Merge "hal: Update to make code more reliable" into av-userspace.lnx.2.0-dev
diff --git a/configs/msm8937/audio_output_policy.conf b/configs/msm8937/audio_output_policy.conf
index 8f70451..3d6b978 100644
--- a/configs/msm8937/audio_output_policy.conf
+++ b/configs/msm8937/audio_output_policy.conf
@@ -32,23 +32,30 @@
bit_width 16
app_type 69936
}
- direct_pcm {
+ direct_pcm_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
- formats AUDIO_FORMAT_PCM_16_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_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|192000
+ bit_width 24
+ app_type 69940
+ }
compress_offload_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
- formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+ formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
sampling_rates 44100|48000|96000|192000
bit_width 16
app_type 69936
}
compress_offload_24 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
- formats AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS
+ formats AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS
sampling_rates 44100|48000|96000|192000
bit_width 24
app_type 69940
diff --git a/configs/msm8937/audio_policy.conf b/configs/msm8937/audio_policy.conf
index f69cfa5..3e59ba2 100644
--- a/configs/msm8937/audio_policy.conf
+++ b/configs/msm8937/audio_policy.conf
@@ -57,14 +57,14 @@
direct_pcm {
sampling_rates 8000|11025|16000|22050|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
- formats AUDIO_FORMAT_PCM_16_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_AUX_DIGITAL
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
}
compress_offload {
sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
- formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+ formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
}
diff --git a/configs/msm8953/audio_output_policy.conf b/configs/msm8953/audio_output_policy.conf
index 8f70451..3d6b978 100644
--- a/configs/msm8953/audio_output_policy.conf
+++ b/configs/msm8953/audio_output_policy.conf
@@ -32,23 +32,30 @@
bit_width 16
app_type 69936
}
- direct_pcm {
+ direct_pcm_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
- formats AUDIO_FORMAT_PCM_16_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_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|192000
+ bit_width 24
+ app_type 69940
+ }
compress_offload_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
- formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+ formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
sampling_rates 44100|48000|96000|192000
bit_width 16
app_type 69936
}
compress_offload_24 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
- formats AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS
+ formats AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS
sampling_rates 44100|48000|96000|192000
bit_width 24
app_type 69940
diff --git a/configs/msm8953/audio_policy.conf b/configs/msm8953/audio_policy.conf
index f69cfa5..3e59ba2 100644
--- a/configs/msm8953/audio_policy.conf
+++ b/configs/msm8953/audio_policy.conf
@@ -57,14 +57,14 @@
direct_pcm {
sampling_rates 8000|11025|16000|22050|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
- formats AUDIO_FORMAT_PCM_16_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY|AUDIO_DEVICE_OUT_AUX_DIGITAL
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
}
compress_offload {
sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
- formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+ formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_AC3|AUDIO_FORMAT_E_AC3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
}
diff --git a/configs/msm8996/audio_output_policy.conf b/configs/msm8996/audio_output_policy.conf
index 2f01bc9..93cd0c2 100644
--- a/configs/msm8996/audio_output_policy.conf
+++ b/configs/msm8996/audio_output_policy.conf
@@ -32,13 +32,20 @@
bit_width 16
app_type 69936
}
- direct_pcm {
+ direct_pcm_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
- formats AUDIO_FORMAT_PCM_16_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_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|192000
+ bit_width 24
+ app_type 69940
+ }
compress_passthrough {
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
@@ -48,14 +55,14 @@
}
compress_offload_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
- formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+ formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
sampling_rates 44100|48000|96000|192000
bit_width 16
app_type 69936
}
compress_offload_24 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
- formats AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO
+ formats AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO
sampling_rates 44100|48000|96000|192000
bit_width 24
app_type 69940
diff --git a/configs/msm8996/audio_policy.conf b/configs/msm8996/audio_policy.conf
index 453aca4..8a3bd30 100644
--- a/configs/msm8996/audio_policy.conf
+++ b/configs/msm8996/audio_policy.conf
@@ -58,16 +58,16 @@
flags AUDIO_OUTPUT_FLAG_DIRECT
}
direct_pcm {
- sampling_rates 44100|48000|64000|88200|96000|176400|192000
+ sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
- formats AUDIO_FORMAT_PCM_16_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
}
compress_offload {
sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
- formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+ formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
}
@@ -99,18 +99,18 @@
formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_AMR_NB|AUDIO_FORMAT_AMR_WB|AUDIO_FORMAT_QCELP|AUDIO_FORMAT_EVRC|AUDIO_FORMAT_EVRCB|AUDIO_FORMAT_EVRCWB|AUDIO_FORMAT_EVRCNW
devices AUDIO_DEVICE_IN_WIRED_HEADSET|AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET|AUDIO_DEVICE_IN_FM_TUNER|AUDIO_DEVICE_IN_VOICE_CALL
}
- float {
- sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|96000|192000
- channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK
- formats AUDIO_FORMAT_PCM_24_BIT_PACKED
- devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC|AUDIO_DEVICE_IN_WIRED_HEADSET
- }
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
formats AUDIO_FORMAT_PCM_16_BIT
devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC
}
+ record_24 {
+ sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|96000|192000
+ channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK
+ formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT|AUDIO_FORMAT_PCM_FLOAT
+ devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC|AUDIO_DEVICE_IN_WIRED_HEADSET
+ }
voice_rx {
sampling_rates 8000|16000|48000
channel_masks AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_MONO
diff --git a/configs/msm8996/audio_policy_configuration.xml b/configs/msm8996/audio_policy_configuration.xml
index d0c4cdd..3f30e32 100644
--- a/configs/msm8996/audio_policy_configuration.xml
+++ b/configs/msm8996/audio_policy_configuration.xml
@@ -152,6 +152,17 @@
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_IN_5POINT1"/>
</mixPort>
+ <mixPort name="record_24" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </mixPort>
<mixPort name="voice_rx" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
@@ -268,6 +279,8 @@
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
+ <route type="mix" sink="record_24"
+ sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
<route type="mix" sink="Telephony Tx"
sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
<route type="mix" sink="voice_rx"
diff --git a/configs/msm8996/msm8996.mk b/configs/msm8996/msm8996.mk
index 23b9213..2b70a81 100644
--- a/configs/msm8996/msm8996.mk
+++ b/configs/msm8996/msm8996.mk
@@ -189,3 +189,9 @@
#enable hw aac encoder by default
PRODUCT_PROPERTY_OVERRIDES += \
qcom.hw.aac.encoder=true
+
+#enable software decoders for ALAC and APE
+PRODUCT_PROPERTY_OVERRIDES += \
+use.qti.sw.alac.decoder=true
+PRODUCT_PROPERTY_OVERRIDES += \
+use.qti.sw.ape.decoder=true
diff --git a/configs/msmcobalt/audio_output_policy.conf b/configs/msmcobalt/audio_output_policy.conf
index cbfd43e..46fa191 100644
--- a/configs/msmcobalt/audio_output_policy.conf
+++ b/configs/msmcobalt/audio_output_policy.conf
@@ -32,13 +32,20 @@
bit_width 16
app_type 69936
}
- direct_pcm {
+ direct_pcm_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
- formats AUDIO_FORMAT_PCM_16_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_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|192000
+ bit_width 24
+ app_type 69940
+ }
compress_offload_16 {
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
diff --git a/configs/msmcobalt/audio_platform_info.xml b/configs/msmcobalt/audio_platform_info.xml
index acc2e4e..6bf14e2 100644
--- a/configs/msmcobalt/audio_platform_info.xml
+++ b/configs/msmcobalt/audio_platform_info.xml
@@ -67,5 +67,15 @@
<!-- followed by perf lock options -->
<param key="perf_lock_opts" value="4, 0x101, 0x704, 0x20F, 0x1E01"/>
</config_params>
+ <backend_names>
+ <device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_SPEAKER_AND_LINE" backend="speaker-and-headphones" interface="SLIMBUS_0_RX-and-SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+ </backend_names>
</audio_platform_info>
diff --git a/configs/msmcobalt/audio_policy.conf b/configs/msmcobalt/audio_policy.conf
index b32bce6..a9f4644 100644
--- a/configs/msmcobalt/audio_policy.conf
+++ b/configs/msmcobalt/audio_policy.conf
@@ -51,16 +51,16 @@
flags AUDIO_OUTPUT_FLAG_DIRECT
}
direct_pcm {
- sampling_rates 44100|48000|64000|88200|96000|176400|192000
+ sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
- formats AUDIO_FORMAT_PCM_16_BIT
+ formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_PROXY
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM
}
compress_offload {
sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|64000|88200|96000|176400|192000
channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_2POINT1|AUDIO_CHANNEL_OUT_QUAD|AUDIO_CHANNEL_OUT_PENTA|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_6POINT1|AUDIO_CHANNEL_OUT_7POINT1
- formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_PCM_16_BIT_OFFLOAD|AUDIO_FORMAT_PCM_24_BIT_OFFLOAD|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
+ formats AUDIO_FORMAT_MP3|AUDIO_FORMAT_FLAC|AUDIO_FORMAT_ALAC|AUDIO_FORMAT_APE|AUDIO_FORMAT_AAC_LC|AUDIO_FORMAT_AAC_HE_V1|AUDIO_FORMAT_AAC_HE_V2|AUDIO_FORMAT_WMA|AUDIO_FORMAT_WMA_PRO|AUDIO_FORMAT_VORBIS|AUDIO_FORMAT_AAC_ADTS_LC|AUDIO_FORMAT_AAC_ADTS_HE_V1|AUDIO_FORMAT_AAC_ADTS_HE_V2
devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_LINE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_PROXY
flags AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING
}
@@ -98,6 +98,12 @@
formats AUDIO_FORMAT_PCM_16_BIT
devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC
}
+ record_24 {
+ sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|96000|192000
+ channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_FRONT_BACK
+ formats AUDIO_FORMAT_PCM_24_BIT_PACKED|AUDIO_FORMAT_PCM_8_24_BIT
+ devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BACK_MIC|AUDIO_DEVICE_IN_WIRED_HEADSET
+ }
voice_rx {
sampling_rates 8000|16000|48000
channel_masks AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_MONO
diff --git a/configs/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index d0c4cdd..3f30e32 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -152,6 +152,17 @@
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_IN_5POINT1"/>
</mixPort>
+ <mixPort name="record_24" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_FLOAT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </mixPort>
<mixPort name="voice_rx" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
@@ -268,6 +279,8 @@
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,Telephony Rx"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
+ <route type="mix" sink="record_24"
+ sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
<route type="mix" sink="Telephony Tx"
sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
<route type="mix" sink="voice_rx"
diff --git a/configs/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index 68f91cb..93d671d 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -94,6 +94,18 @@
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_6_RX Port Mixer SLIM_0_TX" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_4_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="SLIMBUS_4_RX Audio Mixer MultiMedia2" value="0" />
<ctl name="MultiMedia5 Mixer SLIM_0_TX" value="0" />
@@ -127,27 +139,40 @@
<ctl name="HDMI Mixer MultiMedia15" value="0" />
<ctl name="HDMI Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia2" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia2" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia3" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="0" />
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="0" />
<ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
<ctl name="IIR0 INP0 MUX" value="ZERO" />
<ctl name="IIR0 INP1 MUX" value="ZERO" />
@@ -286,8 +311,10 @@
<ctl name="SLIM_0_TX Channels" value="One" />
<ctl name="SLIM_1_TX Channels" value="One" />
<ctl name="SLIM RX0 MUX" value="ZERO" />
+ <ctl name="SLIM RX2 MUX" value="ZERO" />
<ctl name="SLIM RX3 MUX" value="ZERO" />
<ctl name="SLIM RX4 MUX" value="ZERO" />
+ <ctl name="SLIM RX5 MUX" value="ZERO" />
<ctl name="EAR PA Gain" value="G_6_DB" />
<ctl name="SpkrLeft COMP Switch" value="0" />
<ctl name="SpkrRight COMP Switch" value="0" />
@@ -308,6 +335,7 @@
<ctl name="AIF1_CAP Mixer SLIM TX1" value="0"/>
<ctl name="AIF1_CAP Mixer SLIM TX0" value="0"/>
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_6_RX Port Mixer AUX_PCM_UL_TX" value="0" />
<ctl name="HDMI Mixer MultiMedia4" value="0" />
<!-- HFP start -->
<ctl name="HFP_PRI_AUX_UL_HL Switch" value="0" />
@@ -338,6 +366,7 @@
<!-- Voice -->
<ctl name="SLIM_0_RX_Voice Mixer CSVoice" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer CSVoice" value="0" />
<ctl name="Voice_Tx Mixer SLIM_0_TX_Voice" value="0" />
<!-- Voice HDMI -->
<ctl name="HDMI_RX_Voice Mixer CSVoice" value="0" />
@@ -352,6 +381,7 @@
<!-- Voice2 -->
<ctl name="SLIM_0_RX_Voice Mixer Voice2" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer Voice2" value="0" />
<ctl name="Voice2_Tx Mixer SLIM_0_TX_Voice2" value="0" />
<!-- Voice2 HDMI -->
<ctl name="HDMI_RX_Voice Mixer Voice2" value="0" />
@@ -365,6 +395,7 @@
<!-- VoLTE -->
<ctl name="SLIM_0_RX_Voice Mixer VoLTE" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer VoLTE" value="0" />
<ctl name="VoLTE_Tx Mixer SLIM_0_TX_VoLTE" value="0" />
<!-- VoLTE HDMI -->
<ctl name="HDMI_RX_Voice Mixer VoLTE" value="0" />
@@ -378,6 +409,7 @@
<!-- Multimode Voice1 -->
<ctl name="SLIM_0_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode1" value="0" />
<ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="0" />
<!-- Multimode Voice1 HDMI -->
<ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="0" />
@@ -391,6 +423,7 @@
<!-- Multimode Voice2 -->
<ctl name="SLIM_0_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode2" value="0" />
<ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="0" />
<!-- Multimode Voice2 HDMI -->
<ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="0" />
@@ -427,6 +460,7 @@
<!-- compress-voip-call start -->
<ctl name="SLIM_0_RX_Voice Mixer Voip" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer Voip" value="0" />
<ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="0" />
<ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
<ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
@@ -436,6 +470,7 @@
<!-- QCHAT start -->
<ctl name="SLIM_0_RX_Voice Mixer QCHAT" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer QCHAT" value="0" />
<ctl name="QCHAT_Tx Mixer SLIM_0_TX_QCHAT" value="0" />
<ctl name="SLIM_7_RX_Voice Mixer QCHAT" value="0" />
<ctl name="QCHAT_Tx Mixer SLIM_7_TX_QCHAT" value="0" />
@@ -443,6 +478,7 @@
<!-- VoWLAN start -->
<ctl name="SLIM_0_RX_Voice Mixer VoWLAN" value="0" />
+ <ctl name="SLIM_6_RX_Voice Mixer VoWLAN" value="0" />
<ctl name="VoWLAN_Tx Mixer SLIM_0_TX_VoWLAN" value="0" />
<ctl name="HDMI_RX_Voice Mixer VoWLAN" value="0" />
<ctl name="SLIM_7_RX_Voice Mixer VoWLAN" value="0" />
@@ -533,6 +569,7 @@
<!-- ADSP testfwk -->
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
+ <ctl name="SLIMBUS6_DL_HL Switch" value="0" />
<!-- ADSP testfwk end-->
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
@@ -566,7 +603,7 @@
</path>
<path name="echo-reference headphones">
- <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX" />
</path>
<path name="echo-reference headphones-44.1">
@@ -613,6 +650,11 @@
</path>
<path name="deep-buffer-playback headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-headphones">
+ <path name="deep-buffer-playback headphones" />
<path name="deep-buffer-playback" />
</path>
@@ -656,6 +698,11 @@
</path>
<path name="low-latency-playback headphones">
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-headphones">
+ <path name="low-latency-playback headphones" />
<path name="low-latency-playback" />
</path>
@@ -668,7 +715,7 @@
</path>
<path name="audio-ull-playback headphones">
- <path name="audio-ull-playback" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia3" value="1" />
</path>
<path name="audio-ull-playback speaker-and-headphones">
@@ -745,13 +792,18 @@
</path>
<path name="compress-offload-playback headphones">
- <path name="compress-offload-playback" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia4" value="1" />
</path>
<path name="compress-offload-playback headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia4" value="1" />
</path>
+ <path name="compress-offload-playback speaker-and-headphones">
+ <path name="compress-offload-playback headphones" />
+ <path name="compress-offload-playback" />
+ </path>
+
<path name="compress-offload-playback2">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia7" value="1" />
</path>
@@ -788,13 +840,18 @@
</path>
<path name="compress-offload-playback2 headphones">
- <path name="compress-offload-playback2" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia7" value="1" />
</path>
<path name="compress-offload-playback2 headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia7" value="1" />
</path>
+ <path name="compress-offload-playback2 speaker-and-headphones">
+ <path name="compress-offload-playback2 headphones" />
+ <path name="compress-offload-playback2" />
+ </path>
+
<path name="compress-offload-playback3">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia10" value="1" />
</path>
@@ -831,13 +888,18 @@
</path>
<path name="compress-offload-playback3 headphones">
- <path name="compress-offload-playback3" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia10" value="1" />
</path>
<path name="compress-offload-playback3 headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia10" value="1" />
</path>
+ <path name="compress-offload-playback3 speaker-and-headphones">
+ <path name="compress-offload-playback3 headphones" />
+ <path name="compress-offload-playback3" />
+ </path>
+
<path name="compress-offload-playback4">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia11" value="1" />
</path>
@@ -874,13 +936,18 @@
</path>
<path name="compress-offload-playback4 headphones">
- <path name="compress-offload-playback4" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia11" value="1" />
</path>
<path name="compress-offload-playback4 headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia11" value="1" />
</path>
+ <path name="compress-offload-playback4 speaker-and-headphones">
+ <path name="compress-offload-playback4 headphones" />
+ <path name="compress-offload-playback4" />
+ </path>
+
<path name="compress-offload-playback5">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia12" value="1" />
</path>
@@ -917,13 +984,18 @@
</path>
<path name="compress-offload-playback5 headphones">
- <path name="compress-offload-playback5" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia12" value="1" />
</path>
<path name="compress-offload-playback5 headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia12" value="1" />
</path>
+ <path name="compress-offload-playback5 speaker-and-headphones">
+ <path name="compress-offload-playback5 headphones" />
+ <path name="compress-offload-playback5" />
+ </path>
+
<path name="compress-offload-playback6">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia13" value="1" />
</path>
@@ -960,13 +1032,18 @@
</path>
<path name="compress-offload-playback6 headphones">
- <path name="compress-offload-playback6" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia13" value="1" />
</path>
<path name="compress-offload-playback6 headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia13" value="1" />
</path>
+ <path name="compress-offload-playback6 speaker-and-headphones">
+ <path name="compress-offload-playback6 headphones" />
+ <path name="compress-offload-playback6" />
+ </path>
+
<path name="compress-offload-playback7">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia14" value="1" />
</path>
@@ -1003,13 +1080,18 @@
</path>
<path name="compress-offload-playback7 headphones">
- <path name="compress-offload-playback7" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia14" value="1" />
</path>
<path name="compress-offload-playback7 headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia14" value="1" />
</path>
+ <path name="compress-offload-playback7 speaker-and-headphones">
+ <path name="compress-offload-playback7 headphones" />
+ <path name="compress-offload-playback7" />
+ </path>
+
<path name="compress-offload-playback8">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia15" value="1" />
</path>
@@ -1046,13 +1128,18 @@
</path>
<path name="compress-offload-playback8 headphones">
- <path name="compress-offload-playback8" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia15" value="1" />
</path>
<path name="compress-offload-playback8 headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia15" value="1" />
</path>
+ <path name="compress-offload-playback8 speaker-and-headphones">
+ <path name="compress-offload-playback8 headphones" />
+ <path name="compress-offload-playback8" />
+ </path>
+
<path name="compress-offload-playback9">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia16" value="1" />
</path>
@@ -1089,13 +1176,18 @@
</path>
<path name="compress-offload-playback9 headphones">
- <path name="compress-offload-playback9" />
+ <ctl name="SLIMBUS_6_RX Audio Mixer MultiMedia16" value="1" />
</path>
<path name="compress-offload-playback9 headphones-44.1">
<ctl name="SLIMBUS_5_RX Audio Mixer MultiMedia16" value="1" />
</path>
+ <path name="compress-offload-playback9 speaker-and-headphones">
+ <path name="compress-offload-playback9 headphones" />
+ <path name="compress-offload-playback9" />
+ </path>
+
<path name="audio-record">
<ctl name="MultiMedia1 Mixer SLIM_0_TX" value="1" />
</path>
@@ -1189,6 +1281,11 @@
<ctl name="Voice_Tx Mixer AFE_PCM_TX_Voice" value="1" />
</path>
+ <path name="voice-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer CSVoice" value="1" />
+ <ctl name="Voice_Tx Mixer SLIM_0_TX_Voice" value="1" />
+ </path>
+
<path name="voice2-call">
<ctl name="SLIM_0_RX_Voice Mixer Voice2" value="1" />
<ctl name="Voice2_Tx Mixer SLIM_0_TX_Voice2" value="1" />
@@ -1229,6 +1326,11 @@
<path name="voice2-call"/>
</path>
+ <path name="voice2-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer Voice2" value="1" />
+ <ctl name="Voice2_Tx Mixer SLIM_0_TX_Voice2" value="1" />
+ </path>
+
<path name="play-fm">
<ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
<ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="1" />
@@ -1236,7 +1338,9 @@
</path>
<path name="play-fm headphones">
- <path name="play-fm" />
+ <ctl name="Tert MI2S LOOPBACK Volume" value="1" />
+ <ctl name="SLIMBUS_6_RX Port Mixer TERT_MI2S_TX" value="1" />
+ <ctl name="SLIMBUS6_DL_HL Switch" value="1" />
</path>
<path name="incall-rec-uplink">
@@ -1369,16 +1473,34 @@
<ctl name="SLIMBUS_DL_HL Switch" value="1" />
</path>
+ <path name="hfp-sco headphones">
+ <ctl name="HFP_PRI_AUX_UL_HL Switch" value="1" />
+ <ctl name="SLIMBUS_6_RX Port Mixer AUX_PCM_UL_TX" value="1" />
+ <ctl name="AUX_PCM_RX Audio Mixer MultiMedia6" value="1" />
+ <ctl name="MultiMedia6 Mixer SLIM_0_TX" value="1" />
+ <ctl name="SLIMBUS6_DL_HL Switch" value="1" />
+ </path>
+
<path name="hfp-sco-wb">
<ctl name="BT_SCO SampleRate" value="16000" />
<path name="hfp-sco" />
</path>
+ <path name="hfp-sco-wb headphones">
+ <ctl name="AUX PCM SampleRate" value="16000" />
+ <path name="hfp-sco headphones" />
+ </path>
+
<path name="volte-call">
<ctl name="SLIM_0_RX_Voice Mixer VoLTE" value="1" />
<ctl name="VoLTE_Tx Mixer SLIM_0_TX_VoLTE" value="1" />
</path>
+ <path name="volte-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer VoLTE" value="1" />
+ <ctl name="VoLTE_Tx Mixer SLIM_0_TX_VoLTE" value="1" />
+ </path>
+
<path name="volte-call hdmi">
<ctl name="HDMI_RX_Voice Mixer VoLTE" value="1" />
<ctl name="VoLTE_Tx Mixer SLIM_0_TX_VoLTE" value="1" />
@@ -1414,6 +1536,12 @@
<ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="1" />
</path>
+ <path name="compress-voip-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer SLIM_0_TX_Voip" value="1" />
+ </path>
+
+
<path name="compress-voip-call bt-sco">
<ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
<ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="1" />
@@ -1444,6 +1572,11 @@
<ctl name="VoWLAN_Tx Mixer SLIM_0_TX_VoWLAN" value="1" />
</path>
+ <path name="vowlan-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer VoWLAN" value="1" />
+ <ctl name="VoWLAN_Tx Mixer SLIM_0_TX_VoWLAN" value="1" />
+ </path>
+
<path name="vowlan-call hdmi">
<ctl name="HDMI_RX_Voice Mixer VoWLAN" value="1" />
<ctl name="VoWLAN_Tx Mixer SLIM_0_TX_VoWLAN" value="1" />
@@ -1479,6 +1612,11 @@
<ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
</path>
+ <path name="voicemmode1-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
+ </path>
+
<path name="voicemmode1-call hdmi">
<ctl name="HDMI_RX_Voice Mixer VoiceMMode1" value="1" />
<ctl name="VoiceMMode1_Tx Mixer SLIM_0_TX_MMode1" value="1" />
@@ -1514,6 +1652,11 @@
<ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
</path>
+ <path name="voicemmode2-call headphones">
+ <ctl name="SLIM_6_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
+ </path>
+
<path name="voicemmode2-call hdmi">
<ctl name="HDMI_RX_Voice Mixer VoiceMMode2" value="1" />
<ctl name="VoiceMMode2_Tx Mixer SLIM_0_TX_MMode2" value="1" />
@@ -1851,22 +1994,22 @@
</path>
<path name="headphones">
- <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM_0_RX Channels" value="Two" />
- <ctl name="RX INT1_2 MUX" value="RX0" />
- <ctl name="RX INT2_2 MUX" value="RX1" />
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="Two" />
+ <ctl name= "RX INT1_1 MIX1 INP0" value="RX2" />
+ <ctl name= "RX INT2_1 MIX1 INP0" value="RX3" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
</path>
<path name="headphones-44.1">
- <ctl name="SLIM RX3 MUX" value="AIF3_PB" />
<ctl name="SLIM RX4 MUX" value="AIF3_PB" />
+ <ctl name="SLIM RX5 MUX" value="AIF3_PB" />
<ctl name="SLIM_5_RX Channels" value="Two" />
<ctl name="SLIM_5_RX SampleRate" value="KHZ_44P1" />
- <ctl name="RX INT1_1 MIX1 INP0" value="RX3" />
- <ctl name="RX INT2_1 MIX1 INP1" value="RX4" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX4" />
+ <ctl name="RX INT2_1 MIX1 INP1" value="RX5" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="SPL SRC0 MUX" value="SRC_IN_HPHL" />
@@ -1879,8 +2022,8 @@
<path name="true-native-mode">
<ctl name="RX INT1_2 MUX" value="ZERO" />
<ctl name="RX INT2_2 MUX" value="ZERO" />
- <ctl name= "RX INT1_1 MIX1 INP0" value="RX0" />
- <ctl name= "RX INT2_1 MIX1 INP0" value="RX1" />
+ <ctl name= "RX INT1_1 MIX1 INP0" value="RX2" />
+ <ctl name= "RX INT2_1 MIX1 INP0" value="RX3" />
<ctl name= "RX INT1 SPLINE MIX HPHL Native Switch" value="1" />
<ctl name= "RX INT2 SPLINE MIX HPHR Native Switch" value="1" />
</path>
@@ -1939,16 +2082,7 @@
<path name="speaker-and-headphones">
<path name="headphones" />
- <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
- <ctl name="RX INT8_1 MIX1 INP0" value="RX1" />
- <ctl name="SpkrLeft COMP Switch" value="1" />
- <ctl name="SpkrRight COMP Switch" value="1" />
- <ctl name="SpkrLeft BOOST Switch" value="1" />
- <ctl name="SpkrRight BOOST Switch" value="1" />
- <ctl name="SpkrLeft VISENSE Switch" value="1" />
- <ctl name="SpkrRight VISENSE Switch" value="1" />
- <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
- <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+ <path name="speaker" />
</path>
<path name="speaker-and-line">
@@ -1957,16 +2091,7 @@
<path name="speaker-and-headphones-liquid">
<path name="headphones" />
- <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
- <ctl name="RX INT8_1 MIX1 INP0" value="RX1" />
- <ctl name="SpkrLeft COMP Switch" value="1" />
- <ctl name="SpkrRight COMP Switch" value="1" />
- <ctl name="SpkrLeft BOOST Switch" value="1" />
- <ctl name="SpkrRight BOOST Switch" value="1" />
- <ctl name="SpkrLeft VISENSE Switch" value="1" />
- <ctl name="SpkrRight VISENSE Switch" value="1" />
- <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
- <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+ <path name="speaker" />
</path>
<path name="speaker-and-line-liquid">
@@ -1992,11 +2117,11 @@
<ctl name="ANC1 FB MUX" value="ANC_IN_HPHR" />
<ctl name="ADC3 Volume" value="8" />
<ctl name="ADC4 Volume" value="8" />
- <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM_0_RX Channels" value="Two" />
- <ctl name="RX INT1_1 MIX1 INP0" value="RX0" />
- <ctl name="RX INT2_1 MIX1 INP0" value="RX1" />
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="Two" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX2" />
+ <ctl name="RX INT2_1 MIX1 INP0" value="RX3" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX1 Digital Volume" value="81" />
@@ -2009,16 +2134,7 @@
<path name="speaker-and-anc-headphones">
<path name="anc-headphones" />
- <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
- <ctl name="RX INT8_1 MIX1 INP0" value="RX1" />
- <ctl name="SpkrLeft COMP Switch" value="1" />
- <ctl name="SpkrRight COMP Switch" value="1" />
- <ctl name="SpkrLeft BOOST Switch" value="1" />
- <ctl name="SpkrRight BOOST Switch" value="1" />
- <ctl name="SpkrLeft VISENSE Switch" value="1" />
- <ctl name="SpkrRight VISENSE Switch" value="1" />
- <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
- <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+ <path name="speaker" />
</path>
<path name="anc-fb-headphones">
@@ -2028,16 +2144,7 @@
<path name="speaker-and-anc-fb-headphones">
<path name="anc-fb-headphones" />
- <ctl name="RX INT7_1 MIX1 INP0" value="RX0" />
- <ctl name="RX INT8_1 MIX1 INP0" value="RX1" />
- <ctl name="SpkrLeft COMP Switch" value="1" />
- <ctl name="SpkrRight COMP Switch" value="1" />
- <ctl name="SpkrLeft BOOST Switch" value="1" />
- <ctl name="SpkrRight BOOST Switch" value="1" />
- <ctl name="SpkrLeft VISENSE Switch" value="1" />
- <ctl name="SpkrRight VISENSE Switch" value="1" />
- <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
- <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+ <path name="speaker" />
</path>
<path name="voice-anc-headphones">
@@ -2053,11 +2160,11 @@
<ctl name="ANC1 FB MUX" value="ANC_IN_HPHR" />
<ctl name="ADC3 Volume" value="8" />
<ctl name="ADC4 Volume" value="8" />
- <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM_0_RX Channels" value="Two" />
- <ctl name="RX INT1_1 MIX1 INP0" value="RX0" />
- <ctl name="RX INT2_1 MIX1 INP0" value="RX1" />
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="Two" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX2" />
+ <ctl name="RX INT2_1 MIX1 INP0" value="RX3" />
<ctl name="RX HPH Mode" value="CLS_H_LP" />
<ctl name="IIR0 Enable Band1" value="1" />
<ctl name="IIR0 Enable Band2" value="1" />
@@ -2088,11 +2195,11 @@
<ctl name="ANC1 FB MUX" value="ANC_IN_HPHR" />
<ctl name="ADC3 Volume" value="8" />
<ctl name="ADC4 Volume" value="8" />
- <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM RX1 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM_0_RX Channels" value="Two" />
- <ctl name="RX INT1_1 MIX1 INP0" value="RX0" />
- <ctl name="RX INT2_1 MIX1 INP0" value="RX1" />
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM RX3 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="Two" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX2" />
+ <ctl name="RX INT2_1 MIX1 INP0" value="RX3" />
<ctl name="RX HPH Mode" value="CLS_H_LP" />
<ctl name="IIR0 Enable Band1" value="1" />
<ctl name="IIR0 Enable Band2" value="1" />
@@ -2343,9 +2450,9 @@
<!-- TTY devices -->
<path name="tty-headphones">
- <ctl name="SLIM RX0 MUX" value="AIF_MIX1_PB" />
- <ctl name="SLIM_0_RX Channels" value="One" />
- <ctl name="RX INT1_1 MIX1 INP0" value="RX0" />
+ <ctl name="SLIM RX2 MUX" value="AIF4_PB" />
+ <ctl name="SLIM_6_RX Channels" value="One" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX2" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
</path>
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 91d7c0a..3ce83b1 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -40,17 +40,6 @@
#include <cutils/str_parms.h>
-#ifndef PCM_OFFLOAD_ENABLED
-#define AUDIO_FORMAT_PCM_OFFLOAD 0x1A000000UL
-#define AUDIO_FORMAT_PCM_16_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_16_BIT)
-#define AUDIO_FORMAT_PCM_24_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_8_24_BIT)
-#define AUDIO_OFFLOAD_CODEC_FORMAT "music_offload_codec_format"
-#define audio_is_offload_pcm(format) (0)
-#define OFFLOAD_USE_SMALL_BUFFER false
-#else
-#define OFFLOAD_USE_SMALL_BUFFER ((info->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD)
-#endif
-
#ifndef AFE_PROXY_ENABLED
#define AUDIO_DEVICE_OUT_PROXY 0x40000
#endif
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index b3ba2b5..cd9ead7 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -97,6 +97,8 @@
const struct string_to_enum s_format_name_to_enum_table[] = {
STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
STRING_TO_ENUM(AUDIO_FORMAT_MP3),
STRING_TO_ENUM(AUDIO_FORMAT_AAC),
@@ -117,8 +119,6 @@
STRING_TO_ENUM(AUDIO_FORMAT_QCELP),
STRING_TO_ENUM(AUDIO_FORMAT_MP2),
STRING_TO_ENUM(AUDIO_FORMAT_EVRCNW),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT_OFFLOAD),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_OFFLOAD),
STRING_TO_ENUM(AUDIO_FORMAT_FLAC),
STRING_TO_ENUM(AUDIO_FORMAT_ALAC),
STRING_TO_ENUM(AUDIO_FORMAT_APE),
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 8486e18..ad565de 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -74,6 +74,8 @@
#include "sound/asound.h"
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+/*DIRECT PCM has same buffer sizes as DEEP Buffer*/
+#define DIRECT_PCM_NUM_FRAGMENTS 2
/* ToDo: Check and update a proper value in msec */
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
@@ -287,17 +289,17 @@
return ret_val;
}
-static int check_and_set_gapless_mode(struct audio_device *adev) {
-
-
- char value[PROPERTY_VALUE_MAX] = {0};
+static int check_and_set_gapless_mode(struct audio_device *adev, bool enable_gapless)
+{
bool gapless_enabled = false;
const char *mixer_ctl_name = "Compress Gapless Playback";
struct mixer_ctl *ctl;
ALOGV("%s:", __func__);
- property_get("audio.offload.gapless.enabled", value, NULL);
- gapless_enabled = atoi(value) || !strncmp("true", value, 4);
+ gapless_enabled = property_get_bool("audio.offload.gapless.enabled", false);
+
+ /*Disable gapless if its AV playback*/
+ gapless_enabled = gapless_enabled && enable_gapless;
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
@@ -323,8 +325,8 @@
format == AUDIO_FORMAT_AAC_ADTS_LC ||
format == AUDIO_FORMAT_AAC_ADTS_HE_V1 ||
format == AUDIO_FORMAT_AAC_ADTS_HE_V2 ||
- format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD ||
- format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD ||
+ format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+ format == AUDIO_FORMAT_PCM_8_24_BIT ||
format == AUDIO_FORMAT_PCM_16_BIT ||
format == AUDIO_FORMAT_FLAC ||
format == AUDIO_FORMAT_ALAC ||
@@ -351,7 +353,6 @@
case AUDIO_FORMAT_AAC_ADTS:
id = SND_AUDIOCODEC_AAC;
break;
- case AUDIO_FORMAT_PCM_OFFLOAD:
case AUDIO_FORMAT_PCM:
id = SND_AUDIOCODEC_PCM;
break;
@@ -1036,9 +1037,9 @@
if (voice_is_in_call(adev) && adev->mode != AUDIO_MODE_NORMAL) {
vc_usecase = get_usecase_from_list(adev,
get_usecase_id_from_usecase_type(adev, VOICE_CALL));
- if (((vc_usecase) && ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
- (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND))) ||
- (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL)) {
+ if ((vc_usecase) && (((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) ||
+ (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
in_snd_device = vc_usecase->in_snd_device;
out_snd_device = vc_usecase->out_snd_device;
}
@@ -1956,7 +1957,7 @@
for the default max poll time (20s) in the event of an SSR.
Reduce the poll time to observe and deal with SSR faster.
*/
- if (out->use_small_bufs) {
+ if (!out->non_blocking) {
compress_set_max_poll_wait(out->compr, 1000);
}
@@ -1973,7 +1974,7 @@
if (adev->visualizer_start_output != NULL)
adev->visualizer_start_output(out->handle, out->pcm_device_id);
if (adev->offload_effects_start_output != NULL)
- adev->offload_effects_start_output(out->handle, out->pcm_device_id);
+ adev->offload_effects_start_output(out->handle, out->pcm_device_id, adev->mixer);
audio_extn_check_and_set_dts_hpx_state(adev);
}
}
@@ -2061,6 +2062,37 @@
return size;
}
+static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out)
+{
+ uint64_t actual_frames_rendered = 0;
+ size_t kernel_buffer_size = out->compr_config.fragment_size * out->compr_config.fragments;
+
+ /* This adjustment accounts for buffering after app processor.
+ * It is based on estimated DSP latency per use case, rather than exact.
+ */
+ int64_t platform_latency = platform_render_latency(out->usecase) *
+ out->sample_rate / 1000000LL;
+
+ /* not querying actual state of buffering in kernel as it would involve an ioctl call
+ * which then needs protection, this causes delay in TS query for pcm_offload usecase
+ * hence only estimate.
+ */
+ int64_t signed_frames = out->written - kernel_buffer_size;
+
+ signed_frames = signed_frames / (audio_bytes_per_sample(out->format) * popcount(out->channel_mask)) - platform_latency;
+
+ if (signed_frames > 0)
+ actual_frames_rendered = signed_frames;
+
+ ALOGVV("%s signed frames %lld out_written %lld kernel_buffer_size %d"
+ "bytes/sample %zu channel count %d", __func__,(long long int)signed_frames,
+ (long long int)out->written, (int)kernel_buffer_size,
+ audio_bytes_per_sample(out->compr_config.codec->format),
+ popcount(out->channel_mask));
+
+ return actual_frames_rendered;
+}
+
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
@@ -2553,6 +2585,9 @@
out_standby(&out->stream.common);
return ret;
}
+ if ( ret == (ssize_t)bytes && !out->non_blocking)
+ out->written += bytes;
+
if (!out->playback_started && ret >= 0) {
compress_start(out->compr);
audio_extn_dts_eagle_fade(adev, true, out);
@@ -2627,14 +2662,24 @@
*dsp_frames = 0;
if (is_offload_usecase(out->usecase)) {
ssize_t ret = 0;
+
+ /* Below piece of code is not guarded against any lock beacuse audioFliner serializes
+ * this operation and adev_close_output_stream(where out gets reset).
+ */
+ if (!out->non_blocking && (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
+ *dsp_frames = get_actual_pcm_frames_rendered(out);
+ ALOGVV("dsp_frames %d sampleRate %d",(int)*dsp_frames,out->sample_rate);
+ return 0;
+ }
+
lock_output_stream(out);
- if (out->compr != NULL) {
+ if (out->compr != NULL && out->non_blocking) {
ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
&out->sample_rate);
if (ret < 0)
ret = -errno;
ALOGVV("%s rendered frames %d sample_rate %d",
- __func__, *dsp_frames, out->sample_rate);
+ __func__, *dsp_frames, out->sample_rate);
}
pthread_mutex_unlock(&out->lock);
if (-ENETRESET == ret) {
@@ -2686,27 +2731,37 @@
int ret = -1;
unsigned long dsp_frames;
+ /* below piece of code is not guarded against any lock because audioFliner serializes
+ * this operation and adev_close_output_stream( where out gets reset).
+ */
+ if (is_offload_usecase(out->usecase) && !out->non_blocking &&
+ (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
+ *frames = get_actual_pcm_frames_rendered(out);
+ /* this is the best we can do */
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ ALOGVV("frames %lld playedat %lld",(long long int)*frames,
+ timestamp->tv_sec * 1000000LL + timestamp->tv_nsec / 1000);
+ return 0;
+ }
+
lock_output_stream(out);
- if (is_offload_usecase(out->usecase)) {
- if (out->compr != NULL) {
- ret = compress_get_tstamp(out->compr, &dsp_frames,
- &out->sample_rate);
- ALOGVV("%s rendered frames %ld sample_rate %d",
- __func__, dsp_frames, out->sample_rate);
- *frames = dsp_frames;
- if (ret < 0)
- ret = -errno;
- if (-ENETRESET == ret) {
- ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
- set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
- ret = -EINVAL;
- } else
- ret = 0;
-
- /* this is the best we can do */
- clock_gettime(CLOCK_MONOTONIC, timestamp);
- }
+ if (is_offload_usecase(out->usecase) && out->compr != NULL && out->non_blocking) {
+ ret = compress_get_tstamp(out->compr, &dsp_frames,
+ &out->sample_rate);
+ ALOGVV("%s rendered frames %ld sample_rate %d",
+ __func__, dsp_frames, out->sample_rate);
+ *frames = dsp_frames;
+ if (ret < 0)
+ ret = -errno;
+ if (-ENETRESET == ret) {
+ ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+ ret = -EINVAL;
+ } else
+ ret = 0;
+ /* this is the best we can do */
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
} else {
if (out->pcm) {
unsigned int avail;
@@ -2837,6 +2892,7 @@
ALOGD("copl(%p):calling compress flush", out);
lock_output_stream(out);
stop_compressed_output_l(out);
+ out->written = 0;
pthread_mutex_unlock(&out->lock);
ALOGD("copl(%p):out of compress flush", out);
return 0;
@@ -3238,7 +3294,6 @@
out->handle = handle;
out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
out->non_blocking = 0;
- out->use_small_bufs = false;
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
(flags & AUDIO_OUTPUT_FLAG_DIRECT)) {
@@ -3337,6 +3392,9 @@
}
if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ out->stream.pause = out_pause;
+ out->stream.flush = out_flush;
+ out->stream.resume = out_resume;
out->usecase = get_offload_usecase(adev, true);
ALOGV("DIRECT_PCM usecase ... usecase selected %d ", out->usecase);
} else {
@@ -3381,18 +3439,19 @@
out->compr_config.codec->id =
get_snd_codec_id(config->offload_info.format);
- if (((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD)||
- ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM)) {
+ if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
out->compr_config.fragment_size =
platform_get_pcm_offload_buffer_size(&config->offload_info);
+ out->compr_config.fragments = DIRECT_PCM_NUM_FRAGMENTS;
} else if (audio_extn_dolby_is_passthrough_stream(out)) {
out->compr_config.fragment_size =
audio_extn_dolby_get_passt_buffer_size(&config->offload_info);
+ out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
} else {
out->compr_config.fragment_size =
platform_get_compress_offload_buffer_size(&config->offload_info);
+ out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
}
- out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
out->compr_config.codec->sample_rate =
config->offload_info.sample_rate;
out->compr_config.codec->bit_rate =
@@ -3408,16 +3467,12 @@
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS)
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
- if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD)
- out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
- if (config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD)
- out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT)
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
-
- if (out->bit_width == 24) {
+ if (config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_3LE;
+ if (config->offload_info.format == AUDIO_FORMAT_PCM_8_24_BIT)
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
- }
if (config->offload_info.format == AUDIO_FORMAT_FLAC)
out->compr_config.codec->options.flac_dec.sample_size = AUDIO_OUTPUT_BIT_WIDTH;
@@ -3425,14 +3480,6 @@
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
- if (platform_use_small_buffer(&config->offload_info)) {
- //this flag is set from framework only if its for PCM formats
- //no need to check for PCM format again
- out->non_blocking = 0;
- out->use_small_bufs = true;
- ALOGI("Keep write blocking for small buff: non_blockling %d",
- out->non_blocking);
- }
out->send_new_metadata = 1;
out->send_next_track_params = false;
@@ -3446,11 +3493,18 @@
ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
__func__, config->offload_info.version,
config->offload_info.bit_rate);
- //Decide if we need to use gapless mode by default
- if (!audio_extn_dolby_is_passthrough_stream(out)) {
- ALOGV("%s: don't enable gapless for passthrough", __func__);
- check_and_set_gapless_mode(adev);
- }
+
+ /* Disable gapless if any of the following is true
+ * passthrough playback
+ * AV playback
+ * Direct PCM playback
+ */
+ if (audio_extn_dolby_is_passthrough_stream(out) ||
+ config->offload_info.has_video ||
+ out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ check_and_set_gapless_mode(adev, false);
+ } else
+ check_and_set_gapless_mode(adev, true);
if (audio_extn_dolby_is_passthrough_stream(out)) {
out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
@@ -4288,7 +4342,7 @@
ALOGV("%s: DLOPEN successful for %s", __func__,
OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
adev->offload_effects_start_output =
- (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
+ (int (*)(audio_io_handle_t, int, struct mixer *))dlsym(adev->offload_effects_lib,
"offload_effects_bundle_hal_start_output");
adev->offload_effects_stop_output =
(int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 31184d5..6c97840 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -209,7 +209,6 @@
struct stream_app_type_cfg app_type_cfg;
int non_blocking;
- bool use_small_bufs;
int playback_started;
int offload_state;
pthread_cond_t offload_cond;
@@ -343,7 +342,7 @@
int (*visualizer_start_output)(audio_io_handle_t, int);
int (*visualizer_stop_output)(audio_io_handle_t, int);
void *offload_effects_lib;
- int (*offload_effects_start_output)(audio_io_handle_t, int);
+ int (*offload_effects_start_output)(audio_io_handle_t, int, struct mixer *);
int (*offload_effects_stop_output)(audio_io_handle_t, int);
struct sound_card_status snd_card_status;
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 857d9e1..46611fc 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -760,6 +760,7 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
+#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
static bool is_misc_usecase(audio_usecase_t usecase) {
@@ -3651,7 +3652,7 @@
free(kv_pairs);
}
-/* Delay in Us */
+/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
switch (usecase) {
@@ -3659,6 +3660,9 @@
return DEEP_BUFFER_PLATFORM_DELAY;
case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
return LOW_LATENCY_PLATFORM_DELAY;
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD:
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
+ return PCM_OFFLOAD_PLATFORM_DELAY;
default:
return 0;
}
@@ -3846,19 +3850,17 @@
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info)
{
uint32_t fragment_size = 0;
- uint32_t bits_per_sample = 16;
+ uint32_t bytes_per_sample;
uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
- if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
- bits_per_sample = 32;
- }
+ bytes_per_sample = audio_bytes_per_sample(info->format);
//duration is set to 40 ms worth of stereo data at 48Khz
//with 16 bit per sample, modify this when the channel
//configuration is different
fragment_size = (pcm_offload_time
* info->sample_rate
- * (bits_per_sample >> 3)
+ * bytes_per_sample
* popcount(info->channel_mask))/1000;
if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
@@ -3867,23 +3869,18 @@
// To have same PCM samples for all channels, the buffer size requires to
// be multiple of (number of channels * bytes per sample)
// For writes to succeed, the buffer must be written at address which is multiple of 32
- fragment_size = ALIGN(fragment_size, ((bits_per_sample >> 3)* popcount(info->channel_mask) * 32));
+ fragment_size = ALIGN(fragment_size, (bytes_per_sample * popcount(info->channel_mask) * 32));
ALOGI("PCM offload Fragment size to %d bytes", fragment_size);
return fragment_size;
}
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
- return OFFLOAD_USE_SMALL_BUFFER;
-}
-
/*
* configures afe with bit width and Sample Rate
*/
static int platform_set_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- unsigned int bit_width, unsigned int sample_rate)
+ snd_device_t snd_device, unsigned int bit_width,
+ unsigned int sample_rate, audio_format_t format)
{
int ret = 0;
int backend_idx = DEFAULT_CODEC_BACKEND;
@@ -3908,13 +3905,17 @@
}
if (bit_width == 24) {
+ if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+ else
mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else {
mixer_ctl_set_enum_by_string(ctl, "S16_LE");
}
my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:becf: afe: %s mixer set to %d bit", __func__,
- my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
+ ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl,
+ bit_width, format);
}
/*
@@ -4150,11 +4151,13 @@
int new_snd_devices[SND_DEVICE_OUT_END];
int i, num_devices = 1;
bool ret = false;
+ audio_format_t format;
backend_idx = platform_get_backend_index(snd_device);
new_bit_width = usecase->stream.out->bit_width;
new_sample_rate = usecase->stream.out->sample_rate;
+ format = usecase->stream.out->format;
ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
@@ -4171,7 +4174,7 @@
if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
&new_bit_width, &new_sample_rate)) {
platform_set_codec_backend_cfg(adev, new_snd_devices[i],
- new_bit_width, new_sample_rate);
+ new_bit_width, new_sample_rate, format);
ret = true;
}
}
@@ -4691,8 +4694,8 @@
format = DTS_HD;
break;
case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD:
- case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_8_24_BIT:
ALOGV("%s:PCM", __func__);
format = LPCM;
break;
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 5a41b07..2c60f3b 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1119,11 +1119,6 @@
return 0;
}
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
- return false;
-}
-
int platform_get_edid_info(void *platform __unused)
{
return -ENOSYS;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 385d20b..c0e4088 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -667,6 +667,8 @@
{TO_NAME_INDEX(USECASE_INCALL_REC_DOWNLINK)},
{TO_NAME_INDEX(USECASE_INCALL_REC_UPLINK_AND_DOWNLINK)},
{TO_NAME_INDEX(USECASE_AUDIO_HFP_SCO)},
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_FM)},
+ {TO_NAME_INDEX(USECASE_AUDIO_RECORD_FM_VIRTUAL)},
};
#define NO_COLS 2
@@ -757,6 +759,7 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
+#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
bool platform_send_gain_dep_cal(void *platform, int level) {
@@ -1064,7 +1067,6 @@
backend_tag_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
backend_tag_table[SND_DEVICE_IN_CAPTURE_FM] = strdup("capture-fm");
backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
- backend_tag_table[SND_DEVICE_OUT_HEADPHONES] = strdup("headphones");
backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
@@ -3677,7 +3679,7 @@
free(kv_pairs);
}
-/* Delay in Us */
+/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
switch (usecase) {
@@ -3685,6 +3687,9 @@
return DEEP_BUFFER_PLATFORM_DELAY;
case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
return LOW_LATENCY_PLATFORM_DELAY;
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD:
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
+ return PCM_OFFLOAD_PLATFORM_DELAY;
default:
return 0;
}
@@ -3782,44 +3787,38 @@
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info)
{
uint32_t fragment_size = 0;
- uint32_t bits_per_sample = 16;
+ uint32_t bytes_per_sample;
uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
- if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
- bits_per_sample = 32;
- }
+ bytes_per_sample = audio_bytes_per_sample(info->format);
//duration is set to 40 ms worth of stereo data at 48Khz
//with 16 bit per sample, modify this when the channel
//configuration is different
fragment_size = (pcm_offload_time
* info->sample_rate
- * (bits_per_sample >> 3)
+ * bytes_per_sample
* popcount(info->channel_mask))/1000;
if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
else if(fragment_size > MAX_PCM_OFFLOAD_FRAGMENT_SIZE)
fragment_size = MAX_PCM_OFFLOAD_FRAGMENT_SIZE;
+
// To have same PCM samples for all channels, the buffer size requires to
// be multiple of (number of channels * bytes per sample)
// For writes to succeed, the buffer must be written at address which is multiple of 32
- fragment_size = ALIGN(fragment_size, ((bits_per_sample >> 3)* popcount(info->channel_mask) * 32));
+ fragment_size = ALIGN(fragment_size, ((bytes_per_sample) * popcount(info->channel_mask) * 32));
ALOGI("PCM offload Fragment size to %d bytes", fragment_size);
return fragment_size;
}
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
- return OFFLOAD_USE_SMALL_BUFFER;
-}
-
/*
* configures afe with bit width and Sample Rate
*/
static int platform_set_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- unsigned int bit_width, unsigned int sample_rate)
+ snd_device_t snd_device, unsigned int bit_width,
+ unsigned int sample_rate, audio_format_t format)
{
int ret = 0;
int backend_idx = DEFAULT_CODEC_BACKEND;
@@ -3845,13 +3844,16 @@
}
if (bit_width == 24) {
- mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+ else
+ mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else {
mixer_ctl_set_enum_by_string(ctl, "S16_LE");
}
my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:becf: afe: %s mixer set to %d bit", __func__,
- my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
+ ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
}
/*
@@ -4053,11 +4055,13 @@
int i, num_devices = 1;
bool ret = false;
struct platform_data *my_data = (struct platform_data *)adev->platform;
+ audio_format_t format;
backend_idx = platform_get_backend_index(snd_device);
new_bit_width = usecase->stream.out->bit_width;
new_sample_rate = usecase->stream.out->sample_rate;
+ format = usecase->stream.out->format;
ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
@@ -4073,7 +4077,7 @@
if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
&new_bit_width, &new_sample_rate)) {
platform_set_codec_backend_cfg(adev, new_snd_devices[i],
- new_bit_width, new_sample_rate);
+ new_bit_width, new_sample_rate, format);
ret = true;
}
}
@@ -4591,8 +4595,8 @@
format = DTS_HD;
break;
case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD:
- case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_8_24_BIT:
ALOGV("%s:PCM", __func__);
format = LPCM;
break;
diff --git a/hal/platform_api.h b/hal/platform_api.h
index df80a0c..cb177b6 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -109,7 +109,6 @@
struct audio_offload_info_t;
uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info);
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info);
-bool platform_use_small_buffer(audio_offload_info_t* info);
uint32_t platform_get_compress_passthrough_buffer_size(audio_offload_info_t* info);
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 24d8f74..839f1eb 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -460,80 +460,43 @@
return false;
}
- char propValue[PROPERTY_VALUE_MAX];
- bool pcmOffload = false;
-#ifdef PCM_OFFLOAD_ENABLED
- if ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD) {
- bool prop_enabled = false;
- if ((AUDIO_FORMAT_PCM_16_BIT_OFFLOAD == offloadInfo.format) &&
- property_get("audio.offload.pcm.16bit.enable", propValue, NULL)) {
- prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
- }
-
-#ifdef PCM_OFFLOAD_ENABLED_24
- if ((AUDIO_FORMAT_PCM_24_BIT_OFFLOAD == offloadInfo.format) &&
- property_get("audio.offload.pcm.24bit.enable", propValue, NULL)) {
- prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
- }
-#endif
-
- if (prop_enabled) {
- ALOGI("PCM offload property is enabled");
- pcmOffload = true;
- }
-
- if (!pcmOffload) {
- ALOGD("system property not enabled for PCM offload format[%x]",offloadInfo.format);
- return false;
- }
+ //check if it's multi-channel AAC (includes sub formats) and FLAC format
+ if ((popcount(offloadInfo.channel_mask) > 2) &&
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
+ ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
+ return false;
}
-#endif
- if (!pcmOffload) {
-
- bool compressedOffloadDisabled = property_get_bool("audio.offload.compress.disable", false);
- if (compressedOffloadDisabled) {
- ALOGI("compressed offload disabled by audio.offload.compress.disable=%d", compressedOffloadDisabled);
- return false;
- }
-
- //check if it's multi-channel AAC (includes sub formats) and FLAC format
- if ((popcount(offloadInfo.channel_mask) > 2) &&
- (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
- ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
- ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
- return false;
- }
#ifdef AUDIO_EXTN_FORMATS_ENABLED
- //check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
- if ((popcount(offloadInfo.channel_mask) > 2) &&
- (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
- (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && (offloadInfo.sample_rate > 48000)) ||
- (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.sample_rate > 48000)) ||
- (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.sample_rate > 48000)) ||
- ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) {
- ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
- return false;
- }
+ //check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
+ if ((popcount(offloadInfo.channel_mask) > 2) &&
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && (offloadInfo.sample_rate > 48000)) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.sample_rate > 48000)) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.sample_rate > 48000)) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) {
+ ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
+ return false;
+ }
#endif
- //TODO: enable audio offloading with video when ready
- const bool allowOffloadWithVideo =
- property_get_bool("audio.offload.video", false /* default_value */);
- if (offloadInfo.has_video && !allowOffloadWithVideo) {
- ALOGV("isOffloadSupported: has_video == true, returning false");
- return false;
- }
+ //TODO: enable audio offloading with video when ready
+ const bool allowOffloadWithVideo =
+ property_get_bool("audio.offload.video", false /* default_value */);
+ if (offloadInfo.has_video && !allowOffloadWithVideo) {
+ ALOGV("isOffloadSupported: has_video == true, returning false");
+ return false;
+ }
- const bool allowOffloadStreamingWithVideo = property_get_bool("av.streaming.offload.enable",
- false /*default value*/);
- if(offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) {
- ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue );
- return false;
- }
-
+ const bool allowOffloadStreamingWithVideo = property_get_bool("av.streaming.offload.enable",
+ false /*default value*/);
+ if (offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) {
+ ALOGW("offload disabled by av.streaming.offload.enable %d",allowOffloadStreamingWithVideo);
+ return false;
}
//If duration is less than minimum value defined in property, return false
+ char propValue[PROPERTY_VALUE_MAX];
if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
@@ -545,18 +508,18 @@
//do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
if ((offloadInfo.format == AUDIO_FORMAT_MP3) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
- ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS) ||
-#ifdef AUDIO_EXTN_FORMATS_ENABLED
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
- ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))
+ return false;
+
+#ifdef AUDIO_EXTN_FORMATS_ENABLED
+ if (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE) ||
- ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS) ||
-#endif
- pcmOffload)
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))
return false;
-
+#endif
}
// Do not allow offloading if one non offloadable effect is enabled. This prevents from
@@ -1334,6 +1297,19 @@
return false;
}
+bool static isDirectPCMEnabled(int bitWidth)
+{
+ bool directPCMEnabled = false;
+ if (bitWidth == 24 || bitWidth == 32)
+ directPCMEnabled =
+ property_get_bool("audio.offload.pcm.24bit.enable", false);
+ else
+ directPCMEnabled =
+ property_get_bool("audio.offload.pcm.16bit.enable", false);
+
+ return directPCMEnabled;
+}
+
status_t AudioPolicyManagerCustom::getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
@@ -1349,23 +1325,21 @@
audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER;
bool offloadDisabled = property_get_bool("audio.offload.disable", false);
- bool pcmOffloadEnabled = false;
+ uint32_t bitWidth = (audio_bytes_per_sample(format) * 8);
if (offloadDisabled) {
ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled);
}
- //read track offload property only if the global offload switch is off.
- if (!offloadDisabled) {
- pcmOffloadEnabled = property_get_bool("audio.offload.track.enable", false);
- }
+ if (!offloadDisabled && (offloadInfo == NULL) &&
+ isDirectPCMEnabled(bitWidth) &&
+ (flags == AUDIO_OUTPUT_FLAG_NONE)) {
- if (offloadInfo == NULL && pcmOffloadEnabled) {
tOffloadInfo.sample_rate = samplingRate;
tOffloadInfo.channel_mask = channelMask;
tOffloadInfo.format = format;
tOffloadInfo.stream_type = *stream;
- tOffloadInfo.bit_width = 16; //hard coded for PCM_16
+ tOffloadInfo.bit_width = bitWidth;
if (attr != NULL) {
ALOGV("found attribute .. setting usage %d ", attr->usage);
tOffloadInfo.usage = attr->usage;
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index d39a8b7..464bc0d 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -209,7 +209,7 @@
* Interface from audio HAL
*/
__attribute__ ((visibility ("default")))
-int offload_effects_bundle_hal_start_output(audio_io_handle_t output, int pcm_id)
+int offload_effects_bundle_hal_start_output(audio_io_handle_t output, int pcm_id, struct mixer *mixer)
{
int ret = 0;
struct listnode *node;
@@ -245,19 +245,19 @@
/* populate the mixer control to send offload parameters */
snprintf(mixer_string, sizeof(mixer_string),
"%s %d", "Audio Effects Config", out_ctxt->pcm_device_id);
- out_ctxt->mixer = mixer_open(MIXER_CARD);
- if (!out_ctxt->mixer) {
- ALOGE("Failed to open mixer");
+
+ if (!mixer) {
+ ALOGE("Invalid mixer");
out_ctxt->ctl = NULL;
out_ctxt->ref_ctl = NULL;
ret = -EINVAL;
free(out_ctxt);
goto exit;
} else {
+ out_ctxt->mixer = mixer;
out_ctxt->ctl = mixer_get_ctl_by_name(out_ctxt->mixer, mixer_string);
if (!out_ctxt->ctl) {
ALOGE("mixer_get_ctl_by_name failed");
- mixer_close(out_ctxt->mixer);
out_ctxt->mixer = NULL;
ret = -EINVAL;
free(out_ctxt);
@@ -314,9 +314,6 @@
fx_ctxt->ops.stop(fx_ctxt, out_ctxt);
}
- if (out_ctxt->mixer)
- mixer_close(out_ctxt->mixer);
-
list_remove(&out_ctxt->outputs_list_node);
#ifdef DTS_EAGLE