Merge "audio : Add g711 omx component for encoding"
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index 7e5c90f..b7a7a39 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -46,7 +46,7 @@
MM_AUDIO_ENABLED_SAFX := true
TARGET_USES_QCOM_MM_AUDIO := true
AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS := false
-#AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
+AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
##AUDIO_FEATURE_FLAGS
diff --git a/configs/msm8937/sound_trigger_mixer_paths.xml b/configs/msm8937/sound_trigger_mixer_paths.xml
index bbec875..a2ea69e 100644
--- a/configs/msm8937/sound_trigger_mixer_paths.xml
+++ b/configs/msm8937/sound_trigger_mixer_paths.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!--- Copyright (c) 2014, The Linux Foundation. All rights reserved. -->
+<!--- Copyright (c) 2014, 2016, The Linux Foundation. All rights reserved. -->
<!--- -->
<!--- Redistribution and use in source and binary forms, with or without -->
<!--- modification, are permitted provided that the following conditions are -->
@@ -28,48 +28,70 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="TERT_MI2S_TX LSM Function" value="None" />
<path name="listen-voice-wakeup-1">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM1 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM1 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM1 Mixer TERT_MI2S_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM2 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM2 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM2 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-3">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM3 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM3 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM3 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-4">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM4 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM4 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM4 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-5">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM5 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM5 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM5 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-6">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM6 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM6 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM6 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-7">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM7 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM7 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM7 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-8">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM8 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM8 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM8 Mixer TERT_MI2S_TX" value="1" />
</path>
<path name="listen-ape-handset-mic">
diff --git a/configs/msm8937/sound_trigger_mixer_paths_wcd9306.xml b/configs/msm8937/sound_trigger_mixer_paths_wcd9306.xml
index f2e4cb1..bd54837 100644
--- a/configs/msm8937/sound_trigger_mixer_paths_wcd9306.xml
+++ b/configs/msm8937/sound_trigger_mixer_paths_wcd9306.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!--- Copyright (c) 2014, The Linux Foundation. All rights reserved. -->
+<!--- Copyright (c) 2014, 2016, The Linux Foundation. All rights reserved. -->
<!--- -->
<!--- Redistribution and use in source and binary forms, with or without -->
<!--- modification, are permitted provided that the following conditions are -->
@@ -28,14 +28,22 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="SLIMBUS_5_TX LSM Function" value="None" />
<ctl name="AIF1_CAP Mixer SLIM TX1" value="0"/>
<ctl name="LOOPBACK Mode" value="DISABLE" />
@@ -46,51 +54,59 @@
<path name="listen-voice-wakeup-1">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM1 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM2 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-3">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM3 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-4">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM4 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-5">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM5 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-6">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM6 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-7">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM7 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-8">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM8 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-handset-mic">
diff --git a/configs/msm8937/sound_trigger_mixer_paths_wcd9330.xml b/configs/msm8937/sound_trigger_mixer_paths_wcd9330.xml
index 15f0e06..e4dee50 100644
--- a/configs/msm8937/sound_trigger_mixer_paths_wcd9330.xml
+++ b/configs/msm8937/sound_trigger_mixer_paths_wcd9330.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!--- Copyright (c) 2015, The Linux Foundation. All rights reserved.
+<!--- Copyright (c) 2015-2016, The Linux Foundation. All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
@@ -29,14 +29,22 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="SLIMBUS_5_TX LSM Function" value="None" />
<ctl name="MADONOFF Switch" value="0" />
<ctl name="MAD Input" value="DMIC1" />
@@ -44,42 +52,50 @@
<path name="listen-voice-wakeup-1">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-3">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-4">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-5">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-6">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-7">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-8">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-cpe-handset-mic">
<ctl name="MADONOFF Switch" value="1" />
<ctl name="MAD Input" value="DMIC1" />
- <ctl name="CPE AFE MAD Enable" value="1"/>
+ <ctl name="CPE AFE MAD Enable" value="1"/>
</path>
<path name="listen-ape-handset-mic">
diff --git a/configs/msm8937/sound_trigger_mixer_paths_wcd9335.xml b/configs/msm8937/sound_trigger_mixer_paths_wcd9335.xml
index 94d00c5..af630d0 100644
--- a/configs/msm8937/sound_trigger_mixer_paths_wcd9335.xml
+++ b/configs/msm8937/sound_trigger_mixer_paths_wcd9335.xml
@@ -29,14 +29,22 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="SLIMBUS_5_TX LSM Function" value="None" />
<ctl name="MADONOFF Switch" value="0" />
<ctl name="MAD Input" value="DMIC1" />
@@ -52,36 +60,44 @@
<path name="listen-voice-wakeup-1">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-3">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-4">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-5">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-6">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-7">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-8">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-cpe-handset-mic">
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index 2917f9d..3106942 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -46,7 +46,8 @@
MM_AUDIO_ENABLED_SAFX := true
TARGET_USES_QCOM_MM_AUDIO := true
AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS := false
-#AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
+AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
+
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
diff --git a/configs/msm8953/sound_trigger_mixer_paths.xml b/configs/msm8953/sound_trigger_mixer_paths.xml
index bbec875..a2ea69e 100644
--- a/configs/msm8953/sound_trigger_mixer_paths.xml
+++ b/configs/msm8953/sound_trigger_mixer_paths.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!--- Copyright (c) 2014, The Linux Foundation. All rights reserved. -->
+<!--- Copyright (c) 2014, 2016, The Linux Foundation. All rights reserved. -->
<!--- -->
<!--- Redistribution and use in source and binary forms, with or without -->
<!--- modification, are permitted provided that the following conditions are -->
@@ -28,48 +28,70 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="TERT_MI2S_TX LSM Function" value="None" />
<path name="listen-voice-wakeup-1">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM1 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM1 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM1 Mixer TERT_MI2S_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM2 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM2 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM2 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-3">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM3 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM3 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM3 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-4">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM4 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM4 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM4 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-5">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM5 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM5 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM5 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-6">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM6 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM6 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM6 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-7">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM7 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM7 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM7 Mixer TERT_MI2S_TX" value="1" />
</path>
+
<path name="listen-voice-wakeup-8">
<ctl name="TERT_MI2S_TX LSM Function" value="SWAUDIO" />
- <ctl name="LSM8 MUX" value="TERT_MI2S_TX" />
+ <ctl name="LSM8 Port" value="TERT_MI2S_TX" />
+ <ctl name="LSM8 Mixer TERT_MI2S_TX" value="1" />
</path>
<path name="listen-ape-handset-mic">
diff --git a/configs/msm8953/sound_trigger_mixer_paths_wcd9306.xml b/configs/msm8953/sound_trigger_mixer_paths_wcd9306.xml
index f2e4cb1..bd54837 100644
--- a/configs/msm8953/sound_trigger_mixer_paths_wcd9306.xml
+++ b/configs/msm8953/sound_trigger_mixer_paths_wcd9306.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!--- Copyright (c) 2014, The Linux Foundation. All rights reserved. -->
+<!--- Copyright (c) 2014, 2016, The Linux Foundation. All rights reserved. -->
<!--- -->
<!--- Redistribution and use in source and binary forms, with or without -->
<!--- modification, are permitted provided that the following conditions are -->
@@ -28,14 +28,22 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="SLIMBUS_5_TX LSM Function" value="None" />
<ctl name="AIF1_CAP Mixer SLIM TX1" value="0"/>
<ctl name="LOOPBACK Mode" value="DISABLE" />
@@ -46,51 +54,59 @@
<path name="listen-voice-wakeup-1">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM1 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM2 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-3">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM3 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-4">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM4 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-5">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM5 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-6">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM6 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-7">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM7 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-8">
<ctl name="AIF1_CAP Mixer SLIM TX1" value="1"/>
<ctl name="SLIM_0_TX Channels" value="One" />
- <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="SWAUDIO" />
+ <ctl name="LSM8 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-handset-mic">
diff --git a/configs/msm8953/sound_trigger_mixer_paths_wcd9330.xml b/configs/msm8953/sound_trigger_mixer_paths_wcd9330.xml
index b64c4ca..e4dee50 100644
--- a/configs/msm8953/sound_trigger_mixer_paths_wcd9330.xml
+++ b/configs/msm8953/sound_trigger_mixer_paths_wcd9330.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!--- Copyright (c) 2015, The Linux Foundation. All rights reserved.
+<!--- Copyright (c) 2015-2016, The Linux Foundation. All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
@@ -29,14 +29,22 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="SLIMBUS_5_TX LSM Function" value="None" />
<ctl name="MADONOFF Switch" value="0" />
<ctl name="MAD Input" value="DMIC1" />
@@ -44,36 +52,44 @@
<path name="listen-voice-wakeup-1">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-3">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-4">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-5">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-6">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-7">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-8">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-cpe-handset-mic">
diff --git a/configs/msm8953/sound_trigger_mixer_paths_wcd9335.xml b/configs/msm8953/sound_trigger_mixer_paths_wcd9335.xml
index 94d00c5..af630d0 100644
--- a/configs/msm8953/sound_trigger_mixer_paths_wcd9335.xml
+++ b/configs/msm8953/sound_trigger_mixer_paths_wcd9335.xml
@@ -29,14 +29,22 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="SLIMBUS_5_TX LSM Function" value="None" />
<ctl name="MADONOFF Switch" value="0" />
<ctl name="MAD Input" value="DMIC1" />
@@ -52,36 +60,44 @@
<path name="listen-voice-wakeup-1">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-3">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-4">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-5">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-6">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-7">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-8">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-cpe-handset-mic">
diff --git a/configs/msm8996/mixer_paths_tasha.xml b/configs/msm8996/mixer_paths_tasha.xml
index 9f63413..64d6bfb 100644
--- a/configs/msm8996/mixer_paths_tasha.xml
+++ b/configs/msm8996/mixer_paths_tasha.xml
@@ -2278,6 +2278,9 @@
<path name="capture-fm">
</path>
+ <path name="aanc-path">
+ </path>
+
<path name="aanc-handset-mic">
<ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
diff --git a/configs/msm8996/msm8996.mk b/configs/msm8996/msm8996.mk
index 306fa97..5b240e9 100644
--- a/configs/msm8996/msm8996.mk
+++ b/configs/msm8996/msm8996.mk
@@ -46,8 +46,8 @@
MM_AUDIO_ENABLED_SAFX := true
TARGET_USES_QCOM_MM_AUDIO := true
AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS := false
-#AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
-#DOLBY_DDP := true
+AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
+
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
diff --git a/configs/msm8996/sound_trigger_mixer_paths.xml b/configs/msm8996/sound_trigger_mixer_paths.xml
index 0125371..f6c99d3 100644
--- a/configs/msm8996/sound_trigger_mixer_paths.xml
+++ b/configs/msm8996/sound_trigger_mixer_paths.xml
@@ -28,14 +28,22 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" value="None" />
<ctl name="SLIMBUS_5_TX LSM Function" value="None" />
<ctl name="MADONOFF Switch" value="0" />
<ctl name="MAD Input" value="DMIC1" />
@@ -51,36 +59,44 @@
<path name="listen-voice-wakeup-1">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-3">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-4">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-5">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-6">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-7">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-8">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
- <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-cpe-handset-mic">
diff --git a/configs/msm8996/sound_trigger_mixer_paths_wcd9330.xml b/configs/msm8996/sound_trigger_mixer_paths_wcd9330.xml
index 4f8557d..2ad8750 100644
--- a/configs/msm8996/sound_trigger_mixer_paths_wcd9330.xml
+++ b/configs/msm8996/sound_trigger_mixer_paths_wcd9330.xml
@@ -28,51 +28,67 @@
<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="LSM1 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="0" />
+ <ctl name="LSM1 Port" value="None" />
+ <ctl name="LSM2 Port" value="None" />
+ <ctl name="LSM3 Port" value="None" />
+ <ctl name="LSM4 Port" value="None" />
+ <ctl name="LSM5 Port" value="None" />
+ <ctl name="LSM6 Port" value="None" />
+ <ctl name="LSM7 Port" value="None" />
+ <ctl name="LSM8 Port" 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="CPE AFE MAD Enable" value="0"/>
<path name="listen-voice-wakeup-1">
- <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM1 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM1 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-2">
- <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM2 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM2 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-3">
- <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM3 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM3 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-4">
- <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM4 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM4 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-5">
- <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM5 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM5 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-6">
- <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM6 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM6 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-7">
- <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM7 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM7 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-voice-wakeup-8">
- <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM8 Port" value="SLIMBUS_5_TX" />
+ <ctl name="LSM8 Mixer SLIMBUS_5_TX" value="1" />
</path>
<path name="listen-cpe-handset-mic">
diff --git a/configs/msmcobalt/audio_output_policy.conf b/configs/msmcobalt/audio_output_policy.conf
index e60c664..8213f92 100644
--- a/configs/msmcobalt/audio_output_policy.conf
+++ b/configs/msmcobalt/audio_output_policy.conf
@@ -34,18 +34,25 @@
}
direct_pcm_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
- formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT|AUDIO_FORMAT_PCM_32_BIT
sampling_rates 44100|48000|96000|192000
bit_width 16
app_type 69936
}
direct_pcm_24 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
- formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
- sampling_rates 44100|48000|96000|176400|192000|352800
+ formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT|AUDIO_FORMAT_PCM_32_BIT
+ sampling_rates 44100|48000|96000|176400|192000|352800|384000
bit_width 24
app_type 69940
}
+ direct_pcm_32 {
+ flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
+ formats AUDIO_FORMAT_PCM_32_BIT
+ sampling_rates 44100|48000|96000|176400|192000|352800|384000
+ bit_width 32
+ app_type 69942
+ }
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|AUDIO_FORMAT_DSD
diff --git a/configs/msmcobalt/audio_platform_info.xml b/configs/msmcobalt/audio_platform_info.xml
index 07839fd..a1bd9a1 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -67,6 +67,7 @@
<param key="perf_lock_opts" value="4, 0x101, 0x704, 0x20F, 0x1E01"/>
<param key="native_audio_mode" value="src"/>
<param key="input_mic_max_count" value="4"/>
+ <param key="true_32_bit" value="true"/>
<!-- In the below value string, the value indicates sidetone gain in dB -->
<param key="usb_sidetone_gain" value="35"/>
</config_params>
diff --git a/configs/msmcobalt/audio_policy.conf b/configs/msmcobalt/audio_policy.conf
index 70ab311..8fb5676 100644
--- a/configs/msmcobalt/audio_policy.conf
+++ b/configs/msmcobalt/audio_policy.conf
@@ -58,9 +58,9 @@
flags AUDIO_OUTPUT_FLAG_DIRECT
}
direct_pcm {
- sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000|352800
+ sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000|352800|384000
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
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT|AUDIO_FORMAT_PCM_32_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|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
}
@@ -102,7 +102,7 @@
}
surround_sound {
sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000
- channel_masks AUDIO_CHANNEL_IN_5POINT1|AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK|AUDIO_CHANNEL_INDEX_MASK_3|AUDIO_CHANNEL_INDEX_MASK_4
+ channel_masks AUDIO_CHANNEL_IN_5POINT1|AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK|AUDIO_CHANNEL_INDEX_MASK_3|AUDIO_CHANNEL_INDEX_MASK_4|AUDIO_CHANNEL_INDEX_MASK_6
formats AUDIO_FORMAT_PCM_16_BIT
devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC
}
diff --git a/configs/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index 66b7d17..b6d2490 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -89,10 +89,13 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800,384000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800,384000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800,384000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
@@ -176,7 +179,7 @@
<mixPort name="surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3,AUDIO_CHANNEL_INDEX_MASK_4,AUDIO_CHANNEL_IN_5POINT1"/>
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK,AUDIO_CHANNEL_INDEX_MASK_3,AUDIO_CHANNEL_INDEX_MASK_4,AUDIO_CHANNEL_IN_5POINT1,AUDIO_CHANNEL_INDEX_MASK_6"/>
</mixPort>
<mixPort name="record_24" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
diff --git a/configs/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index efd275d..23c0f06 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -2293,6 +2293,12 @@
<path name="capture-fm">
</path>
+ <path name="aanc-path">
+ <ctl name="ADC MUX10" value="DMIC" />
+ <ctl name="DMIC MUX10" value="DMIC4" />
+ <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
+ </path>
+
<path name="aanc-handset-mic">
<ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
@@ -2309,9 +2315,6 @@
<ctl name="ADC MUX7" value="DMIC" />
<ctl name="DMIC MUX7" value="DMIC0" />
<ctl name="IIR0 INP0 MUX" value="DEC6" />
- <ctl name="ADC MUX10" value="DMIC" />
- <ctl name="DMIC MUX10" value="DMIC4" />
- <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
</path>
<!-- Dual MIC devices -->
diff --git a/configs/msmcobalt/mixer_paths_tavil.xml b/configs/msmcobalt/mixer_paths_tavil.xml
index 29212f9..555555d 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -2068,6 +2068,12 @@
<path name="capture-fm">
</path>
+ <path name="aanc-path">
+ <ctl name="ADC MUX10" value="DMIC" />
+ <ctl name="DMIC MUX10" value="DMIC4" />
+ <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
+ </path>
+
<path name="aanc-handset-mic">
<ctl name="AIF1_CAP Mixer SLIM TX6" value="1" />
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
@@ -2084,9 +2090,6 @@
<ctl name="ADC MUX7" value="DMIC" />
<ctl name="DMIC MUX7" value="DMIC0" />
<ctl name="IIR0 INP0 MUX" value="DEC6" />
- <ctl name="ADC MUX10" value="DMIC" />
- <ctl name="DMIC MUX10" value="DMIC4" />
- <ctl name="ANC0 FB MUX" value="ANC_IN_EAR_SPKR" />
</path>
<!-- Dual MIC devices -->
diff --git a/configs/msmcobalt/msmcobalt.mk b/configs/msmcobalt/msmcobalt.mk
index 9aa0322..b7684ed 100644
--- a/configs/msmcobalt/msmcobalt.mk
+++ b/configs/msmcobalt/msmcobalt.mk
@@ -22,8 +22,7 @@
AUDIO_FEATURE_ENABLED_HDMI_PASSTHROUGH := true
#AUDIO_FEATURE_ENABLED_KEEP_ALIVE := true
AUDIO_FEATURE_ENABLED_DISPLAY_PORT := true
-#AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
-#DOLBY_DDP := true
+AUDIO_FEATURE_ENABLED_DS2_DOLBY_DAP := true
AUDIO_FEATURE_ENABLED_HFP := true
AUDIO_FEATURE_ENABLED_INCALL_MUSIC := false
AUDIO_FEATURE_ENABLED_MULTI_VOICE_SESSIONS := true
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
index e72cb76..28d0f75 100644
--- a/hal/audio_extn/a2dp.c
+++ b/hal/audio_extn/a2dp.c
@@ -645,6 +645,35 @@
return ret;
}
+static void reset_a2dp_enc_config_params()
+{
+ int ret =0;
+
+ struct mixer_ctl *ctl_enc_config, *ctrl_bit_format;
+ struct sbc_enc_cfg_t dummy_reset_config;
+
+ memset(&dummy_reset_config, 0x0, sizeof(struct sbc_enc_cfg_t));
+ ctl_enc_config = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ENC_CONFIG_BLOCK);
+ if (!ctl_enc_config) {
+ ALOGE(" ERROR a2dp encoder format mixer control not identifed");
+ } else {
+ ret = mixer_ctl_set_array(ctl_enc_config, (void *)&dummy_reset_config,
+ sizeof(struct sbc_enc_cfg_t));
+ a2dp.bt_encoder_format = ENC_MEDIA_FMT_NONE;
+ }
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ENC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ ALOGE(" ERROR bit format CONFIG data mixer control not identifed");
+ } else {
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set bit format to encoder", __func__);
+ }
+ }
+}
+
int audio_extn_a2dp_stop_playback()
{
int ret =0;
@@ -659,35 +688,13 @@
a2dp.a2dp_total_active_session_request--;
if ( a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
- struct mixer_ctl *ctl_enc_config, *ctrl_bit_format;
- struct sbc_enc_cfg_t dummy_reset_config;
-
ALOGV("calling BT module stream stop");
ret = a2dp.audio_stop_stream();
if (ret < 0)
ALOGE("stop stream to BT IPC lib failed");
else
ALOGV("stop steam to BT IPC lib successful");
- memset(&dummy_reset_config, 0x0, sizeof(struct sbc_enc_cfg_t));
- ctl_enc_config = mixer_get_ctl_by_name(a2dp.adev->mixer,
- MIXER_ENC_CONFIG_BLOCK);
- if (!ctl_enc_config) {
- ALOGE(" ERROR a2dp encoder format mixer control not identifed");
- } else {
- ret = mixer_ctl_set_array(ctl_enc_config, (void *)&dummy_reset_config,
- sizeof(struct sbc_enc_cfg_t));
- a2dp.bt_encoder_format = ENC_MEDIA_FMT_NONE;
- }
- ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
- MIXER_ENC_BIT_FORMAT);
- if (!ctrl_bit_format) {
- ALOGE(" ERROR bit format CONFIG data mixer control not identifed");
- } else {
- ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
- if (ret != 0) {
- ALOGE("%s: Failed to set bit format to encoder", __func__);
- }
- }
+ reset_a2dp_enc_config_params();
}
if(!a2dp.a2dp_total_active_session_request)
a2dp.a2dp_started = false;
@@ -724,6 +731,7 @@
val = atoi(value);
if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
ALOGV("Received device dis- connect request");
+ reset_a2dp_enc_config_params();
close_a2dp_output();
}
goto param_handled;
@@ -735,6 +743,7 @@
if ((!strncmp(value,"true",sizeof(value)))) {
ALOGD("Setting a2dp to suspend state");
a2dp.a2dp_suspended = true;
+ reset_a2dp_enc_config_params();
if(a2dp.audio_suspend_stream)
a2dp.audio_suspend_stream();
} else if (a2dp.a2dp_suspended == true) {
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 083b925..0f38b82 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -185,7 +185,7 @@
return ret;
}
-static void check_and_set_ext_disp_connection_status(const struct audio_device *adev,
+static void audio_extn_ext_disp_set_parameters(const struct audio_device *adev,
struct str_parms *parms)
{
char value[32] = {0};
@@ -204,13 +204,14 @@
&& (atoi(value) & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
//params = "disconnect=1024" for external display disconnection.
update_ext_disp_sysfs_node(adev, 0);
+ ALOGV("invalidate cached edid");
+ platform_invalidate_hdmi_config(adev->platform);
} else {
// handle ext disp devices only
return;
}
}
-
#ifndef FM_POWER_OPT
#define audio_extn_fm_set_parameters(adev, parms) (0)
#else
@@ -772,7 +773,7 @@
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_ext_disp_connection_status(adev, parms);
+ audio_extn_ext_disp_set_parameters(adev, parms);
if (adev->offload_effects_set_parameters != NULL)
adev->offload_effects_set_parameters(parms);
}
@@ -1144,3 +1145,52 @@
}
}
#endif /* KPI_OPTIMIZE_ENABLED */
+
+static int audio_extn_set_multichannel_mask(struct audio_device *adev,
+ struct stream_in *in,
+ struct audio_config *config,
+ bool *channel_mask_updated)
+{
+ int ret = -EINVAL;
+ int channel_count = audio_channel_count_from_in_mask(in->channel_mask);
+ *channel_mask_updated = false;
+
+ int max_mic_count = platform_get_max_mic_count(adev->platform);
+ /* validate input params*/
+ if ((channel_count == 6) &&
+ (in->format == AUDIO_FORMAT_PCM_16_BIT)) {
+
+ switch (max_mic_count) {
+ case 4:
+ config->channel_mask = AUDIO_CHANNEL_INDEX_MASK_4;
+ break;
+ case 3:
+ config->channel_mask = AUDIO_CHANNEL_INDEX_MASK_3;
+ break;
+ case 2:
+ config->channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ break;
+ default:
+ config->channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ break;
+ }
+ ret = 0;
+ *channel_mask_updated = true;
+ }
+ return ret;
+}
+
+int audio_extn_check_and_set_multichannel_usecase(struct audio_device *adev,
+ struct stream_in *in,
+ struct audio_config *config,
+ bool *update_params)
+{
+ bool ssr_supported = false;
+ ssr_supported = audio_extn_ssr_check_usecase(in);
+ if (ssr_supported) {
+ return audio_extn_ssr_set_usecase(in, config, update_params);
+ } else {
+ return audio_extn_set_multichannel_mask(adev, in, config,
+ update_params);
+ }
+}
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index cd9763e..07714f6 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -194,17 +194,21 @@
#endif
#ifndef SSR_ENABLED
-#define audio_extn_ssr_check_and_set_usecase(in) (-1)
-#define audio_extn_ssr_init(in, num_out_chan) (0)
-#define audio_extn_ssr_deinit() (0)
-#define audio_extn_ssr_update_enabled() (0)
-#define audio_extn_ssr_get_enabled() (0)
-#define audio_extn_ssr_read(stream, buffer, bytes) (0)
-#define audio_extn_ssr_set_parameters(adev, parms) (0)
-#define audio_extn_ssr_get_parameters(adev, parms, reply) (0)
-#define audio_extn_ssr_get_stream() (0)
+#define audio_extn_ssr_check_usecase(in) (0)
+#define audio_extn_ssr_set_usecase(in, config, channel_mask_updated) (0)
+#define audio_extn_ssr_init(in, num_out_chan) (0)
+#define audio_extn_ssr_deinit() (0)
+#define audio_extn_ssr_update_enabled() (0)
+#define audio_extn_ssr_get_enabled() (0)
+#define audio_extn_ssr_read(stream, buffer, bytes) (0)
+#define audio_extn_ssr_set_parameters(adev, parms) (0)
+#define audio_extn_ssr_get_parameters(adev, parms, reply) (0)
+#define audio_extn_ssr_get_stream() (0)
#else
-int audio_extn_ssr_check_and_set_usecase(struct stream_in *in);
+bool audio_extn_ssr_check_usecase(struct stream_in *in);
+int audio_extn_ssr_set_usecase(struct stream_in *in,
+ struct audio_config *config,
+ bool *channel_mask_updated);
int32_t audio_extn_ssr_init(struct stream_in *in,
int num_out_chan);
int32_t audio_extn_ssr_deinit();
@@ -219,6 +223,10 @@
struct str_parms *reply);
struct stream_in *audio_extn_ssr_get_stream();
#endif
+int audio_extn_check_and_set_multichannel_usecase(struct audio_device *adev,
+ struct stream_in *in,
+ struct audio_config *config,
+ bool *update_params);
#ifndef HW_VARIANTS_ENABLED
#define hw_info_init(snd_card_name) (0)
diff --git a/hal/audio_extn/dolby.c b/hal/audio_extn/dolby.c
index f07c66a..b958bf6 100644
--- a/hal/audio_extn/dolby.c
+++ b/hal/audio_extn/dolby.c
@@ -484,9 +484,7 @@
};
int audio_extn_dap_hal_init(int snd_card) {
- char c_dmid[128] = {0};
- void *handle = NULL;
- int i_dmid, ret = -EINVAL;
+ int ret = -EINVAL;
dap_hal_device_be_id_map_t device_be_id_map;
ALOGV("%s: opening DAP HAL lib\n", __func__);
@@ -532,9 +530,7 @@
void audio_extn_dolby_ds2_set_endpoint(struct audio_device *adev) {
struct listnode *node;
struct audio_usecase *usecase;
- struct mixer_ctl *ctl;
- const char *mixer_ctl_name = "DS1 DAP Endpoint";
- int endpoint = 0, ret;
+ int endpoint = 0;
bool send = false;
list_for_each(node, &adev->usecase_list) {
@@ -587,7 +583,7 @@
return 0;
}
-int audio_extn_dolby_set_dap_bypass(struct audio_device *adev, int state) {
+int audio_extn_dolby_set_dap_bypass(struct audio_device *adev __unused, int state) {
ALOGV("%s: state %d", __func__, state);
if (ds2extnmod.dap_hal_set_hw_info) {
@@ -599,12 +595,12 @@
return 0;
}
-void audio_extn_dolby_set_license(struct audio_device *adev)
+void audio_extn_dolby_set_license(struct audio_device *adev __unused)
{
int i_key=0;
char c_key[128] = {0};
char c_dmid[128] = {0};
- int i_dmid, ret = -EINVAL;
+ int i_dmid;
struct dolby_param_license dolby_license;
#ifdef DOLBY_ACDB_LICENSE
@@ -631,7 +627,7 @@
void audio_extn_ds2_set_parameters(struct audio_device *adev,
struct str_parms *parms)
{
- int val, ret;
+ int ret;
char value[32]={0};
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SND_CARD_STATUS, value,
diff --git a/hal/audio_extn/ssr.c b/hal/audio_extn/ssr.c
index f55f3ce..51a6a26 100644
--- a/hal/audio_extn/ssr.c
+++ b/hal/audio_extn/ssr.c
@@ -331,33 +331,55 @@
return false;
}
-int audio_extn_ssr_check_and_set_usecase(struct stream_in *in)
-{
- int ret = -1;
+bool audio_extn_ssr_check_usecase(struct stream_in *in) {
+ int ret = false;
int channel_count = audio_channel_count_from_in_mask(in->channel_mask);
audio_devices_t devices = in->device;
audio_source_t source = in->source;
- /* validate input params
- * only stereo and 5:1 channel config is supported
- * only AUDIO_DEVICE_IN_BUILTIN_MIC, AUDIO_DEVICE_IN_BACK_MIC supports 3 mics */
- if (audio_extn_ssr_get_enabled() &&
- ((channel_count == 2) || (channel_count == 6)) &&
- ((AUDIO_SOURCE_MIC == source) || (AUDIO_SOURCE_CAMCORDER == source)) &&
- ((AUDIO_DEVICE_IN_BUILTIN_MIC == devices) || (AUDIO_DEVICE_IN_BACK_MIC == devices)) &&
- (in->format == AUDIO_FORMAT_PCM_16_BIT)) {
-
- ALOGD("%s: Found SSR use case starting SSR lib with channel_count :%d",
+ if ((audio_extn_ssr_get_enabled()) &&
+ ((channel_count == 2) || (channel_count == 6)) &&
+ ((AUDIO_SOURCE_MIC == source) || (AUDIO_SOURCE_CAMCORDER == source)) &&
+ ((AUDIO_DEVICE_IN_BUILTIN_MIC == devices) || (AUDIO_DEVICE_IN_BACK_MIC == devices)) &&
+ (in->format == AUDIO_FORMAT_PCM_16_BIT)) {
+ ALOGD("%s: SSR enabled with channel_count :%d",
__func__, channel_count);
+ ret = true;
+ }
+ return ret;
+}
- if (!audio_extn_ssr_init(in, channel_count)) {
- ALOGD("%s: Created SSR session succesfully", __func__);
+int audio_extn_ssr_set_usecase(struct stream_in *in,
+ struct audio_config *config,
+ bool *update_params)
+{
+ int ret = -EINVAL;
+ int channel_count = audio_channel_count_from_in_mask(in->channel_mask);
+ audio_channel_representation_t representation =
+ audio_channel_mask_get_representation(in->channel_mask);
+ *update_params = false;
+
+ if (audio_extn_ssr_check_usecase(in)) {
+
+ if (representation == AUDIO_CHANNEL_REPRESENTATION_INDEX) {
+ /* update params in case channel representation index.
+ * on returning error, flinger will retry with supported representation passed
+ */
+ ALOGD("%s: SSR supports only channel representation position, channel_mask(%#x)"
+ ,__func__, config->channel_mask);
+ config->channel_mask = AUDIO_CHANNEL_IN_5POINT1;
ret = 0;
+ *update_params = true;
} else {
- ALOGE("%s: Unable to start SSR record session", __func__);
+ if (!audio_extn_ssr_init(in, channel_count)) {
+ ALOGD("%s: Created SSR session succesfully", __func__);
+ ret = 0;
+ } else {
+ ALOGE("%s: Unable to start SSR record session", __func__);
+ }
}
- }
- return ret;
+ }
+ return ret;
}
static void pcm_buffer_queue_push(struct pcm_buffer_queue **queue,
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 3816748..3a12adf 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -1206,6 +1206,19 @@
return NULL;
}
+struct stream_in *get_next_active_input(const struct audio_device *adev)
+{
+ struct audio_usecase *usecase;
+ struct listnode *node;
+
+ list_for_each_reverse(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == PCM_CAPTURE)
+ return usecase->stream.in;
+ }
+ return NULL;
+}
+
/*
* is a true native playback active
*/
@@ -1424,6 +1437,10 @@
if (voice_is_call_state_active(adev) ||
voice_extn_compress_voip_is_started(adev))
voice_set_sidetone(adev, usecase->out_snd_device, false);
+
+ /* Disable aanc only if voice call exists */
+ if (voice_is_call_state_active(adev))
+ voice_check_and_update_aanc_path(adev, usecase->out_snd_device, false);
}
/* Disable current sound devices */
@@ -1494,6 +1511,10 @@
enable_audio_route(adev, usecase);
if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
+ /* Enable aanc only if voice call exists */
+ if (voice_is_call_state_active(adev))
+ voice_check_and_update_aanc_path(adev, out_snd_device, true);
+
/* Enable sidetone only if other voice/voip call already exists */
if (voice_is_call_state_active(adev) ||
voice_extn_compress_voip_is_started(adev))
@@ -1519,7 +1540,7 @@
struct audio_usecase *uc_info;
struct audio_device *adev = in->dev;
- adev->active_input = NULL;
+ adev->active_input = get_next_active_input(adev);
ALOGV("%s: enter: usecase(%d: %s)", __func__,
in->usecase, use_case_table[in->usecase]);
@@ -1666,7 +1687,7 @@
audio_extn_perf_lock_release(&adev->perf_lock_handle);
stop_input_stream(in);
error_config:
- adev->active_input = NULL;
+ adev->active_input = get_next_active_input(adev);
/*
* sleep 50ms to allow sufficient time for kernel
* drivers to recover incases like SSR.
@@ -2452,7 +2473,15 @@
(val == AUDIO_DEVICE_NONE)) {
val = AUDIO_DEVICE_OUT_SPEAKER;
}
-
+ /* To avoid a2dp to sco overlapping force route BT usecases
+ * to speaker based on Phone state
+ */
+ if ((val & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) &&
+ ((adev->mode == AUDIO_MODE_RINGTONE) ||
+ (adev->mode == AUDIO_MODE_IN_CALL))) {
+ ALOGD("Forcing a2dp routing to speaker for ring/call mode");
+ val = AUDIO_DEVICE_OUT_SPEAKER;
+ }
/*
* select_devices() call below switches all the usecases on the same
* backend to the new device. Refer to check_usecases_codec_backend() in
@@ -4251,10 +4280,13 @@
ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
- if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
- ALOGV("invalidate cached edid");
- platform_invalidate_hdmi_config(adev->platform);
- } else if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
+ /*
+ * The HDMI / Displayport disconnect handling has been moved to
+ * audio extension to ensure that its parameters are not
+ * invalidated prior to updating sysfs of the disconnect event
+ * Invalidate will be handled by audio_extn_ext_disp_set_parameters()
+ */
+ if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
!(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) {
ret = str_parms_get_str(parms, "card", value, sizeof(value));
if (ret >= 0) {
@@ -4440,6 +4472,7 @@
int ret = 0, buffer_size, frame_size;
int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
bool is_low_latency = false;
+ bool channel_mask_updated = false;
*stream_in = NULL;
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) {
@@ -4534,7 +4567,14 @@
in->config.channels = channel_count;
in->config.rate = config->sample_rate;
in->sample_rate = config->sample_rate;
- } else if (!audio_extn_ssr_check_and_set_usecase(in)) {
+ } else if (!audio_extn_check_and_set_multichannel_usecase(adev,
+ in, config, &channel_mask_updated)) {
+ if (channel_mask_updated == true) {
+ ALOGD("%s: return error to retry with updated channel mask (%#x)",
+ __func__, config->channel_mask);
+ ret = -EINVAL;
+ goto err_open;
+ }
ALOGD("%s: created surround sound session succesfully",__func__);
} else if (audio_extn_compr_cap_enabled() &&
audio_extn_compr_cap_format_supported(config->format) &&
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index a5cc804..a42f984 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -780,8 +780,8 @@
static int msm_device_to_be_id_external_codec [][NO_COLS] = {
{AUDIO_DEVICE_OUT_EARPIECE , 2},
{AUDIO_DEVICE_OUT_SPEAKER , 2},
- {AUDIO_DEVICE_OUT_WIRED_HEADSET , 2},
- {AUDIO_DEVICE_OUT_WIRED_HEADPHONE , 2},
+ {AUDIO_DEVICE_OUT_WIRED_HEADSET , 41},
+ {AUDIO_DEVICE_OUT_WIRED_HEADPHONE , 41},
{AUDIO_DEVICE_OUT_BLUETOOTH_SCO , 11},
{AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET , 11},
{AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT , 11},
@@ -2924,6 +2924,7 @@
*num_devices = 2;
new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
+ status = true;
}
ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
@@ -5622,6 +5623,14 @@
return 0;
}
+void platform_update_aanc_path(struct audio_device *adev __unused,
+ snd_device_t out_snd_device __unused,
+ bool enable __unused,
+ char *str __unused)
+{
+ return;
+}
+
bool platform_check_codec_dsd_support(void *platform __unused)
{
return false;
@@ -5671,3 +5680,8 @@
{
return -ENOSYS;
}
+
+int platform_get_max_mic_count(void *platform) {
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->max_mic_count;
+}
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index b5a4f11..e025772 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1315,6 +1315,14 @@
return 0;
}
+void platform_update_aanc_path(struct audio_device *adev __unused,
+ snd_device_t out_snd_device __unused,
+ bool enable __unused,
+ char *str __unused)
+{
+ return;
+}
+
bool platform_check_codec_dsd_support(void *platform __unused)
{
return false;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index fa67342..e947f91 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -795,8 +795,8 @@
static int msm_device_to_be_id [][NO_COLS] = {
{AUDIO_DEVICE_OUT_EARPIECE , 2},
{AUDIO_DEVICE_OUT_SPEAKER , 2},
- {AUDIO_DEVICE_OUT_WIRED_HEADSET , 2},
- {AUDIO_DEVICE_OUT_WIRED_HEADPHONE , 2},
+ {AUDIO_DEVICE_OUT_WIRED_HEADSET , 41},
+ {AUDIO_DEVICE_OUT_WIRED_HEADPHONE , 41},
{AUDIO_DEVICE_OUT_BLUETOOTH_SCO , 11},
{AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET , 11},
{AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT , 11},
@@ -2742,9 +2742,9 @@
*num_devices = 2;
new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
+ status = true;
}
-
ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
snd_device, *num_devices, *new_snd_devices);
@@ -4697,10 +4697,31 @@
ALOGD("%s:becf: afe: true napb active set rate to 44.1 khz",
__func__);
}
- } else if ((OUTPUT_SAMPLING_RATE_44100 == sample_rate) &&
- (na_mode != NATIVE_AUDIO_MODE_MULTIPLE_44_1)) {
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: napb not active - set (48k) default rate",
+ } else if (na_mode != NATIVE_AUDIO_MODE_MULTIPLE_44_1) {
+ /*
+ * Map native sampling rates to upper limit range
+ * if multiple of native sampling rates are not supported.
+ * This check also indicates that this is not tavil codec
+ * And 32bit/384kHz is only supported on tavil
+ * Hence reset 32b/384kHz to 24b/192kHz.
+ */
+ switch (sample_rate) {
+ case 44100:
+ sample_rate = 48000;
+ break;
+ case 88200:
+ sample_rate = 96000;
+ break;
+ case 176400:
+ case 352800:
+ case 384000:
+ sample_rate = 192000;
+ break;
+ }
+ if (bit_width > 24)
+ bit_width = 24;
+
+ ALOGD("%s:becf: afe: napb not active - set non fractional rate",
__func__);
}
} else if ((usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) ||
@@ -4751,23 +4772,6 @@
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);
@@ -5627,10 +5631,14 @@
{
int ret;
if (out_snd_device == SND_DEVICE_OUT_USB_HEADSET) {
+ if (property_get_bool("audio.debug.usb.disable_sidetone", 0)) {
+ ALOGI("Debug: Disable sidetone");
+ } else {
ret = audio_extn_usb_enable_sidetone(out_snd_device, enable);
if (ret)
ALOGI("%s: usb device %d does not support device sidetone\n",
__func__, out_snd_device);
+ }
} else {
ALOGV("%s: sidetone out device(%d) mixer cmd = %s\n",
__func__, out_snd_device, str);
@@ -5643,6 +5651,22 @@
return 0;
}
+void platform_update_aanc_path(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable,
+ char *str)
+{
+ ALOGD("%s: aanc out device(%d) mixer cmd = %s, enable = %d\n",
+ __func__, out_snd_device, str, enable);
+
+ if (enable)
+ audio_route_apply_and_update_path(adev->audio_route, str);
+ else
+ audio_route_reset_and_update_path(adev->audio_route, str);
+
+ return;
+}
+
static void make_cal_cfg(acdb_audio_cal_cfg_t* cal, int acdb_dev_id,
int acdb_device_type, int app_type, int topology_id,
int sample_rate, uint32_t module_id, uint32_t param_id, bool persist)
@@ -5792,3 +5816,8 @@
ERROR_RETURN:
return ret;
}
+
+int platform_get_max_mic_count(void *platform) {
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->max_mic_count;
+}
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 7dcd1b6..61f42de 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -157,12 +157,17 @@
snd_device_t out_snd_device,
bool enable,
char * str);
+void platform_update_aanc_path(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable,
+ char * str);
bool platform_supports_true_32bit();
bool platform_check_if_backend_has_to_be_disabled(snd_device_t new_snd_device, snd_device_t cuurent_snd_device);
bool platform_check_codec_dsd_support(void *platform);
bool platform_check_codec_asrc_support(void *platform);
int platform_get_backend_index(snd_device_t snd_device);
int platform_get_ext_disp_type(void *platform);
+void platform_invalidate_hdmi_config(void *platform);
int platform_send_audio_cal(void* platform, int acdb_dev_id, int acdb_device_type,
int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
@@ -181,4 +186,5 @@
void* data, int* length);
unsigned char* platform_get_license(void* platform, int* size);
+int platform_get_max_mic_count(void *platform);
#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/voice.c b/hal/voice.c
index b84c7b7..ca3098b 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -95,6 +95,39 @@
return;
}
+static bool voice_is_aanc_device(snd_device_t out_device,
+ char *mixer_path)
+{
+ bool is_aanc_dev;
+
+ switch (out_device) {
+ case SND_DEVICE_OUT_ANC_HANDSET:
+ is_aanc_dev = true;
+ strlcpy(mixer_path, "aanc-path", MIXER_PATH_MAX_LENGTH);
+ break;
+ default:
+ is_aanc_dev = false;
+ break;
+ }
+
+ return is_aanc_dev;
+}
+
+void voice_check_and_update_aanc_path(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable)
+{
+ char mixer_path[MIXER_PATH_MAX_LENGTH];
+
+ ALOGV("%s: %s, out_snd_device: %d\n",
+ __func__, (enable ? "enable" : "disable"), out_snd_device);
+
+ if (voice_is_aanc_device(out_snd_device, mixer_path))
+ platform_update_aanc_path(adev, out_snd_device, enable, mixer_path);
+
+ return;
+}
+
int voice_stop_usecase(struct audio_device *adev, audio_usecase_t usecase_id)
{
int ret = 0;
@@ -125,6 +158,10 @@
if (!voice_is_call_state_active(adev))
voice_set_sidetone(adev, uc_info->out_snd_device, false);
+ /* Disable aanc only when no calls are active */
+ if (!voice_is_call_state_active(adev))
+ voice_check_and_update_aanc_path(adev, uc_info->out_snd_device, false);
+
ret = platform_stop_voice_call(adev->platform, session->vsid);
/* 1. Close the PCM devices */
@@ -229,6 +266,10 @@
pcm_start(session->pcm_tx);
pcm_start(session->pcm_rx);
+ /* Enable aanc only when no calls are active */
+ if (!voice_is_call_state_active(adev))
+ voice_check_and_update_aanc_path(adev, uc_info->out_snd_device, true);
+
/* Enable sidetone only when no calls are already active */
if (!voice_is_call_state_active(adev))
voice_set_sidetone(adev, uc_info->out_snd_device, true);
diff --git a/hal/voice.h b/hal/voice.h
index efe48d8..3ae42a8 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -97,5 +97,8 @@
void voice_set_sidetone(struct audio_device *adev,
snd_device_t out_snd_device,
bool enable);
+void voice_check_and_update_aanc_path(struct audio_device *adev,
+ snd_device_t out_snd_device,
+ bool enable);
bool voice_is_call_state_active(struct audio_device *adev);
#endif //VOICE_H
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
index 854eaee..f9913c4 100644
--- a/policy_hal/Android.mk
+++ b/policy_hal/Android.mk
@@ -1,3 +1,16 @@
+# This file was modified by Dolby Laboratories, Inc. The portions of the
+# code that are surrounded by "DOLBY..." are copyrighted and
+# licensed separately, as follows:
+#
+# (C) 2016 Dolby Laboratories, Inc.
+# All rights reserved.
+#
+# This program is protected under international and U.S. Copyright laws as
+# an unpublished work. This program is confidential and proprietary to the
+# copyright owners. Reproduction or disclosure, in whole or in part, or the
+# production of derivative works therefrom without the express permission of
+# the copyright owners is prohibited.
+#
ifneq ($(USE_LEGACY_AUDIO_POLICY), 1)
ifeq ($(USE_CUSTOM_AUDIO_POLICY), 1)
LOCAL_PATH := $(call my-dir)
@@ -64,6 +77,11 @@
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FM_POWER_OPT)),true)
LOCAL_CFLAGS += -DFM_POWER_OPT
endif
+# DOLBY_START
+ifeq ($(strip $(DOLBY_ENABLE)),true)
+LOCAL_CFLAGS += $(dolby_cflags)
+endif
+# DOLBY_END
ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
LOCAL_CFLAGS += -DUSE_XML_AUDIO_POLICY_CONF
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index d06929c..8bfea9a 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -2231,40 +2231,6 @@
ALOGD("USE_XML_AUDIO_POLICY_CONF is FALSE");
#endif
- //TODO: Check the new logic to parse policy conf and update the below code
- // Need this when SSR encoding is enabled
- char ssr_enabled[PROPERTY_VALUE_MAX] = {0};
- bool prop_ssr_enabled = false;
-
- if (property_get("ro.qc.sdk.audio.ssr", ssr_enabled, NULL)) {
- prop_ssr_enabled = atoi(ssr_enabled) || !strncmp("true", ssr_enabled, 4);
- }
-
- for (size_t i = 0; i < mHwModules.size(); i++) {
- ALOGV("Hw module %zu", i);
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
- const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
- AudioProfileVector profiles = inProfile->getAudioProfiles();
- for (size_t k = 0; k < profiles.size(); k++){
- ChannelsVector channels = profiles[k]->getChannels();
- for (size_t x = 0; x < channels.size(); x++) {
- audio_channel_mask_t channelMask = channels[x];
- ALOGV("Channel Mask %x size %zu", channelMask,
- channels.size());
- if (AUDIO_CHANNEL_IN_5POINT1 == channelMask) {
- if (!prop_ssr_enabled) {
- ALOGI("removing AUDIO_CHANNEL_IN_5POINT1 from"
- " input profile as SSR(surround sound record)"
- " is not supported on this chipset variant");
- channels.removeItemsAt(x, 1);
- ALOGV("Channel Mask size now %zu",
- channels.size());
- }
- }
- }
- }
- }
- }
#ifdef RECORD_PLAY_CONCURRENCY
mIsInputRequestOnProgress = false;
#endif
diff --git a/post_proc/volume_listener.c b/post_proc/volume_listener.c
index e1dd026..7b60248 100644
--- a/post_proc/volume_listener.c
+++ b/post_proc/volume_listener.c
@@ -697,24 +697,31 @@
struct listnode *node = NULL;
vol_listener_context_t *context = NULL;
vol_listener_context_t *recv_contex = (vol_listener_context_t *)handle;
- int status = -1;
+ int status = -EINVAL;
bool recompute_flag = false;
int active_stream_count = 0;
+ uint32_t session_id;
+ uint32_t stream_type;
+ effect_uuid_t uuid;
+
ALOGV("%s context %p", __func__, handle);
- if (recv_contex == NULL || recv_contex->desc == NULL) {
- ALOGE("%s: Got invalid handle while release, DO NOTHING ", __func__);
+
+ if (recv_contex == NULL) {
return status;
}
-
pthread_mutex_lock(&vol_listner_init_lock);
+ session_id = recv_contex->session_id;
+ stream_type = recv_contex->stream_type;
+ uuid = recv_contex->desc->uuid;
// check if the handle/context provided is valid
list_for_each(node, &vol_effect_list) {
context = node_to_item(node, struct vol_listener_context_s, effect_list_node);
- if ((memcmp(&(context->desc->uuid), &(recv_contex->desc->uuid), sizeof(effect_uuid_t)) == 0)
- && (context->session_id == recv_contex->session_id)
- && (context->stream_type == recv_contex->stream_type)) {
+ if ((memcmp(&(context->desc->uuid), &uuid, sizeof(effect_uuid_t)) == 0)
+ && (context->session_id == session_id)
+ && (context->stream_type == stream_type)) {
ALOGV("--- Found something to remove ---");
+ list_remove(node);
PRINT_STREAM_TYPE(context->stream_type);
if (context->dev_id == AUDIO_DEVICE_OUT_SPEAKER) {
recompute_flag = true;
@@ -730,6 +737,8 @@
if (status != 0) {
ALOGE("something wrong ... <<<--- Found NOTHING to remove ... ???? --->>>>>");
+ pthread_mutex_unlock(&vol_listner_init_lock);
+ return status;
}
// if there are no active streams, reset cal and volume level
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
index 8426945..f334719 100644
--- a/qahw_api/test/qahw_playback_test.c
+++ b/qahw_api/test/qahw_playback_test.c
@@ -18,188 +18,471 @@
/* Test app to play audio at the HAL layer */
+#include <getopt.h>
+#include <pthread.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include "qahw_api.h"
#include "qahw_defs.h"
+
#define nullptr NULL
-#define WAV 1
-#define MP3 2
+FILE * log_file = NULL;
+const char *log_filename = NULL;
+float vol_level = 0.01;
+
+enum {
+ FILE_WAV = 1,
+ FILE_MP3,
+ FILE_AAC,
+ FILE_AAC_ADTS
+};
+
+typedef enum {
+ AAC_LC = 1,
+ AAC_HE_V1,
+ AAC_HE_V2
+} aac_format_type_t;
+
+static pthread_mutex_t write_lock = PTHREAD_MUTEX_INITIALIZER;
+static pthread_cond_t write_cond = PTHREAD_COND_INITIALIZER;
+static pthread_mutex_t drain_lock = PTHREAD_MUTEX_INITIALIZER;
+static pthread_cond_t drain_cond = PTHREAD_COND_INITIALIZER;
+
+
+int async_callback(qahw_stream_callback_event_t event, void *param,
+ void *cookie)
+{
+ switch (event) {
+ case QAHW_STREAM_CBK_EVENT_WRITE_READY:
+ fprintf(log_file, "QAHW_STREAM_CBK_EVENT_DRAIN_READY\n");
+ pthread_mutex_lock(&write_lock);
+ pthread_cond_signal(&write_cond);
+ pthread_mutex_unlock(&write_lock);
+ break;
+ case QAHW_STREAM_CBK_EVENT_DRAIN_READY:
+ fprintf(log_file, "QAHW_STREAM_CBK_EVENT_DRAIN_READY\n");
+ pthread_mutex_lock(&drain_lock);
+ pthread_cond_signal(&drain_cond);
+ pthread_mutex_unlock(&drain_lock);
+ default:
+ break;
+ }
+ return 0;
+}
+
+
+int write_to_hal(qahw_stream_handle_t* out_handle, char *data,
+ size_t bytes)
+{
+ ssize_t ret;
+ pthread_mutex_lock(&write_lock);
+ qahw_out_buffer_t out_buf;
+
+ memset(&out_buf,0, sizeof(qahw_out_buffer_t));
+ out_buf.buffer = data;
+ out_buf.bytes = bytes;
+
+ ret = qahw_out_write(out_handle, &out_buf);
+ if (ret < 0 || ret == bytes) {
+ fprintf(log_file, "Writing data to hal failed or full write %ld, %ld\n",
+ ret, bytes);
+ } else if (ret != bytes) {
+ fprintf(log_file, "ret %ld, bytes %ld\n", ret, bytes);
+ fprintf(log_file, "Waiting for event write ready\n");
+ pthread_cond_wait(&write_cond, &write_lock);
+ fprintf(log_file, "out of wait for event write ready\n");
+ }
+
+ pthread_mutex_unlock(&write_lock);
+ return ret;
+}
/* Play audio from a WAV file.
+ *
+ * Parameters:
+ * out_stream: A pointer to the output audio stream.
+ * in_file: A pointer to a SNDFILE object.
+ * config: A pointer to struct that contains audio configuration data.
+ *
+ * Returns: An int which has a non-negative number on success.
+ */
- Parameters:
- out_stream: A pointer to the output audio stream.
- in_file: A pointer to a SNDFILE object.
- config: A pointer to struct that contains audio configuration data.
+int play_file(qahw_stream_handle_t* out_handle, FILE* in_file,
+ bool is_offload) {
+ int rc = 0;
+ int offset = 0;
+ size_t bytes_wanted = 0;
+ size_t write_length = 0;
+ size_t bytes_remaining = 0;
+ size_t bytes_written = 0;
+ size_t bytes_read = 0;
+ char *data = NULL;
+ qahw_out_buffer_t out_buf;
+ bool exit = false;
- Returns: An int which has a non-negative number on success.
-*/
+ if (is_offload) {
+ fprintf(log_file, "Set callback for offload stream\n");
+ qahw_out_set_callback(out_handle, async_callback, NULL);
+ }
-int play_file(qahw_stream_handle_t* out_handle, FILE* in_file) {
+ rc = qahw_out_set_volume(out_handle, vol_level, vol_level);
+ if (rc < 0)
+ fprintf(log_file, "unable to set volume");
- int rc = 0;
- size_t frames_read = 1;
- size_t bytes_wanted ;
- char *data = NULL;
- qahw_out_buffer_t out_buf;
+ bytes_wanted = qahw_out_get_buffer_size(out_handle);
+ data = (char *) malloc (bytes_wanted);
+ if (data == NULL) {
+ fprintf(log_file, "calloc failed!!\n");
+ return -ENOMEM;
+ }
- bytes_wanted = qahw_out_get_buffer_size(out_handle);
- data = (char *) malloc (bytes_wanted);
- if (data == NULL) {
- printf("calloc failed!!\n");
- return -ENOMEM;
- }
+ while (!exit) {
+ if (!bytes_remaining) {
+ bytes_read = fread(data, 1, bytes_wanted, in_file);
+ fprintf(log_file, "fread from file %ld\n", bytes_read);
+ if (bytes_read <= 0) {
+ if (feof(in_file)) {
+ fprintf(log_file, "End of file");
+ if (is_offload) {
+ pthread_mutex_lock(&drain_lock);
+ if (is_offload) {
+ qahw_out_drain(out_handle, QAHW_DRAIN_ALL);
+ pthread_cond_wait(&drain_cond, &drain_lock);
+ fprintf(log_file, "Out of compress drain\n");
+ }
+ pthread_mutex_unlock(&drain_lock);
+ }
+ } else {
+ fprintf(log_file, "Error in fread --%d\n", ferror(in_file));
+ fprintf(stderr, "Error in fread --%d\n", ferror(in_file));
+ }
+ exit = true;
+ continue;
+ }
+ bytes_remaining = write_length = bytes_read;
+ }
- while(frames_read != 0) {
- frames_read = fread(data, bytes_wanted , 1, in_file);
- if (frames_read < 1) {
- if (feof(in_file))
- break;
- else
- printf("Error in fread --%d\n",ferror(in_file));
- }
- memset(&out_buf,0, sizeof(qahw_out_buffer_t));
- out_buf.buffer = data;
- out_buf.bytes = frames_read * bytes_wanted;
- rc = qahw_out_write(out_handle, &out_buf);
- if (rc < 0) {
- printf("Writing data to hal failed %d \n",rc);
- break;
- }
- }
- return rc;
+ offset = write_length - bytes_remaining;
+ fprintf(log_file, "bytes_remaining %ld, offset %d, write length %ld\n",
+ bytes_remaining, offset, write_length);
+ bytes_written = write_to_hal(out_handle, data+offset, bytes_remaining);
+ bytes_remaining -= bytes_written;
+ fprintf(log_file, "bytes_written %ld, bytes_remaining %ld\n",
+ bytes_written, bytes_remaining);
+ }
+
+ return rc;
}
-// Prints usage information if input arguments are missing.
-void Usage() {
- fprintf(stderr, "Usage:hal_play [device] [filename] [filetype]\n"
- "device: hex value representing the audio device (see "
- "system/media/audio/include/system/audio.h)\n"
- "filename must be passed as an argument.\n"
- "filetype (1:WAV 2:MP3) \n");
+bool is_valid_aac_format_type(aac_format_type_t format_type)
+{
+ bool valid_format_type = false;
+
+ switch (format_type) {
+ case AAC_LC:
+ case AAC_HE_V1:
+ case AAC_HE_V2:
+ valid_format_type = true;
+ break;
+ default:
+ break;
+ }
+ return valid_format_type;
+}
+
+/*
+ * Obtain aac format (refer audio.h) for format type entered.
+ */
+
+audio_format_t get_aac_format(int filetype, aac_format_type_t format_type)
+{
+ audio_format_t aac_format = AUDIO_FORMAT_AAC_ADTS_LC; /* default aac frmt*/
+
+ if (filetype == FILE_AAC_ADTS) {
+ switch (format_type) {
+ case AAC_LC:
+ aac_format = AUDIO_FORMAT_AAC_ADTS_LC;
+ break;
+ case AAC_HE_V1:
+ aac_format = AUDIO_FORMAT_AAC_ADTS_HE_V1;
+ break;
+ case AAC_HE_V2:
+ aac_format = AUDIO_FORMAT_AAC_ADTS_HE_V2;
+ break;
+ default:
+ break;
+ }
+ } else if (filetype == FILE_AAC) {
+ switch (format_type) {
+ case AAC_LC:
+ aac_format = AUDIO_FORMAT_AAC_LC;
+ break;
+ case AAC_HE_V1:
+ aac_format = AUDIO_FORMAT_AAC_HE_V1;
+ break;
+ case AAC_HE_V2:
+ aac_format = AUDIO_FORMAT_AAC_HE_V2;
+ break;
+ default:
+ break;
+ }
+ } else {
+ fprintf(log_file, "Invalid filetype provided %d\n", filetype);
+ fprintf(stderr, "Invalid filetype provided %d\n", filetype);
+ }
+
+ fprintf(log_file, "aac format %d\n", aac_format);
+ return aac_format;
+}
+
+void usage() {
+ printf(" \n Command \n");
+ printf(" \n hal_play_test <file path> - path of file to be played\n");
+ printf(" \n Options\n");
+ printf(" -r --sample-rate <sampling rate> - Required for Non-WAV streams\n");
+ printf(" For AAC-HE pls specify half the sample rate\n\n");
+ printf(" -c --channel count <channels> - Required for Non-WAV streams\n\n");
+ printf(" -v --volume <float volume level> - Volume level float value between 0.0 - 1.0.\n");
+ printf(" -d --device <decimal value> - see system/media/audio/include/system/audio.h for device values\n");
+ printf(" Optional Argument and Default value is 2, i.e Speaker\n\n");
+ printf(" -t --file-type <file type> - 1:WAV 2:MP3 3:AAC 4:AAC_ADTS\n");
+ printf(" Required for non WAV formats\n\n");
+ printf(" -a --aac-type <aac type> - Required for AAC streams\n");
+ printf(" 1: LC 2: HE_V1 3: HE_V2\n\n");
+ printf(" -l --log-file <FILEPATH> - File path for debug msg, to print\n");
+ printf(" on console use stdout or 1 \n\n");
+ printf(" \n Examples \n");
+ printf(" hal_play_test /etc/Anukoledenadu.wav -> plays Wav stream with default params\n\n");
+ printf(" hal_play_test /etc/MateRani.mp3 -t 2 -d 2 -v 0.01 -r 44100 -c 2 \n");
+ printf(" -> plays MP3 stream(-t = 2) on speaker device(-d = 2)\n");
+ printf(" -> 2 channels and 44100 sample rate\n\n");
+ printf(" hal_play_test /etc/AACLC-71-48000Hz-384000bps.aac -t 4 -d 2 -v 0.05 -r 48000 -c 2 -a 1 \n");
+ printf(" -> plays AAC-ADTS stream(-t = 4) on speaker device(-d = 2)\n");
+ printf(" -> AAC format type is LC(-a = 1)\n");
+ printf(" -> 2 channels and 48000 sample rate\n\n");
+ printf(" hal_play_test /etc/AACHE-adts-stereo-32000KHz-128000Kbps.aac -t 4 -d 2 -v 0.05 -r 16000 -c 2 -a 3 \n");
+ printf(" -> plays AAC-ADTS stream(-t = 4) on speaker device(-d = 2)\n");
+ printf(" -> AAC format type is HE V2(-a = 3)\n");
+ printf(" -> 2 channels and 16000 sample rate\n");
+ printf(" -> note that the sample rate is half the actual sample rate\n\n");
}
int main(int argc, char* argv[]) {
- if (argc < 4) {
- Usage();
- return -1;
- }
- // Process command line arguments.
- FILE *filestream = NULL;
- char header[44] = {0};
- int sample_rate = 0;
- int channels = 0;
- const int audio_device_base = 16;
- char* filename = nullptr;
- int filetype;
- qahw_module_handle_t *qahw_mod_handle;
- const char *mod_name = "audio.primary";
- uint32_t desired_output_device = strtol(
- argv[1], nullptr /* look at full string*/, audio_device_base);
+ FILE *file_stream = NULL;
+ char header[44] = {0};
+ char* filename = nullptr;
+ qahw_module_handle_t *qahw_mod_handle;
+ const char *mod_name = "audio.primary";
+ qahw_stream_handle_t* out_handle = nullptr;
+ int rc = 0;
- filename = argv[2];
- filetype = atoi (argv[3]);
+ /*
+ * Default values
+ */
+ int filetype = FILE_WAV;
+ int sample_rate = 44100;
+ int channels = 2;
+ const int audio_device_base = 0x2;/* spkr device*/
+ aac_format_type_t format_type = AAC_LC;
+ log_file = stdout;
+ audio_devices_t output_device = AUDIO_DEVICE_OUT_SPEAKER;
- printf("Starting audio hal tests.\n");
- int rc = 0;
+ struct option long_options[] = {
+ /* These options set a flag. */
+ {"device", required_argument, 0, 'd'},
+ {"sample-rate", required_argument, 0, 'r'},
+ {"channels", required_argument, 0, 'c'},
+ {"volume", required_argument, 0, 'v'},
+ {"log-file", required_argument, 0, 'l'},
+ {"file-type", required_argument, 0, 't'},
+ {"aac-type", required_argument, 0, 'a'},
+ {"help", no_argument, 0, 'h'},
+ {0, 0, 0, 0}
+ };
- qahw_mod_handle = qahw_load_module(mod_name);
+ int opt = 0;
+ int option_index = 0;
+ while ((opt = getopt_long(argc,
+ argv,
+ "-r:c:d:v:l::t:a:h",
+ long_options,
+ &option_index)) != -1) {
+ switch (opt) {
+ case 'r':
+ sample_rate = atoi(optarg);
+ break;
+ case 'c':;
+ channels = atoi(optarg);
+ break;
+ case 'd':
+ output_device = atoi(optarg);
+ break;
+ case 'v':
+ vol_level = atof(optarg);
+ break;
+ case 'l':
+ /*
+ * Fix Me: unable to log to a given file.
+ */
+ log_filename = optarg;
+ if((log_file = fopen(log_filename,"wb"))== NULL) {
+ fprintf(stderr, "Cannot open log file %s\n", log_filename);
+ /*
+ * continue to log to std out.
+ */
+ log_file = stdout;
+ }
- // Set to a high number so it doesn't interfere with existing stream handles
- audio_io_handle_t handle = 0x999;
- audio_devices_t output_device =
- (audio_devices_t)desired_output_device;
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
- audio_config_t config;
+ break;
+ case 't':
+ filetype = atoi(optarg);
+ break;
+ case 'a':
+ format_type = atoi(optarg);
+ break;
+ case 'h':
+ usage();
+ return 0;
+ break;
+ }
+ }
- memset(&config, 0, sizeof(audio_config_t));
+ filename = argv[1];
+ if((file_stream = fopen(filename, "r"))== NULL) {
+ fprintf(stderr, "Cannot Open Audio File %s\n", filename);
+ goto EXIT;
+ }
- if (filename) {
- printf("filename-----%s\n",filename);
- filestream = fopen (filename,"r");
- if (filestream == NULL) {
- printf("failed to open\n");
- exit(0);
- }
- }
+ /*
+ * Set to a high number so it doesn't interfere with existing stream handles
+ */
- switch (filetype) {
- case WAV:
- //Read the wave header
- rc = fread (header, 44 , 1, filestream);
- if (rc != 1) {
- printf("Error .Fread failed\n");
- exit(0);
- }
- if (strncmp (header,"RIFF",4) && strncmp (header+8, "WAVE",4)) {
- printf("Not a wave format\n");
- exit (1);
- }
- memcpy (&channels, &header[22], 2);
- memcpy (&sample_rate, &header[24], 4);
- config.channel_mask = audio_channel_out_mask_from_count(channels);
- config.offload_info.channel_mask = config.channel_mask;
- config.offload_info.sample_rate = sample_rate;
- config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT;
- break;
- case MP3:
- printf("Enter Number of channels:");
- scanf ("%d",&channels);
- config.channel_mask = audio_channel_out_mask_from_count(channels);
- printf("\nEnter Sample Rate:");
- scanf ("%d",&sample_rate);
- config.offload_info.channel_mask = config.channel_mask;
- config.offload_info.sample_rate = sample_rate;
- config.offload_info.format = AUDIO_FORMAT_MP3;
- break;
- default:
- printf("Does not support given filetype\n");
- Usage();
- exit (0);
- }
- config.offload_info.version = AUDIO_OFFLOAD_INFO_VERSION_CURRENT;
- config.offload_info.size = sizeof(audio_offload_info_t);
+ audio_io_handle_t handle = 0x999;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
- printf("Now playing to output_device=%d sample_rate=%d \n",output_device,
- config.offload_info.sample_rate);
- const char* stream_name = "output_stream";
+ fprintf(stdout, "Playing:%s\n", filename);
+ fprintf(stdout, "File Type:%d\n", filetype);
+ fprintf(stdout, "Sample Rate:%d\n", sample_rate);
+ fprintf(stdout, "Channels:%d\n", channels);
+ fprintf(stdout, "Log file:%s\n", log_filename);
+ fprintf(stdout, "Volume level:%f\n", vol_level);
+ fprintf(stdout, "Output Device:%d\n", output_device);
+ fprintf(stdout, "Format Type:%d\n", format_type);
- // Open audio output stream.
- qahw_stream_handle_t* out_handle = nullptr;
- printf("calling open_out_put_stream:\n");
- rc = qahw_open_output_stream(qahw_mod_handle, handle, output_device,
- flags, &config, &out_handle,
- stream_name);
- printf("open output stream is sucess:%d out_handhle %p\n",rc,out_handle);
- if (rc) {
- printf("could not open output stream %d \n",rc);
- return -1;
- }
+ fprintf(stdout, "Starting audio hal tests.\n");
- play_file(out_handle, filestream);
+ qahw_mod_handle = qahw_load_module(mod_name);
- // Close output stream and device.
- rc = qahw_out_standby(out_handle);
- if (rc) {
- printf("out standby failed %d \n",rc);
- }
+ audio_config_t config;
+ memset(&config, 0, sizeof(audio_config_t));
- rc = qahw_close_output_stream(out_handle);
- if (rc) {
- printf("could not close output stream %d \n",rc);
- }
+ switch (filetype) {
+ case FILE_WAV:
+ /*
+ * Read the wave header
+ */
+ rc = fread (header, 44 , 1, file_stream);
+ if (rc != 1) {
+ fprintf(stdout, "Error .Fread failed\n");
+ exit(0);
+ }
+ if (strncmp (header, "RIFF", 4) && strncmp (header+8, "WAVE", 4)) {
+ fprintf(stdout, "Not a wave format\n");
+ exit (1);
+ }
+ memcpy (&channels, &header[22], 2);
+ memcpy (&sample_rate, &header[24], 4);
+ config.channel_mask = audio_channel_out_mask_from_count(channels);
+ config.offload_info.channel_mask = config.channel_mask;
+ config.offload_info.sample_rate = sample_rate;
+ config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT;
+ break;
- rc = qahw_unload_module(qahw_mod_handle);
- if (rc) {
- printf("could not unload hal %d \n",rc);
- return -1;
- }
+ case FILE_MP3:
+ config.channel_mask = audio_channel_out_mask_from_count(channels);
+ config.offload_info.channel_mask = config.channel_mask;
+ config.sample_rate = sample_rate;
+ config.offload_info.sample_rate = sample_rate;
+ config.offload_info.format = AUDIO_FORMAT_MP3;
+ flags |= AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+ break;
- printf("Done with hal tests \n");
- return 0;
+ case FILE_AAC:
+ case FILE_AAC_ADTS:
+ config.channel_mask = audio_channel_out_mask_from_count(channels);
+ config.offload_info.channel_mask = config.channel_mask;
+ config.sample_rate = sample_rate;
+ config.offload_info.sample_rate = sample_rate;
+ if (!is_valid_aac_format_type(format_type)) {
+ fprintf(log_file, "Invalid format type for AAC %d\n", format_type);
+ goto EXIT;
+ }
+ config.offload_info.format = get_aac_format(filetype, format_type);
+ flags |= AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+ break;
+
+
+ default:
+ fprintf(stderr, "Does not support given filetype\n");
+ usage();
+ return 0;
+ }
+ config.offload_info.version = AUDIO_OFFLOAD_INFO_VERSION_CURRENT;
+ config.offload_info.size = sizeof(audio_offload_info_t);
+
+ fprintf(log_file, "Now playing to output_device=%d sample_rate=%d \n"
+ , output_device, config.offload_info.sample_rate);
+ const char* stream_name = "output_stream";
+
+ fprintf(log_file, "calling open_out_put_stream:\n");
+ rc = qahw_open_output_stream(qahw_mod_handle,
+ handle,
+ output_device,
+ flags,
+ &config,
+ &out_handle,
+ stream_name);
+ fprintf(log_file, "open output stream is sucess:%d out_handhle %p\n"
+ , rc, out_handle);
+ if (rc) {
+ fprintf(stdout, "could not open output stream %d \n", rc);
+ goto EXIT;
+ }
+
+ play_file(out_handle,
+ file_stream,
+ (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD));
+
+EXIT:
+
+ if (out_handle != nullptr) {
+ rc = qahw_out_standby(out_handle);
+ if (rc) {
+ fprintf(stdout, "out standby failed %d \n", rc);
+ }
+
+ rc = qahw_close_output_stream(out_handle);
+ if (rc) {
+ fprintf(stdout, "could not close output stream %d \n", rc);
+ }
+
+ rc = qahw_unload_module(qahw_mod_handle);
+ if (rc) {
+ fprintf(stdout, "could not unload hal %d \n", rc);
+ return -1;
+ }
+ }
+
+ if ((log_file != stdout) && (log_file != nullptr))
+ fclose(log_file);
+
+ if (file_stream != nullptr)
+ fclose(file_stream);
+
+ fprintf(stdout, "\nBYE BYE\n");
+ return 0;
}