Merge "policy_hal: add bitrate restriction for wma/wma_pro/wma_lossless" into audio-userspace.lnx.2.1-dev
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/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index b7da238..4bde15c 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>
@@ -143,10 +141,6 @@
samplingRates="2822400,5644800"
channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
</mixPort>
- <mixPort name="voice_tx" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
- </mixPort>
<mixPort name="voip_rx" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_VOIP_RX">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
@@ -174,10 +168,6 @@
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>
@@ -218,10 +208,6 @@
<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"/>
@@ -246,6 +232,10 @@
<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"
@@ -271,9 +261,11 @@
<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 -->
@@ -296,16 +288,14 @@
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"
@@ -339,7 +329,24 @@
</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/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index eb5a150..d096d1f 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -1641,15 +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" />
@@ -1668,6 +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" />
@@ -1768,11 +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>
@@ -1825,7 +1825,7 @@
</path>
<path name="handset-mic">
- <path name="dmic1" />
+ <path name="dmic3" />
</path>
<path name="handset-mic-db">
@@ -1852,10 +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">
@@ -1867,7 +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>
@@ -2159,13 +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>
@@ -2175,10 +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>
@@ -2187,10 +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>
@@ -2262,7 +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" />
@@ -2286,16 +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">
@@ -2373,7 +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">
diff --git a/configs/msmcobalt/mixer_paths_tavil.xml b/configs/msmcobalt/mixer_paths_tavil.xml
index 98b9bbf..dab6cac 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -45,6 +45,22 @@
<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="DEC5 Volume" value="84" />
+ <ctl name="DEC6 Volume" value="84" />
+ <ctl name="DEC7 Volume" value="84" />
+ <ctl name="DEC8 Volume" value="84" />
+ <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="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" />
@@ -92,36 +108,46 @@
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia5" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia7" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia8" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia8" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia8" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia12" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia13" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_2_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="0" />
@@ -147,9 +173,13 @@
<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" />
@@ -289,12 +319,22 @@
<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" />
@@ -309,6 +349,16 @@
<ctl name="SpkrRight VISENSE Switch" value="0" />
<ctl name="SpkrLeft SWR DAC_Port Switch" value="0" />
<ctl name="SpkrRight SWR DAC_Port Switch" value="0" />
+
+ <ctl name="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" />
@@ -316,6 +366,10 @@
<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="AIF1_CAP Mixer SLIM TX0" value="0" />
@@ -565,6 +619,10 @@
<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" />
@@ -613,6 +671,10 @@
<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" />
@@ -661,6 +723,10 @@
<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" />
@@ -709,6 +775,10 @@
<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" />
@@ -757,6 +827,10 @@
<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" />
@@ -805,6 +879,10 @@
<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" />
@@ -853,6 +931,10 @@
<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" />
@@ -901,6 +983,10 @@
<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" />
@@ -949,6 +1035,10 @@
<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" />
@@ -1297,33 +1387,51 @@
<!-- 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">
@@ -1385,11 +1493,11 @@
</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">
@@ -1438,7 +1546,7 @@
</path>
<path name="handset-mic">
- <path name="dmic1" />
+ <path name="dmic3" />
</path>
<path name="handset-mic-db">
@@ -1452,6 +1560,19 @@
</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">
@@ -1461,11 +1582,38 @@
<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" />
</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">
@@ -1618,9 +1766,27 @@
<!-- 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">
@@ -1684,6 +1850,15 @@
</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">
@@ -1696,6 +1871,23 @@
<!-- 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">
diff --git a/configs/msmcobalt/msmcobalt.mk b/configs/msmcobalt/msmcobalt.mk
index 43aeb1a..9b74162 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
@@ -200,4 +200,9 @@
PRODUCT_PROPERTY_OVERRIDES += \
use.qti.sw.alac.decoder=true
PRODUCT_PROPERTY_OVERRIDES += \
-use.qti.sw.ape.decoder=true
\ No newline at end of file
+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..3c75b8e
--- /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" "DMIC0" />
+ <ctl name="MAD_SEL MUX" "SPE" />
+ <ctl name="MAD_INP MUX" "MAD" />
+ <ctl name="MAD_CPE1 Switch" 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..1f90bd5 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 +54,18 @@
<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>
+
+ <!-- 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 +73,8 @@
<!-- 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 adee78b..705e5e8 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -285,6 +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 += $(TARGET_OUT_HEADERS)/mm-audio/sound_trigger
@@ -307,6 +315,11 @@
LOCAL_COPY_HEADERS_TO := mm-audio
LOCAL_COPY_HEADERS := audio_extn/audio_defs.h
+ifeq ($(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/audio_extn.h b/hal/audio_extn/audio_extn.h
index d186a5f..0fd7d3a 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -401,6 +401,10 @@
#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)
@@ -419,8 +423,6 @@
#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);
@@ -586,4 +588,17 @@
#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/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/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 af399a1..b617407 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -82,6 +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
@@ -501,6 +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)
@@ -541,6 +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;
@@ -616,6 +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;
@@ -767,7 +802,8 @@
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;
}
@@ -824,6 +860,11 @@
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);
@@ -895,7 +936,9 @@
((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_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));
@@ -1166,6 +1209,28 @@
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)
{
@@ -2537,6 +2602,14 @@
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)
{
@@ -2555,6 +2628,20 @@
* 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;
@@ -3666,12 +3753,24 @@
__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);
@@ -3681,6 +3780,10 @@
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) {
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index ee28157..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 {
@@ -389,6 +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,
@@ -410,6 +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 738df09..9c6cc6f 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -2448,7 +2448,7 @@
return ret;
}
-static int platform_get_backend_index(snd_device_t snd_device)
+int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
@@ -5408,3 +5408,19 @@
}
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 dcd351a..6c89d0a 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -198,9 +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
@@ -208,6 +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,
@@ -356,7 +361,8 @@
enum {
LEGACY_PCM = 0,
PASSTHROUGH,
- PASSTHROUGH_CONVERT
+ PASSTHROUGH_CONVERT,
+ PASSTHROUGH_DSD
};
/*
* ID for setting mute and lateny on the device side
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 e42af8c..07060b6 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -112,6 +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 fc706f8..7d6f02b 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -247,6 +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] = {
@@ -334,6 +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",
@@ -451,6 +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,
@@ -568,6 +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)},
@@ -1102,11 +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");
@@ -1715,6 +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 =
@@ -1745,6 +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 =
@@ -1907,6 +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;
@@ -2092,7 +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__,
@@ -2107,6 +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;
@@ -2159,6 +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",
@@ -2238,7 +2298,7 @@
return ret;
}
-static int platform_get_backend_index(snd_device_t snd_device)
+int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
@@ -2247,6 +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;
@@ -2764,6 +2827,12 @@
} 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) {
@@ -4070,14 +4139,6 @@
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;
@@ -4096,14 +4157,24 @@
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;
@@ -4180,6 +4251,17 @@
}
}
+ 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;
}
@@ -4400,6 +4482,24 @@
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);
@@ -4440,6 +4540,17 @@
/*this is populated by check_codec_backend_cfg hence set default value to false*/
backend_cfg.passthrough_enabled = false;
+ /* 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,
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 48bfb2b..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,
@@ -192,13 +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,
@@ -447,7 +453,8 @@
enum {
LEGACY_PCM = 0,
PASSTHROUGH,
- PASSTHROUGH_CONVERT
+ PASSTHROUGH_CONVERT,
+ PASSTHROUGH_DSD
};
/*
* ID for setting mute and lateny on the device side
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