE_OUT_LINE)
+ ret = true;
+
+ return ret;
+}
+
+int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
- if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+ if (snd_device >= SND_DEVICE_OUT_BEGIN && snd_device < SND_DEVICE_OUT_END) {
if (backend_tag_table[snd_device] != NULL) {
if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
sizeof("headphones-44.1")) == 0)
port = HEADPHONE_44_1_BACKEND;
+ else if (strncmp(backend_tag_table[snd_device], "headphones-dsd",
+ sizeof("headphones-dsd")) == 0)
+ port = DSD_NATIVE_BACKEND;
else if (strncmp(backend_tag_table[snd_device], "headphones",
sizeof("headphones")) == 0)
port = HEADPHONE_BACKEND;
else if (strcmp(backend_tag_table[snd_device], "hdmi") == 0)
port = HDMI_RX_BACKEND;
+ else if (strcmp(backend_tag_table[snd_device], "display-port") == 0)
+ port = DISP_PORT_RX_BACKEND;
else if (strcmp(backend_tag_table[snd_device], "usb-headphones") == 0)
port = USB_AUDIO_RX_BACKEND;
}
+ } else if (snd_device >= SND_DEVICE_IN_BEGIN && snd_device < SND_DEVICE_IN_END) {
+ port = DEFAULT_CODEC_TX_BACKEND;
+ if (backend_tag_table[snd_device] != NULL) {
+ if (strcmp(backend_tag_table[snd_device], "usb-headset-mic") == 0)
+ port = USB_AUDIO_TX_BACKEND;
+ }
} else {
- ALOGV("%s:napb: Invalid device - %d ", __func__, snd_device);
+ ALOGW("%s:napb: Invalid device - %d ", __func__, snd_device);
}
- ALOGV("%s:napb: backend port - %d snd_device %d", __func__, port, snd_device);
+ ALOGV("%s:napb: backend port - %d device - %d ", __func__, port, snd_device);
return port;
}
@@ -2311,7 +2450,9 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
audio_extn_spkr_prot_is_enabled()) {
if (my_data->is_vbat_speaker)
acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2346,9 +2487,16 @@
if (my_data->acdb_send_voice_cal == NULL) {
ALOGE("%s: dlsym error for acdb_send_voice_call", __func__);
} else {
- if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
- audio_extn_spkr_prot_is_enabled())
- out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ if (audio_extn_spkr_prot_is_enabled()) {
+ if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
+ }
acdb_rx_id = acdb_device_table[out_snd_device];
acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2375,7 +2523,9 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
audio_extn_spkr_prot_is_enabled()) {
if (my_data->is_vbat_speaker)
acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2576,20 +2726,68 @@
new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
new_snd_devices[1] = SND_DEVICE_OUT_HDMI;
status = true;
+ } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT &&
+ !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_DISPLAY_PORT)) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+ new_snd_devices[1] = SND_DEVICE_OUT_DISPLAY_PORT;
+ status = true;
} else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET &&
!platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_USB_HEADSET)) {
*num_devices = 2;
new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
new_snd_devices[1] = SND_DEVICE_OUT_USB_HEADSET;
status = true;
+ } else if (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+ new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
}
+
ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
snd_device, *num_devices, *new_snd_devices);
return status;
}
+int platform_get_ext_disp_type(void *platform)
+{
+ int disp_type;
+ struct platform_data *my_data = (struct platform_data *)platform;
+
+ if (my_data->ext_disp_type != EXT_DISPLAY_TYPE_NONE) {
+ ALOGD("%s: Returning cached ext disp type:%s",
+ __func__, (my_data->ext_disp_type == EXT_DISPLAY_TYPE_DP) ? "DisplayPort" : "HDMI");
+ return my_data->ext_disp_type;
+ }
+
+#ifdef DISPLAY_PORT_ENABLED
+ struct audio_device *adev = my_data->adev;
+ struct mixer_ctl *ctl;
+ char *mixer_ctl_name = "External Display Type";
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ disp_type = mixer_ctl_get_value(ctl, 0);
+ if (disp_type == EXT_DISPLAY_TYPE_NONE) {
+ ALOGE("%s: Invalid external display type: %d", __func__, disp_type);
+ return -EINVAL;
+ }
+#else
+ disp_type = EXT_DISPLAY_TYPE_HDMI;
+#endif
+
+ my_data->ext_disp_type = disp_type;
+ ALOGD("%s: ext disp type:%s", __func__, (disp_type == EXT_DISPLAY_TYPE_DP) ? "DisplayPort" : "HDMI");
+ return disp_type;
+}
+
snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -2635,13 +2833,26 @@
snd_device = SND_DEVICE_OUT_SPEAKER_AND_LINE;
} else if (devices == (AUDIO_DEVICE_OUT_AUX_DIGITAL |
AUDIO_DEVICE_OUT_SPEAKER)) {
- snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ snd_device = SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT;
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ goto exit;
+ }
} else if (devices == (AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
AUDIO_DEVICE_OUT_SPEAKER)) {
snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
} else if (devices == (AUDIO_DEVICE_OUT_USB_DEVICE |
AUDIO_DEVICE_OUT_SPEAKER)) {
snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
+ } else if ((devices & AUDIO_DEVICE_OUT_SPEAKER) &&
+ (devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
+ snd_device = SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP;
} else {
ALOGE("%s: Invalid combo device(%#x)", __func__, devices);
goto exit;
@@ -2693,10 +2904,19 @@
else
snd_device = SND_DEVICE_OUT_BT_SCO;
} else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
- if (my_data->is_vbat_speaker)
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
- else
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ if (my_data->is_vbat_speaker) {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT;
+ } else {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2;
+ }
+ } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ snd_device = SND_DEVICE_OUT_BT_A2DP;
} else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
snd_device = SND_DEVICE_OUT_USB_HEADSET;
@@ -2716,7 +2936,8 @@
}
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
- devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+ devices & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+ devices & AUDIO_DEVICE_OUT_LINE) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET
&& audio_extn_get_anc_enabled()) {
if (audio_extn_should_use_fb_anc())
@@ -2726,6 +2947,12 @@
} else if (NATIVE_AUDIO_MODE_SRC == na_mode &&
OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+ } else if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode &&
+ (sample_rate % OUTPUT_SAMPLING_RATE_44100 == 0) &&
+ (out->format != AUDIO_FORMAT_DSD)) {
+ snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+ } else if (out->format == AUDIO_FORMAT_DSD) {
+ snd_device = SND_DEVICE_OUT_HEADPHONES_DSD;
} else
snd_device = SND_DEVICE_OUT_HEADPHONES;
} else if (devices & AUDIO_DEVICE_OUT_LINE) {
@@ -2746,8 +2973,20 @@
snd_device = SND_DEVICE_OUT_BT_SCO_WB;
else
snd_device = SND_DEVICE_OUT_BT_SCO;
+ } else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ snd_device = SND_DEVICE_OUT_BT_A2DP;
} else if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
- snd_device = SND_DEVICE_OUT_HDMI ;
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ snd_device = SND_DEVICE_OUT_HDMI;
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ snd_device = SND_DEVICE_OUT_DISPLAY_PORT;
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ goto exit;
+ }
} else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
ALOGD("%s: setting USB hadset channel capability(2) for Proxy", __func__);
@@ -2896,7 +3135,7 @@
if (my_data->fluence_in_voice_rec && channel_count == 1) {
if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
(my_data->source_mic_type & SOURCE_QUAD_MIC)) {
- snd_device = SND_DEVICE_IN_HANDSET_QMIC;
+ snd_device = SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE;
} else if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
(my_data->source_mic_type & SOURCE_THREE_MIC)) {
snd_device = SND_DEVICE_IN_HANDSET_TMIC;
@@ -3153,7 +3392,7 @@
struct audio_device *adev = my_data->adev;
struct mixer_ctl *ctl;
const char *channel_cnt_str = NULL;
- const char *mixer_ctl_name = "HDMI_RX Channels";
+ char *mixer_ctl_name;
switch (channel_count) {
case 8:
channel_cnt_str = "Eight"; break;
@@ -3170,13 +3409,26 @@
default:
channel_cnt_str = "Two"; break;
}
+
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ mixer_ctl_name = "HDMI_RX Channels";
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ mixer_ctl_name = "Display Port RX Channels";
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ return -EINVAL;
+ }
+
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
- ALOGV("HDMI channel count: %s", channel_cnt_str);
+ ALOGV("Ext disp channel count: %s", channel_cnt_str);
mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
return 0;
}
@@ -3558,6 +3810,16 @@
}
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER, value, len);
+ if (err >= 0) {
+ if (!strncmp("left", value, sizeof("left")))
+ my_data->mono_speaker = SPKR_1;
+ else if (!strncmp("right", value, sizeof("right")))
+ my_data->mono_speaker = SPKR_2;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER);
+ }
+
err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE,
value, len);
if (err >= 0) {
@@ -3850,8 +4112,8 @@
!strncmp("true", propValue, 4);
}
- if (prop_playback_enabled && (voice_is_in_call(my_data->adev) ||
- (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev)))) {
+ if ((prop_playback_enabled && (voice_is_in_call(my_data->adev))) ||
+ (SND_CARD_STATE_OFFLINE == get_snd_card_state(my_data->adev))) {
char *decoder_mime_type = value;
//check if unsupported mime type or not
@@ -3875,6 +4137,54 @@
free(kv_pairs);
}
+unsigned char* platform_get_license(void *platform, int *size)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ char value[PROPERTY_VALUE_MAX] = {0};
+ acdb_audio_cal_cfg_t cal;
+ unsigned char *dptr = NULL;
+ int ret=0;
+ uint32_t param_len;
+
+ if (platform == NULL) {
+ ALOGE("[%s] received null pointer %d ",__func__, __LINE__);
+ ret = -EINVAL;
+ goto done;
+ }
+ memset(&cal, 0, sizeof(cal));
+ cal.persist = 1;
+ cal.cal_type = AUDIO_CORE_METAINFO_CAL_TYPE;
+ if (!property_get("audio.qaf.acdbid", value , "") && !atoi(value)) {
+ ALOGE("[%s] audio.qaf.acdbid is not set %d ",__func__, __LINE__);
+ ret = -EINVAL;
+ goto done;
+ }
+ cal.acdb_dev_id = (uint32_t) atoi (value);
+ param_len = MAX_SET_CAL_BYTE_SIZE;
+ dptr = (unsigned char*) calloc(param_len, sizeof(unsigned char*));
+ if (dptr == NULL) {
+ ALOGE("[%s] Memory allocation failed for length %d",__func__,param_len);
+ ret = -ENOMEM;
+ goto done;
+ }
+ if (my_data->acdb_get_audio_cal != NULL) {
+ ret = my_data->acdb_get_audio_cal((void*)&cal, (void*)dptr, ¶m_len);
+ ALOGE("%s, ret[%d], param_len[%d] line %d", __func__, ret, param_len, __LINE__);
+ if (ret == 0) {
+ *size = param_len;
+ return dptr;
+ } else {
+ *size = 0;
+ }
+ }
+done:
+ if (dptr != NULL)
+ free(dptr);
+
+ return NULL;
+}
+
+/* Delay in Us */
/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
@@ -3886,6 +4196,8 @@
case USECASE_AUDIO_PLAYBACK_OFFLOAD:
case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
return PCM_OFFLOAD_PLATFORM_DELAY;
+ case USECASE_AUDIO_PLAYBACK_ULL:
+ return ULL_PLATFORM_DELAY;
default:
return 0;
}
@@ -3906,7 +4218,9 @@
if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
(snd_device < SND_DEVICE_IN_END) &&
(snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
- (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
needs_event = true;
return needs_event;
@@ -3924,7 +4238,9 @@
if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
(snd_device < SND_DEVICE_IN_END) &&
(snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
- (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
needs_event = true;
return needs_event;
@@ -3956,18 +4272,21 @@
fragment_size = info->offload_buffer_size;
}
- // For FLAC use max size since it is loss less, and has sampling rates
- // upto 192kHZ
- if (info != NULL && !info->has_video &&
- info->format == AUDIO_FORMAT_FLAC) {
- fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
- ALOGV("FLAC fragment size %d", fragment_size);
- }
-
- if (info != NULL && info->has_video && info->is_streaming) {
- fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
- ALOGV("%s: offload fragment size reduced for AV streaming to %d",
- __func__, fragment_size);
+ if (info != NULL && !info->has_video) {
+ if (info->is_streaming) {
+ fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
+ ALOGV("%s: offload fragment size reduced for AV streaming to %d",
+ __func__, fragment_size);
+ } else if (info->format == AUDIO_FORMAT_FLAC) {
+ fragment_size = FLAC_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ ALOGV("FLAC fragment size %d", fragment_size);
+ } else if (info->format == AUDIO_FORMAT_DSD) {
+ fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ if((property_get("audio.native.dsd.buffer.size.kb", value, "")) &&
+ atoi(value))
+ fragment_size = atoi(value) * 1024;
+ ALOGV("DSD fragment size %d", fragment_size);
+ }
}
fragment_size = ALIGN( fragment_size, 1024);
@@ -3981,26 +4300,54 @@
}
/*
+ * return backend_idx on which voice call is active
+ */
+static int platform_get_voice_call_backend(struct audio_device* adev)
+{
+ struct audio_usecase *uc = NULL;
+ struct listnode *node;
+ snd_device_t out_snd_device = SND_DEVICE_NONE;
+
+ int backend_idx = -1;
+
+ if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
+ list_for_each(node, &adev->usecase_list) {
+ uc = node_to_item(node, struct audio_usecase, list);
+ if (uc && (uc->type == VOICE_CALL || uc->type == VOIP_CALL) && uc->stream.out) {
+ out_snd_device = platform_get_output_snd_device(adev->platform, uc->stream.out);
+ backend_idx = platform_get_backend_index(out_snd_device);
+ break;
+ }
+ }
+ }
+ return backend_idx;
+}
+
+/*
* configures afe with bit width and Sample Rate
*/
static int platform_set_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device, unsigned int bit_width,
- unsigned int sample_rate, audio_format_t format)
+ snd_device_t snd_device, struct audio_backend_cfg backend_cfg)
{
int ret = 0;
- int backend_idx = DEFAULT_CODEC_BACKEND;
+ int backend_idx = platform_get_backend_index(snd_device);
struct platform_data *my_data = (struct platform_data *)adev->platform;
-
backend_idx = platform_get_backend_index(snd_device);
+ unsigned int bit_width = backend_cfg.bit_width;
+ unsigned int sample_rate = backend_cfg.sample_rate;
+ unsigned int channels = backend_cfg.channels;
+ audio_format_t format = backend_cfg.format;
+ bool passthrough_enabled = backend_cfg.passthrough_enabled;
- ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
- ", backend_idx %d device (%s)", __func__, bit_width, sample_rate, backend_idx,
+ ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+ ", backend_idx %d device (%s)", __func__, bit_width,
+ sample_rate, channels, backend_idx,
platform_get_snd_device_name(snd_device));
if (bit_width !=
my_data->current_backend_cfg[backend_idx].bit_width) {
- struct mixer_ctl *ctl;
+ struct mixer_ctl *ctl = NULL;
ctl = mixer_get_ctl_by_name(adev->mixer,
my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
if (!ctl) {
@@ -4012,31 +4359,30 @@
if (bit_width == 24) {
if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
- mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+ ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
else
- mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else if (bit_width == 32) {
- mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ ret = mixer_ctl_set_enum_by_string(ctl, "S32_LE");
} else {
- mixer_ctl_set_enum_by_string(ctl, "S16_LE");
+ ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
}
- my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
- my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+ if ( ret < 0) {
+ ALOGE("%s:becf: afe: fail for %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+ } else {
+ my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
+ ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+ }
+ /* set the ret as 0 and not pass back to upper layer */
+ ret = 0;
}
- /*
- * Backend sample rate configuration follows:
- * 16 bit playback - 48khz for streams at any valid sample rate
- * 24 bit playback - 48khz for stream sample rate less than 48khz
- * 24 bit playback - 96khz for sample rate range of 48khz to 96khz
- * 24 bit playback - 192khz for sample rate range of 96khz to 192 khz
- * Upper limit is inclusive in the sample rate range.
- */
if (sample_rate !=
my_data->current_backend_cfg[backend_idx].sample_rate) {
char *rate_str = NULL;
- struct mixer_ctl *ctl;
+ struct mixer_ctl *ctl = NULL;
switch (sample_rate) {
case 8000:
@@ -4051,14 +4397,24 @@
rate_str = "KHZ_44P1";
break;
case 64000:
- case 88200:
case 96000:
rate_str = "KHZ_96";
break;
+ case 88200:
+ rate_str = "KHZ_88P2";
+ break;
case 176400:
+ rate_str = "KHZ_176P4";
+ break;
case 192000:
rate_str = "KHZ_192";
break;
+ case 352800:
+ rate_str = "KHZ_352P8";
+ break;
+ case 384000:
+ rate_str = "KHZ_384";
+ break;
default:
rate_str = "KHZ_48";
break;
@@ -4078,48 +4434,208 @@
mixer_ctl_set_enum_by_string(ctl, rate_str);
my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
}
+ if ((my_data->current_backend_cfg[backend_idx].channels_mixer_ctl) &&
+ (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+ struct mixer_ctl *ctl = NULL;
+ char *channel_cnt_str = NULL;
+
+ switch (channels) {
+ case 8:
+ channel_cnt_str = "Eight"; break;
+ case 7:
+ channel_cnt_str = "Seven"; break;
+ case 6:
+ channel_cnt_str = "Six"; break;
+ case 5:
+ channel_cnt_str = "Five"; break;
+ case 4:
+ channel_cnt_str = "Four"; break;
+ case 3:
+ channel_cnt_str = "Three"; break;
+ case 1:
+ channel_cnt_str = "One"; break;
+ case 2:
+ default:
+ channel_cnt_str = "Two"; break;
+ }
+
+ ctl = mixer_get_ctl_by_name(adev->mixer,
+ my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+ if (!ctl) {
+ ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+ __func__,
+ my_data->current_backend_cfg[backend_idx].channels_mixer_ctl);
+ return -EINVAL;
+ }
+ mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
+ my_data->current_backend_cfg[backend_idx].channels = channels;
+
+ if (backend_idx == HDMI_RX_BACKEND)
+ platform_set_edid_channels_configuration(adev->platform, channels);
+
+ ALOGD("%s:becf: afe: %s set to %s", __func__,
+ my_data->current_backend_cfg[backend_idx].channels_mixer_ctl,
+ channel_cnt_str);
+ }
+
+ bool set_ext_disp_format = false;
+ char *ext_disp_format = NULL;
+
+ if (backend_idx == HDMI_RX_BACKEND) {
+ ext_disp_format = "HDMI RX Format";
+ set_ext_disp_format = true;
+ } else if (backend_idx == DISP_PORT_RX_BACKEND) {
+ ext_disp_format = "Display Port Rx Format";
+ set_ext_disp_format = true;
+ } else {
+ ALOGV("%s: Format doesnt have to be set", __func__);
+ }
+
+ if (set_ext_disp_format) {
+ struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, ext_disp_format);
+ if (!ctl) {
+ ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+ __func__, ext_disp_format);
+ return -EINVAL;
+ }
+
+ if (passthrough_enabled) {
+ ALOGD("%s:Ext display compress format", __func__);
+ mixer_ctl_set_enum_by_string(ctl, "Compr");
+ } else {
+ ALOGD("%s: Ext display PCM format", __func__);
+ mixer_ctl_set_enum_by_string(ctl, "LPCM");
+ }
+ }
+
+ if (snd_device == SND_DEVICE_OUT_HEADPHONES || snd_device ==
+ SND_DEVICE_OUT_HEADPHONES_44_1) {
+ if (sample_rate > 48000 ||
+ (bit_width >= 24 && (sample_rate == 48000 || sample_rate == 44100))) {
+ ALOGV("%s: apply HPH HQ mode\n", __func__);
+ audio_route_apply_and_update_path(adev->audio_route, "hph-highquality-mode");
+ } else {
+ ALOGV("%s: apply HPH LP mode\n", __func__);
+ audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
+ }
+ }
return ret;
}
/*
+ *Validate the selected bit_width, sample_rate and channels using the edid
+ *of the connected sink device.
+ */
+static void platform_check_hdmi_backend_cfg(struct audio_device* adev,
+ struct audio_usecase* usecase,
+ int backend_idx,
+ struct audio_backend_cfg *hdmi_backend_cfg)
+{
+ unsigned int bit_width;
+ unsigned int sample_rate;
+ unsigned int channels, max_supported_channels = 0;
+ struct platform_data *my_data = (struct platform_data *)adev->platform;
+ edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+ bool passthrough_enabled = false;
+
+ bit_width = hdmi_backend_cfg->bit_width;
+ sample_rate = hdmi_backend_cfg->sample_rate;
+ channels = hdmi_backend_cfg->channels;
+
+
+ ALOGI("%s:becf: HDMI: bitwidth %d, samplerate %d, channels %d"
+ ", usecase = %d", __func__, bit_width,
+ sample_rate, channels, usecase->id);
+
+ if (audio_extn_passthru_is_enabled() && audio_extn_passthru_is_active()
+ && (usecase->stream.out->compr_config.codec->compr_passthr != 0)) {
+ passthrough_enabled = true;
+ ALOGI("passthrough is enabled for this stream");
+ }
+
+ // For voice calls use default configuration i.e. 16b/48K, only applicable to
+ // default backend
+ if (!passthrough_enabled) {
+
+ max_supported_channels = platform_edid_get_max_channels(my_data);
+
+ //Check EDID info for supported samplerate
+ if (!edid_is_supported_sr(edid_info,sample_rate)) {
+ //reset to current sample rate
+ sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
+ }
+
+ //Check EDID info for supported bit width
+ if (!edid_is_supported_bps(edid_info,bit_width)) {
+ //reset to current sample rate
+ bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
+ }
+
+ if (channels > max_supported_channels)
+ channels = max_supported_channels;
+
+ } else {
+ /*During pass through set default bit width and channels*/
+ channels = DEFAULT_HDMI_OUT_CHANNELS;
+ if ((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) ||
+ (usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC))
+ sample_rate = sample_rate * 4 ;
+
+ bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ /* We force route so that the BE format can be set to Compr */
+ }
+
+ ALOGI("%s:becf: afe: HDMI backend: passthrough %d updated bit width: %d and sample rate: %d"
+ "channels %d", __func__, passthrough_enabled , bit_width,
+ sample_rate, channels);
+
+ hdmi_backend_cfg->bit_width = bit_width;
+ hdmi_backend_cfg->sample_rate = sample_rate;
+ hdmi_backend_cfg->channels = channels;
+ hdmi_backend_cfg->passthrough_enabled = passthrough_enabled;
+}
+
+/*
* goes through all the current usecases and picks the highest
* bitwidth & samplerate
*/
static bool platform_check_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase* usecase,
snd_device_t snd_device,
- unsigned int* new_bit_width,
- unsigned int* new_sample_rate)
+ struct audio_backend_cfg *backend_cfg)
{
bool backend_change = false;
struct listnode *node;
unsigned int bit_width;
unsigned int sample_rate;
+ unsigned int channels;
+ bool passthrough_enabled = false;
int backend_idx = DEFAULT_CODEC_BACKEND;
struct platform_data *my_data = (struct platform_data *)adev->platform;
int na_mode = platform_get_native_support();
- edid_audio_info *edid_info = (edid_audio_info *)my_data->edid_info;
+ bool channels_updated = false;
backend_idx = platform_get_backend_index(snd_device);
- bit_width = *new_bit_width;
- sample_rate = *new_sample_rate;
+ bit_width = backend_cfg->bit_width;
+ sample_rate = backend_cfg->sample_rate;
+ channels = backend_cfg->channels;
- ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
+ ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
", backend_idx %d usecase = %d device (%s)", __func__, bit_width,
- sample_rate, backend_idx, usecase->id,
+ sample_rate, channels, backend_idx, usecase->id,
platform_get_snd_device_name(snd_device));
// For voice calls use default configuration i.e. 16b/48K, only applicable to
// default backend
// force routing is not required here, caller will do it anyway
- if ((voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
- backend_idx == DEFAULT_CODEC_BACKEND) {
+ if (backend_idx == platform_get_voice_call_backend(adev)) {
ALOGW("%s:becf: afe:Use default bw and sr for voice/voip calls ",
__func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ channels = CODEC_BACKEND_DEFAULT_CHANNELS;
} else {
/*
* The backend should be configured at highest bit width and/or
@@ -4136,11 +4652,12 @@
uc = node_to_item(node, struct audio_usecase, list);
struct stream_out *out = (struct stream_out*) uc->stream.out;
if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+ unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
- "(%d) device %s", __func__, i++, use_case_table[uc->id],
+ "(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
uc->id, out->sample_rate,
- out->bit_width,
+ out->bit_width, out_channels,
platform_get_snd_device_name(uc->out_snd_device));
if (platform_check_backends_match(snd_device, uc->out_snd_device)) {
@@ -4150,86 +4667,124 @@
sample_rate = out->sample_rate;
if (out->sample_rate < OUTPUT_SAMPLING_RATE_44100)
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ if (channels < out_channels)
+ channels = out_channels;
}
}
}
}
- if (audio_is_true_native_stream_active(adev)) {
- if (check_hdset_combo_device(snd_device)) {
- /*
- * In true native mode Tasha has a limitation that one port at 44.1 khz
- * cannot drive both spkr and hdset, to simiplify the solution lets
- * move the AFE to 48khzwhen a ring tone selects combo device.
- */
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
- ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
- __func__);
- } else {
- /*
- * in single BE mode, if native audio playback
- * is active then it will take priority
- */
- sample_rate = OUTPUT_SAMPLING_RATE_44100;
- ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
- __func__);
+ /* Native playback is preferred for Headphone/HS device over 192Khz */
+ if (codec_device_supports_native_playback(usecase->devices)) {
+ if (audio_is_true_native_stream_active(adev)) {
+ if (check_hdset_combo_device(snd_device)) {
+ /*
+ * In true native mode Tasha has a limitation that one port at 44.1 khz
+ * cannot drive both spkr and hdset, to simiplify the solution lets
+ * move the AFE to 48khzwhen a ring tone selects combo device.
+ * or if NATIVE playback is not enabled.
+ */
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ ALOGD("%s:becf: afe: port has to run at 48k for a combo device",
+ __func__);
+ } else {
+ /*
+ * in single BE mode, if native audio playback
+ * is active then it will take priority
+ */
+ sample_rate = OUTPUT_SAMPLING_RATE_44100;
+ ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
+ __func__);
+ }
+ } else if ((OUTPUT_SAMPLING_RATE_44100 == sample_rate) &&
+ (na_mode != NATIVE_AUDIO_MODE_MULTIPLE_44_1)) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
+ __func__);
}
- }
+ } else if ((usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) ||
+ (usecase->devices & AUDIO_DEVICE_OUT_EARPIECE) ) {
- /*
- * hifi playback not supported on spkr devices, limit the Sample Rate
- * to 48 khz.
- */
- if (SND_DEVICE_OUT_SPEAKER == snd_device ||
- SND_DEVICE_OUT_SPEAKER_WSA == snd_device ||
- SND_DEVICE_OUT_SPEAKER_VBAT == snd_device) {
+ if (bit_width >= 24) {
+ bit_width = platform_get_snd_device_bit_width(SND_DEVICE_OUT_SPEAKER);
+ ALOGD("%s:becf: afe: reset bitwidth to %d (based on supported"
+ " value for this platform)", __func__, bit_width);
+ }
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: playback on speaker device Configure afe to "
+ ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
"default Sample Rate(48k)", __func__);
}
- /*
- * native playback is not enabled.Configure afe to default Sample Rate(48k)
- */
- if (NATIVE_AUDIO_MODE_INVALID == na_mode &&
- OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
- __func__);
- }
-
if (backend_idx == USB_AUDIO_RX_BACKEND) {
- unsigned int channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
- audio_extn_usb_is_config_supported(&bit_width, &sample_rate, channels);
+ audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, true);
ALOGV("%s: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
__func__, bit_width, sample_rate, channels);
+ if (channels != my_data->current_backend_cfg[backend_idx].channels)
+ channels_updated = true;
}
- if (backend_idx == HDMI_RX_BACKEND) {
- //Check EDID info for supported samplerate
- if (!edid_is_supported_sr(edid_info,sample_rate)) {
- //reset to current sample rate
- sample_rate = my_data->current_backend_cfg[backend_idx].sample_rate;
- }
- //Check EDID info for supported bit widhth
- if (!edid_is_supported_bps(edid_info,bit_width)) {
- //reset to current sample rate
- bit_width = my_data->current_backend_cfg[backend_idx].bit_width;
+ if (backend_idx == HDMI_RX_BACKEND || backend_idx == DISP_PORT_RX_BACKEND) {
+ struct audio_backend_cfg hdmi_backend_cfg;
+ hdmi_backend_cfg.bit_width = bit_width;
+ hdmi_backend_cfg.sample_rate = sample_rate;
+ hdmi_backend_cfg.channels = channels;
+ hdmi_backend_cfg.passthrough_enabled = false;
+
+ /*
+ * HDMI does not support 384Khz/32bit playback hence configure BE to 24b/192Khz
+ * TODO: Instead have the validation against edid return the next best match
+ */
+ if (bit_width > 24)
+ hdmi_backend_cfg.bit_width = 24;
+ if (sample_rate > 192000)
+ hdmi_backend_cfg.sample_rate = 192000;
+
+ platform_check_hdmi_backend_cfg(adev, usecase, backend_idx, &hdmi_backend_cfg);
+
+ bit_width = hdmi_backend_cfg.bit_width;
+ sample_rate = hdmi_backend_cfg.sample_rate;
+ channels = hdmi_backend_cfg.channels;
+ passthrough_enabled = hdmi_backend_cfg.passthrough_enabled;
+
+ if (channels != my_data->current_backend_cfg[backend_idx].channels)
+ channels_updated = true;
+ }
+
+ /*
+ * Map native sampling rates to upper limit range
+ * if multiple of native sampling rates are not supported.
+ */
+ if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 != na_mode) {
+ switch (sample_rate) {
+ case 88200:
+ sample_rate = 96000;
+ break;
+ case 176400:
+ sample_rate = 192000;
+ break;
+ case 352800:
+ sample_rate = 192000;
+ break;
}
}
+
ALOGI("%s:becf: afe: Codec selected backend: %d updated bit width: %d and sample rate: %d",
__func__, backend_idx , bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
- (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate)) {
- *new_bit_width = bit_width;
- *new_sample_rate = sample_rate;
+ (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+ passthrough_enabled || channels_updated) {
+ backend_cfg->bit_width = bit_width;
+ backend_cfg->sample_rate = sample_rate;
+ backend_cfg->channels = channels;
+ backend_cfg->passthrough_enabled = passthrough_enabled;
backend_change = true;
- ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d new sample rate: %d",
- __func__, *new_bit_width, *new_sample_rate);
+ ALOGI("%s:becf: afe: Codec backend needs to be updated. new bit width: %d"
+ "new sample rate: %d new channels: %d",
+ __func__, backend_cfg->bit_width, backend_cfg->sample_rate, backend_cfg->channels);
}
return backend_change;
@@ -4238,158 +4793,50 @@
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase *usecase, snd_device_t snd_device)
{
- unsigned int new_bit_width;
- unsigned int new_sample_rate;
int backend_idx = DEFAULT_CODEC_BACKEND;
int new_snd_devices[SND_DEVICE_OUT_END];
int i, num_devices = 1;
bool ret = false;
struct platform_data *my_data = (struct platform_data *)adev->platform;
- audio_format_t format;
+ struct audio_backend_cfg backend_cfg;
backend_idx = platform_get_backend_index(snd_device);
- new_bit_width = usecase->stream.out->bit_width;
- new_sample_rate = usecase->stream.out->sample_rate;
- format = usecase->stream.out->format;
+ backend_cfg.bit_width = usecase->stream.out->bit_width;
+ backend_cfg.sample_rate = usecase->stream.out->sample_rate;
+ backend_cfg.format = usecase->stream.out->format;
+ backend_cfg.channels = audio_channel_count_from_out_mask(usecase->stream.out->channel_mask);
+ /*this is populated by check_codec_backend_cfg hence set default value to false*/
+ backend_cfg.passthrough_enabled = false;
- ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
- ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
- new_sample_rate, backend_idx, usecase->id,
+ /* Set Backend sampling rate to 176.4 for DSD64 and
+ * 352.8Khz for DSD128.
+ * Set Bit Width to 16
+ */
+ if ((backend_idx == DSD_NATIVE_BACKEND) && (backend_cfg.format == AUDIO_FORMAT_DSD)) {
+ backend_cfg.bit_width = 16;
+ if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD64)
+ backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+ else if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD128)
+ backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
+ }
+ ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
+ ", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
+ backend_cfg.sample_rate, backend_cfg.channels, backend_idx, usecase->id,
platform_get_snd_device_name(snd_device));
-
if (!platform_can_split_snd_device(my_data, snd_device, &num_devices, new_snd_devices))
new_snd_devices[0] = snd_device;
for (i = 0; i < num_devices; i++) {
ALOGI("%s: new_snd_devices[%d] is %d", __func__, i, new_snd_devices[i]);
- if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
- &new_bit_width, &new_sample_rate)) {
- platform_set_codec_backend_cfg(adev, new_snd_devices[i],
- new_bit_width, new_sample_rate, format);
- ret = true;
+ if ((platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
+ &backend_cfg))) {
+ platform_set_codec_backend_cfg(adev, new_snd_devices[i],
+ backend_cfg);
+ ret = true;
}
}
-
- return ret;
-}
-
-/*
- * configures afe with bit width and Sample Rate
- */
-
-int platform_set_capture_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- unsigned int bit_width, unsigned int sample_rate,
- audio_format_t format)
-{
- int ret = 0;
- int backend_idx = DEFAULT_CODEC_BACKEND;
- struct platform_data *my_data = (struct platform_data *)adev->platform;
-
- ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
- __func__, bit_width, sample_rate, backend_idx,
- platform_get_snd_device_name(snd_device));
-
- if (bit_width !=
- my_data->current_tx_backend_cfg[backend_idx].bit_width) {
-
- struct mixer_ctl *ctl = NULL;
- ctl = mixer_get_ctl_by_name(adev->mixer,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- if (!ctl) {
- ALOGE("%s:txbecf: afe: Could not get ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- return -EINVAL;
- }
- if (bit_width == 24) {
- if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
- ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
- else
- ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
- } else {
- ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
- }
-
- if (ret < 0) {
- ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- return -EINVAL;
- }
-
- my_data->current_tx_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:txbecf: afe: %s mixer set to %d bit", __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
- }
-
- /*
- * Backend sample rate configuration follows:
- * 16 bit record - 48khz for streams at any valid sample rate
- * 24 bit record - 48khz for stream sample rate less than 48khz
- * 24 bit record - 96khz for sample rate range of 48khz to 96khz
- * 24 bit record - 192khz for sample rate range of 96khz to 192 khz
- * Upper limit is inclusive in the sample rate range.
- */
- // TODO: This has to be more dynamic based on policy file
-
- if (sample_rate != my_data->current_tx_backend_cfg[(int)backend_idx].sample_rate) {
- /*
- * sample rate update is needed only for hifi audio enabled platforms
- */
- char *rate_str = NULL;
- struct mixer_ctl *ctl = NULL;
-
- switch (sample_rate) {
- case 8000:
- case 11025:
- case 16000:
- case 22050:
- case 32000:
- case 44100:
- case 48000:
- rate_str = "KHZ_48";
- break;
- case 64000:
- case 88200:
- case 96000:
- rate_str = "KHZ_96";
- break;
- case 176400:
- case 192000:
- rate_str = "KHZ_192";
- break;
- default:
- rate_str = "KHZ_48";
- break;
- }
-
- ctl = mixer_get_ctl_by_name(adev->mixer,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-
- if (!ctl) {
- ALOGE("%s:txbecf: afe: Could not get ctl to set the Sample Rate for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
- return -EINVAL;
- }
-
- ALOGD("%s:txbecf: afe: %s set to %s", __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl,
- rate_str);
- ret = mixer_ctl_set_enum_by_string(ctl, rate_str);
- if (ret < 0) {
- ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
- return -EINVAL;
- }
-
- my_data->current_tx_backend_cfg[backend_idx].sample_rate = sample_rate;
- }
-
return ret;
}
@@ -4397,43 +4844,54 @@
* goes through all the current usecases and picks the highest
* bitwidth & samplerate
*/
-bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
- unsigned int* new_bit_width,
- unsigned int* new_sample_rate)
+static bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
+ int backend_idx,
+ struct audio_backend_cfg *backend_cfg)
{
bool backend_change = false;
unsigned int bit_width;
unsigned int sample_rate;
- int backend_idx = DEFAULT_CODEC_BACKEND;
+ unsigned int channels;
struct platform_data *my_data = (struct platform_data *)adev->platform;
- bit_width = *new_bit_width;
- sample_rate = *new_sample_rate;
+ bit_width = backend_cfg->bit_width;
+ sample_rate = backend_cfg->sample_rate;
+ channels = backend_cfg->channels;
ALOGI("%s:txbecf: afe: Codec selected backend: %d current bit width: %d and "
- "sample rate: %d",__func__,backend_idx, bit_width, sample_rate);
+ "sample rate: %d, channels %d",__func__,backend_idx, bit_width,
+ sample_rate, channels);
// For voice calls use default configuration i.e. 16b/48K, only applicable to
// default backend
// force routing is not required here, caller will do it anyway
if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
- ALOGW("%s:txbecf: afe:Use default bw and sr for voice/voip calls and "
+ ALOGW("%s:txbecf: afe: Use default bw and sr for voice/voip calls and "
"for unprocessed/camera source", __func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
+ if (backend_idx == USB_AUDIO_TX_BACKEND) {
+ audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, false);
+ ALOGV("%s:txbecf: afe: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
+ __func__, bit_width, sample_rate, channels);
+ }
ALOGI("%s:txbecf: afe: Codec selected backend: %d updated bit width: %d and "
"sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
- if ((bit_width != my_data->current_tx_backend_cfg[backend_idx].bit_width) ||
- (sample_rate != my_data->current_tx_backend_cfg[backend_idx].sample_rate)) {
- *new_bit_width = bit_width;
- *new_sample_rate = sample_rate;
+ if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
+ (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+ (channels != my_data->current_backend_cfg[backend_idx].channels)) {
+ backend_cfg->bit_width = bit_width;
+ backend_cfg->sample_rate= sample_rate;
+ backend_cfg->channels = channels;
backend_change = true;
ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
- "new sample rate: %d", __func__, *new_bit_width, *new_sample_rate);
+ "new sample rate: %d new channel: %d",
+ __func__, backend_cfg->bit_width,
+ backend_cfg->sample_rate, backend_cfg->channels);
}
return backend_change;
@@ -4442,28 +4900,34 @@
bool platform_check_and_set_capture_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase *usecase, snd_device_t snd_device)
{
- unsigned int new_bit_width;
- unsigned int new_sample_rate;
- audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
- int backend_idx = DEFAULT_CODEC_BACKEND;
+ int backend_idx = platform_get_backend_index(snd_device);
int ret = 0;
+ struct audio_backend_cfg backend_cfg;
+
+ backend_cfg.passthrough_enabled = false;
if(usecase->type == PCM_CAPTURE) {
- new_sample_rate = usecase->stream.in->sample_rate;
- new_bit_width = usecase->stream.in->bit_width;
- format = usecase->stream.in->format;
+ backend_cfg.sample_rate= usecase->stream.in->sample_rate;
+ backend_cfg.bit_width= usecase->stream.in->bit_width;
+ backend_cfg.format= usecase->stream.in->format;
+ backend_cfg.channels = audio_channel_count_from_in_mask(usecase->stream.in->channel_mask);
} else {
- new_bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
- new_sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ backend_cfg.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ backend_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ backend_cfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ backend_cfg.channels = 1;
}
- ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d"
- ", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
- new_sample_rate, backend_idx, usecase->id,
+ ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, channel %d"
+ ", backend_idx %d usecase = %d device (%s)", __func__,
+ backend_cfg.bit_width,
+ backend_cfg.sample_rate,
+ backend_cfg.channels,
+ backend_idx, usecase->id,
platform_get_snd_device_name(snd_device));
- if (platform_check_capture_codec_backend_cfg(adev, &new_bit_width,
- &new_sample_rate)) {
- ret = platform_set_capture_codec_backend_cfg(adev, snd_device,
- new_bit_width, new_sample_rate, format);
+ if (platform_check_capture_codec_backend_cfg(adev, backend_idx,
+ &backend_cfg)) {
+ ret = platform_set_codec_backend_cfg(adev, snd_device,
+ backend_cfg);
if(!ret)
return true;
}
@@ -4626,7 +5090,7 @@
struct audio_device *adev = my_data->adev;
char block[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE];
int ret, count;
-
+ char *mix_ctl_name;
struct mixer_ctl *ctl;
char edid_data[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE + 1] = {0};
edid_audio_info *info;
@@ -4636,17 +5100,28 @@
return 0;
}
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ mix_ctl_name = "HDMI EDID";
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ mix_ctl_name = "Display Port EDID";
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ return -EINVAL;
+ }
+
if (my_data->edid_info == NULL) {
my_data->edid_info =
(struct edid_audio_info *)calloc(1, sizeof(struct edid_audio_info));
}
info = my_data->edid_info;
-
- ctl = mixer_get_ctl_by_name(adev->mixer, AUDIO_DATA_BLOCK_MIXER_CTL);
+ ctl = mixer_get_ctl_by_name(adev->mixer, mix_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
- __func__, AUDIO_DATA_BLOCK_MIXER_CTL);
+ __func__, mix_ctl_name);
goto fail;
}
@@ -4667,7 +5142,7 @@
memcpy(&edid_data[1], block, count);
if (!edid_get_sink_caps(info, edid_data)) {
- ALOGE("%s: Failed to get HDMI sink capabilities", __func__);
+ ALOGE("%s: Failed to get extn disp sink capabilities", __func__);
goto fail;
}
my_data->edid_valid = true;
@@ -4686,16 +5161,28 @@
int platform_set_channel_allocation(void *platform, int channel_alloc)
{
struct mixer_ctl *ctl;
- const char *mixer_ctl_name = "HDMI RX CA";
+ char *mixer_ctl_name;
int ret;
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
+ switch(my_data->ext_disp_type) {
+ case EXT_DISPLAY_TYPE_HDMI:
+ mixer_ctl_name = "HDMI RX CA";
+ break;
+ case EXT_DISPLAY_TYPE_DP:
+ mixer_ctl_name = "Display Port RX CA";
+ break;
+ default:
+ ALOGE("%s: Invalid disp_type %d", __func__, my_data->ext_disp_type);
+ return -EINVAL;
+ }
+
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
- ret = EINVAL;
+ return -EINVAL;
}
ALOGD(":%s channel allocation = 0x%x", __func__, channel_alloc);
ret = mixer_ctl_set_value(ctl, 0, channel_alloc);
@@ -4927,16 +5414,25 @@
void platform_invalidate_hdmi_config(void * platform)
{
- //reset HDMI EDID info
+ //reset ext display EDID info
struct platform_data *my_data = (struct platform_data *)platform;
my_data->edid_valid = false;
if (my_data->edid_info) {
memset(my_data->edid_info, 0, sizeof(struct edid_audio_info));
}
- //reset HDMI_RX_BACKEND to default values
- my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ if (my_data->ext_disp_type == EXT_DISPLAY_TYPE_HDMI) {
+ //reset HDMI_RX_BACKEND to default values
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ } else {
+ //reset Display port BACKEND to default values
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
+ my_data->current_backend_cfg[DISP_PORT_RX_BACKEND].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ }
+ my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
}
int platform_set_mixer_control(struct stream_out *out, const char * mixer_ctl_name,
@@ -4955,90 +5451,6 @@
return mixer_ctl_set_enum_by_string(ctl, mixer_val);
}
-static int set_mixer_control(struct mixer *mixer,
- const char * mixer_ctl_name,
- const char *mixer_val)
-{
- struct mixer_ctl *ctl;
- ALOGD("setting mixer ctl %s with value %s", mixer_ctl_name, mixer_val);
- ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
- if (!ctl) {
- ALOGE("%s: could not get ctl for mixer cmd - %s",
- __func__, mixer_ctl_name);
- return -EINVAL;
- }
-
- return mixer_ctl_set_enum_by_string(ctl, mixer_val);
-}
-
-int platform_set_hdmi_config(void *platform, uint32_t channel_count,
- uint32_t sample_rate, bool enable_passthrough)
-{
- struct platform_data *my_data = (struct platform_data *)platform;
- struct audio_device *adev = my_data->adev;
- const char *hdmi_format_ctrl = "HDMI RX Format";
- const char *hdmi_rate_ctrl = "HDMI_RX SampleRate";
- const char *hdmi_chans_ctrl = "HDMI_RX Channels";
- const char *channel_cnt_str = NULL;
-
- ALOGI("%s ch[%d] sr[%d], pthru[%d]", __func__,
- channel_count, sample_rate, enable_passthrough);
-
- switch (channel_count) {
- case 8:
- channel_cnt_str = "Eight"; break;
- case 7:
- channel_cnt_str = "Seven"; break;
- case 6:
- channel_cnt_str = "Six"; break;
- case 5:
- channel_cnt_str = "Five"; break;
- case 4:
- channel_cnt_str = "Four"; break;
- case 3:
- channel_cnt_str = "Three"; break;
- default:
- channel_cnt_str = "Two"; break;
- }
- ALOGV("%s: HDMI channel count: %s", __func__, channel_cnt_str);
- set_mixer_control(adev->mixer, hdmi_chans_ctrl, channel_cnt_str);
-
- if (enable_passthrough) {
- ALOGD("%s:HDMI compress format", __func__);
- set_mixer_control(adev->mixer, hdmi_format_ctrl, "Compr");
- } else {
- ALOGD("%s: HDMI PCM format", __func__);
- set_mixer_control(adev->mixer, hdmi_format_ctrl, "LPCM");
- }
-
- switch (sample_rate) {
- case 32000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_32");
- break;
- case 44100:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_44P1");
- break;
- case 96000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_96");
- break;
- case 128000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_128");
- break;
- case 176400:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_176_4");
- break;
- case 192000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_192");
- break;
- default:
- case 48000:
- set_mixer_control(adev->mixer, hdmi_rate_ctrl, "KHZ_48");
- break;
- }
-
- return 0;
-}
-
int platform_set_device_params(struct stream_out *out, int param, int value)
{
struct audio_device *adev = out->dev;
@@ -5073,7 +5485,9 @@
if (snd_device == SND_DEVICE_OUT_SPEAKER ||
snd_device == SND_DEVICE_OUT_SPEAKER_VBAT ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
- snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) {
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2) {
ret = true;
}
@@ -5091,12 +5505,18 @@
case SND_DEVICE_OUT_VOICE_SPEAKER:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED);
+ break;
case SND_DEVICE_OUT_SPEAKER_VBAT:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
break;
case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT);
break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT);
+ break;
default:
acdb_id = -EINVAL;
break;
@@ -5114,14 +5534,34 @@
return SND_DEVICE_OUT_SPEAKER_PROTECTED;
case SND_DEVICE_OUT_VOICE_SPEAKER:
return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
case SND_DEVICE_OUT_SPEAKER_VBAT:
return SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
default:
return snd_device;
}
}
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device)
+{
+ switch(snd_device) {
+ case SND_DEVICE_OUT_SPEAKER_PROTECTED:
+ case SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2;
+ default:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ }
+}
int platform_spkr_prot_is_wsa_analog_mode(void *adev __unused)
{
@@ -5202,3 +5642,153 @@
}
return 0;
}
+
+static void make_cal_cfg(acdb_audio_cal_cfg_t* cal, int acdb_dev_id,
+ int acdb_device_type, int app_type, int topology_id,
+ int sample_rate, uint32_t module_id, uint32_t param_id, bool persist)
+{
+ int persist_send_flags = 1;
+
+ if (!cal) {
+ return;
+ }
+
+ if (persist)
+ persist_send_flags |= 0x2;
+
+ memset(cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+ cal->persist = persist;
+ cal->app_type = app_type;
+ cal->acdb_dev_id = acdb_dev_id;
+ cal->sampling_rate = sample_rate;
+ cal->topo_id = topology_id;
+ //if module and param id is set to 0, the whole blob will be stored
+ //or sent to the DSP
+ cal->module_id = module_id;
+ cal->param_id = param_id;
+ cal->cal_type = acdb_device_type;
+ cal->persist = persist;
+
+}
+
+int platform_send_audio_cal(void* platform, int acdb_dev_id,
+ int acdb_device_type, int app_type, int topology_id, int sample_rate,
+ uint32_t module_id, uint32_t param_id, void* data, int length, bool persist)
+{
+ int ret = 0;
+ struct platform_data *my_data = (struct platform_data *)platform;
+ acdb_audio_cal_cfg_t cal;
+ memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+ if (!my_data) {
+ ret = -EINVAL;
+ goto ERROR_RETURN;
+ }
+
+ make_cal_cfg(&cal, acdb_dev_id, acdb_device_type, app_type, topology_id,
+ sample_rate, module_id, param_id, true);
+
+ if (my_data->acdb_set_audio_cal) {
+ // persist audio cal in local cache
+ if (persist) {
+ ret = my_data->acdb_set_audio_cal((void*)&cal, data, (uint32_t)length);
+ }
+ // send audio cal to dsp
+ if (ret == 0) {
+ cal.persist = false;
+ ret = my_data->acdb_set_audio_cal((void*)&cal, data, (uint32_t)length);
+ if (persist && (ret != 0)) {
+ ALOGV("[%s] audio cal stored with success, ignore set cal failure", __func__);
+ ret = 0;
+ }
+ }
+ }
+
+ERROR_RETURN:
+ return ret;
+}
+
+int platform_get_audio_cal(void* platform, int acdb_dev_id,
+ int acdb_device_type, int app_type, int topology_id,
+ int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int* length, bool persist)
+{
+ int ret = 0;
+ struct platform_data *my_data = (struct platform_data *)platform;
+ acdb_audio_cal_cfg_t cal;
+ memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+ if (!my_data) {
+ ret = -EINVAL;
+ goto ERROR_RETURN;
+ }
+
+ make_cal_cfg(&cal, acdb_dev_id, acdb_device_type, app_type, topology_id,
+ sample_rate, module_id, param_id, false);
+
+ if (my_data->acdb_get_audio_cal) {
+ // get cal from dsp
+ ret = my_data->acdb_get_audio_cal((void*)&cal, data, (uint32_t*)length);
+ // get cached cal if prevoius attempt fails and persist flag is set
+ if ((ret != 0) && persist) {
+ cal.persist = true;
+ ret = my_data->acdb_get_audio_cal((void*)&cal, data, (uint32_t*)length);
+ }
+ }
+
+ERROR_RETURN:
+ return ret;
+}
+
+int platform_store_audio_cal(void* platform, int acdb_dev_id,
+ int acdb_device_type, int app_type, int topology_id,
+ int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int length)
+{
+ int ret = 0;
+ struct platform_data *my_data = (struct platform_data *)platform;
+ acdb_audio_cal_cfg_t cal;
+ memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+ if (!my_data) {
+ ret = -EINVAL;
+ goto ERROR_RETURN;
+ }
+
+ make_cal_cfg(&cal, acdb_dev_id, acdb_device_type, app_type, topology_id,
+ sample_rate, module_id, param_id, true);
+
+ if (my_data->acdb_set_audio_cal) {
+ ret = my_data->acdb_set_audio_cal((void*)&cal, data, (uint32_t)length);
+ }
+
+ERROR_RETURN:
+ return ret;
+}
+
+int platform_retrieve_audio_cal(void* platform, int acdb_dev_id,
+ int acdb_device_type, int app_type, int topology_id,
+ int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int* length)
+{
+ int ret = 0;
+ struct platform_data *my_data = (struct platform_data *)platform;
+ acdb_audio_cal_cfg_t cal;
+ memset(&cal, 0, sizeof(acdb_audio_cal_cfg_t));
+
+ if (!my_data) {
+ ret = -EINVAL;
+ goto ERROR_RETURN;
+ }
+
+ make_cal_cfg(&cal, acdb_dev_id, acdb_device_type, app_type, topology_id,
+ sample_rate, module_id, param_id, true);
+
+ if (my_data->acdb_get_audio_cal) {
+ ret = my_data->acdb_get_audio_cal((void*)&cal, data, (uint32_t*)length);
+ }
+
+ERROR_RETURN:
+ return ret;
+}
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 019678a..c231843 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -39,6 +39,11 @@
SOURCE_QUAD_MIC = 0x8, /* Target contains 4 mics */
};
+enum {
+ SPKR_1,
+ SPKR_2
+};
+
/*
* Below are the devices for which is back end is same, SLIMBUS_0_RX.
* All these devices are handled by the internal HW codec. We can
@@ -80,6 +85,7 @@
SND_DEVICE_OUT_SPEAKER_VBAT,
SND_DEVICE_OUT_LINE,
SND_DEVICE_OUT_HEADPHONES,
+ SND_DEVICE_OUT_HEADPHONES_DSD,
SND_DEVICE_OUT_HEADPHONES_44_1,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_LINE,
@@ -88,12 +94,18 @@
SND_DEVICE_OUT_VOICE_HANDSET,
SND_DEVICE_OUT_VOICE_SPEAKER,
SND_DEVICE_OUT_VOICE_SPEAKER_VBAT,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT,
SND_DEVICE_OUT_VOICE_HEADPHONES,
SND_DEVICE_OUT_VOICE_LINE,
SND_DEVICE_OUT_HDMI,
SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+ SND_DEVICE_OUT_DISPLAY_PORT,
+ SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT,
SND_DEVICE_OUT_BT_SCO,
SND_DEVICE_OUT_BT_SCO_WB,
+ SND_DEVICE_OUT_BT_A2DP,
+ SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
@@ -110,10 +122,13 @@
SND_DEVICE_OUT_ANC_HANDSET,
SND_DEVICE_OUT_SPEAKER_PROTECTED,
SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED,
SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT,
SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT,
SND_DEVICE_OUT_SPEAKER_WSA,
SND_DEVICE_OUT_VOICE_SPEAKER_WSA,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA,
SND_DEVICE_OUT_END,
/*
@@ -167,6 +182,8 @@
SND_DEVICE_IN_HANDSET_STEREO_DMIC,
SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2,
SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE,
SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE,
SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE,
@@ -177,6 +194,7 @@
SND_DEVICE_IN_SPEAKER_QMIC_AEC,
SND_DEVICE_IN_SPEAKER_QMIC_NS,
SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS,
+ SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE,
SND_DEVICE_IN_THREE_MIC,
SND_DEVICE_IN_HANDSET_TMIC,
SND_DEVICE_IN_UNPROCESSED_MIC,
@@ -189,19 +207,29 @@
SND_DEVICE_MAX = SND_DEVICE_IN_END,
};
-
+#define INPUT_SAMPLING_RATE_DSD64 2822400
+#define INPUT_SAMPLING_RATE_DSD128 5644800
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
#define OUTPUT_SAMPLING_RATE_44100 44100
-#define MAX_CODEC_TX_BACKENDS 1
+#define OUTPUT_SAMPLING_RATE_DSD64 176400
+#define OUTPUT_SAMPLING_RATE_DSD128 352800
enum {
DEFAULT_CODEC_BACKEND,
SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,
+ DSD_NATIVE_BACKEND,
+ SLIMBUS_2_RX = DSD_NATIVE_BACKEND,
HEADPHONE_44_1_BACKEND,
SLIMBUS_5_RX = HEADPHONE_44_1_BACKEND,
HEADPHONE_BACKEND,
SLIMBUS_6_RX = HEADPHONE_BACKEND,
HDMI_RX_BACKEND,
+ DISP_PORT_RX_BACKEND,
USB_AUDIO_RX_BACKEND,
+ MAX_RX_CODEC_BACKENDS = USB_AUDIO_RX_BACKEND,
+ /* TX BE follows RX BE */
+ SLIMBUS_0_TX,
+ DEFAULT_CODEC_TX_BACKEND = SLIMBUS_0_TX,
+ USB_AUDIO_TX_BACKEND,
MAX_CODEC_BACKENDS
};
@@ -229,7 +257,14 @@
* We should take care of returning proper size when AudioFlinger queries for
* the buffer size of an input/output stream
*/
+
+/* for 384Khz output below period size corresponds to 20ms worth duration of buffer,
+ * current implementation can support buffer size of 40ms duration
+ * for 32b/384Khz/stereo output.
+ */
#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1920
+#define DEEP_BUFFER_OUTPUT_PERIOD_DURATION 40 /* 40 milisecs */
+
#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 2
#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
@@ -264,6 +299,7 @@
#define AUDIO_RECORD_PCM_DEVICE 0
#define MULTIMEDIA2_PCM_DEVICE 1
#define MULTIMEDIA3_PCM_DEVICE 4
+#define MULTIMEDIA9_PCM_DEVICE 32
#define FM_PLAYBACK_PCM_DEVICE 5
#define FM_CAPTURE_PCM_DEVICE 6
#define HFP_PCM_RX 5
@@ -444,7 +480,8 @@
enum {
LEGACY_PCM = 0,
PASSTHROUGH,
- PASSTHROUGH_CONVERT
+ PASSTHROUGH_CONVERT,
+ PASSTHROUGH_DSD
};
/*
* ID for setting mute and lateny on the device side
@@ -460,4 +497,13 @@
char device_name[100];
char interface_name[100];
};
+
+struct audio_backend_cfg {
+ unsigned int sample_rate;
+ unsigned int channels;
+ unsigned int bit_width;
+ bool passthrough_enabled;
+ audio_format_t format;
+};
+
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 0bb73f3..7dcd1b6 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -23,10 +23,14 @@
#define CODEC_BACKEND_DEFAULT_BIT_WIDTH 16
#define CODEC_BACKEND_DEFAULT_SAMPLE_RATE 48000
+#define CODEC_BACKEND_DEFAULT_CHANNELS 2
+#define CODEC_BACKEND_DEFAULT_TX_CHANNELS 1
+
enum {
NATIVE_AUDIO_MODE_SRC = 1,
NATIVE_AUDIO_MODE_TRUE_44_1,
+ NATIVE_AUDIO_MODE_MULTIPLE_44_1,
NATIVE_AUDIO_MODE_INVALID
};
@@ -36,6 +40,8 @@
int na_mode;
} native_audio_prop;
+enum card_status_t;
+
void *platform_init(struct audio_device *adev);
void platform_deinit(void *platform);
const char *platform_get_snd_device_name(snd_device_t snd_device);
@@ -139,6 +145,7 @@
bool platform_can_enable_spkr_prot_on_device(snd_device_t snd_device);
int platform_get_spkr_prot_acdb_id(snd_device_t snd_device);
int platform_get_spkr_prot_snd_device(snd_device_t snd_device);
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device);
int platform_spkr_prot_is_wsa_analog_mode(void *adev);
bool platform_can_split_snd_device(void *platform,
snd_device_t snd_device,
@@ -151,4 +158,27 @@
bool enable,
char * str);
bool platform_supports_true_32bit();
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device, snd_device_t cuurent_snd_device);
+bool platform_check_codec_dsd_support(void *platform);
+bool platform_check_codec_asrc_support(void *platform);
+int platform_get_backend_index(snd_device_t snd_device);
+int platform_get_ext_disp_type(void *platform);
+
+int platform_send_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
+ int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int length, bool persist);
+
+int platform_get_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
+ int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int* length, bool persist);
+
+int platform_store_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
+ int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int length);
+
+int platform_retrieve_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
+ int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
+ void* data, int* length);
+
+unsigned char* platform_get_license(void* platform, int* size);
#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/voice.c b/hal/voice.c
index f86483e..b84c7b7 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -417,11 +417,13 @@
int err = 0;
adev->voice.mic_mute = state;
- if (adev->mode == AUDIO_MODE_IN_CALL)
+ if (audio_extn_hfp_is_active(adev)) {
+ err = hfp_set_mic_mute(adev, state);
+ } else if (adev->mode == AUDIO_MODE_IN_CALL) {
err = platform_set_mic_mute(adev->platform, state);
- if (adev->mode == AUDIO_MODE_IN_COMMUNICATION)
+ } else if (adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
err = voice_extn_compress_voip_set_mic_mute(adev, state);
-
+ }
return err;
}
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index 7293485..3222e0b 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -244,6 +244,7 @@
{
int ret = 0;
struct audio_usecase *uc_info;
+ struct listnode *node;
ALOGD("%s: enter, out_stream_count=%d, in_stream_count=%d",
__func__, voip_data.out_stream_count, voip_data.in_stream_count);
@@ -277,6 +278,12 @@
list_remove(&uc_info->list);
free(uc_info);
+
+ // restore device for other active usecases
+ list_for_each(node, &adev->usecase_list) {
+ uc_info = node_to_item(node, struct audio_usecase, list);
+ select_devices(adev, uc_info->id);
+ }
} else
ALOGV("%s: NO-OP because out_stream_count=%d, in_stream_count=%d",
__func__, voip_data.out_stream_count, voip_data.in_stream_count);
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index b89c82c..d06929c 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -435,6 +435,48 @@
return BAD_VALUE;
}
+bool AudioPolicyManagerCustom::isInvalidationOfMusicStreamNeeded(routing_strategy strategy)
+{
+ if (strategy == STRATEGY_MEDIA) {
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> newOutputDesc = mOutputs.valueAt(i);
+ if (newOutputDesc->mFormat == AUDIO_FORMAT_DSD)
+ return false;
+ }
+ }
+ return true;
+}
+
+void AudioPolicyManagerCustom::checkOutputForStrategy(routing_strategy strategy)
+{
+ audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
+ audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mOutputs);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+ // also take into account external policy-related changes: add all outputs which are
+ // associated with policies in the "before" and "after" output vectors
+ ALOGV("checkOutputForStrategy(): policy related outputs");
+ for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
+ const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ srcOutputs.add(desc->mIoHandle);
+ ALOGV(" previous outputs: adding %d", desc->mIoHandle);
+ }
+ }
+ for (size_t i = 0 ; i < mOutputs.size() ; i++) {
+ const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ dstOutputs.add(desc->mIoHandle);
+ ALOGV(" new outputs: adding %d", desc->mIoHandle);
+ }
+ }
+
+ if (!vectorsEqual(srcOutputs,dstOutputs) && isInvalidationOfMusicStreamNeeded(strategy)) {
+ AudioPolicyManager::checkOutputForStrategy(strategy);
+ }
+}
+
// This function checks for the parameters which can be offloaded.
// This can be enhanced depending on the capability of the DSP and policy
// of the system.
@@ -464,6 +506,11 @@
}
}
#endif
+ if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
+ isInCall() && (offloadInfo.format == AUDIO_FORMAT_DSD)) {
+ ALOGD("blocking DSD compress offload on call mode");
+ return false;
+ }
#ifdef RECORD_PLAY_CONCURRENCY
char recConcPropValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;
@@ -511,6 +558,14 @@
ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
return false;
}
+
+ if ((((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.bit_rate > MAX_BITRATE_WMA)) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_PRO)) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_LOSSLESS))){
+ //Safely choose the min bitrate as threshold and leave the restriction to NT decoder as we can't distinguish wma pro and wma lossless here.
+ ALOGD("offload disabled for WMA/WMA_PRO/WMA_LOSSLESS clips with bit rate over maximum supported value");
+ return false;
+ }
#endif
//TODO: enable audio offloading with video when ready
const bool allowOffloadWithVideo =
@@ -549,6 +604,7 @@
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DSD) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))
return false;
#endif
@@ -563,16 +619,7 @@
if (mEffects.isNonOffloadableEffectEnabled()) {
return false;
}
- // Check for soundcard status
- String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
- String8("SND_CARD_STATUS"));
- AudioParameter result = AudioParameter(valueStr);
- int isonline = 0;
- if ((result.getInt(String8("SND_CARD_STATUS"), isonline) == NO_ERROR)
- && !isonline) {
- ALOGD("copl: soundcard is offline rejecting offload request");
- return false;
- }
+
// See if there is a profile to support this.
// AUDIO_DEVICE_NONE
sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
@@ -846,6 +893,26 @@
}
#endif
+
+ sp<SwAudioOutputDescriptor> outputDesc = NULL;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ outputDesc = mOutputs.valueAt(i);
+ if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("voice_conc:ouput desc / profile is NULL");
+ continue;
+ }
+
+ if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
+ (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+ (outputDesc->mFormat == AUDIO_FORMAT_DSD)) {
+ ALOGD("voice_conc:calling closeOutput on call mode for DSD COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ // call invalidate for music, so that DSD compress will fallback to deep-buffer.
+ mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+ }
+
+ }
+
#ifdef RECORD_PLAY_CONCURRENCY
char recConcPropValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;
@@ -1073,6 +1140,7 @@
outputDesc->sharesHwModuleWith(desc) &&
(newDevice != desc->device())) {
audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/);
+ bool force = desc->device() != dev;
uint32_t delayMs;
if (dev == prevDevice) {
delayMs = 0;
@@ -1081,7 +1149,7 @@
}
setOutputDevice(desc,
dev,
- true,
+ force,
delayMs);
}
}
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index dfda1c9..00da599 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -39,6 +39,10 @@
#ifndef AUDIO_EXTN_AFE_PROXY_ENABLED
#define AUDIO_DEVICE_OUT_PROXY 0x1000000
#endif
+
+#define MAX_BITRATE_WMA 384000
+#define MAX_BITRATE_WMA_PRO 1536000
+#define MAX_BITRATE_WMA_LOSSLESS 1152000
// ----------------------------------------------------------------------------
class AudioPolicyManagerCustom: public AudioPolicyManager
@@ -92,6 +96,14 @@
// see getDeviceForStrategy() for the use of fromCache parameter
audio_devices_t getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
bool fromCache);
+
+ // avoid invalidation for active music stream on previous outputs
+ // which is supported on the new device.
+ bool isInvalidationOfMusicStreamNeeded(routing_strategy strategy);
+
+ // Must be called before updateDevicesAndOutputs()
+ void checkOutputForStrategy(routing_strategy strategy);
+
// returns true if given output is direct output
bool isDirectOutput(audio_io_handle_t output);
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index 464bc0d..fd5ee8c 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -852,8 +852,9 @@
if (pCmdData == NULL ||
cmdSize < (int)(sizeof(effect_param_t) + sizeof(uint32_t)) ||
pReplyData == NULL ||
- *replySize < (int)(sizeof(effect_param_t) + sizeof(uint32_t) +
- sizeof(uint16_t))) {
+ *replySize < (int)(sizeof(effect_param_t) + sizeof(uint32_t) + sizeof(uint16_t)) ||
+ // constrain memcpy below
+ ((effect_param_t *)pCmdData)->psize > *replySize - sizeof(effect_param_t)) {
status = -EINVAL;
ALOGW("EFFECT_CMD_GET_PARAM invalid command cmdSize %d *replySize %d",
cmdSize, *replySize);
diff --git a/qahw_api/Android.mk b/qahw_api/Android.mk
new file mode 100644
index 0000000..8c99c5b
--- /dev/null
+++ b/qahw_api/Android.mk
@@ -0,0 +1,32 @@
+ifeq ($(strip $(BOARD_SUPPORTS_QAHW)),true)
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+libqahw-inc := $(LOCAL_PATH)/inc
+
+LOCAL_MODULE := libqahw
+LOCAL_MODULE_TAGS := optional
+LOCAL_MODULE_OWNER := qti
+LOCAL_C_INCLUDES := $(libqahw-inc)
+
+LOCAL_SRC_FILES := \
+ src/qahw.c
+
+LOCAL_SHARED_LIBRARIES := \
+ liblog \
+ libcutils \
+ libhardware
+
+LOCAL_COPY_HEADERS_TO := mm-audio/qahw_api/inc
+LOCAL_COPY_HEADERS := inc/qahw_api.h
+LOCAL_COPY_HEADERS += inc/qahw_defs.h
+
+LOCAL_PRELINK_MODULE := false
+
+include $(BUILD_SHARED_LIBRARY)
+
+#test app compilation
+include $(LOCAL_PATH)/test/Android.mk
+endif
diff --git a/qahw_api/inc/qahw_api.h b/qahw_api/inc/qahw_api.h
new file mode 100644
index 0000000..17f6f5f
--- /dev/null
+++ b/qahw_api/inc/qahw_api.h
@@ -0,0 +1,420 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2011 The Android Open Source Project *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef QTI_AUDIO_HAL_API_H
+#define QTI_AUDIO_HAL_API_H
+
+#include <stdint.h>
+#include <strings.h>
+#include <sys/cdefs.h>
+#include <sys/types.h>
+#include <sys/time.h>
+#include <cutils/bitops.h>
+#include <system/audio.h>
+#include "qahw_defs.h"
+
+__BEGIN_DECLS
+/*
+ * Helper macros for module implementors.
+ *
+ * The derived modules should provide convenience macros for supported
+ * versions so that implementations can explicitly specify module
+ * versions at definition time.
+ */
+
+#define QAHW_MAKE_API_VERSION(maj,min) \
+ ((((maj) & 0xff) << 8) | ((min) & 0xff))
+
+/* First generation of audio devices had version hardcoded to 0. all devices with
+ * versions < 1.0 will be considered of first generation API.
+ */
+#define QAHW_MODULE_API_VERSION_0_0 QAHW_MAKE_API_VERSION(0, 0)
+
+/* Minimal QTI audio HAL version supported by the audio framework */
+#define QAHW_MODULE_API_VERSION_MIN QAHW_MODULE_API_VERSION_0_0
+
+/**
+ * List of known audio HAL modules. This is the base name of the audio HAL
+ * library composed of the "audio." prefix, one of the base names below and
+ * a suffix specific to the device.
+ * e.g: audio.primary.goldfish.so or audio.a2dp.default.so
+ */
+
+#define QAHW_MODULE_ID_PRIMARY "audio.primary"
+#define QAHW_MODULE_ID_A2DP "audio.a2dp"
+#define QAHW_MODULE_ID_USB "audio.usb"
+
+typedef void qahw_module_handle_t;
+typedef void qahw_stream_handle_t;
+
+/**************************************/
+/* Output stream specific APIs **/
+
+/*
+ * This method creates and opens the audio hardware output stream.
+ * The "address" parameter qualifies the "devices" audio device type if needed.
+ * The format format depends on the device type:
+ * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
+ * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y"
+ * - Other devices may use a number or any other string.
+ */
+
+int qahw_open_output_stream(qahw_module_handle_t *hw_module,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ qahw_stream_handle_t **out_handle,
+ const char *address);
+
+int qahw_close_output_stream(qahw_stream_handle_t *out_handle);
+
+
+/*
+ * Return the sampling rate in Hz - eg. 44100.
+ */
+uint32_t qahw_out_get_sample_rate(const qahw_stream_handle_t *stream);
+
+/*
+ * use set_parameters with key QAHW_PARAMETER_STREAM_SAMPLING_RATE
+ */
+int qahw_out_set_sample_rate(qahw_stream_handle_t *stream, uint32_t rate);
+
+/*
+ * Return size of input/output buffer in bytes for this stream - eg. 4800.
+ * It should be a multiple of the frame size. See also get_input_buffer_size.
+ */
+size_t qahw_out_get_buffer_size(const qahw_stream_handle_t *stream);
+
+/*
+ * Return the channel mask -
+ * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
+ */
+audio_channel_mask_t qahw_out_get_channels(const qahw_stream_handle_t *stream);
+
+/*
+ * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
+ */
+audio_format_t qahw_out_get_format(const qahw_stream_handle_t *stream);
+
+/*
+ * Put the audio hardware input/output into standby mode.
+ * Driver should exit from standby mode at the next I/O operation.
+ * Returns 0 on success and <0 on failure.
+ */
+int qahw_out_standby(qahw_stream_handle_t *stream);
+
+/*
+ * set/get audio stream parameters. The function accepts a list of
+ * parameter key value pairs in the form: key1=value1;key2=value2;...
+ *
+ * Some keys are reserved for standard parameters (See AudioParameter class)
+ *
+ * If the implementation does not accept a parameter change while
+ * the output is active but the parameter is acceptable otherwise, it must
+ * return -ENOSYS.
+ *
+ * The audio flinger will put the stream in standby and then change the
+ * parameter value.
+ */
+int qahw_out_set_parameters(qahw_stream_handle_t *stream, const char*kv_pairs);
+
+/*
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+char* qahw_out_get_parameters(const qahw_stream_handle_t *stream,
+ const char *keys);
+
+/*
+ * Return the audio hardware driver estimated latency in milliseconds.
+ */
+uint32_t qahw_out_get_latency(const qahw_stream_handle_t *stream);
+
+/*
+ * Use this method in situations where audio mixing is done in the
+ * hardware. This method serves as a direct interface with hardware,
+ * allowing you to directly set the volume as apposed to via the framework.
+ * This method might produce multiple PCM outputs or hardware accelerated
+ * codecs, such as MP3 or AAC.
+ */
+int qahw_out_set_volume(qahw_stream_handle_t *stream, float left, float right);
+
+/*
+ * Write audio buffer present in meta_data starting from offset
+ * along with timestamp to driver. Returns number of bytes
+ * written or a negative status_t. If at least one frame was written successfully
+ * prior to the error, it is suggested that the driver return that successful
+ * (short) byte count and then return an error in the subsequent call.
+ * timestamp is only sent driver is session has been opened with timestamp flag
+ * otherwise its ignored.
+ *
+ * If set_callback() has previously been called to enable non-blocking mode
+ * the write() is not allowed to block. It must write only the number of
+ * bytes that currently fit in the driver/hardware buffer and then return
+ * this byte count. If this is less than the requested write size the
+ * callback function must be called when more space is available in the
+ * driver/hardware buffer.
+ */
+ssize_t qahw_out_write(qahw_stream_handle_t *stream,
+ qahw_out_buffer_t *out_buf);
+
+/*
+ * return the number of audio frames written by the audio dsp to DAC since
+ * the output has exited standby
+ */
+int qahw_out_get_render_position(const qahw_stream_handle_t *stream,
+ uint32_t *dsp_frames);
+
+/*
+ * set the callback function for notifying completion of non-blocking
+ * write and drain.
+ * Calling this function implies that all future rite() and drain()
+ * must be non-blocking and use the callback to signal completion.
+ */
+int qahw_out_set_callback(qahw_stream_handle_t *stream,
+ qahw_stream_callback_t callback,
+ void *cookie);
+
+/*
+ * Notifies to the audio driver to stop playback however the queued buffers are
+ * retained by the hardware. Useful for implementing pause/resume. Empty implementation
+ * if not supported however should be implemented for hardware with non-trivial
+ * latency. In the pause state audio hardware could still be using power. User may
+ * consider calling suspend after a timeout.
+ *
+ * Implementation of this function is mandatory for offloaded playback.
+ */
+int qahw_out_pause(qahw_stream_handle_t *out_handle);
+
+/*
+ * Notifies to the audio driver to resume playback following a pause.
+ * Returns error if called without matching pause.
+ *
+ * Implementation of this function is mandatory for offloaded playback.
+ */
+int qahw_out_resume(qahw_stream_handle_t *out_handle);
+
+/*
+ * Requests notification when data buffered by the driver/hardware has
+ * been played. If set_callback() has previously been called to enable
+ * non-blocking mode, the drain() must not block, instead it should return
+ * quickly and completion of the drain is notified through the callback.
+ * If set_callback() has not been called, the drain() must block until
+ * completion.
+ * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
+ * data has been played.
+ * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all
+ * data for the current track has played to allow time for the framework
+ * to perform a gapless track switch.
+ *
+ * Drain must return immediately on stop() and flush() call
+ *
+ * Implementation of this function is mandatory for offloaded playback.
+ */
+int qahw_out_drain(qahw_stream_handle_t *out_handle, qahw_drain_type_t type);
+
+/*
+ * Notifies to the audio driver to flush the queued data. Stream must already
+ * be paused before calling flush().
+ *
+ * Implementation of this function is mandatory for offloaded playback.
+ */
+int qahw_out_flush(qahw_stream_handle_t *out_handle);
+
+/*
+ * Return a recent count of the number of audio frames presented to an external observer.
+ * This excludes frames which have been written but are still in the pipeline.
+ * The count is not reset to zero when output enters standby.
+ * Also returns the value of CLOCK_MONOTONIC as of this presentation count.
+ * The returned count is expected to be 'recent',
+ * but does not need to be the most recent possible value.
+ * However, the associated time should correspond to whatever count is returned.
+ * Example: assume that N+M frames have been presented, where M is a 'small' number.
+ * Then it is permissible to return N instead of N+M,
+ * and the timestamp should correspond to N rather than N+M.
+ * The terms 'recent' and 'small' are not defined.
+ * They reflect the quality of the implementation.
+ *
+ * 3.0 and higher only.
+ */
+int qahw_out_get_presentation_position(const qahw_stream_handle_t *out_handle,
+ uint64_t *frames, struct timespec *timestamp);
+
+/* Input stream specific APIs */
+
+/* This method creates and opens the audio hardware input stream */
+int qahw_open_input_stream(qahw_module_handle_t *hw_module,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ qahw_stream_handle_t **stream_in,
+ audio_input_flags_t flags,
+ const char *address,
+ audio_source_t source);
+
+int qahw_close_input_stream(qahw_stream_handle_t *in_handle);
+
+
+/*
+ * Return the sampling rate in Hz - eg. 44100.
+ */
+uint32_t qahw_in_get_sample_rate(const qahw_stream_handle_t *in_handle);
+
+/*
+ * currently unused - use set_parameters with key
+ * QAHW_PARAMETER_STREAM_SAMPLING_RATE
+ */
+int qahw_in_set_sample_rate(qahw_stream_handle_t *in_handle, uint32_t rate);
+
+/*
+ * Return size of input/output buffer in bytes for this stream - eg. 4800.
+ * It should be a multiple of the frame size. See also get_input_buffer_size.
+ */
+size_t qahw_in_get_buffer_size(const qahw_stream_handle_t *in_handle);
+
+/*
+ * Return the channel mask -
+ * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
+ */
+audio_channel_mask_t qahw_in_get_channels(const qahw_stream_handle_t *in_handle);
+
+/*
+ * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
+ */
+audio_format_t qahw_in_get_format(const qahw_stream_handle_t *in_handle);
+
+/*
+ * currently unused - use set_parameters with key
+ * QAHW_PARAMETER_STREAM_FORMAT
+ */
+int qahw_in_set_format(qahw_stream_handle_t *in_handle, audio_format_t format);
+
+/*
+ * Put the audio hardware input/output into standby mode.
+ * Driver should exit from standby mode at the next I/O operation.
+ * Returns 0 on success and <0 on failure.
+ */
+int qahw_in_standby(qahw_stream_handle_t *in_handle);
+
+/*
+ * set/get audio stream parameters. The function accepts a list of
+ * parameter key value pairs in the form: key1=value1;key2=value2;...
+ *
+ * Some keys are reserved for standard parameters (See AudioParameter class)
+ *
+ * If the implementation does not accept a parameter change while
+ * the output is active but the parameter is acceptable otherwise, it must
+ * return -ENOSYS.
+ *
+ * The audio flinger will put the stream in standby and then change the
+ * parameter value.
+ */
+int qahw_in_set_parameters(qahw_stream_handle_t *in_handle, const char *kv_pairs);
+
+/*
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+char* qahw_in_get_parameters(const qahw_stream_handle_t *in_handle,
+ const char *keys);
+/*
+ * Read audio buffer in from audio driver. Returns number of bytes read, or a
+ * negative status_t. meta_data structure is filled buffer pointer, start
+ * offset and valid catpure timestamp (if session is opened with timetamp flag)
+ * and buffer. if at least one frame was read prior to the error,
+ * read should return that byte count and then return an error in the
+ * subsequent call.
+ */
+ssize_t qahw_in_read(qahw_stream_handle_t *in_handle,
+ qahw_in_buffer_t *in_buf);
+/*
+ * Return the amount of input frames lost in the audio driver since the
+ * last call of this function.
+ * Audio driver is expected to reset the value to 0 and restart counting
+ * upon returning the current value by this function call.
+ * Such loss typically occurs when the user space process is blocked
+ * longer than the capacity of audio driver buffers.
+ *
+ * Unit: the number of input audio frames
+ */
+uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle);
+
+/*
+ * Return a recent count of the number of audio frames received and
+ * the clock time associated with that frame count.
+ *
+ * frames is the total frame count received. This should be as early in
+ * the capture pipeline as possible. In general,
+ * frames should be non-negative and should not go "backwards".
+ *
+ * time is the clock MONOTONIC time when frames was measured. In general,
+ * time should be a positive quantity and should not go "backwards".
+ *
+ * The status returned is 0 on success, -ENOSYS if the device is not
+ * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
+ */
+int qahw_in_get_capture_position(const qahw_stream_handle_t *in_handle,
+ int64_t *frames, int64_t *time);
+
+/* Module specific APIs */
+
+/* convenience API for opening and closing an audio HAL module */
+qahw_module_handle_t *qahw_load_module(const char *hw_module_id);
+
+int qahw_unload_module(qahw_module_handle_t *hw_module);
+
+/*
+ * check to see if the audio hardware interface has been initialized.
+ * returns 0 on success, -ENODEV on failure.
+ */
+int qahw_init_check(const qahw_module_handle_t *hw_module);
+
+/* set the audio volume of a voice call. Range is between 0.0 and 1.0 */
+int qahw_set_voice_volume(qahw_module_handle_t *hw_module, float volume);
+
+/*
+ * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
+ * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
+ * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
+ */
+int qahw_set_mode(qahw_module_handle_t *hw_module, audio_mode_t mode);
+
+/* set/get global audio parameters */
+int qahw_set_parameters(qahw_module_handle_t *hw_module, const char *kv_pairs);
+
+/*
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+char* qahw_get_parameters(const qahw_module_handle_t *hw_module,
+ const char *keys);
+
+/* Returns audio input buffer size according to parameters passed or
+ * 0 if one of the parameters is not supported.
+ * See also get_buffer_size which is for a particular stream.
+ */
+size_t qahw_get_input_buffer_size(const qahw_module_handle_t *hw_module,
+ const struct audio_config *config);
+
+/*returns current QTI HAL version */
+int qahw_get_version();
+
+__END_DECLS
+
+#endif // QTI_AUDIO_HAL_API_H
diff --git a/qahw_api/inc/qahw_defs.h b/qahw_api/inc/qahw_defs.h
new file mode 100644
index 0000000..4441435
--- /dev/null
+++ b/qahw_api/inc/qahw_defs.h
@@ -0,0 +1,215 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2011 The Android Open Source Project *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <sys/cdefs.h>
+#include <stdint.h>
+
+#ifndef QTI_AUDIO_HAL_DEFS_H
+#define QTI_AUDIO_HAL_DEFS_H
+
+__BEGIN_DECLS
+
+/**************************************/
+
+/**
+ * standard audio parameters that the HAL may need to handle
+ */
+
+/**
+ * audio device parameters
+ */
+
+/* BT SCO Noise Reduction + Echo Cancellation parameters */
+#define QAHW_PARAMETER_KEY_BT_NREC "bt_headset_nrec"
+#define QAHW_PARAMETER_VALUE_ON "on"
+#define QAHW_PARAMETER_VALUE_OFF "off"
+
+/* TTY mode selection */
+#define QAHW_PARAMETER_KEY_TTY_MODE "tty_mode"
+#define QAHW_PARAMETER_VALUE_TTY_OFF "tty_off"
+#define QAHW_PARAMETER_VALUE_TTY_VCO "tty_vco"
+#define QAHW_PARAMETER_VALUE_TTY_HCO "tty_hco"
+#define QAHW_PARAMETER_VALUE_TTY_FULL "tty_full"
+
+/* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off
+ Strings must be in sync with CallFeaturesSetting.java */
+#define QAHW_PARAMETER_KEY_HAC "HACSetting"
+#define QAHW_PARAMETER_VALUE_HAC_ON "ON"
+#define QAHW_PARAMETER_VALUE_HAC_OFF "OFF"
+
+/* A2DP sink address set by framework */
+#define QAHW_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
+
+/* A2DP source address set by framework */
+#define QAHW_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address"
+
+/* Screen state */
+#define QAHW_PARAMETER_KEY_SCREEN_STATE "screen_state"
+
+/* Bluetooth SCO wideband */
+#define QAHW_PARAMETER_KEY_BT_SCO_WB "bt_wbs"
+
+/* Get a new HW synchronization source identifier.
+ * Return a valid source (positive integer) or AUDIO_HW_SYNC_INVALID if an error occurs
+ * or no HW sync is available. */
+#define QAHW_PARAMETER_HW_AV_SYNC "hw_av_sync"
+
+/**
+ * audio stream parameters
+ */
+
+#define QAHW_PARAMETER_STREAM_ROUTING "routing" /* audio_devices_t */
+#define QAHW_PARAMETER_STREAM_FORMAT "format" /* audio_format_t */
+#define QAHW_PARAMETER_STREAM_CHANNELS "channels" /* audio_channel_mask_t */
+#define QAHW_PARAMETER_STREAM_FRAME_COUNT "frame_count" /* size_t */
+#define QAHW_PARAMETER_STREAM_INPUT_SOURCE "input_source" /* audio_source_t */
+#define QAHW_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" /* uint32_t */
+
+#define QAHW_PARAMETER_DEVICE_CONNECT "connect" /* audio_devices_t */
+#define QAHW_PARAMETER_DEVICE_DISCONNECT "disconnect" /* audio_devices_t */
+
+/* Query supported formats. The response is a '|' separated list of strings from
+ * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */
+#define QAHW_PARAMETER_STREAM_SUP_FORMATS "sup_formats"
+
+/* Query supported channel masks. The response is a '|' separated list of
+ * strings from audio_channel_mask_t enum
+ * e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */
+#define QAHW_PARAMETER_STREAM_SUP_CHANNELS "sup_channels"
+
+/* Query supported sampling rates. The response is a '|' separated list of
+ * integer values e.g: "sup_sampling_rates=44100|48000" */
+#define QAHW_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates"
+
+/* Set the HW synchronization source for an output stream. */
+#define QAHW_PARAMETER_STREAM_HW_AV_SYNC "hw_av_sync"
+
+/* Enable mono audio playback if 1, else should be 0. */
+#define QAHW_PARAMETER_MONO_OUTPUT "mono_output"
+
+/**
+ * audio codec parameters
+ */
+
+#define QAHW_OFFLOAD_CODEC_PARAMS "music_offload_codec_param"
+#define QAHW_OFFLOAD_CODEC_BIT_PER_SAMPLE "music_offload_bit_per_sample"
+#define QAHW_OFFLOAD_CODEC_BIT_RATE "music_offload_bit_rate"
+#define QAHW_OFFLOAD_CODEC_AVG_BIT_RATE "music_offload_avg_bit_rate"
+#define QAHW_OFFLOAD_CODEC_ID "music_offload_codec_id"
+#define QAHW_OFFLOAD_CODEC_BLOCK_ALIGN "music_offload_block_align"
+#define QAHW_OFFLOAD_CODEC_SAMPLE_RATE "music_offload_sample_rate"
+#define QAHW_OFFLOAD_CODEC_ENCODE_OPTION "music_offload_encode_option"
+#define QAHW_OFFLOAD_CODEC_NUM_CHANNEL "music_offload_num_channels"
+#define QAHW_OFFLOAD_CODEC_DOWN_SAMPLING "music_offload_down_sampling"
+#define QAHW_OFFLOAD_CODEC_DELAY_SAMPLES "delay_samples"
+#define QAHW_OFFLOAD_CODEC_PADDING_SAMPLES "padding_samples"
+
+/**
+ * extended audio codec parameters
+ */
+
+#define QAHW_OFFLOAD_CODEC_WMA_FORMAT_TAG "music_offload_wma_format_tag"
+#define QAHW_OFFLOAD_CODEC_WMA_BLOCK_ALIGN "music_offload_wma_block_align"
+#define QAHW_OFFLOAD_CODEC_WMA_BIT_PER_SAMPLE "music_offload_wma_bit_per_sample"
+#define QAHW_OFFLOAD_CODEC_WMA_CHANNEL_MASK "music_offload_wma_channel_mask"
+#define QAHW_OFFLOAD_CODEC_WMA_ENCODE_OPTION "music_offload_wma_encode_option"
+#define QAHW_OFFLOAD_CODEC_WMA_ENCODE_OPTION1 "music_offload_wma_encode_option1"
+#define QAHW_OFFLOAD_CODEC_WMA_ENCODE_OPTION2 "music_offload_wma_encode_option2"
+
+#define QAHW_OFFLOAD_CODEC_FLAC_MIN_BLK_SIZE "music_offload_flac_min_blk_size"
+#define QAHW_OFFLOAD_CODEC_FLAC_MAX_BLK_SIZE "music_offload_flac_max_blk_size"
+#define QAHW_OFFLOAD_CODEC_FLAC_MIN_FRAME_SIZE "music_offload_flac_min_frame_size"
+#define QAHW_OFFLOAD_CODEC_FLAC_MAX_FRAME_SIZE "music_offload_flac_max_frame_size"
+
+#define QAHW_OFFLOAD_CODEC_ALAC_FRAME_LENGTH "music_offload_alac_frame_length"
+#define QAHW_OFFLOAD_CODEC_ALAC_COMPATIBLE_VERSION "music_offload_alac_compatible_version"
+#define QAHW_OFFLOAD_CODEC_ALAC_BIT_DEPTH "music_offload_alac_bit_depth"
+#define QAHW_OFFLOAD_CODEC_ALAC_PB "music_offload_alac_pb"
+#define QAHW_OFFLOAD_CODEC_ALAC_MB "music_offload_alac_mb"
+#define QAHW_OFFLOAD_CODEC_ALAC_KB "music_offload_alac_kb"
+#define QAHW_OFFLOAD_CODEC_ALAC_NUM_CHANNELS "music_offload_alac_num_channels"
+#define QAHW_OFFLOAD_CODEC_ALAC_MAX_RUN "music_offload_alac_max_run"
+#define QAHW_OFFLOAD_CODEC_ALAC_MAX_FRAME_BYTES "music_offload_alac_max_frame_bytes"
+#define QAHW_OFFLOAD_CODEC_ALAC_AVG_BIT_RATE "music_offload_alac_avg_bit_rate"
+#define QAHW_OFFLOAD_CODEC_ALAC_SAMPLING_RATE "music_offload_alac_sampling_rate"
+#define QAHW_OFFLOAD_CODEC_ALAC_CHANNEL_LAYOUT_TAG "music_offload_alac_channel_layout_tag"
+
+#define QAHW_OFFLOAD_CODEC_APE_COMPATIBLE_VERSION "music_offload_ape_compatible_version"
+#define QAHW_OFFLOAD_CODEC_APE_COMPRESSION_LEVEL "music_offload_ape_compression_level"
+#define QAHW_OFFLOAD_CODEC_APE_FORMAT_FLAGS "music_offload_ape_format_flags"
+#define QAHW_OFFLOAD_CODEC_APE_BLOCKS_PER_FRAME "music_offload_ape_blocks_per_frame"
+#define QAHW_OFFLOAD_CODEC_APE_FINAL_FRAME_BLOCKS "music_offload_ape_final_frame_blocks"
+#define QAHW_OFFLOAD_CODEC_APE_TOTAL_FRAMES "music_offload_ape_total_frames"
+#define QAHW_OFFLOAD_CODEC_APE_BITS_PER_SAMPLE "music_offload_ape_bits_per_sample"
+#define QAHW_OFFLOAD_CODEC_APE_NUM_CHANNELS "music_offload_ape_num_channels"
+#define QAHW_OFFLOAD_CODEC_APE_SAMPLE_RATE "music_offload_ape_sample_rate"
+#define QAHW_OFFLOAD_CODEC_APE_SEEK_TABLE_PRESENT "music_offload_seek_table_present"
+
+#define QAHW_OFFLOAD_CODEC_VORBIS_BITSTREAM_FMT "music_offload_vorbis_bitstream_fmt"
+
+/* Query fm volume */
+#define QAHW_PARAMETER_KEY_FM_VOLUME "fm_volume"
+
+/* Query if a2dp is supported */
+#define QAHW_PARAMETER_KEY_HANDLE_A2DP_DEVICE "isA2dpDeviceSupported"
+
+/* type of asynchronous write callback events. Mutually exclusive */
+typedef enum {
+ QAHW_STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */
+ QAHW_STREAM_CBK_EVENT_DRAIN_READY /* drain completed */
+} qahw_stream_callback_event_t;
+
+typedef int qahw_stream_callback_t(qahw_stream_callback_event_t event,
+ void *param,
+ void *cookie);
+
+/* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
+typedef enum {
+ QAHW_DRAIN_ALL, /* drain() returns when all data has been played */
+ QAHW_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data
+ from the current track has been played to
+ give time for gapless track switch */
+} qahw_drain_type_t;
+
+/* meta data flags */
+/*TBD: Extend this based on stb requirement*/
+typedef enum {
+ QAHW_META_DATA_FLAGS_NONE = 0,
+} qahw_meta_data_flags_t;
+
+typedef struct {
+ const void *buffer; /* write buffer pointer */
+ size_t bytes; /* size of buffer */
+ size_t offset; /* offset in buffer from where valid byte starts */
+ int64_t *timestamp; /* timestmap */
+ qahw_meta_data_flags_t flags; /* meta data flags */
+ uint32_t reserved[64]; /*reserved for future */
+} qahw_out_buffer_t;
+
+typedef struct {
+ void *buffer; /* read buffer pointer */
+ size_t bytes; /* size of buffer */
+ size_t offset; /* offset in buffer from where valid byte starts */
+ int64_t *timestamp; /* timestmap */
+ uint32_t reserved[64]; /*reserved for future */
+} qahw_in_buffer_t;
+
+__END_DECLS
+
+#endif // QTI_AUDIO_HAL_DEFS_H
+
diff --git a/qahw_api/src/qahw.c b/qahw_api/src/qahw.c
new file mode 100644
index 0000000..06bcee1
--- /dev/null
+++ b/qahw_api/src/qahw.c
@@ -0,0 +1,1513 @@
+/*
+* Copyright (c) 2016, The Linux Foundation. All rights reserved.
+*
+* Redistribution and use in source and binary forms, with or without
+* modification, are permitted provided that the following conditions are
+* met:
+* * Redistributions of source code must retain the above copyright
+* notice, this list of conditions and the following disclaimer.
+* * Redistributions in binary form must reproduce the above
+* copyright notice, this list of conditions and the following
+* disclaimer in the documentation and/or other materials provided
+* with the distribution.
+* * Neither the name of The Linux Foundation nor the names of its
+* contributors may be used to endorse or promote products derived
+* from this software without specific prior written permission.
+*
+* THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+* ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+* BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+* OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+* IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#define LOG_TAG "qahw"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <utils/Log.h>
+#include <stdlib.h>
+#include <cutils/list.h>
+
+#include <hardware/audio.h>
+#include "qahw_api.h"
+
+#define NO_ERROR 0
+#define MAX_MODULE_NAME_LENGTH 100
+
+/*
+ * The current HAL API version.
+ */
+#define QAHW_MODULE_API_VERSION_CURRENT QAHW_MODULE_API_VERSION_0_0
+
+typedef struct {
+ audio_hw_device_t *audio_device;
+ char module_name[MAX_MODULE_NAME_LENGTH];
+ struct listnode module_list;
+ struct listnode in_list;
+ struct listnode out_list;
+ pthread_mutex_t lock;
+ uint32_t ref_count;
+} qahw_module_t;
+
+typedef struct {
+ qahw_module_t *module;
+ struct listnode module_list;
+ pthread_mutex_t lock;
+} qahw_module_instances_t;
+
+typedef struct {
+ audio_stream_out_t *stream;
+ qahw_module_t *module;
+ struct listnode list;
+ pthread_mutex_t lock;
+} qahw_stream_out_t;
+
+typedef struct {
+ audio_stream_in_t *stream;
+ qahw_module_t *module;
+ struct listnode list;
+ pthread_mutex_t lock;
+} qahw_stream_in_t;
+
+typedef enum {
+ STREAM_DIR_IN,
+ STREAM_DIR_OUT,
+} qahw_stream_direction_t;
+
+static struct listnode qahw_module_list;
+static int qahw_list_count;
+static pthread_mutex_t qahw_module_init_lock = PTHREAD_MUTEX_INITIALIZER;
+
+/** Start of internal functions */
+/******************************************************************************/
+
+/* call this function without anylock held */
+static bool is_valid_qahw_stream(void *qahw_stream,
+ qahw_stream_direction_t dir)
+{
+
+ int is_valid = false;
+ struct listnode *module_node = NULL;
+ struct listnode *stream_node = NULL;
+ struct listnode *list_node = NULL;
+ void *stream = NULL;
+ qahw_module_t *qahw_module = NULL;
+
+ if (qahw_stream == NULL) {
+ ALOGE("%s:: Invalid stream", __func__);
+ goto exit;
+ }
+
+ if ((dir != STREAM_DIR_OUT) && (dir != STREAM_DIR_IN)) {
+ ALOGE("%s:: Invalid stream direction %d", __func__, dir);
+ goto exit;
+ }
+
+ /* go through all the modules and check for valid stream */
+ pthread_mutex_lock(&qahw_module_init_lock);
+ list_for_each(module_node, &qahw_module_list) {
+ qahw_module = node_to_item(module_node, qahw_module_t, module_list);
+ pthread_mutex_lock(&qahw_module->lock);
+ if(dir == STREAM_DIR_OUT)
+ list_node = &qahw_module->out_list;
+ else
+ list_node = &qahw_module->in_list;
+ list_for_each(stream_node, list_node) {
+ if(dir == STREAM_DIR_OUT)
+ stream = (void *)node_to_item(stream_node,
+ qahw_stream_out_t,
+ list);
+ else
+ stream = (void *)node_to_item(stream_node,
+ qahw_stream_in_t,
+ list);
+ if(stream == qahw_stream) {
+ is_valid = true;
+ break;
+ }
+ }
+ pthread_mutex_unlock(&qahw_module->lock);
+ if(is_valid)
+ break;
+ }
+ pthread_mutex_unlock(&qahw_module_init_lock);
+
+exit:
+ return is_valid;
+}
+
+/* call this fucntion with ahw_module_init_lock held*/
+static qahw_module_t* get_qahw_module_by_ptr(qahw_module_t *qahw_module)
+{
+ struct listnode *node = NULL;
+ qahw_module_t *module = NULL, *module_temp = NULL;
+
+ if (qahw_module == NULL)
+ goto exit;
+
+ list_for_each(node, &qahw_module_list) {
+ module_temp = node_to_item(node, qahw_module_t, module_list);
+ if (module_temp == qahw_module) {
+ module = module_temp;
+ break;
+ }
+ }
+exit:
+ return module;
+}
+
+/* call this function with qahw_module_init_lock held*/
+static qahw_module_t* get_qahw_module_by_name(const char *qahw_name)
+{
+ struct listnode *node = NULL;
+ qahw_module_t *module = NULL, *module_temp = NULL;
+
+ if (qahw_name == NULL)
+ goto exit;
+
+ list_for_each(node, &qahw_module_list) {
+ module_temp = node_to_item(node, qahw_module_t, module_list);
+ if(!strncmp(qahw_name, module_temp->module_name, MAX_MODULE_NAME_LENGTH)) {
+ module = module_temp;
+ break;
+ }
+ }
+exit:
+ return module;
+}
+/* End of of internal functions */
+
+/*
+ * Return the sampling rate in Hz - eg. 44100.
+ */
+uint32_t qahw_out_get_sample_rate(const qahw_stream_handle_t *out_handle)
+{
+ uint32_t rate = 0;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGV("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->common.get_sample_rate)
+ rate = out->common.get_sample_rate(&out->common);
+ else
+ ALOGW("%s not supported", __func__);
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rate;
+}
+
+/*
+ * currently unused - use set_parameters with key
+ * AUDIO_PARAMETER_STREAM_SAMPLING_RATE
+ */
+int qahw_out_set_sample_rate(qahw_stream_handle_t *out_handle, uint32_t rate)
+{
+ int32_t rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->common.set_sample_rate) {
+ rc = out->common.set_sample_rate(&out->common, rate);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+exit:
+ return rc;
+}
+
+size_t qahw_out_get_buffer_size(const qahw_stream_handle_t *out_handle)
+{
+ size_t buf_size = 0;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->common.get_buffer_size) {
+ buf_size = out->common.get_buffer_size(&out->common);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return buf_size;
+}
+
+audio_channel_mask_t qahw_out_get_channels(const qahw_stream_handle_t *out_handle)
+{
+ audio_channel_mask_t ch_mask = 0;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->common.get_channels) {
+ ch_mask = out->common.get_channels(&out->common);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return ch_mask;
+}
+
+audio_format_t qahw_out_get_format(const qahw_stream_handle_t *out_handle)
+{
+ audio_format_t format = AUDIO_FORMAT_INVALID;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->common.get_format) {
+ format = out->common.get_format(&out->common);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return format;
+}
+
+int qahw_out_standby(qahw_stream_handle_t *out_handle)
+{
+ int32_t rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->common.standby) {
+ rc = out->common.standby(&out->common);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+int qahw_out_set_parameters(qahw_stream_handle_t *out_handle, const char *kv_pairs)
+{
+ int rc = NO_ERROR;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ rc = -EINVAL;
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->common.set_parameters) {
+ rc = out->common.set_parameters(&out->common, kv_pairs);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+char *qahw_out_get_parameters(const qahw_stream_handle_t *out_handle,
+ const char *keys)
+{
+ char *str_param = NULL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->common.get_parameters) {
+ str_param = out->common.get_parameters(&out->common, keys);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return str_param;
+}
+
+uint32_t qahw_out_get_latency(const qahw_stream_handle_t *out_handle)
+{
+ uint32_t latency = 0;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->get_latency) {
+ latency = out->get_latency(out);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return latency;
+}
+
+int qahw_out_set_volume(qahw_stream_handle_t *out_handle, float left, float right)
+{
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->set_volume) {
+ rc = out->set_volume(out, left, right);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+ssize_t qahw_out_write(qahw_stream_handle_t *out_handle,
+ qahw_out_buffer_t *out_buf)
+{
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if ((out_buf == NULL) || (out_buf->buffer == NULL)) {
+ ALOGE("%s::Invalid meta data %p", __func__, out_buf);
+ goto exit;
+ }
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ /*TBD:: validate other meta data parameters */
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->write) {
+ rc = out->write(out, out_buf->buffer, out_buf->bytes);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+exit:
+ return rc;
+}
+
+int qahw_out_get_render_position(const qahw_stream_handle_t *out_handle,
+ uint32_t *dsp_frames)
+{
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->get_render_position) {
+ rc = out->get_render_position(out, dsp_frames);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+exit:
+ return rc;
+}
+
+int qahw_out_set_callback(qahw_stream_handle_t *out_handle,
+ qahw_stream_callback_t callback,
+ void *cookie)
+{
+ /*TBD:load hal func pointer and call */
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->set_callback) {
+ rc = out->set_callback(out, (stream_callback_t)callback, cookie);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+int qahw_out_pause(qahw_stream_handle_t *out_handle)
+{
+ /*TBD:load hal func pointer and call */
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->pause) {
+ rc = out->pause(out);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+int qahw_out_resume(qahw_stream_handle_t *out_handle)
+{
+ /*TBD:load hal func pointer and call */
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->resume) {
+ rc = out->resume(out);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+int qahw_out_drain(qahw_stream_handle_t *out_handle, qahw_drain_type_t type )
+{
+ /*TBD:load hal func pointer and call */
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->drain) {
+ rc = out->drain(out,(audio_drain_type_t)type);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+int qahw_out_flush(qahw_stream_handle_t *out_handle)
+{
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->flush) {
+ rc = out->flush(out);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+int qahw_out_get_presentation_position(const qahw_stream_handle_t *out_handle,
+ uint64_t *frames, struct timespec *timestamp)
+{
+ int rc = -EINVAL;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ audio_stream_out_t *out = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ out = qahw_stream_out->stream;
+ if (out->get_presentation_position) {
+ rc = out->get_presentation_position(out, frames, timestamp);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+exit:
+ return rc;
+}
+
+/* Input stream specific APIs */
+uint32_t qahw_in_get_sample_rate(const qahw_stream_handle_t *in_handle)
+{
+ uint32_t rate = 0;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.get_sample_rate) {
+ rate = in->common.get_sample_rate(&in->common);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return rate;
+}
+
+/*
+ * currently unused - use set_parameters with key
+ * AUDIO_PARAMETER_STREAM_SAMPLING_RATE
+ */
+int qahw_in_set_sample_rate(qahw_stream_handle_t *in_handle, uint32_t rate)
+{
+ int rc = -EINVAL;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.set_sample_rate) {
+ rc = in->common.set_sample_rate(&in->common, rate);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return rc;
+}
+
+size_t qahw_in_get_buffer_size(const qahw_stream_handle_t *in_handle)
+{
+ size_t buf_size = 0;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.get_sample_rate) {
+ buf_size = in->common.get_buffer_size(&in->common);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return buf_size;
+}
+
+
+audio_channel_mask_t qahw_in_get_channels(const qahw_stream_handle_t *in_handle)
+{
+ audio_channel_mask_t ch_mask = 0;;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.get_channels) {
+ ch_mask = in->common.get_channels(&in->common);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return ch_mask;
+}
+
+audio_format_t qahw_in_get_format(const qahw_stream_handle_t *in_handle)
+{
+ audio_format_t format = AUDIO_FORMAT_INVALID;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.get_format) {
+ format = in->common.get_format(&in->common);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return format;
+}
+
+/*
+ * currently unused - use set_parameters with key
+ * AUDIO_PARAMETER_STREAM_FORMAT
+ */
+int qahw_in_set_format(qahw_stream_handle_t *in_handle, audio_format_t format)
+{
+ int rc = -EINVAL;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.set_format) {
+ rc = in->common.set_format(&in->common, format);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return rc;
+}
+
+int qahw_in_standby(qahw_stream_handle_t *in_handle)
+{
+ int rc = -EINVAL;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.standby) {
+ rc = in->common.standby(&in->common);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return rc;
+}
+
+/*
+ * set/get audio stream parameters. The function accepts a list of
+ * parameter key value pairs in the form: key1=value1;key2=value2;...
+ *
+ * Some keys are reserved for standard parameters (See AudioParameter class)
+ *
+ * If the implementation does not accept a parameter change while
+ * the output is active but the parameter is acceptable otherwise, it must
+ * return -ENOSYS.
+ *
+ * The audio flinger will put the stream in standby and then change the
+ * parameter value.
+ */
+int qahw_in_set_parameters(qahw_stream_handle_t *in_handle, const char *kv_pairs)
+{
+ int rc = -EINVAL;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.set_parameters) {
+ rc = in->common.set_parameters(&in->common, kv_pairs);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+exit:
+ return rc;
+}
+
+/*
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+char * qahw_in_get_parameters(const qahw_stream_handle_t *in_handle,
+ const char *keys)
+{
+ char *str_param = NULL;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->common.get_parameters) {
+ str_param = in->common.get_parameters(&in->common, keys);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return str_param;
+}
+
+/*
+ * Read audio buffer in from audio driver. Returns number of bytes read, or a
+ * negative status_t. If at least one frame was read prior to the error,
+ * read should return that byte count and then return an error in the subsequent call.
+ */
+ssize_t qahw_in_read(qahw_stream_handle_t *in_handle,
+ qahw_in_buffer_t *in_buf)
+{
+ int rc = -EINVAL;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if ((in_buf == NULL) || (in_buf->buffer == NULL)) {
+ ALOGE("%s::Invalid meta data %p", __func__, in_buf);
+ goto exit;
+ }
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ /*TBD:: call HAL timestamp read API*/
+ if (in->read) {
+ rc = in->read(in, in_buf->buffer, in_buf->bytes);
+ in_buf->offset = 0;
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return rc;
+}
+
+/*
+ * Return the amount of input frames lost in the audio driver since the
+ * last call of this function.
+ * Audio driver is expected to reset the value to 0 and restart counting
+ * upon returning the current value by this function call.
+ * Such loss typically occurs when the user space process is blocked
+ * longer than the capacity of audio driver buffers.
+ *
+ * Unit: the number of input audio frames
+ */
+uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle)
+{
+ uint32_t rc = 0;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ in = qahw_stream_in->stream;
+ if (in->get_input_frames_lost) {
+ rc = in->get_input_frames_lost(in);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+exit:
+ return rc;
+}
+
+/*
+ * Return a recent count of the number of audio frames received and
+ * the clock time associated with that frame count.
+ *
+ * frames is the total frame count received. This should be as early in
+ * the capture pipeline as possible. In general,
+ * frames should be non-negative and should not go "backwards".
+ *
+ * time is the clock MONOTONIC time when frames was measured. In general,
+ * time should be a positive quantity and should not go "backwards".
+ *
+ * The status returned is 0 on success, -ENOSYS if the device is not
+ * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
+ */
+int qahw_in_get_capture_position(const qahw_stream_handle_t *in_handle __unused,
+ int64_t *frames __unused, int64_t *time __unused)
+{
+ /*TBD:: do we need this*/
+ return -ENOSYS;
+}
+
+/*
+ * check to see if the audio hardware interface has been initialized.
+ * returns 0 on success, -ENODEV on failure.
+ */
+int qahw_init_check(const qahw_module_handle_t *hw_module)
+{
+ int rc = -EINVAL;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp;
+
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ pthread_mutex_unlock(&qahw_module_init_lock);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ if (qahw_module->audio_device->init_check) {
+ rc = qahw_module->audio_device->init_check(qahw_module->audio_device);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+ return rc;
+}
+/* set the audio volume of a voice call. Range is between 0.0 and 1.0 */
+int qahw_set_voice_volume(qahw_module_handle_t *hw_module, float volume)
+{
+ int rc = -EINVAL;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp;
+
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ pthread_mutex_unlock(&qahw_module_init_lock);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ if (qahw_module->audio_device->set_voice_volume) {
+ rc = qahw_module->audio_device->set_voice_volume(qahw_module->audio_device,
+ volume);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+ return rc;
+}
+
+/*
+ * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
+ * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
+ * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
+ */
+int qahw_set_mode(qahw_module_handle_t *hw_module, audio_mode_t mode)
+{
+ int rc = -EINVAL;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp;
+
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ pthread_mutex_unlock(&qahw_module_init_lock);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ if (qahw_module->audio_device->set_mode) {
+ rc = qahw_module->audio_device->set_mode(qahw_module->audio_device,
+ mode);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+ return rc;
+}
+
+/* set/get global audio parameters */
+int qahw_set_parameters(qahw_module_handle_t *hw_module, const char *kv_pairs)
+{
+ int rc = -EINVAL;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp;
+ audio_hw_device_t *audio_device;
+
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ pthread_mutex_unlock(&qahw_module_init_lock);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ audio_device = qahw_module->audio_device;
+ if (qahw_module->audio_device->set_parameters) {
+ rc = audio_device->set_parameters(qahw_module->audio_device, kv_pairs);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+ return rc;
+}
+
+/*
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+char * qahw_get_parameters(const qahw_module_handle_t *hw_module,
+ const char *keys)
+{
+ char *str_param = NULL;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp;
+ audio_hw_device_t *audio_device;
+
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ pthread_mutex_unlock(&qahw_module_init_lock);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ audio_device = qahw_module->audio_device;
+ if (qahw_module->audio_device->get_parameters) {
+ str_param = audio_device->get_parameters(qahw_module->audio_device, keys);
+ } else {
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+ return str_param;
+}
+
+/* Returns audio input buffer size according to parameters passed or
+ * 0 if one of the parameters is not supported.
+ * See also get_buffer_size which is for a particular stream.
+ */
+size_t qahw_get_input_buffer_size(const qahw_module_handle_t *hw_module,
+ const struct audio_config *config)
+{
+ size_t rc = 0;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp;
+ audio_hw_device_t *audio_device;
+
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ pthread_mutex_unlock(&qahw_module_init_lock);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ audio_device = qahw_module->audio_device;
+ if (qahw_module->audio_device->get_input_buffer_size) {
+ rc = audio_device->get_input_buffer_size(qahw_module->audio_device,
+ config);
+ } else {
+ rc = -ENOSYS;
+ ALOGW("%s not supported", __func__);
+ }
+ pthread_mutex_unlock(&qahw_module->lock);
+
+exit:
+ return rc;
+}
+
+/*
+ * This method creates and opens the audio hardware output stream.
+ * The "address" parameter qualifies the "devices" audio device type if needed.
+ * The format format depends on the device type:
+ * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
+ * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y"
+ * - Other devices may use a number or any other string.
+ */
+int qahw_open_output_stream(qahw_module_handle_t *hw_module,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ qahw_stream_handle_t **out_handle,
+ const char *address)
+{
+ int rc = -EINVAL;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp = NULL;
+ audio_hw_device_t *audio_device = NULL;
+ qahw_stream_out_t *qahw_stream_out = NULL;
+
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ pthread_mutex_unlock(&qahw_module_init_lock);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ return rc;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ audio_device = qahw_module->audio_device;
+ qahw_stream_out = (qahw_stream_out_t *)calloc(1, sizeof(qahw_stream_out_t));
+ if (qahw_stream_out == NULL) {
+ ALOGE("%s:: calloc failed for out stream_out_t",__func__);
+ rc = -ENOMEM;
+ goto exit;
+ }
+
+ rc = audio_device->open_output_stream(audio_device,
+ handle,
+ devices,
+ flags,
+ config,
+ &qahw_stream_out->stream,
+ address);
+ if (rc) {
+ ALOGE("%s::open output stream failed %d",__func__, rc);
+ free(qahw_stream_out);
+ } else {
+ qahw_stream_out->module = hw_module;
+ *out_handle = (void *)qahw_stream_out;
+ pthread_mutex_init(&qahw_stream_out->lock, (const pthread_mutexattr_t *)NULL);
+ list_add_tail(&qahw_module->out_list, &qahw_stream_out->list);
+ }
+
+exit:
+ pthread_mutex_unlock(&qahw_module->lock);
+ return rc;
+}
+
+int qahw_close_output_stream(qahw_stream_handle_t *out_handle)
+{
+
+ int rc = 0;
+ qahw_stream_out_t *qahw_stream_out = (qahw_stream_out_t *)out_handle;
+ qahw_module_t *qahw_module = NULL;
+ audio_hw_device_t *audio_device = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_out, STREAM_DIR_OUT)) {
+ ALOGE("%s::Invalid out handle %p", __func__, out_handle);
+ rc = -EINVAL;
+ goto exit;
+ }
+
+ ALOGV("%s::calling device close_output_stream %p", __func__, out_handle);
+ pthread_mutex_lock(&qahw_stream_out->lock);
+ qahw_module = qahw_stream_out->module;
+ audio_device = qahw_module->audio_device;
+ audio_device->close_output_stream(audio_device,
+ qahw_stream_out->stream);
+
+ pthread_mutex_lock(&qahw_module->lock);
+ list_remove(&qahw_stream_out->list);
+ pthread_mutex_unlock(&qahw_module->lock);
+
+ pthread_mutex_unlock(&qahw_stream_out->lock);
+
+ pthread_mutex_destroy(&qahw_stream_out->lock);
+ free(qahw_stream_out);
+
+exit:
+ return rc;
+}
+
+/* This method creates and opens the audio hardware input stream */
+int qahw_open_input_stream(qahw_module_handle_t *hw_module,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ qahw_stream_handle_t **in_handle,
+ audio_input_flags_t flags,
+ const char *address,
+ audio_source_t source)
+{
+ int rc = -EINVAL;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp = NULL;
+ audio_hw_device_t *audio_device = NULL;
+ qahw_stream_in_t *qahw_stream_in = NULL;
+
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ pthread_mutex_unlock(&qahw_module_init_lock);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ return rc;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ audio_device = qahw_module->audio_device;
+ qahw_stream_in = (qahw_stream_in_t *)calloc(1, sizeof(qahw_stream_in_t));
+ if (qahw_stream_in == NULL) {
+ ALOGE("%s:: calloc failed for in stream_in_t",__func__);
+ rc = -ENOMEM;
+ goto exit;
+ }
+
+ rc = audio_device->open_input_stream(audio_device,
+ handle,
+ devices,
+ config,
+ &qahw_stream_in->stream,
+ flags,
+ address,
+ source);
+ if (rc) {
+ ALOGE("%s::open input stream failed %d",__func__, rc);
+ free(qahw_stream_in);
+ } else {
+ qahw_stream_in->module = hw_module;
+ *in_handle = (void *)qahw_stream_in;
+ pthread_mutex_init(&qahw_stream_in->lock, (const pthread_mutexattr_t *)NULL);
+ list_add_tail(&qahw_module->in_list, &qahw_stream_in->list);
+ }
+
+exit:
+ pthread_mutex_unlock(&qahw_module->lock);
+ return rc;
+}
+
+int qahw_close_input_stream(qahw_stream_handle_t *in_handle)
+{
+ int rc = 0;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ qahw_module_t *qahw_module = NULL;
+ audio_hw_device_t *audio_device = NULL;
+
+ if (!is_valid_qahw_stream((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ rc = -EINVAL;
+ goto exit;
+ }
+
+ ALOGV("%s:: calling device close_input_stream %p", __func__, in_handle);
+ pthread_mutex_lock(&qahw_stream_in->lock);
+ qahw_module = qahw_stream_in->module;
+ audio_device = qahw_module->audio_device;
+ audio_device->close_input_stream(audio_device,
+ qahw_stream_in->stream);
+
+ pthread_mutex_lock(&qahw_module->lock);
+ list_remove(&qahw_stream_in->list);
+ pthread_mutex_unlock(&qahw_module->lock);
+
+ pthread_mutex_unlock(&qahw_stream_in->lock);
+
+ pthread_mutex_destroy(&qahw_stream_in->lock);
+ free(qahw_stream_in);
+
+exit:
+ return rc;
+}
+
+/*returns current QTI HAL verison */
+int qahw_get_version() {
+ return QAHW_MODULE_API_VERSION_CURRENT;
+}
+
+/* convenience API for opening and closing an audio HAL module */
+
+qahw_module_handle_t *qahw_load_module(const char *hw_module_id)
+{
+ int rc = -EINVAL;
+ qahw_module_handle_t *qahw_mod_handle = NULL;
+ qahw_module_t *qahw_module = NULL;
+ char *ahal_name = NULL;
+ const hw_module_t* module = NULL;
+ audio_hw_device_t* audio_device = NULL;
+
+ if (hw_module_id == NULL) {
+ ALOGE("%s::module id is NULL",__func__);
+ goto exit;
+ }
+
+ if (!strcmp(hw_module_id, QAHW_MODULE_ID_PRIMARY)) {
+ ahal_name = "primary";
+ } else if (!strcmp(hw_module_id, QAHW_MODULE_ID_A2DP)) {
+ ahal_name = "a2dp";
+ } else if (!strcmp(hw_module_id, QAHW_MODULE_ID_USB)) {
+ ahal_name = "usb";
+ } else {
+ ALOGE("%s::Invalid Module id %s", __func__, hw_module_id);
+ goto exit;
+ }
+
+ /* return exiting module ptr if already loaded */
+ pthread_mutex_lock(&qahw_module_init_lock);
+ if (qahw_list_count > 0) {
+ qahw_module = get_qahw_module_by_name(hw_module_id);
+ if(qahw_module != NULL) {
+ qahw_mod_handle = (void *)qahw_module;
+ pthread_mutex_lock(&qahw_module->lock);
+ qahw_module->ref_count++;
+ pthread_mutex_unlock(&qahw_module->lock);
+ goto error_exit;
+ }
+ }
+
+ rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, ahal_name, &module);
+ if(rc) {
+ ALOGE("%s::HAL Loading failed %d", __func__, rc);
+ goto error_exit;
+ }
+
+ rc = audio_hw_device_open(module, &audio_device);
+ if(rc) {
+ ALOGE("%s::HAL Device open failed %d", __func__, rc);
+ goto error_exit;
+ }
+
+ qahw_module = (qahw_module_t *)calloc(1, sizeof(qahw_module_t));
+ if(qahw_module == NULL) {
+ ALOGE("%s::calloc failed", __func__);
+ audio_hw_device_close(audio_device);
+ goto error_exit;
+ }
+ ALOGD("%s::Loaded HAL %s module %p", __func__, ahal_name, qahw_module);
+
+ if (!qahw_list_count)
+ list_init(&qahw_module_list);
+ qahw_list_count++;
+
+ pthread_mutex_init(&qahw_module->lock, (const pthread_mutexattr_t *) NULL);
+ pthread_mutex_lock(&qahw_module->lock);
+ qahw_module->ref_count++;
+ pthread_mutex_unlock(&qahw_module->lock);
+
+ list_init(&qahw_module->out_list);
+ list_init(&qahw_module->in_list);
+
+ /* update qahw_module */
+ qahw_module->audio_device = audio_device;
+ strlcpy(&qahw_module->module_name[0], hw_module_id, MAX_MODULE_NAME_LENGTH);
+
+ qahw_mod_handle = (void *)qahw_module;
+
+ /* Add module list to global module list */
+ list_add_tail(&qahw_module_list, &qahw_module->module_list);
+
+
+error_exit:
+ pthread_mutex_unlock(&qahw_module_init_lock);
+
+exit:
+ return qahw_mod_handle;
+}
+
+int qahw_unload_module(qahw_module_handle_t *hw_module)
+{
+ int rc = -EINVAL;
+ bool is_empty = false;
+ qahw_module_t *qahw_module = (qahw_module_t *)hw_module;
+ qahw_module_t *qahw_module_temp = NULL;
+
+ /* close HW device if its valid and all the streams on
+ * it is closed
+ */
+ pthread_mutex_lock(&qahw_module_init_lock);
+ qahw_module_temp = get_qahw_module_by_ptr(qahw_module);
+ if (qahw_module_temp == NULL) {
+ ALOGE("%s:: invalid hw module %p", __func__, qahw_module);
+ goto error_exit;
+ }
+
+ pthread_mutex_lock(&qahw_module->lock);
+ qahw_module->ref_count--;
+ if (qahw_module->ref_count > 0) {
+ rc = 0;
+ ALOGE("%s:: skipping module unload of %p count %d", __func__,
+ qahw_module,
+ qahw_module->ref_count);
+ pthread_mutex_unlock(&qahw_module->lock);
+ goto error_exit;
+ }
+
+ is_empty = (list_empty(&qahw_module->out_list) &&
+ list_empty(&qahw_module->in_list));
+ if (is_empty) {
+ rc = audio_hw_device_close(qahw_module->audio_device);
+ if(rc) {
+ ALOGE("%s::HAL Device close failed Error %d Module %p",__func__,
+ rc, qahw_module);
+ rc = 0;
+ }
+ qahw_list_count--;
+ list_remove(&qahw_module->module_list);
+ pthread_mutex_unlock(&qahw_module->lock);
+ pthread_mutex_destroy(&qahw_module->lock);
+ free(qahw_module);
+ } else {
+ pthread_mutex_unlock(&qahw_module->lock);
+ ALOGE("%s::failed as all the streams on this module"
+ "is not closed", __func__);
+ rc = -EINVAL;
+ }
+
+error_exit:
+ pthread_mutex_unlock(&qahw_module_init_lock);
+
+exit:
+ return rc;
+}
+
+__END_DECLS
diff --git a/qahw_api/test/Android.mk b/qahw_api/test/Android.mk
new file mode 100644
index 0000000..1688069
--- /dev/null
+++ b/qahw_api/test/Android.mk
@@ -0,0 +1,40 @@
+LOCAL_PATH := $(call my-dir)
+
+# audio_hal_playback_test
+# ==============================================================================
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES := qahw_playback_test.c
+LOCAL_MODULE := hal_play_test
+
+hal-play-inc = $(TARGET_OUT_HEADERS)/mm-audio/qahw_api/inc
+
+LOCAL_CFLAGS += -Wall -Werror -Wno-sign-compare
+
+LOCAL_SHARED_LIBRARIES := \
+ libaudioutils\
+ libqahw \
+ libutils
+
+LOCAL_32_BIT_ONLY := true
+
+LOCAL_C_INCLUDES += $(hal-play-inc)
+
+include $(BUILD_EXECUTABLE)
+
+# audio_hal_multi_record_test
+# ==============================================================================
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES := qahw_multi_record_test.c
+LOCAL_MODULE := hal_rec_test
+LOCAL_CFLAGS += -Wall -Werror -Wno-sign-compare
+LOCAL_SHARED_LIBRARIES := \
+ libaudioutils \
+ libqahw \
+ libutils
+
+LOCAL_32_BIT_ONLY := true
+
+hal-rec-inc = $(TARGET_OUT_HEADERS)/mm-audio/qahw_api/inc
+
+LOCAL_C_INCLUDES += $(hal-rec-inc)
+include $(BUILD_EXECUTABLE)
diff --git a/qahw_api/test/qahw_multi_record_test.c b/qahw_api/test/qahw_multi_record_test.c
new file mode 100644
index 0000000..07dede7
--- /dev/null
+++ b/qahw_api/test/qahw_multi_record_test.c
@@ -0,0 +1,488 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2015 The Android Open Source Project *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/* Test app to record multiple audio sessions at the HAL layer */
+
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include <pthread.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include "qahw_api.h"
+#include "qahw_defs.h"
+
+struct audio_config_params {
+ qahw_module_handle_t *qahw_mod_handle;
+ audio_io_handle_t handle;
+ audio_devices_t input_device;
+ audio_config_t config;
+ audio_input_flags_t flags;
+ const char* kStreamName ;
+ audio_source_t kInputSource;
+ char output_filename[256];
+ double loopTime;
+ char profile[50];
+};
+
+#define SOUNDFOCUS_PARAMS "SoundFocus.start_angles;SoundFocus.enable_sectors;" \
+ "SoundFocus.gain_step"
+#define SOURCETRACK_PARAMS "SourceTrack.vad;SourceTrack.doa_speech;SourceTrack.doa_noise;"\
+ "SourceTrack.polar_activity;ssr.noise_level;ssr.noise_level_after_ns"
+int sourcetrack_done = 0;
+static pthread_mutex_t glock;
+pthread_cond_t gcond;
+int tests_running;
+bool gerror;
+
+void *read_sourcetrack_data(void* data)
+{
+ char kvpair_soundfocus[200] = SOUNDFOCUS_PARAMS;
+ char kvpair_sourcetrack[200] = SOURCETRACK_PARAMS;
+ char *string = NULL;
+ char *token = NULL;
+ char choice = '\0';
+ int i =0;
+ qahw_module_handle_t *qawh_module_handle =
+ (qahw_module_handle_t *)data;
+
+ while (1) {
+ printf("\nGet SoundFocus Params from app");
+ string = qahw_get_parameters(qawh_module_handle, kvpair_soundfocus);
+ if (!string) {
+ printf("Error.Failed Get SoundFocus Params\n");
+ } else {
+ token = strtok (string , "=");
+ while (token) {
+ if (*token == 'S') {
+ choice = *(token + 11);
+ token = strtok (NULL,",;");
+ i=0;
+ }
+ switch (choice) {
+ case 'g':
+ printf ("\nSoundFocus.gain_step=%s",token);
+ break;
+ case 'e':
+ printf ("\nSoundFocus.enable_sectors[%d]=%s",i,token);
+ i++;
+ break;
+ case 's':
+ printf ("\nSoundFocus.start_angles[%d]=%s",i,token);
+ i++;
+ break;
+ }
+ token = strtok (NULL,",;=");
+ }
+ }
+ choice = '\0';
+ printf ("\nGet SourceTracking Params from app");
+ string = qahw_get_parameters(qawh_module_handle, kvpair_sourcetrack);
+ if (!string) {
+ printf ("Error.Failed Get SourceTrack Params\n");
+ } else {
+ token = strtok (string , "=");
+ while (token) {
+ if (*token == 'S') {
+ choice = *(token + 12);
+ if (choice == 'd')
+ choice = *(token + 16);
+ token = strtok (NULL,",;");
+ i=0;
+ }
+ switch (choice) {
+ case 'p':
+ printf ("\nSourceTrack.polar_activity=%s,",token);
+ choice = '\0';
+ break;
+ case 'v':
+ printf ("\nSourceTrack.vad[%d]=%s",i,token);
+ i++;
+ break;
+ case 's':
+ printf ("\nSourceTrack.doa_speech=%s",token);
+ break;
+ case 'n':
+ printf ("\nSourceTrack.doa_noise[%d]=%s",i,token);
+ i++;
+ break;
+ default :
+ printf ("%s,",token);
+ break;
+ }
+ token = strtok (NULL,",;=");
+ }
+ }
+ if (sourcetrack_done == 1)
+ return NULL;
+ }
+}
+
+void *start_input(void *thread_param)
+{
+ int rc = 0;
+ struct audio_config_params* params = (struct audio_config_params*) thread_param;
+ qahw_module_handle_t *qahw_mod_handle = params->qahw_mod_handle;
+
+ // Open audio input stream.
+ qahw_stream_handle_t* in_handle = NULL;
+
+ rc = qahw_open_input_stream(qahw_mod_handle,
+ params->handle, params->input_device,
+ ¶ms->config, &in_handle,
+ params->flags, params->kStreamName,
+ params->kInputSource);
+ if (rc) {
+ printf("ERROR :::: Could not open input stream.\n" );
+ pthread_mutex_lock(&glock);
+ gerror = true;
+ pthread_cond_signal(&gcond);
+ pthread_mutex_unlock(&glock);
+ pthread_exit(0);
+ }
+
+ // Get buffer size to get upper bound on data to read from the HAL.
+ size_t buffer_size;
+ buffer_size = qahw_in_get_buffer_size(in_handle);
+ char *buffer;
+ buffer = (char *)calloc(1, buffer_size);
+ if (buffer == NULL) {
+ printf("calloc failed!!\n");
+ pthread_mutex_lock(&glock);
+ gerror = true;
+ pthread_cond_signal(&gcond);
+ pthread_mutex_unlock(&glock);
+ pthread_exit(0);
+ }
+
+ printf("input opened, buffer = %p, size %zun",
+ buffer, buffer_size);
+
+ int num_channels = audio_channel_count_from_in_mask(params->config.channel_mask);
+
+ time_t start_time = time(0);
+ ssize_t bytes_read = -1;
+ char param[100] = "audio_stream_profile=";
+ qahw_in_buffer_t in_buf;
+
+ // set profile for the recording session
+ strlcat(param, params->profile, sizeof(param));
+ qahw_in_set_parameters(in_handle, param);
+
+ printf("\nPlease speak into the microphone for %lf seconds.\n", params->loopTime);
+
+ FILE *fd = fopen(params->output_filename,"w");
+ if (fd == NULL) {
+ printf("File open failed \n");
+ pthread_mutex_lock(&glock);
+ gerror = true;
+ pthread_cond_signal(&gcond);
+ pthread_mutex_unlock(&glock);
+ pthread_exit(0);
+ }
+ pthread_mutex_lock(&glock);
+ tests_running++;
+ pthread_cond_signal(&gcond);
+ pthread_mutex_unlock(&glock);
+ memset(&in_buf,0, sizeof(qahw_in_buffer_t));
+
+ while(true) {
+ in_buf.buffer = buffer;
+ in_buf.bytes = buffer_size;
+ bytes_read = qahw_in_read(in_handle, &in_buf);
+ fwrite(in_buf.buffer, sizeof(char), buffer_size, fd);
+ if(difftime(time(0), start_time) > params->loopTime) {
+ printf("\nTest completed.\n");
+ break;
+ }
+ }
+
+ printf("closing input");
+
+ // Close output stream and device.
+ rc = qahw_in_standby(in_handle);
+ if (rc) {
+ printf("out standby failed %d \n",rc);
+ }
+
+ rc = qahw_close_input_stream(in_handle);
+ if (rc) {
+ printf("could not close input stream %d \n",rc);
+ }
+
+ // Print instructions to access the file.
+ printf("\nThe audio recording has been saved to %s. Please use adb pull to get "
+ "the file and play it using audacity. The audio data has the "
+ "following characteristics:\nsample rate: %i\nformat: %d\n"
+ "num channels: %i\n",
+ params->output_filename, params->config.sample_rate,
+ params->config.format, num_channels);
+
+ pthread_mutex_lock(&glock);
+ tests_running--;
+ pthread_cond_signal(&gcond);
+ pthread_mutex_unlock(&glock);
+ pthread_exit(0);
+ return NULL;
+}
+
+int read_config_params_from_user(struct audio_config_params *thread_param, int rec_session) {
+ int channels = 0, format = 0, sample_rate = 0,source = 0, device = 0;
+
+ thread_param->kStreamName = "input_stream";
+
+ printf(" \n Enter input device (4->built-in mic, 16->wired_headset .. etc) ::::: ");
+ scanf(" %d", &device);
+ if (device & AUDIO_DEVICE_IN_BUILTIN_MIC)
+ thread_param->input_device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ else if (device & AUDIO_DEVICE_IN_WIRED_HEADSET)
+ thread_param->input_device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+
+ printf(" \n Enter the channels (1 -mono, 2 -stereo and 4 -quad channels) ::::: ");
+ scanf(" %d", &channels);
+ if (channels == 1) {
+ thread_param->config.channel_mask = AUDIO_CHANNEL_IN_MONO;
+ } else if (channels == 2) {
+ thread_param->config.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ } else if (channels == 4) {
+ thread_param->config.channel_mask = AUDIO_CHANNEL_INDEX_MASK_4;
+ } else {
+ gerror = true;
+ printf("\nINVALID channels");
+ return -1;
+ }
+
+ printf(" \n Enter the format (16 - 16 bit recording, 24 - 24 bit recording) ::::: ");
+ scanf(" %d", &format);
+ if (format == 16) {
+ thread_param->config.format = AUDIO_FORMAT_PCM_16_BIT;
+ } else if (format == 24) {
+ thread_param->config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ } else {
+ gerror = true;
+ printf("\n INVALID format");
+ return -1;
+ }
+
+ printf(" \n Enter the sample rate (48000, 16000 etc) :::: ");
+ scanf(" %d", &sample_rate);
+ thread_param->config.sample_rate = sample_rate;
+
+#ifdef MULTIRECORD_SUPPOT
+ printf(" \n Enter profile (none, record_fluence, record_mec, record_unprocessed etc) :::: ");
+ scanf(" %s", thread_param->profile);
+#else
+ thread_param->flags = (audio_input_flags_t)AUDIO_INPUT_FLAG_NONE;
+#endif
+ printf("\n Enter the audio source ( ref: system/media/audio/include/system/audio.h) :::: ");
+ scanf(" %d", &source);
+ thread_param->kInputSource = (audio_source_t)source;
+
+ if (rec_session == 1) {
+ thread_param->handle = 0x999;
+ strcpy(thread_param->output_filename, "/data/rec1.raw");
+ } else if (rec_session == 2) {
+ thread_param->handle = 0x998;
+ strcpy(thread_param->output_filename, "/data/rec2.raw");
+ } else if (rec_session == 3) {
+ thread_param->handle = 0x997;
+ strcpy(thread_param->output_filename, "/data/rec3.raw");
+ } else if (rec_session == 4) {
+ thread_param->handle = 0x996;
+ strcpy(thread_param->output_filename, "/data/rec4.raw");
+ }
+
+ printf("\n Enter the record duration in seconds :::: ");
+ scanf(" %lf", &thread_param->loopTime);
+ return 0;
+}
+
+int main() {
+ int max_recordings_requested = 0, source_track = 0;
+ int thread_active[4] = {0};
+ qahw_module_handle_t *qahw_mod_handle;
+ const char *mod_name = "audio.primary";
+
+ pthread_cond_init(&gcond, (const pthread_condattr_t *) NULL);
+
+ qahw_mod_handle = qahw_load_module(mod_name);
+ if(qahw_mod_handle == NULL) {
+ printf(" qahw_load_module failed");
+ return -1;
+ }
+#ifdef MULTIRECORD_SUPPOT
+ printf("Starting audio hal multi recording test. \n");
+ printf(" Enter number of record sessions to be started \n");
+ printf(" (Maximum of 4 record sessions are allowed):::: ");
+ scanf(" %d", &max_recordings_requested);
+#else
+ max_recordings_requested = 1;
+#endif
+ printf(" \n Source Tracking enabled ??? ( 1 - Enable 0 - Disable)::: ");
+ scanf(" %d", &source_track);
+
+ struct audio_config_params thread1_params, thread2_params;
+ struct audio_config_params thread3_params, thread4_params;
+
+ switch (max_recordings_requested) {
+ case 4:
+ printf(" Enter the config params for fourth record session \n");
+ thread4_params.qahw_mod_handle = qahw_mod_handle;
+ read_config_params_from_user( &thread4_params, 4);
+ thread_active[3] = 1;
+ printf(" \n");
+ case 3:
+ printf(" Enter the config params for third record session \n");
+ thread3_params.qahw_mod_handle = qahw_mod_handle;
+ read_config_params_from_user( &thread3_params, 3);
+ thread_active[2] = 1;
+ printf(" \n");
+ case 2:
+ printf(" Enter the config params for second record session \n");
+ thread2_params.qahw_mod_handle = qahw_mod_handle;
+ read_config_params_from_user( &thread2_params, 2);
+ thread_active[1] = 1;
+ printf(" \n");
+ case 1:
+ printf(" Enter the config params for first record session \n");
+ thread1_params.qahw_mod_handle = qahw_mod_handle;
+ read_config_params_from_user( &thread1_params, 1);
+ thread_active[0] = 1;
+ printf(" \n");
+ break;
+ default:
+ printf(" INVALID input -- Max record sessions supported is 4 -exit \n");
+ gerror = true;
+ break;
+ }
+
+ pthread_t tid[4];
+ pthread_t sourcetrack_thread;
+ int ret = -1;
+
+ if (thread_active[0] == 1) {
+ printf("\n Create first record thread \n");
+ ret = pthread_create(&tid[0], NULL, start_input, (void *)&thread1_params);
+ if (ret) {
+ gerror = true;
+ printf(" Failed to create first record thread \n ");
+ thread_active[0] = 0;
+ }
+ }
+ if (thread_active[1] == 1) {
+ printf("Create second record thread \n");
+ ret = pthread_create(&tid[1], NULL, start_input, (void *)&thread2_params);
+ if (ret) {
+ gerror = true;
+ printf(" Failed to create second record thread \n ");
+ thread_active[1] = 0;
+ }
+ }
+ if (thread_active[2] == 1) {
+ printf("Create third record thread \n");
+ ret = pthread_create(&tid[2], NULL, start_input, (void *)&thread3_params);
+ if (ret) {
+ gerror = true;
+ printf(" Failed to create third record thread \n ");
+ thread_active[2] = 0;
+ }
+ }
+ if (thread_active[3] == 1) {
+ printf("Create fourth record thread \n");
+ ret = pthread_create(&tid[3], NULL, start_input, (void *)&thread4_params);
+ if (ret) {
+ gerror = true;
+ printf(" Failed to create fourth record thread \n ");
+ thread_active[3] = 0;
+ }
+ }
+ if (source_track && max_recordings_requested) {
+ printf("Create source tracking thread \n");
+ ret = pthread_create(&sourcetrack_thread,
+ NULL, read_sourcetrack_data,
+ (void *)qahw_mod_handle);
+ if (ret) {
+ printf(" Failed to create source tracking thread \n ");
+ source_track = 0;
+ }
+ }
+
+ // set bad mic param
+ while (max_recordings_requested && !source_track) {
+ bool test_completed = false;
+
+ pthread_mutex_lock(&glock);
+ if (!tests_running && !gerror)
+ pthread_cond_wait(&gcond, &glock);
+ test_completed = (tests_running == 0);
+ gerror = true;
+ pthread_mutex_unlock(&glock);
+
+ if (test_completed)
+ break;
+#ifdef MULTIRECORD_SUPPOT
+ char ch;
+ printf("\n Bad mic test required (y/n):::");
+ scanf(" %c", &ch);
+ if (ch == 'y' || ch == 'Y') {
+ int bad_mic_ch_index, ret;
+ char param[100] = "bad_mic_channel_index=";
+ printf("\nEnter bad mic channel index (1, 2, 4 ...):::");
+ scanf(" %d", &bad_mic_ch_index);
+ snprintf(param, sizeof(param), "%s%d", param, bad_mic_ch_index);
+ ret = qahw_set_parameters(qahw_mod_handle, param);
+ printf("param %s set to hal with return value %d\n", param, ret);
+ } else {
+ break;
+ }
+#endif
+ }
+
+ printf(" Waiting for threads exit \n");
+ if (thread_active[0] == 1) {
+ pthread_join(tid[0], NULL);
+ printf("after first record thread exit \n");
+ }
+ if (thread_active[1] == 1) {
+ pthread_join(tid[1], NULL);
+ printf("after second record thread exit \n");
+ }
+ if (thread_active[2] == 1) {
+ pthread_join(tid[2], NULL);
+ printf("after third record thread exit \n");
+ }
+ if (thread_active[3] == 1) {
+ pthread_join(tid[3], NULL);
+ printf("after fourth record thread exit \n");
+ }
+ if (source_track) {
+ sourcetrack_done = 1;
+ pthread_join(sourcetrack_thread,NULL);
+ printf("after source tracking thread exit \n");
+ }
+
+ ret = qahw_unload_module(qahw_mod_handle);
+ if (ret) {
+ printf("could not unload hal %d \n",ret);
+ }
+
+
+ printf("Done with hal record test \n");
+ pthread_cond_destroy(&gcond);
+ return 0;
+}
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
new file mode 100644
index 0000000..8426945
--- /dev/null
+++ b/qahw_api/test/qahw_playback_test.c
@@ -0,0 +1,205 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2015 The Android Open Source Project *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/* Test app to play audio at the HAL layer */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <errno.h>
+#include "qahw_api.h"
+#include "qahw_defs.h"
+#define nullptr NULL
+#define WAV 1
+#define MP3 2
+
+
+/* Play audio from a WAV file.
+
+ Parameters:
+ out_stream: A pointer to the output audio stream.
+ in_file: A pointer to a SNDFILE object.
+ config: A pointer to struct that contains audio configuration data.
+
+ Returns: An int which has a non-negative number on success.
+*/
+
+int play_file(qahw_stream_handle_t* out_handle, FILE* in_file) {
+
+ int rc = 0;
+ size_t frames_read = 1;
+ size_t bytes_wanted ;
+ char *data = NULL;
+ qahw_out_buffer_t out_buf;
+
+ bytes_wanted = qahw_out_get_buffer_size(out_handle);
+ data = (char *) malloc (bytes_wanted);
+ if (data == NULL) {
+ printf("calloc failed!!\n");
+ return -ENOMEM;
+ }
+
+ while(frames_read != 0) {
+ frames_read = fread(data, bytes_wanted , 1, in_file);
+ if (frames_read < 1) {
+ if (feof(in_file))
+ break;
+ else
+ printf("Error in fread --%d\n",ferror(in_file));
+ }
+ memset(&out_buf,0, sizeof(qahw_out_buffer_t));
+ out_buf.buffer = data;
+ out_buf.bytes = frames_read * bytes_wanted;
+ rc = qahw_out_write(out_handle, &out_buf);
+ if (rc < 0) {
+ printf("Writing data to hal failed %d \n",rc);
+ break;
+ }
+ }
+ return rc;
+}
+
+// Prints usage information if input arguments are missing.
+void Usage() {
+ fprintf(stderr, "Usage:hal_play [device] [filename] [filetype]\n"
+ "device: hex value representing the audio device (see "
+ "system/media/audio/include/system/audio.h)\n"
+ "filename must be passed as an argument.\n"
+ "filetype (1:WAV 2:MP3) \n");
+}
+
+int main(int argc, char* argv[]) {
+ if (argc < 4) {
+ Usage();
+ return -1;
+ }
+ // Process command line arguments.
+ FILE *filestream = NULL;
+ char header[44] = {0};
+ int sample_rate = 0;
+ int channels = 0;
+ const int audio_device_base = 16;
+ char* filename = nullptr;
+ int filetype;
+ qahw_module_handle_t *qahw_mod_handle;
+ const char *mod_name = "audio.primary";
+
+ uint32_t desired_output_device = strtol(
+ argv[1], nullptr /* look at full string*/, audio_device_base);
+
+ filename = argv[2];
+ filetype = atoi (argv[3]);
+
+ printf("Starting audio hal tests.\n");
+ int rc = 0;
+
+ qahw_mod_handle = qahw_load_module(mod_name);
+
+ // Set to a high number so it doesn't interfere with existing stream handles
+ audio_io_handle_t handle = 0x999;
+ audio_devices_t output_device =
+ (audio_devices_t)desired_output_device;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
+ audio_config_t config;
+
+ memset(&config, 0, sizeof(audio_config_t));
+
+ if (filename) {
+ printf("filename-----%s\n",filename);
+ filestream = fopen (filename,"r");
+ if (filestream == NULL) {
+ printf("failed to open\n");
+ exit(0);
+ }
+ }
+
+ switch (filetype) {
+ case WAV:
+ //Read the wave header
+ rc = fread (header, 44 , 1, filestream);
+ if (rc != 1) {
+ printf("Error .Fread failed\n");
+ exit(0);
+ }
+ if (strncmp (header,"RIFF",4) && strncmp (header+8, "WAVE",4)) {
+ printf("Not a wave format\n");
+ exit (1);
+ }
+ memcpy (&channels, &header[22], 2);
+ memcpy (&sample_rate, &header[24], 4);
+ config.channel_mask = audio_channel_out_mask_from_count(channels);
+ config.offload_info.channel_mask = config.channel_mask;
+ config.offload_info.sample_rate = sample_rate;
+ config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT;
+ break;
+ case MP3:
+ printf("Enter Number of channels:");
+ scanf ("%d",&channels);
+ config.channel_mask = audio_channel_out_mask_from_count(channels);
+ printf("\nEnter Sample Rate:");
+ scanf ("%d",&sample_rate);
+ config.offload_info.channel_mask = config.channel_mask;
+ config.offload_info.sample_rate = sample_rate;
+ config.offload_info.format = AUDIO_FORMAT_MP3;
+ break;
+ default:
+ printf("Does not support given filetype\n");
+ Usage();
+ exit (0);
+ }
+ config.offload_info.version = AUDIO_OFFLOAD_INFO_VERSION_CURRENT;
+ config.offload_info.size = sizeof(audio_offload_info_t);
+
+ printf("Now playing to output_device=%d sample_rate=%d \n",output_device,
+ config.offload_info.sample_rate);
+ const char* stream_name = "output_stream";
+
+ // Open audio output stream.
+ qahw_stream_handle_t* out_handle = nullptr;
+ printf("calling open_out_put_stream:\n");
+ rc = qahw_open_output_stream(qahw_mod_handle, handle, output_device,
+ flags, &config, &out_handle,
+ stream_name);
+ printf("open output stream is sucess:%d out_handhle %p\n",rc,out_handle);
+ if (rc) {
+ printf("could not open output stream %d \n",rc);
+ return -1;
+ }
+
+ play_file(out_handle, filestream);
+
+ // Close output stream and device.
+ rc = qahw_out_standby(out_handle);
+ if (rc) {
+ printf("out standby failed %d \n",rc);
+ }
+
+ rc = qahw_close_output_stream(out_handle);
+ if (rc) {
+ printf("could not close output stream %d \n",rc);
+ }
+
+ rc = qahw_unload_module(qahw_mod_handle);
+ if (rc) {
+ printf("could not unload hal %d \n",rc);
+ return -1;
+ }
+
+ printf("Done with hal tests \n");
+ return 0;
+}
diff --git a/visualizer/Android.mk b/visualizer/Android.mk
index 87d4987..c9795ea 100644
--- a/visualizer/Android.mk
+++ b/visualizer/Android.mk
@@ -21,6 +21,10 @@
LOCAL_CFLAGS+= -O2 -fvisibility=hidden
+ifneq ($(filter msmcobalt,$(TARGET_BOARD_PLATFORM)),)
+ LOCAL_CFLAGS += -DPLATFORM_MSMCOBALT
+endif
+
LOCAL_SHARED_LIBRARIES := \
libcutils \
liblog \
diff --git a/visualizer/offload_visualizer.c b/visualizer/offload_visualizer.c
index b2f0952..f49c434 100644
--- a/visualizer/offload_visualizer.c
+++ b/visualizer/offload_visualizer.c
@@ -179,7 +179,11 @@
#define MIXER_CARD 0
#define SOUND_CARD 0
+#ifdef PLATFORM_MSMCOBALT
+#define CAPTURE_DEVICE 7
+#else
#define CAPTURE_DEVICE 8
+#endif
/* Proxy port supports only MMAP read and those fixed parameters*/
#define AUDIO_CAPTURE_CHANNEL_COUNT 2
diff --git a/voice_processing/voice_processing.c b/voice_processing/voice_processing.c
index 1e1e123..610bee6 100644
--- a/voice_processing/voice_processing.c
+++ b/voice_processing/voice_processing.c
@@ -565,7 +565,9 @@
if (pCmdData == NULL ||
cmdSize < (int)sizeof(effect_param_t) ||
pReplyData == NULL ||
- *replySize < (int)sizeof(effect_param_t)) {
+ *replySize < (int)sizeof(effect_param_t) ||
+ // constrain memcpy below
+ ((effect_param_t *)pCmdData)->psize > *replySize - sizeof(effect_param_t)) {
ALOGV("fx_command() EFFECT_CMD_GET_PARAM invalid args");
return -EINVAL;
}