Merge "DO NOT MERGE Fix AudioEffect reply overflow" into audio-userspace.lnx.2.1-dev
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index 1202aba..735aa35 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -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/mixer_paths_mtp.xml b/configs/msm8953/mixer_paths_mtp.xml
index 42a9e68..b9fc59a 100644
--- a/configs/msm8953/mixer_paths_mtp.xml
+++ b/configs/msm8953/mixer_paths_mtp.xml
@@ -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..486f49a 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" />
@@ -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_skum.xml b/configs/msm8953/mixer_paths_qrd_skum.xml
index 8343847..6aa8e1e 100644
--- a/configs/msm8953/mixer_paths_qrd_skum.xml
+++ b/configs/msm8953/mixer_paths_qrd_skum.xml
@@ -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/msm8953.mk b/configs/msm8953/msm8953.mk
index e646646..7da4800 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -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/msm8996.mk b/configs/msm8996/msm8996.mk
index 3b83c24..71705cb 100644
--- a/configs/msm8996/msm8996.mk
+++ b/configs/msm8996/msm8996.mk
@@ -196,3 +196,7 @@
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/audio_output_policy.conf b/configs/msmcobalt/audio_output_policy.conf
index 67d79bf..1bbaad2 100644
--- a/configs/msmcobalt/audio_output_policy.conf
+++ b/configs/msmcobalt/audio_output_policy.conf
@@ -48,8 +48,8 @@
}
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 512e8ee..f5547dc 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -55,6 +55,8 @@
<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" />
</pcm_ids>
<config_params>
<param key="spkr_1_tz_name" value="wsatz.13"/>
@@ -70,9 +72,12 @@
<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..166b9b6 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 {
@@ -61,14 +61,21 @@
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_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 235c157..451c85e 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -51,11 +51,9 @@
<attachedDevices>
<item>Earpiece</item>
<item>Speaker</item>
- <item>Telephony Tx</item>
<item>Built-In Mic</item>
<item>Built-In Back Mic</item>
<item>FM Tuner</item>
- <item>Telephony Rx</item>
</attachedDevices>
<defaultOutputDevice>Speaker</defaultOutputDevice>
<mixPorts>
@@ -118,6 +116,21 @@
<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"/>
@@ -137,9 +150,11 @@
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="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"/>
+ <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="voip_rx" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_VOIP_RX">
@@ -168,10 +183,6 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3,AUDIO_CHANNEL_INDEX_MASK_4"/>
</mixPort>
- <mixPort name="voice_rx" role="sink">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
- </mixPort>
</mixPorts>
<devicePorts>
@@ -212,10 +223,6 @@
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
</devicePort>
- <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
- </devicePort>
<devicePort tagName="HDMI" type="AUDIO_DEVICE_OUT_AUX_DIGITAL" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="dynamic"/>
@@ -228,6 +235,22 @@
<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"
@@ -253,9 +276,11 @@
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
</devicePort>
- <devicePort tagName="Telephony Rx" type="AUDIO_DEVICE_IN_TELEPHONY_RX" role="source">
+ <devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ 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 -->
@@ -265,11 +290,11 @@
<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,voip_rx"/>
+ 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,voip_rx"/>
+ 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,voip_rx"/>
+ 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"
@@ -278,25 +303,65 @@
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="Telephony Tx"
- sources="voice_tx"/>
+ <route type="mix" sink="USB Device Out"
+ sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_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"/>
<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="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_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index 860d014..d096d1f 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -548,6 +548,11 @@
<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" />
@@ -614,7 +619,7 @@
</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>
@@ -657,7 +662,7 @@
</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>
@@ -689,7 +694,7 @@
</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 +702,7 @@
</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 +711,15 @@
</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="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>
@@ -724,11 +729,11 @@
</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">
@@ -760,7 +765,7 @@
</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>
@@ -808,7 +813,7 @@
</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>
@@ -856,7 +861,7 @@
</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>
@@ -904,7 +909,7 @@
</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>
@@ -952,7 +957,7 @@
</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>
@@ -1000,7 +1005,7 @@
</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>
@@ -1048,7 +1053,7 @@
</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>
@@ -1096,7 +1101,7 @@
</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>
@@ -1144,7 +1149,7 @@
</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>
@@ -1192,7 +1197,7 @@
</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 +1214,7 @@
</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 +1223,24 @@
</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 +1398,12 @@
</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 +1424,7 @@
</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 +1464,7 @@
</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 +1504,7 @@
</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 +1544,7 @@
</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 +1641,6 @@
</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 +1659,15 @@
<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" />
@@ -1763,11 +1768,11 @@
</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>
@@ -1820,7 +1825,7 @@
</path>
<path name="handset-mic">
- <path name="dmic1" />
+ <path name="dmic3" />
</path>
<path name="handset-mic-db">
@@ -1847,10 +1852,10 @@
<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">
@@ -1862,7 +1867,7 @@
<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="DMIC MUX10" value="DMIC2" />
<ctl name="ANC0 FB MUX" value="ANC_IN_EAR" />
<ctl name="ANC EAR Enable Switch" value="1" />
</path>
@@ -1871,8 +1876,8 @@
<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>
@@ -1902,6 +1907,14 @@
<ctl name= "RX INT2 SPLINE MIX HPHR Native Switch" value="1" />
</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">
<path name="headphones" />
</path>
@@ -2146,13 +2159,13 @@
<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" />
</path>
@@ -2162,10 +2175,10 @@
<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 +2187,10 @@
<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 +2262,7 @@
<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 +2286,16 @@
<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 +2373,7 @@
<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 +2389,122 @@
<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 1c92421..b9b0b03 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -45,11 +45,34 @@
<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="DEC0 Volume" value="84" />
+ <ctl name="DEC2 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" />
+ <ctl name="ADC1 Volume" value="12" />
+ <ctl name="ADC2 Volume" value="12" />
+ <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="ADC MUX5" value="AMIC" />
+ <ctl name="ADC MUX6" value="AMIC" />
+ <ctl name="ADC MUX7" value="AMIC" />
+ <ctl name="ADC MUX8" value="AMIC" />
+ <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="AMIC MUX0" value="ZERO" />
+ <ctl name="AMIC MUX6" value="ZERO" />
<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" />
@@ -67,6 +90,9 @@
<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" />
@@ -88,33 +114,46 @@
<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 +161,7 @@
<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" />
@@ -139,8 +179,13 @@
<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" />
@@ -262,6 +307,12 @@
<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="0" />
+ <!-- split a2dp end-->
+
<!-- ADSP testfwk -->
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
<ctl name="SLIMBUS6_DL_HL Switch" value="0" />
@@ -272,10 +323,28 @@
<!-- Codec 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="RX INT1_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT2_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT1_2 MUX" value="ZERO" />
+ <ctl name="RX INT2_2 MUX" value="ZERO" />
<ctl name="RX INT7_1 MIX1 INP0" value="ZERO" />
<ctl name="RX INT8_1 MIX1 INP0" value="ZERO" />
+ <ctl name="COMP1 Switch" value="1" />
+ <ctl name="COMP2 Switch" value="1" />
<ctl name="COMP7 Switch" value="0" />
<ctl name="COMP8 Switch" value="0" />
<ctl name="SpkrLeft COMP Switch" value="0" />
@@ -287,8 +356,28 @@
<ctl name="SpkrLeft SWR DAC_Port Switch" value="0" />
<ctl name="SpkrRight SWR DAC_Port Switch" value="0" />
- <ctl name="AIF1_CAP Mixer SLIM TX0" value="0" />
- <ctl name="AIF1_CAP Mixer SLIM TX2" value="0" />
+ <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="RX INT1 SEC MIX HPHL Switch" value="0" />
+ <ctl name="ASRC1 MUX" value="ZERO" />
+ <ctl name="RX INT2 SEC MIX HPHR Switch" 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="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 INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="CDC_IF TX0 MUX" value="ZERO" />
<ctl name="CDC_IF TX2 MUX" value="ZERO" />
<ctl name="ADC MUX0" value="ZERO" />
@@ -296,8 +385,6 @@
<ctl name="DMIC MUX0" value="ZERO" />
<ctl name="DMIC MUX2" value="ZERO" />
- <ctl name="DEC0 Volume" value="0" />
- <ctl name="DEC2 Volume" value="0" />
<ctl name="RX7 Digital Volume" value="84" />
<ctl name="RX8 Digital Volume" value="84" />
@@ -330,6 +417,7 @@
</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">
@@ -357,7 +445,7 @@
</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>
@@ -400,7 +488,7 @@
</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>
@@ -432,7 +520,7 @@
</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">
@@ -440,7 +528,7 @@
</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">
@@ -449,15 +537,15 @@
</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="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>
@@ -467,11 +555,11 @@
</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">
@@ -503,7 +591,7 @@
</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>
@@ -533,6 +621,10 @@
<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" />
@@ -551,7 +643,7 @@
</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>
@@ -581,6 +673,10 @@
<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" />
@@ -599,7 +695,7 @@
</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>
@@ -629,6 +725,10 @@
<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" />
@@ -647,7 +747,7 @@
</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>
@@ -677,6 +777,10 @@
<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" />
@@ -695,7 +799,7 @@
</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>
@@ -725,6 +829,10 @@
<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" />
@@ -743,7 +851,7 @@
</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>
@@ -773,6 +881,10 @@
<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" />
@@ -791,7 +903,7 @@
</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>
@@ -821,6 +933,10 @@
<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" />
@@ -839,7 +955,7 @@
</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>
@@ -869,6 +985,10 @@
<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" />
@@ -887,7 +1007,7 @@
</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>
@@ -917,6 +1037,10 @@
<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" />
@@ -935,7 +1059,7 @@
</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>
@@ -952,7 +1076,7 @@
</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>
@@ -961,24 +1085,24 @@
</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">
@@ -1150,7 +1274,7 @@
</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>
@@ -1190,7 +1314,7 @@
</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>
@@ -1230,7 +1354,7 @@
</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>
@@ -1258,40 +1382,68 @@
<!-- 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" />
</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" />
</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" />
</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" />
</path>
<path name="dmic3">
- <ctl name="AIF1_CAP Mixer SLIM TX2" value="1" />
- <ctl name="CDC_IF TX2 MUX" value="DEC2" />
+ <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 MUX2" value="DMIC" />
- <ctl name="DMIC MUX2" value="DMIC2" />
- <ctl name="DEC2 Volume" value="84" />
+ <ctl name="ADC MUX7" value="DMIC" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
</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" />
</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" />
</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" />
</path>
<path name="speaker">
@@ -1353,22 +1505,32 @@
</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">
@@ -1396,7 +1558,7 @@
</path>
<path name="handset-mic">
- <path name="dmic1" />
+ <path name="dmic3" />
</path>
<path name="handset-mic-db">
@@ -1410,15 +1572,60 @@
</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">
</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="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="true-native-mode">
@@ -1571,9 +1778,27 @@
<!-- 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">
@@ -1637,6 +1862,15 @@
</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">
@@ -1649,6 +1883,23 @@
<!-- 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">
@@ -1715,4 +1966,122 @@
<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="2" />
+ </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 05be352..bb6ee95 100644
--- a/configs/msmcobalt/msmcobalt.mk
+++ b/configs/msmcobalt/msmcobalt.mk
@@ -4,7 +4,7 @@
BOARD_USES_ALSA_AUDIO := true
USE_CUSTOM_AUDIO_POLICY := 1
USE_XML_AUDIO_POLICY_CONF := 1
-BOARD_SUPPORTS_SOUND_TRIGGER := true
+BOARD_SUPPORTS_SOUND_TRIGGER_HAL := true
AUDIO_USE_LL_AS_PRIMARY_OUTPUT := true
AUDIO_FEATURE_ENABLED_VBAT_MONITOR := true
@@ -19,7 +19,7 @@
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_DS2_DOLBY_DAP := true
#DOLBY_DDP := true
@@ -52,6 +52,7 @@
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
+AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
##AUDIO_FEATURE_FLAGS
#Audio Specific device overlays
@@ -78,7 +79,9 @@
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 +169,9 @@
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 +189,21 @@
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..d12b62f
--- /dev/null
+++ b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
@@ -0,0 +1,115 @@
+<?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_BROADCAST Switch" value="1" />
+ <ctl name="TX13 INP MUX" value="MAD_BRDCST" />
+ <ctl name="AIF4_MAD Mixer SLIM TX13" value="1" />
+ <ctl name="MAD Input" value="DMIC0" />
+ </path>
+
+</mixer>
diff --git a/configs/msmcobalt/sound_trigger_platform_info.xml b/configs/msmcobalt/sound_trigger_platform_info.xml
index b92ea48..7ce74aa 100644
--- a/configs/msmcobalt/sound_trigger_platform_info.xml
+++ b/configs/msmcobalt/sound_trigger_platform_info.xml
@@ -29,8 +29,7 @@
<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_cpe_sessions="2" />
<param max_ape_sessions="8" />
<param enable_failure_detection="false" />
</common_config>
@@ -41,11 +40,12 @@
<param DEVICE_HANDSET_CPE_ECPP_ACDB_ID="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" />
@@ -54,7 +54,27 @@
<param max_ape_users="10" />
<param sample_rate="16000" />
- <!-- Module and param ids with which the algorithm is integrated in firmware -->
+ <gcs_uid>
+ <param uid="0x1" />
+ <param did="0x4" />
+ <param load_sound_model_ids="0x00012C0D, 0x0, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x0, 0x00012C28" />
+ <param operation_mode_ids="0x00012C0D, 0x0, 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x0, 0x00012C29" />
+ <param capture_event_ids="0x00020013, 0x0,0x00020015" />
+ </gcs_uid>
+ <gcs_uid>
+ <param uid="0x2" />
+ <param did="0x4" />
+ <param load_sound_model_ids="0x00012C0D, 0x1, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x1, 0x00012C28" />
+ <param operation_mode_ids="0x00012C0D, 0x1 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x1, 0x00012C29" />
+ <param capture_event_ids="0x00020013, 0x1,0x00020015" />
+ </gcs_uid>
+
+ <!-- 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" />
@@ -62,7 +82,8 @@
<!-- 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>
diff --git a/hal/Android.mk b/hal/Android.mk
index 83787e3..705e5e8 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -243,6 +243,11 @@
LOCAL_SRC_FILES += audio_extn/source_track.c
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_SPLIT_A2DP)),true)
+ LOCAL_CFLAGS += -DSPLIT_A2DP_ENABLED
+ LOCAL_SRC_FILES += audio_extn/a2dp.c
+endif
+
LOCAL_SHARED_LIBRARIES := \
liblog \
libcutils \
@@ -251,6 +256,7 @@
libaudioroute \
libdl \
libaudioutils \
+ libhardware \
libexpat
LOCAL_C_INCLUDES += \
@@ -279,6 +285,14 @@
endif
ifeq ($(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 += $(TARGET_OUT_HEADERS)/mm-audio/sound_trigger
@@ -301,6 +315,11 @@
LOCAL_COPY_HEADERS_TO := mm-audio
LOCAL_COPY_HEADERS := audio_extn/audio_defs.h
+ifeq ($(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..7293ded
--- /dev/null
+++ b/hal/audio_extn/a2dp.c
@@ -0,0 +1,705 @@
+/*
+* 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 MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0
+#define MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3
+#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_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;
+ void *enc_config_data;
+ 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;
+ uint32_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_config_data = NULL;
+ 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_config_data = NULL;
+ 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;
+ 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;
+ } else
+ is_configured = true;
+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;
+ 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;
+ }
+ is_configured = true;
+fail:
+ return is_configured;
+}
+
+/* API to configure APTX HD DSP encoder
+ * TODO: ADD 24 bit configuration support
+ */
+bool configure_aptx_hd_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
+{
+ struct mixer_ctl *ctl_enc_data;
+ 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;
+ }
+ is_configured = true;
+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;
+ 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;
+ 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;
+ }
+ if (aac_bt_cfg->format_flag)
+ aac_dsp_cfg.aac_fmt_flag = MEDIA_FMT_AAC_FORMAT_FLAG_RAW;
+ else
+ aac_dsp_cfg.aac_fmt_flag = MEDIA_FMT_AAC_FORMAT_FLAG_ADTS;
+ 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;
+ } else
+ is_configured = true;
+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;
+ 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;
+ }
+ }
+ 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_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_config_data = NULL;
+ 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..569b4b2 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -755,6 +755,7 @@
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);
@@ -762,6 +763,7 @@
audio_extn_pm_set_parameters(parms);
audio_extn_source_track_set_parameters(adev, parms);
audio_extn_fbsp_set_parameters(parms);
+ audio_extn_keep_alive_set_parameters(adev, parms);
check_and_set_hdmi_connection_status(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..e8caeee 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -171,6 +171,22 @@
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)
+#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);
+#endif
+
#ifndef SSR_ENABLED
#define audio_extn_ssr_check_and_set_usecase(in) (0)
#define audio_extn_ssr_init(in, num_out_chan) (0)
@@ -387,6 +403,10 @@
#endif
+#ifndef AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
+#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH 0x10000
+#endif
+
#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 +425,6 @@
#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);
@@ -572,4 +590,17 @@
#endif
+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/keep_alive.c b/hal/audio_extn/keep_alive.c
index 1a4f135..60e7eef 100644
--- a/hal/audio_extn/keep_alive.c
+++ b/hal/audio_extn/keep_alive.c
@@ -38,7 +38,7 @@
#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 +52,9 @@
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 +90,8 @@
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) {
@@ -143,6 +147,27 @@
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 +176,20 @@
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) {
@@ -170,6 +197,14 @@
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 */
@@ -202,9 +237,15 @@
platform_get_default_app_type(adev->platform),
acdb_dev_id, sample_rate);
mixer_ctl_set_array(ctl, app_type_cfg, len);
+ /*Configure HDMI Backend with default values, this as well
+ *helps reconfigure HDMI backend after passthrough
+ */
+ 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");
/*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;
@@ -232,13 +273,13 @@
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);
}
@@ -290,6 +331,7 @@
struct listnode *item;
uint8_t * silence = NULL;
int32_t bytes = 0;
+ struct timespec ts;
while (true) {
pthread_mutex_lock(&ka.lock);
@@ -328,9 +370,17 @@
* 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/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/utils.c b/hal/audio_extn/utils.c
index e3f1b6c..26c43b4 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
};
@@ -515,6 +514,21 @@
__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;
+ }
+
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 8660b5a..43c83de 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
@@ -284,6 +316,116 @@
//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 (out->routing_change) {
+ out->routing_change = false;
+ // must be checked for backward compatibility
+ if (adev->adm_on_routing_change)
+ adev->adm_on_routing_change(adev->adm_data, out->handle);
+ }
+
+ 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 (in->routing_change) {
+ in->routing_change = false;
+ if (adev->adm_on_routing_change)
+ adev->adm_on_routing_change(adev->adm_data, in->capture_handle);
+ }
+
+ 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 +502,7 @@
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 +511,12 @@
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 +543,9 @@
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 +621,36 @@
return 0;
}
+/*
+ * Enable ASRC mode if native or DSD stream is active.
+ */
+static void audio_check_and_set_asrc_mode(struct audio_device *adev, snd_device_t snd_device)
+{
+ if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
+ !adev->asrc_mode_enabled) {
+ 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) {
+ if((platform_get_backend_index(uc->out_snd_device) == HEADPHONE_44_1_BACKEND) ||
+ (platform_get_backend_index(uc->out_snd_device) == DSD_NATIVE_BACKEND)) {
+ ALOGD("%s:DSD or native stream detected enabling asrcmode in hardware",
+ __func__);
+ audio_route_apply_and_update_path(adev->audio_route,
+ "asrc-mode");
+ adev->asrc_mode_enabled = true;
+ break;
+ }
+ }
+ }
+ }
+}
+
int pcm_ioctl(struct pcm *pcm, int request, ...)
{
va_list ap;
@@ -568,7 +750,6 @@
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 +769,13 @@
}
} 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,
@@ -613,7 +801,8 @@
audio_route_apply_and_update_path(adev->audio_route,
"true-native-mode");
adev->native_playback_enabled = true;
- }
+ } else
+ audio_check_and_set_asrc_mode(adev, snd_device);
}
return 0;
}
@@ -644,6 +833,7 @@
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);
@@ -656,6 +846,9 @@
audio_route_reset_and_update_path(adev->audio_route, device_name);
}
+ if (SND_DEVICE_OUT_BT_A2DP == snd_device)
+ audio_extn_a2dp_stop_playback();
+
if (snd_device == SND_DEVICE_OUT_HDMI)
adev->is_channel_status_set = false;
else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
@@ -665,6 +858,11 @@
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__);
+ audio_route_reset_and_update_path(adev->audio_route, "asrc-mode");
+ adev->asrc_mode_enabled = false;
}
audio_extn_dev_arbi_release(snd_device);
@@ -685,7 +883,7 @@
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 +903,15 @@
*/
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 +930,13 @@
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) ||
+ (force_restart_session)) &&
+ (platform_check_backends_match(snd_device, usecase->out_snd_device)||
+ (platform_check_codec_asrc_support(adev->platform) && !adev->asrc_mode_enabled &&
+ platform_check_if_backend_has_to_be_disabled(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));
@@ -997,6 +1207,28 @@
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__);
+ }
+ }
+ return active;
+}
static bool force_device_switch(struct audio_usecase *usecase)
{
@@ -1016,6 +1248,14 @@
}
}
+ // 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 +1300,8 @@
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 +1309,8 @@
} 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;
@@ -1182,10 +1425,6 @@
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;
@@ -1206,6 +1445,13 @@
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 +1570,8 @@
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 +1602,13 @@
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 +1869,6 @@
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 +1909,14 @@
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 +1958,6 @@
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 +1969,16 @@
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 +1989,8 @@
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 +2071,20 @@
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);
@@ -2141,7 +2222,7 @@
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 +2253,6 @@
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 +2264,13 @@
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 +2364,17 @@
(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
@@ -2304,7 +2395,9 @@
* playback to headset.
*/
if (val != 0) {
- out->devices = val;
+ 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)) {
@@ -2319,11 +2412,16 @@
}
if (!out->standby) {
- audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
- adev->perf_lock_opts,
- adev->perf_lock_opts_size);
+ if (!same_dev) {
+ ALOGV("update routing change");
+ out->routing_change = true;
+ 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);
+ if (!same_dev)
+ audio_extn_perf_lock_release(&adev->perf_lock_handle);
}
}
@@ -2478,11 +2576,21 @@
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 +2600,14 @@
return latency;
}
+static float AmpToDb(float amplification)
+{
+ if (amplification == 0) {
+ return DSD_VOLUME_MIN_DB;
+ }
+ return 20 * log10(amplification);
+}
+
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
@@ -2510,6 +2626,20 @@
* 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 +2686,11 @@
/* 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 +2698,10 @@
}
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 +2723,6 @@
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)){
@@ -2679,12 +2810,19 @@
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 +2839,14 @@
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);
}
}
@@ -2939,7 +3076,6 @@
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);
@@ -3011,8 +3147,8 @@
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 +3179,6 @@
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 +3192,16 @@
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 +3258,11 @@
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");
+ in->routing_change = true;
ret = select_devices(adev, in->usecase);
+ }
}
}
@@ -3212,19 +3348,24 @@
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 +3387,7 @@
}
}
- 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
@@ -3514,7 +3654,12 @@
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 +3676,7 @@
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 */
@@ -3614,12 +3759,24 @@
__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->format == AUDIO_FORMAT_DSD) ||
config->offload_info.has_video ||
out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
check_and_set_gapless_mode(adev, false);
@@ -3629,6 +3786,10 @@
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;
+ }
} else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
if (ret != 0) {
@@ -3661,7 +3822,9 @@
} else {
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;
@@ -3748,6 +3911,7 @@
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() */
@@ -3945,7 +4109,7 @@
ALOGV("cache new edid");
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 +4133,7 @@
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 +4143,26 @@
}
}
+ 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 +4370,21 @@
#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 +4460,17 @@
}
}
- 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)) &&
@@ -4535,6 +4729,14 @@
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 +4768,16 @@
}
}
+ 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..664d1fc 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[].
@@ -162,6 +167,7 @@
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 {
@@ -228,6 +234,10 @@
audio_format_t hal_op_format;
void *convert_buffer;
+ bool realtime;
+ int af_period_multiplier;
+ bool routing_change;
+
struct audio_device *dev;
};
@@ -252,6 +262,10 @@
bool is_st_session_active;
int sample_rate;
int bit_width;
+ bool realtime;
+ int af_period_multiplier;
+ bool routing_change;
+
struct audio_device *dev;
};
@@ -308,6 +322,12 @@
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 +381,10 @@
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 +395,7 @@
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,
@@ -392,6 +417,8 @@
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);
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 500a28d..b41e040 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -192,8 +192,10 @@
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};
@@ -346,6 +348,8 @@
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
[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",
@@ -465,6 +469,8 @@
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 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,
@@ -586,6 +592,8 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
{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)},
@@ -767,6 +775,7 @@
#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) {
@@ -1205,6 +1214,8 @@
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_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");
@@ -1847,6 +1858,9 @@
/* 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);
@@ -1915,6 +1929,8 @@
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");
ret = audio_extn_utils_get_codec_version(snd_card_name,
my_data->adev->snd_card,
@@ -2420,8 +2436,19 @@
return ret;
}
+int check_44100_support_device(audio_devices_t out_device)
+{
+ int ret = true;
-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 = false;
+
+ return ret;
+}
+
+int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
@@ -2803,6 +2830,10 @@
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__,
@@ -2875,6 +2906,9 @@
} 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;
@@ -2925,6 +2959,8 @@
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;
@@ -3008,6 +3044,8 @@
snd_device = SND_DEVICE_OUT_BT_SCO;
} else if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
snd_device = SND_DEVICE_OUT_HDMI ;
+ } 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__);
@@ -3787,8 +3825,8 @@
!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
@@ -3825,6 +3863,8 @@
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;
}
@@ -4014,16 +4054,21 @@
* 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 !=
@@ -4115,19 +4160,146 @@
mixer_ctl_set_enum_by_string(ctl, rate_str);
my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
}
+ if ((backend_idx == HDMI_RX_BACKEND) &&
+ (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;
+ 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;
+ 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);
+ }
+
+ if (backend_idx == HDMI_RX_BACKEND) {
+ const char *hdmi_format_ctrl = "HDMI RX Format";
+ struct mixer_ctl *ctl;
+ ctl = mixer_get_ctl_by_name(adev->mixer,hdmi_format_ctrl);
+
+ if (!ctl) {
+ ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+ __func__, hdmi_format_ctrl);
+ return -EINVAL;
+ }
+
+ if (passthrough_enabled) {
+ ALOGD("%s:HDMI compress format", __func__);
+ mixer_ctl_set_enum_by_string(ctl, "Compr");
+ } else {
+ ALOGD("%s: HDMI 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,
+ 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[HDMI_RX_BACKEND].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[HDMI_RX_BACKEND].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 +4307,27 @@
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) {
+ usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
ALOGW("%s:becf: afe:Use default bw and sr for voice/voip calls ",
__func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
@@ -4173,11 +4348,12 @@
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)) {
@@ -4187,6 +4363,8 @@
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;
}
}
}
@@ -4215,14 +4393,12 @@
}
/*
- * hifi playback not supported on spkr devices, limit the Sample Rate
+ * hifi playback not supported on non-44.1-support 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 (check_44100_support_device(usecase->devices)) {
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: playback on speaker device Configure afe to "
+ ALOGD("%s:becf: afe: playback on non-44.1-support device Configure afe to "
"default Sample Rate(48k)", __func__);
}
@@ -4246,16 +4422,21 @@
}
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;
+
+ platform_check_hdmi_backend_cfg(adev, usecase, &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
@@ -4286,13 +4467,16 @@
// 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 +4485,24 @@
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 +4513,9 @@
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;
}
}
@@ -5000,6 +5185,7 @@
//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;
+ my_data->current_backend_cfg[HDMI_RX_BACKEND].channels = DEFAULT_HDMI_OUT_CHANNELS;
}
int platform_set_mixer_control(struct stream_out *out, const char * mixer_ctl_name,
@@ -5018,91 +5204,6 @@
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;
@@ -5311,3 +5412,19 @@
}
return 0;
}
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+ return false;
+}
+
+bool platform_check_codec_asrc_support(void *platform __unused)
+{
+ return false;
+}
+
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device __unused,
+ snd_device_t cuurent_snd_device __unused)
+{
+ return false;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 756c749..6c89d0a 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -98,6 +98,8 @@
SND_DEVICE_OUT_SPEAKER_AND_HDMI,
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,
@@ -196,9 +198,12 @@
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
@@ -206,6 +211,8 @@
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,
@@ -354,7 +361,8 @@
enum {
LEGACY_PCM = 0,
PASSTHROUGH,
- PASSTHROUGH_CONVERT
+ PASSTHROUGH_CONVERT,
+ PASSTHROUGH_DSD
};
/*
* ID for setting mute and lateny on the device side
@@ -370,4 +378,13 @@
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..e5d42bd 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1257,3 +1257,24 @@
}
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;
+}
+
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device __unused,
+ snd_device_t cuurent_snd_device __unused)
+{
+ return false;
+}
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index aab5436..07060b6 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
@@ -64,6 +64,8 @@
SND_DEVICE_OUT_HDMI,
SND_DEVICE_OUT_SPEAKER_AND_HDMI,
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 +112,12 @@
#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/platform.c b/hal/msm8974/platform.c
index b98cc73..24bee89 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -188,8 +188,10 @@
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};
@@ -245,6 +247,8 @@
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] = {
@@ -332,6 +336,7 @@
[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",
@@ -347,6 +352,8 @@
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
[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",
@@ -423,6 +430,7 @@
[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 +454,7 @@
[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,
@@ -460,6 +469,8 @@
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 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,
@@ -535,6 +546,7 @@
[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 +572,7 @@
{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)},
@@ -575,6 +588,8 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
{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)},
@@ -646,6 +661,7 @@
{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)},
@@ -783,6 +799,7 @@
#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 +1068,8 @@
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;
}
@@ -1089,9 +1107,13 @@
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_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");
@@ -1672,6 +1694,9 @@
/* 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);
@@ -1697,6 +1722,11 @@
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 =
@@ -1727,6 +1757,13 @@
}
}
+ 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 =
@@ -1735,6 +1772,8 @@
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[USB_AUDIO_RX_BACKEND].bitwidth_mixer_ctl =
strdup("USB_AUDIO_RX Format");
@@ -1887,6 +1926,32 @@
return result;
}
+bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device,
+ snd_device_t cuurent_snd_device)
+{
+ bool result = false;
+
+ ALOGV("%s: current snd device = %s, new snd device = %s", __func__,
+ platform_get_snd_device_name(cuurent_snd_device),
+ platform_get_snd_device_name(new_snd_device));
+
+ if ((new_snd_device < SND_DEVICE_MIN) || (new_snd_device >= SND_DEVICE_OUT_END) ||
+ (cuurent_snd_device < SND_DEVICE_MIN) || (cuurent_snd_device >= SND_DEVICE_OUT_END)) {
+ ALOGE("%s: Invalid snd_device",__func__);
+ return false;
+ }
+
+ if (cuurent_snd_device == SND_DEVICE_OUT_HEADPHONES &&
+ (new_snd_device == SND_DEVICE_OUT_HEADPHONES_44_1 ||
+ new_snd_device == SND_DEVICE_OUT_HEADPHONES_DSD)) {
+ result = true;
+ }
+
+ ALOGV("%s: Need to disable current backend %s, %d",
+ __func__, platform_get_snd_device_name(cuurent_snd_device), result);
+ return result;
+}
+
int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type)
{
int device_id;
@@ -2072,7 +2137,8 @@
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 +2153,18 @@
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 +2217,8 @@
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",
@@ -2205,7 +2285,20 @@
return ret;
}
-static int platform_get_backend_index(snd_device_t snd_device)
+
+int check_44100_support_device(audio_devices_t out_device)
+{
+ int ret = true;
+
+ if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+ out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+ out_device & AUDIO_DEVICE_OUT_LINE)
+ ret = false;
+
+ return ret;
+}
+
+int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
@@ -2214,6 +2307,9 @@
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;
@@ -2582,8 +2678,13 @@
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);
@@ -2642,6 +2743,9 @@
} 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;
@@ -2697,6 +2801,8 @@
snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
else
snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ } 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;
@@ -2726,6 +2832,12 @@
} 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) {
@@ -2746,6 +2858,8 @@
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 ;
} else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
@@ -2896,7 +3010,7 @@
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;
@@ -3850,8 +3964,8 @@
!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
@@ -3886,6 +4000,8 @@
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;
}
@@ -3984,17 +4100,20 @@
* 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;
-
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 !=
@@ -4025,14 +4144,6 @@
my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
}
- /*
- * 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;
@@ -4051,14 +4162,24 @@
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;
@@ -4078,44 +4199,185 @@
mixer_ctl_set_enum_by_string(ctl, rate_str);
my_data->current_backend_cfg[backend_idx].sample_rate = sample_rate;
}
+ if ((backend_idx == HDMI_RX_BACKEND) &&
+ (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;
+ 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;
+ 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);
+ }
+
+ if (backend_idx == HDMI_RX_BACKEND) {
+ const char *hdmi_format_ctrl = "HDMI RX Format";
+ struct mixer_ctl *ctl;
+ ctl = mixer_get_ctl_by_name(adev->mixer,hdmi_format_ctrl);
+
+ if (!ctl) {
+ ALOGE("%s:becf: afe: Could not get ctl for mixer command - %s",
+ __func__, hdmi_format_ctrl);
+ return -EINVAL;
+ }
+
+ if (passthrough_enabled) {
+ ALOGD("%s:HDMI compress format", __func__);
+ mixer_ctl_set_enum_by_string(ctl, "Compr");
+ } else {
+ ALOGD("%s: HDMI 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 || (sample_rate == 48000 && bit_width >= 24)) {
+ 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,
+ 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[HDMI_RX_BACKEND].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[HDMI_RX_BACKEND].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) {
+ usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
ALOGW("%s:becf: afe:Use default bw and sr for voice/voip calls ",
__func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
@@ -4136,11 +4398,12 @@
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)) {
@@ -4150,6 +4413,8 @@
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;
}
}
}
@@ -4178,14 +4443,12 @@
}
/*
- * hifi playback not supported on spkr devices, limit the Sample Rate
+ * hifi playback not supported on non-44.1-support 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 (check_44100_support_device(usecase->devices)) {
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: playback on speaker device Configure afe to "
+ ALOGD("%s:becf: afe: playback on non-44.1-support device Configure afe to "
"default Sample Rate(48k)", __func__);
}
@@ -4207,29 +4470,57 @@
}
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;
+
+ platform_check_hdmi_backend_cfg(adev, usecase, &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;
@@ -4238,40 +4529,50 @@
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;
}
@@ -4936,6 +5237,7 @@
//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;
}
@@ -4955,90 +5257,6 @@
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;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 019678a..9394ef8 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -80,6 +80,7 @@
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,
@@ -94,6 +95,8 @@
SND_DEVICE_OUT_SPEAKER_AND_HDMI,
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,
@@ -177,6 +180,7 @@
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,
@@ -189,13 +193,18 @@
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_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,
@@ -444,7 +453,8 @@
enum {
LEGACY_PCM = 0,
PASSTHROUGH,
- PASSTHROUGH_CONVERT
+ PASSTHROUGH_CONVERT,
+ PASSTHROUGH_DSD
};
/*
* ID for setting mute and lateny on the device side
@@ -460,4 +470,13 @@
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..ec64206 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -27,6 +27,7 @@
enum {
NATIVE_AUDIO_MODE_SRC = 1,
NATIVE_AUDIO_MODE_TRUE_44_1,
+ NATIVE_AUDIO_MODE_MULTIPLE_44_1,
NATIVE_AUDIO_MODE_INVALID
};
@@ -36,6 +37,8 @@
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);
@@ -151,4 +154,8 @@
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);
#endif // AUDIO_PLATFORM_API_H
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 122ac14..015ea25 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -511,6 +511,14 @@
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 =
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index dfda1c9..deef57d 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