Merge "audio: add the property audio.offload.min.duration.secs"
diff --git a/configs/atoll/atoll.mk b/configs/atoll/atoll.mk
index 3432cdc..5acb773 100644
--- a/configs/atoll/atoll.mk
+++ b/configs/atoll/atoll.mk
@@ -69,6 +69,8 @@
AUDIO_FEATURE_ENABLED_BATTERY_LISTENER := true
##AUDIO_FEATURE_FLAGS
+BOARD_SUPPORTS_OPENSOURCE_STHAL := true
+
AUDIO_HARDWARE := audio.a2dp.default
AUDIO_HARDWARE += audio.usb.default
AUDIO_HARDWARE += audio.r_submix.default
@@ -167,6 +169,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/mixer_paths_wcd9375.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_wcd9375.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_qrd.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/mixer_paths_wcd9375qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_wcd9375qrd.xml \
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
diff --git a/configs/atoll/audio_policy_configuration.xml b/configs/atoll/audio_policy_configuration.xml
index 5a251c2..a6d7eef 100644
--- a/configs/atoll/audio_policy_configuration.xml
+++ b/configs/atoll/audio_policy_configuration.xml
@@ -266,17 +266,20 @@
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
@@ -319,27 +322,8 @@
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
</devicePort>
<devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
<devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
</devicePorts>
diff --git a/configs/atoll/mixer_paths_idp.xml b/configs/atoll/mixer_paths_idp.xml
index 860a253..433e1a8 100644
--- a/configs/atoll/mixer_paths_idp.xml
+++ b/configs/atoll/mixer_paths_idp.xml
@@ -2130,6 +2130,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/atoll/mixer_paths_qrd.xml b/configs/atoll/mixer_paths_qrd.xml
index 5efd1aa..8719bf1 100644
--- a/configs/atoll/mixer_paths_qrd.xml
+++ b/configs/atoll/mixer_paths_qrd.xml
@@ -2151,6 +2151,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/atoll/mixer_paths_wcd9375.xml b/configs/atoll/mixer_paths_wcd9375.xml
index 680f445..c4d2af7 100644
--- a/configs/atoll/mixer_paths_wcd9375.xml
+++ b/configs/atoll/mixer_paths_wcd9375.xml
@@ -2155,6 +2155,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/atoll/mixer_paths_wcd9375qrd.xml b/configs/atoll/mixer_paths_wcd9375qrd.xml
index 758a1d3..aee360c 100644
--- a/configs/atoll/mixer_paths_wcd9375qrd.xml
+++ b/configs/atoll/mixer_paths_wcd9375qrd.xml
@@ -2237,6 +2237,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/kona/audio_platform_info.xml b/configs/kona/audio_platform_info.xml
index 5b12029..346041f 100644
--- a/configs/kona/audio_platform_info.xml
+++ b/configs/kona/audio_platform_info.xml
@@ -110,6 +110,7 @@
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="in" id="36" />
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="out" id="36" />
<usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="23" />
+ <usecase name="USECASE_AUDIO_RECORD_COMPRESS2" type="in" id="37" />
</pcm_ids>
<config_params>
<param key="spkr_1_tz_name" value="wsatz.13"/>
@@ -145,6 +146,7 @@
<device name="SND_DEVICE_OUT_SPEAKER_AND_LINE" backend="speaker-and-headphones" interface="WSA_CDC_DMA_RX_0-and-RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET" backend="speaker-and-headphones" interface="WSA_CDC_DMA_RX_0-and-RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_HEADSET" backend="headset" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_ANC_HEADSET" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
@@ -160,6 +162,7 @@
<device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="WSA_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT" interface="WSA_CDC_DMA_RX_0"/>
@@ -196,10 +199,16 @@
<device name="SND_DEVICE_IN_SPEAKER_DMIC_AEC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_SPEAKER_DMIC_NS" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_HEADSET_MIC" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" interface="TX_CDC_DMA_TX_3"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_REC_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_AEC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
<device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_HDMI_MIC" interface="HDMI"/>
<device name="SND_DEVICE_IN_BT_SCO_MIC" interface="SLIMBUS_7_TX"/>
<device name="SND_DEVICE_IN_BT_SCO_MIC_NREC" interface="SLIMBUS_7_TX"/>
diff --git a/configs/kona/audio_platform_info_intcodec.xml b/configs/kona/audio_platform_info_intcodec.xml
index 2aa4b27..2d22400 100644
--- a/configs/kona/audio_platform_info_intcodec.xml
+++ b/configs/kona/audio_platform_info_intcodec.xml
@@ -94,6 +94,7 @@
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="in" id="36" />
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="out" id="36" />
<usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="23" />
+ <usecase name="USECASE_AUDIO_RECORD_COMPRESS2" type="in" id="37" />
</pcm_ids>
<config_params>
<!-- In the below value string, the value indicates default mono -->
@@ -149,6 +150,7 @@
<device name="SND_DEVICE_OUT_SPEAKER_AND_LINE" backend="speaker-and-headphones" interface="WSA_CDC_DMA_RX_0-and-RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET" backend="speaker-and-headphones" interface="WSA_CDC_DMA_RX_0-and-RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_HEADSET" backend="headset" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_ANC_HEADSET" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
@@ -164,6 +166,7 @@
<device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="WSA_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT" interface="WSA_CDC_DMA_RX_0"/>
@@ -214,11 +217,17 @@
<device name="SND_DEVICE_IN_SPEAKER_DMIC_NS_SB" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS_SB" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_HEADSET_MIC" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" interface="TX_CDC_DMA_TX_3"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_REC_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_AEC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
<device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_SB" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_HDMI_MIC" interface="HDMI"/>
<device name="SND_DEVICE_IN_BT_SCO_MIC" interface="SLIMBUS_7_TX"/>
<device name="SND_DEVICE_IN_BT_SCO_MIC_NREC" interface="SLIMBUS_7_TX"/>
diff --git a/configs/kona/audio_platform_info_qrd.xml b/configs/kona/audio_platform_info_qrd.xml
index 7ff0f17..bcf63aa 100644
--- a/configs/kona/audio_platform_info_qrd.xml
+++ b/configs/kona/audio_platform_info_qrd.xml
@@ -92,6 +92,7 @@
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="in" id="36" />
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="out" id="36" />
<usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="23" />
+ <usecase name="USECASE_AUDIO_RECORD_COMPRESS2" type="in" id="37" />
</pcm_ids>
<config_params>
<!-- In the below value string, the value indicates default mono -->
@@ -160,6 +161,7 @@
<device name="SND_DEVICE_OUT_SPEAKER_AND_LINE" backend="speaker-and-headphones" interface="WSA_CDC_DMA_RX_0-and-RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET" backend="speaker-and-headphones" interface="WSA_CDC_DMA_RX_0-and-RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_HEADSET" backend="headset" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_ANC_HEADSET" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="RX_CDC_DMA_RX_0"/>
@@ -175,6 +177,7 @@
<device name="SND_DEVICE_OUT_VOICE_HANDSET" backend="handset" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="WSA_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT" interface="WSA_CDC_DMA_RX_0"/>
@@ -225,11 +228,17 @@
<device name="SND_DEVICE_IN_SPEAKER_DMIC_NS_SB" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS_SB" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_HEADSET_MIC" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" interface="TX_CDC_DMA_TX_3"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_VOICE_REC_HEADSET_MIC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_AEC" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
+ <device name="SND_DEVICE_IN_HEADSET_MIC_FLUENCE" backend="headset-mic" interface="TX_CDC_DMA_TX_4"/>
<device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_SB" interface="TX_CDC_DMA_TX_3"/>
- <device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_IN_HDMI_MIC" interface="HDMI"/>
<device name="SND_DEVICE_IN_BT_SCO_MIC" interface="SLIMBUS_7_TX"/>
<device name="SND_DEVICE_IN_BT_SCO_MIC_NREC" interface="SLIMBUS_7_TX"/>
diff --git a/configs/kona/audio_policy_configuration.xml b/configs/kona/audio_policy_configuration.xml
index 657b5d1..8bb3328 100644
--- a/configs/kona/audio_policy_configuration.xml
+++ b/configs/kona/audio_policy_configuration.xml
@@ -184,7 +184,7 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
channelMasks="AUDIO_CHANNEL_IN_5POINT1,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</mixPort>
- <mixPort name="record_24" role="sink">
+ <mixPort name="record_24" role="sink" maxOpenCount="2" maxActiveCount="2">
<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,AUDIO_CHANNEL_INDEX_MASK_3,AUDIO_CHANNEL_INDEX_MASK_4"/>
@@ -311,27 +311,8 @@
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
</devicePort>
<devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
<devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
</devicePorts>
diff --git a/configs/kona/kona.mk b/configs/kona/kona.mk
index 2de2f3b..33a678f 100644
--- a/configs/kona/kona.mk
+++ b/configs/kona/kona.mk
@@ -32,6 +32,7 @@
AUDIO_FEATURE_ENABLED_ALAC_OFFLOAD := true
AUDIO_FEATURE_ENABLED_APE_OFFLOAD := true
AUDIO_FEATURE_ENABLED_AAC_ADTS_OFFLOAD := true
+AUDIO_FEATURE_ENABLED_MPEGH_SW_DECODER := true
AUDIO_FEATURE_ENABLED_PROXY_DEVICE := true
AUDIO_FEATURE_ENABLED_SSR := true
AUDIO_FEATURE_ENABLED_DTS_EAGLE := false
@@ -179,12 +180,15 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/audio_platform_info_intcodec.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_platform_info_intcodec.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/sound_trigger_mixer_paths.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/sound_trigger_mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths_qrd.xml \
+ vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/sound_trigger_mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths_cdp.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/mixer_paths.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_qrd.xml \
+ vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_cdp.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/audio_configs_stock.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs_stock.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/audio_tuning_mixer.txt:$(TARGET_COPY_OUT_VENDOR)/etc/audio_tuning_mixer.txt \
- frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -208,6 +212,14 @@
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio_hal.period_size=192
+##Ambisonic Capture
+PRODUCT_PROPERTY_OVERRIDES += \
+vendor.audio.ambisonic.capture=true \
+vendor.audio.ambisonic.auto.profile=true
+
+PRODUCT_PROPERTY_OVERRIDES += \
+vendor.audio.apptype.multirec.enabled=true
+
##fluencetype can be "fluence" or "fluencepro" or "none"
PRODUCT_PROPERTY_OVERRIDES += \
ro.vendor.audio.sdk.fluencetype=none\
@@ -289,7 +301,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -389,18 +401,17 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=true \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=true \
vendor.audio.feature.compress_meta_data.enable=true \
vendor.audio.feature.compr_voip.enable=false \
-vendor.audio.feature.concurrent_capture.enable=false \
+vendor.audio.feature.concurrent_capture.enable=true \
vendor.audio.feature.custom_stereo.enable=true \
vendor.audio.feature.display_port.enable=true \
vendor.audio.feature.dsm_feedback.enable=false \
vendor.audio.feature.dynamic_ecns.enable=true \
-vendor.audio.feature.ext_hw_plugin.enable=true \
+vendor.audio.feature.ext_hw_plugin.enable=false \
vendor.audio.feature.external_dsp.enable=false \
vendor.audio.feature.external_speaker.enable=false \
vendor.audio.feature.external_speaker_tfa.enable=false \
@@ -445,6 +456,15 @@
vendor.qti.hardware.audiohalext@1.0-impl \
vendor.qti.hardware.audiohalext-utils
+# enable audio hidl hal 5.0
+PRODUCT_PACKAGES += \
+ android.hardware.audio@5.0 \
+ android.hardware.audio.common@5.0 \
+ android.hardware.audio.common@5.0-util \
+ android.hardware.audio@5.0-impl \
+ android.hardware.audio.effect@5.0 \
+ android.hardware.audio.effect@5.0-impl
+
PRODUCT_PACKAGES_ENG += \
VoicePrintTest \
VoicePrintDemo
diff --git a/configs/kona/mixer_paths.xml b/configs/kona/mixer_paths.xml
index 3e40870..3949c7c 100644
--- a/configs/kona/mixer_paths.xml
+++ b/configs/kona/mixer_paths.xml
@@ -61,11 +61,16 @@
<ctl name="MultiMedia5 Mixer AFE_PCM_TX" value="0" />
<ctl name="MultiMedia5 Mixer TX_CDC_DMA_TX_3" value="0" />
<ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_4" value="0" />
<ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
<ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
<ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_4" value="0" />
<ctl name="MultiMedia10 Mixer SLIM_7_TX" value="0" />
<ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_4" value="0" />
+ <ctl name="MultiMedia17 Mixer SLIM_7_TX" value="0" />
<ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
<ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
<ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
@@ -129,6 +134,7 @@
<ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia17 Mixer USB_AUDIO_TX" value="0" />
<ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
<ctl name="WSA_CDC_DMA_RX_0 Channels" value="One" />
<ctl name="RX_CDC_DMA_RX_0 Channels" value="One" />
@@ -175,6 +181,7 @@
<ctl name="WSA_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="0" />
<ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="0" />
<ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode1" value="0" />
<!-- Multimode Voice1 BTSCO -->
<ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="0" />
<ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="0" />
@@ -191,6 +198,7 @@
<ctl name="WSA_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="0" />
<ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="0" />
<ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode2" value="0" />
<!-- Multimode Voice2 BTSCO -->
<ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="0" />
<ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="0" />
@@ -423,6 +431,7 @@
<!-- defaults for mmap record -->
<ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_4" value="0" />
<ctl name="MultiMedia16 Mixer SLIM_7_TX" value="0" />
<ctl name="MultiMedia16 Mixer TERT_MI2S_TX" value="0" />
<ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="0" />
@@ -440,6 +449,10 @@
<ctl name="AUDIO_REF_EC_UL1 MUX" value="RX_CDC_DMA_RX_0" />
</path>
+ <path name="echo-reference headset">
+ <path name="echo-reference headphones" />
+ </path>
+
<path name="echo-reference display-port">
<ctl name="AUDIO_REF_EC_UL1 MUX" value="DISPLAY_PORT" />
</path>
@@ -507,6 +520,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia1" value="1" />
</path>
+ <path name="deep-buffer-playback headset">
+ <path name="deep-buffer-playback headphones" />
+ </path>
+
<path name="deep-buffer-playback speaker-and-headphones">
<path name="deep-buffer-playback headphones" />
<path name="deep-buffer-playback" />
@@ -578,6 +595,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia5" value="1" />
</path>
+ <path name="low-latency-playback headset">
+ <path name="low-latency-playback headphones" />
+ </path>
+
<path name="low-latency-playback speaker-and-headphones">
<path name="low-latency-playback headphones" />
<path name="low-latency-playback" />
@@ -613,6 +634,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia8" value="1" />
</path>
+ <path name="audio-ull-playback headset">
+ <path name="audio-ull-playback headphones" />
+ </path>
+
<path name="audio-ull-playback speaker-and-headphones">
<path name="audio-ull-playback" />
<path name="audio-ull-playback headphones" />
@@ -730,6 +755,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
</path>
+ <path name="compress-offload-playback headset">
+ <path name="compress-offload-playback headphones" />
+ </path>
+
<path name="compress-offload-playback headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
</path>
@@ -805,6 +834,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
</path>
+ <path name="compress-offload-playback2 headset">
+ <path name="compress-offload-playback2 headphones" />
+ </path>
+
<path name="compress-offload-playback2 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
@@ -881,6 +914,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
</path>
+ <path name="compress-offload-playback3 headset">
+ <path name="compress-offload-playback3 headphones" />
+ </path>
+
<path name="compress-offload-playback3 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
</path>
@@ -957,6 +994,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
</path>
+ <path name="compress-offload-playback4 headset">
+ <path name="compress-offload-playback4 headphones" />
+ </path>
+
<path name="compress-offload-playback4 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
</path>
@@ -1032,6 +1073,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
</path>
+ <path name="compress-offload-playback5 headset">
+ <path name="compress-offload-playback5 headphones" />
+ </path>
+
<path name="compress-offload-playback5 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
</path>
@@ -1107,6 +1152,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
</path>
+ <path name="compress-offload-playback6 headset">
+ <path name="compress-offload-playback6 headphones" />
+ </path>
+
<path name="compress-offload-playback6 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
</path>
@@ -1182,6 +1231,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
</path>
+ <path name="compress-offload-playback7 headset">
+ <path name="compress-offload-playback7 headphones" />
+ </path>
+
<path name="compress-offload-playback7 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
</path>
@@ -1257,6 +1310,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
</path>
+ <path name="compress-offload-playback8 headset">
+ <path name="compress-offload-playback8 headphones" />
+ </path>
+
<path name="compress-offload-playback8 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
</path>
@@ -1332,6 +1389,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
</path>
+ <path name="compress-offload-playback9 headset">
+ <path name="compress-offload-playback9 headphones" />
+ </path>
+
<path name="compress-offload-playback9 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
</path>
@@ -1381,6 +1442,10 @@
<path name="audio-record bt-sco" />
</path>
+ <path name="audio-record headset-mic">
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
<path name="audio-record capture-fm">
<ctl name="MultiMedia1 Mixer TERT_MI2S_TX" value="1" />
</path>
@@ -1406,6 +1471,31 @@
<ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
</path>
+ <path name="audio-record-compress2">
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="audio-record-compress2 bt-sco">
+ <ctl name="MultiMedia17 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-record-compress2 bt-sco" />
+ </path>
+
+ <path name="audio-record-compress2 bt-sco-swb">
+ <path name="audio-record-compress2 bt-sco" />
+ </path>
+
+ <path name="audio-record-compress2 usb-headset-mic">
+ <ctl name="MultiMedia17 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress2 headset-mic">
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
<path name="low-latency-record">
<ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_3" value="1" />
</path>
@@ -1447,6 +1537,10 @@
<ctl name="RX_CDC_DMA_RX_0_DL_HL Switch" value="1" />
</path>
+ <path name="play-fm headset">
+ <path name="play-fm headphones" />
+ </path>
+
<path name="incall-rec-uplink">
<ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
</path>
@@ -1471,6 +1565,10 @@
<path name="incall-rec-uplink" />
</path>
+ <path name="incall-rec-uplink headset-mic">
+ <path name="incall-rec-uplink" />
+ </path>
+
<path name="incall-rec-uplink-compress">
<ctl name="MultiMedia8 Mixer VOC_REC_UL" value="1" />
</path>
@@ -1495,6 +1593,10 @@
<path name="incall-rec-uplink-compress" />
</path>
+ <path name="incall-rec-uplink-compress headset-mic">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
<path name="incall-rec-downlink">
<ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
</path>
@@ -1519,6 +1621,10 @@
<path name="incall-rec-downlink" />
</path>
+ <path name="incall-rec-downlink headset-mic">
+ <path name="incall-rec-downlink" />
+ </path>
+
<path name="incall-rec-downlink-compress">
<ctl name="MultiMedia8 Mixer VOC_REC_DL" value="1" />
</path>
@@ -1543,6 +1649,10 @@
<path name="incall-rec-downlink-compress" />
</path>
+ <path name="incall-rec-downlink-compress headset-mic">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
<path name="incall-rec-uplink-and-downlink">
<path name="incall-rec-uplink" />
<path name="incall-rec-downlink" />
@@ -1568,6 +1678,10 @@
<path name="incall-rec-uplink-and-downlink" />
</path>
+ <path name="incall-rec-uplink-and-downlink headset-mic">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
<path name="incall-rec-uplink-and-downlink-compress">
<path name="incall-rec-uplink-compress" />
<path name="incall-rec-downlink-compress" />
@@ -1593,6 +1707,10 @@
<path name="incall-rec-uplink-and-downlink-compress" />
</path>
+ <path name="incall-rec-uplink-and-downlink-compress headset-mic">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
<path name="hfp-sco">
</path>
@@ -1662,6 +1780,11 @@
<ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="1" />
</path>
+ <path name="voicemmode1-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode1" value="1" />
+ </path>
+
<path name="voicemmode1-call bt-sco">
<ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="1" />
<ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="1" />
@@ -1711,6 +1834,11 @@
<ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="1" />
</path>
+ <path name="voicemmode2-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_4_MMode2" value="1" />
+ </path>
+
<path name="voicemmode2-call bt-sco">
<ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="1" />
<ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="1" />
@@ -1819,6 +1947,10 @@
<path name="audio-record-voip bt-sco" />
</path>
+ <path name="audio-record-voip headset-mic">
+ <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
<path name="spkr-rx-calib">
<ctl name="WSA_CDC_DMA_RX_0_DL_HL Switch" value="1" />
</path>
@@ -1828,10 +1960,10 @@
<!-- These are actual sound device specific mixer settings -->
<path name="amic2">
- <ctl name="TX DEC0 MUX" value="SWR_MIC" />
- <ctl name="TX SMIC MUX0" value="ADC1" />
- <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
- <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="TX DEC5 MUX" value="SWR_MIC" />
+ <ctl name="TX SMIC MUX5" value="ADC1" />
+ <ctl name="TX_CDC_DMA_TX_4 Channels" value="One" />
+ <ctl name="TX_AIF2_CAP Mixer DEC5" value="1" />
<ctl name="ADC2_MIXER Switch" value="1" />
<ctl name="ADC2 MUX" value="INP2" />
</path>
@@ -2027,8 +2159,6 @@
<ctl name="RX INT1_1 MIX1 INP0" value="RX1" />
<ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
- <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
- <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
<ctl name="RX_COMP1 Switch" value="1" />
<ctl name="RX_COMP2 Switch" value="1" />
<ctl name="HPHL_COMP Switch" value="1" />
@@ -2056,7 +2186,6 @@
<path name="headset-mic">
<path name="amic2" />
- <ctl name="TX_DEC0 Volume" value="84" />
</path>
<path name="headset-mic-liquid">
@@ -2113,6 +2242,10 @@
<path name="headphones" />
</path>
+ <path name="voice-headset">
+ <path name="headphones" />
+ </path>
+
<path name="voice-line">
<path name="voice-headphones" />
</path>
@@ -2648,6 +2781,10 @@
<ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
</path>
+ <path name="mmap-record headset-mic">
+ <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
<path name="hifi-playback display-port">
<ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
</path>
@@ -2728,6 +2865,10 @@
<path name="incall_music_uplink" />
</path>
+ <path name="incall_music_uplink headset">
+ <path name="incall_music_uplink" />
+ </path>
+
<path name="incall_music_uplink speaker-and-headphones">
<path name="incall_music_uplink" />
</path>
diff --git a/configs/kona/mixer_paths_cdp.xml b/configs/kona/mixer_paths_cdp.xml
new file mode 100644
index 0000000..67041e6
--- /dev/null
+++ b/configs/kona/mixer_paths_cdp.xml
@@ -0,0 +1,2916 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!-- Copyright (c) 2015-2019, The Linux Foundation. All rights reserved. -->
+<!-- -->
+<!-- Redistribution and use in source and binary forms, with or without -->
+<!-- modification, are permitted provided that the following conditions are -->
+<!-- met: -->
+<!-- * Redistributions of source code must retain the above copyright -->
+<!-- notice, this list of conditions and the following disclaimer. -->
+<!-- * Redistributions in binary form must reproduce the above -->
+<!-- copyright notice, this list of conditions and the following -->
+<!-- disclaimer in the documentation and/or other materials provided -->
+<!-- with the distribution. -->
+<!-- * Neither the name of The Linux Foundation nor the names of its -->
+<!-- contributors may be used to endorse or promote products derived -->
+<!-- from this software without specific prior written permission. -->
+<!-- -->
+<!-- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED -->
+<!-- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF -->
+<!-- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT -->
+<!-- ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS -->
+<!-- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR -->
+<!-- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF -->
+<!-- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR -->
+<!-- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, -->
+<!-- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE -->
+<!-- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
+<!-- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -->
+<mixer>
+ <!-- These are the initial mixer settings -->
+ <ctl name="Voice Rx Device Mute" id="0" value="0" />
+ <ctl name="Voice Rx Device Mute" id="1" value="-1" />
+ <ctl name="Voice Rx Device Mute" id="2" value="20" />
+ <ctl name="Voice Tx Mute" id="0" value="0" />
+ <ctl name="Voice Tx Mute" id="1" value="-1" />
+ <ctl name="Voice Tx Mute" id="2" value="500" />
+ <ctl name="Voice Rx Gain" id="0" value="0" />
+ <ctl name="Voice Rx Gain" id="1" value="-1" />
+ <ctl name="Voice Rx Gain" id="2" value="20" />
+ <ctl name="Voice Sidetone Enable" value="0" />
+ <ctl name="Voip Tx Mute" id="0" value="0" />
+ <ctl name="Voip Tx Mute" id="1" value="500" />
+ <ctl name="Voip Rx Gain" id="0" value="0" />
+ <ctl name="Voip Rx Gain" id="1" value="20" />
+ <ctl name="Voip Mode Config" value="12" />
+ <ctl name="Voip Rate Config" value="0" />
+ <ctl name="Voip Evrc Min Max Rate Config" id="0" value="1" />
+ <ctl name="Voip Evrc Min Max Rate Config" id="1" value="4" />
+ <ctl name="Voip Dtx Mode" value="0" />
+ <ctl name="TTY Mode" value="OFF" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="MultiMedia5 Mixer AFE_PCM_TX" value="0" />
+ <ctl name="MultiMedia5 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_4" value="0" />
+ <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_4" value="0" />
+ <ctl name="MultiMedia10 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_4" value="0" />
+ <ctl name="MultiMedia17 Mixer SLIM_7_TX" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia6" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia8" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="0" />
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia1" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia2" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia3" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia5" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia8" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia1" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia2" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia3" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia5" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia8" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia3" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia17 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <ctl name="WSA_CDC_DMA_RX_0 Channels" value="One" />
+ <ctl name="RX_CDC_DMA_RX_0 Channels" value="One" />
+ <ctl name="VI_FEED_TX Channels" value="Two" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="0" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Format" value="UNPACKED" />
+ <ctl name="WSA_CDC_DMA_TX_0 Format" value="UNPACKED" />
+ <ctl name="RX_CDC_DMA_RX_0 Format" value="UNPACKED" />
+ <ctl name="TX_CDC_DMA_TX_3 Format" value="UNPACKED" />
+ <!-- HFP start -->
+ <ctl name="HFP_SLIM7_UL_HL Switch" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Port Mixer SLIM_7_TX" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Port Mixer SLIM_7_TX" value="0" />
+ <!-- HFP end -->
+ <!-- echo reference -->
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="None" />
+ <!-- usb headset -->
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia12" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia13" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia14" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia15" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="0" />
+ <ctl name="MultiMedia1 Mixer AFE_PCM_TX" value="0" />
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
+ <!-- usb headset end -->
+ <!-- fm -->
+ <ctl name="Tert MI2S LOOPBACK Volume" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0 Port Mixer TERT_MI2S_TX" value="0" />
+ <ctl name="WSA_CDC_DMA_RX_0_DL_HL Switch" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0 Port Mixer TERT_MI2S_TX" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0_DL_HL Switch" value="0" />
+ <ctl name="MultiMedia1 Mixer TERT_MI2S_TX" value="0" />
+ <ctl name="MultiMedia2 Mixer TERT_MI2S_TX" value="0" />
+ <!-- fm end -->
+
+ <!-- Multimode Voice1 -->
+ <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode1" value="0" />
+ <!-- Multimode Voice1 BTSCO -->
+ <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="0" />
+ <!-- Multimode Voice1 USB headset -->
+ <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer AFE_PCM_TX_MMode1" value="0" />
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="0" />
+ <!-- Multimode Voice1 Display-Port -->
+ <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode1" value="0" />
+ <!-- Miltimode Voice1 end-->
+
+ <!-- Multimode Voice2 -->
+ <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode2" value="0" />
+ <!-- Multimode Voice2 BTSCO -->
+ <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="0" />
+ <!-- Multimode Voice2 USB headset -->
+ <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="VoiceMMode2_Tx Mixer AFE_PCM_TX_MMode2" value="0" />
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="0" />
+ <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="0" />
+ <!-- Multimode Voice2 Display-Port -->
+ <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode2" value="0" />
+ <!-- Multimode Voice2 end-->
+
+ <!-- Voice external ec. reference -->
+ <ctl name="VOC_EXT_EC MUX" value="NONE" />
+ <!-- Voice external ec. reference end -->
+
+ <!-- RT Proxy Cal -->
+ <ctl name="RT_PROXY_1_RX SetCalMode" value="CAL_MODE_NONE" />
+ <ctl name="RT_PROXY_1_TX SetCalMode" value="CAL_MODE_NONE" />
+ <!-- RT Proxy Cal end -->
+
+ <!-- Incall Recording -->
+ <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="0" />
+ <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="0" />
+ <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="0" />
+ <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="0" />
+ <!-- Incall Recording End -->
+
+ <!-- Incall Music -->
+ <ctl name="Incall_Music Audio Mixer MultiMedia2" value="0" />
+ <ctl name="Incall_Music Audio Mixer MultiMedia9" value="0" />
+ <!-- Incall Music End -->
+
+ <!-- compress-voip-call start -->
+ <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="0" />
+ <ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
+ <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
+ <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
+ <ctl name="Voip_Tx Mixer AFE_PCM_TX_Voip" value="0" />
+ <ctl name="USB_AUDIO_RX_Voice Mixer Voip" value="0" />
+ <ctl name="Voip_Tx Mixer USB_AUDIO_TX_Voip" value="0" />
+ <!-- compress-voip-call end-->
+
+ <!-- Audio BTSCO -->
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="0" />
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia6" value="0" />
+ <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
+ <!-- audio record compress-->
+ <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia8 Mixer AFE_PCM_TX" value="0" />
+ <!-- audio record compress end-->
+
+ <!-- split a2dp -->
+ <ctl name="BT SampleRate" value="KHZ_8" />
+ <ctl name="AFE Input Channels" value="Zero" />
+ <ctl name="SLIM7_RX ADM Channels" value="Zero" />
+ <!-- split a2dp end-->
+
+ <!-- ADSP testfwk -->
+ <ctl name="WSA_CDC_DMA_RX_0_DL_HL Switch" value="0" />
+ <ctl name="RX_CDC_DMA_RX_0_DL_HL Switch" value="0" />
+ <!-- ADSP testfwk end-->
+
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia3" value="0" />
+
+ <!-- Codec controls -->
+ <!-- WSA controls -->
+ <ctl name="WSA RX0 MUX" value="ZERO" />
+ <ctl name="WSA RX1 MUX" value="ZERO" />
+ <ctl name="WSA_RX0 INP0" value="ZERO" />
+ <ctl name="WSA_RX1 INP0" value="ZERO" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="0" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="0" />
+ <ctl name="WSA_RX_0_VI_FB_LCH_MUX" value="ZERO" />
+ <ctl name="WSA_RX_0_VI_FB_RCH_MUX" value="ZERO" />
+ <ctl name="SpkrLeft COMP Switch" value="0" />
+ <ctl name="SpkrRight COMP Switch" value="0" />
+ <ctl name="SpkrLeft BOOST Switch" value="0" />
+ <ctl name="SpkrRight BOOST Switch" value="0" />
+ <ctl name="SpkrLeft VISENSE Switch" value="0" />
+ <ctl name="SpkrRight VISENSE Switch" value="0" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="0" />
+ <ctl name="SpkrRight SWR DAC_Port Switch" value="0" />
+
+ <!-- RX Controls -->
+ <ctl name="RX_MACRO RX0 MUX" value="ZERO" />
+ <ctl name="RX_MACRO RX1 MUX" value="ZERO" />
+ <ctl name="RX_CDC_DMA_RX_0 Channels" value="One" />
+ <ctl name="RX INT0_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT0_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT0_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP1" value="ZERO" />
+ <ctl name="RX INT1_1 MIX1 INP2" value="ZERO" />
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia1" value="0" />
+ <ctl name="RX INT0 DEM MUX" value="NORMAL_DSM_OUT" />
+ <ctl name="RX INT1 DEM MUX" value="NORMAL_DSM_OUT" />
+ <ctl name="RX_COMP1 Switch" value="0" />
+ <ctl name="RX_COMP2 Switch" value="0" />
+ <ctl name="HPHL_COMP Switch" value="0" />
+ <ctl name="HPHR_COMP Switch" value="0" />
+ <ctl name="EAR_RDAC Switch" value="0" />
+ <ctl name="HPHL_RDAC Switch" value="0" />
+ <ctl name="HPHR_RDAC Switch" value="0" />
+ <ctl name="AUX_RDAC Switch" value="0" />
+ <ctl name="RDAC3_MUX" value="ZERO" />
+ <ctl name="RX_EAR Mode" value="OFF" />
+
+ <!-- TX Controls -->
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="ADC1_MIXER Switch" value="0" />
+ <ctl name="ADC2_MIXER Switch" value="0" />
+ <ctl name="ADC2 MUX" value="ZERO" />
+ <ctl name="ADC3_MIXER Switch" value="0" />
+ <ctl name="ADC3 MUX" value="ZERO" />
+ <ctl name="ADC4_MIXER Switch" value="0" />
+ <ctl name="ADC4 MUX" value="ZERO" />
+ <ctl name="TX_AIF1_CAP Mixer DEC0" value="0" />
+ <ctl name="TX_AIF1_CAP Mixer DEC1" value="0" />
+ <ctl name="TX_AIF1_CAP Mixer DEC2" value="0" />
+ <ctl name="TX_AIF1_CAP Mixer DEC3" value="0" />
+ <ctl name="TX_AIF1_CAP Mixer DEC4" value="0" />
+ <ctl name="TX_AIF1_CAP Mixer DEC5" value="0" />
+ <ctl name="TX_AIF1_CAP Mixer DEC6" value="0" />
+ <ctl name="TX_AIF1_CAP Mixer DEC7" value="0" />
+ <ctl name="TX DEC0 MUX" value="MSM_DMIC" />
+ <ctl name="TX DMIC MUX0" value="ZERO" />
+ <ctl name="TX SMIC MUX0" value="ZERO" />
+ <ctl name="TX DEC1 MUX" value="MSM_DMIC" />
+ <ctl name="TX DMIC MUX1" value="ZERO" />
+ <ctl name="TX SMIC MUX1" value="ZERO" />
+ <ctl name="TX DEC2 MUX" value="MSM_DMIC" />
+ <ctl name="TX DMIC MUX2" value="ZERO" />
+ <ctl name="TX SMIC MUX2" value="ZERO" />
+ <ctl name="TX DEC3 MUX" value="MSM_DMIC" />
+ <ctl name="TX DMIC MUX3" value="ZERO" />
+ <ctl name="TX SMIC MUX3" value="ZERO" />
+ <ctl name="TX DEC4 MUX" value="MSM_DMIC" />
+ <ctl name="TX DMIC MUX4" value="ZERO" />
+ <ctl name="TX SMIC MUX4" value="ZERO" />
+ <ctl name="TX DEC5 MUX" value="MSM_DMIC" />
+ <ctl name="TX DMIC MUX5" value="ZERO" />
+ <ctl name="TX SMIC MUX5" value="ZERO" />
+ <ctl name="TX DEC6 MUX" value="MSM_DMIC" />
+ <ctl name="TX DMIC MUX6" value="ZERO" />
+ <ctl name="TX SMIC MUX6" value="ZERO" />
+ <ctl name="TX DEC7 MUX" value="MSM_DMIC" />
+ <ctl name="TX DMIC MUX7" value="ZERO" />
+ <ctl name="TX SMIC MUX7" value="ZERO" />
+
+ <!-- Volume controls -->
+ <ctl name="WSA_RX0 Digital Volume" value="84" />
+ <ctl name="WSA_RX1 Digital Volume" value="84" />
+ <ctl name="RX_RX0 Digital Volume" value="84" />
+ <ctl name="RX_RX1 Digital Volume" value="84" />
+ <ctl name="RX_RX2 Digital Volume" value="84" />
+ <ctl name="HPHL Volume" value="20" />
+ <ctl name="HPHR Volume" value="20" />
+ <ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
+
+ <ctl name="TX_DEC0 Volume" value="102" />
+ <ctl name="TX_DEC1 Volume" value="102" />
+ <ctl name="TX_DEC2 Volume" value="102" />
+ <ctl name="TX_DEC3 Volume" value="102" />
+ <ctl name="TX_DEC4 Volume" value="84" />
+ <ctl name="TX_DEC5 Volume" value="84" />
+ <ctl name="TX_DEC6 Volume" value="84" />
+ <ctl name="TX_DEC7 Volume" value="84" />
+
+ <ctl name="ADC1 Volume" value="12" />
+ <ctl name="ADC2 Volume" value="12" />
+ <ctl name="ADC3 Volume" value="12" />
+
+ <!-- Compander controls -->
+ <ctl name="WSA_COMP1 Switch" value="0" />
+ <ctl name="WSA_COMP2 Switch" value="0" />
+ <ctl name="COMP7 Switch" value="0" />
+ <ctl name="COMP8 Switch" value="0" />
+
+ <!-- Headphone class-H mode -->
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
+
+ <!-- IIR/voice anc -->
+ <ctl name="IIR0 Band1" id ="0" value="268435456" />
+ <ctl name="IIR0 Band1" id ="1" value="0" />
+ <ctl name="IIR0 Band1" id ="2" value="0" />
+ <ctl name="IIR0 Band1" id ="3" value="0" />
+ <ctl name="IIR0 Band1" id ="4" value="0" />
+ <ctl name="IIR0 Band2" id ="0" value="268435456" />
+ <ctl name="IIR0 Band2" id ="1" value="0" />
+ <ctl name="IIR0 Band2" id ="2" value="0" />
+ <ctl name="IIR0 Band2" id ="3" value="0" />
+ <ctl name="IIR0 Band2" id ="4" value="0" />
+ <ctl name="IIR0 Band3" id ="0" value="268435456" />
+ <ctl name="IIR0 Band3" id ="1" value="0" />
+ <ctl name="IIR0 Band3" id ="2" value="0" />
+ <ctl name="IIR0 Band3" id ="3" value="0" />
+ <ctl name="IIR0 Band3" id ="4" value="0" />
+ <ctl name="IIR0 Band4" id ="0" value="268435456" />
+ <ctl name="IIR0 Band4" id ="1" value="0" />
+ <ctl name="IIR0 Band4" id ="2" value="0" />
+ <ctl name="IIR0 Band4" id ="3" value="0" />
+ <ctl name="IIR0 Band4" id ="4" value="0" />
+ <ctl name="IIR0 Band5" id ="0" value="268435456" />
+ <ctl name="IIR0 Band5" id ="1" value="0" />
+ <ctl name="IIR0 Band5" id ="2" value="0" />
+ <ctl name="IIR0 Band5" id ="3" value="0" />
+ <ctl name="IIR0 Band5" id ="4" value="0" />
+ <ctl name="IIR0 Enable Band1" value="0" />
+ <ctl name="IIR0 Enable Band2" value="0" />
+ <ctl name="IIR0 Enable Band3" value="0" />
+ <ctl name="IIR0 Enable Band4" value="0" />
+ <ctl name="IIR0 Enable Band5" value="0" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="IIR0 INP0 MUX" value="ZERO" />
+ <ctl name="IIR0 INP1 MUX" value="ZERO" />
+ <ctl name="IIR0 INP2 MUX" value="ZERO" />
+ <ctl name="IIR0 INP3 MUX" value="ZERO" />
+
+ <!-- vbat related data -->
+ <ctl name="GSM mode Enable" value="OFF" />
+ <ctl name="WSA_Softclip0 Enable" value="0" />
+ <ctl name="WSA_Softclip1 Enable" value="0" />
+ <ctl name="WSA_RX INT0 VBAT WSA RX0 VBAT Enable" value="0" />
+ <ctl name="WSA_RX INT1 VBAT WSA RX1 VBAT Enable" value="0" />
+
+ <!-- Codec controls end -->
+
+ <!-- defaults for mmap record -->
+ <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_4" value="0" />
+ <ctl name="MultiMedia16 Mixer SLIM_7_TX" value="0" />
+ <ctl name="MultiMedia16 Mixer TERT_MI2S_TX" value="0" />
+ <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="0" />
+
+ <!-- These are audio route (FE to BE) specific mixer settings -->
+ <path name="gsm-mode">
+ <ctl name="GSM mode Enable" value="ON" />
+ </path>
+
+ <path name="echo-reference">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="WSA_CDC_DMA_RX_0" />
+ </path>
+
+ <path name="echo-reference headphones">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="RX_CDC_DMA_RX_0" />
+ </path>
+
+ <path name="echo-reference headset">
+ <path name="echo-reference headphones" />
+ </path>
+
+ <path name="echo-reference display-port">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="DISPLAY_PORT" />
+ </path>
+
+ <path name="echo-reference headphones-44.1">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="RX_CDC_DMA_RX_0" />
+ </path>
+
+ <path name="echo-reference-voip">
+ <ctl name="AUDIO_REF_EC_UL10 MUX" value="WSA_CDC_DMA_RX_0" />
+ </path>
+
+ <path name="echo-reference-voip headphones">
+ <ctl name="AUDIO_REF_EC_UL10 MUX" value="RX_CDC_DMA_RX_0" />
+ </path>
+
+ <path name="deep-buffer-playback">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-protected">
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-display-port">
+ <path name="deep-buffer-playback display-port" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="deep-buffer-playback bt-sco" />
+ </path>
+
+ <path name="deep-buffer-playback bt-sco-swb">
+ <path name="deep-buffer-playback bt-sco" />
+ </path>
+
+ <path name="deep-buffer-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-usb-headphones">
+ <path name="deep-buffer-playback usb-headphones" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback headset">
+ <path name="deep-buffer-playback headphones" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-headphones">
+ <path name="deep-buffer-playback headphones" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-bt-sco">
+ <path name="deep-buffer-playback bt-sco" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-bt-sco-wb">
+ <path name="deep-buffer-playback bt-sco-wb" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-bt-sco-swb">
+ <path name="deep-buffer-playback bt-sco-swb" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="low-latency-playback">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback speaker-protected">
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="low-latency-playback bt-sco" />
+ </path>
+
+ <path name="low-latency-playback bt-sco-swb">
+ <path name="low-latency-playback bt-sco" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-display-port">
+ <path name="low-latency-playback display-port" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-usb-headphones">
+ <path name="low-latency-playback usb-headphones" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="low-latency-playback headset">
+ <path name="low-latency-playback headphones" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-headphones">
+ <path name="low-latency-playback headphones" />
+ <path name="low-latency-playback" />
+ </path>
+ <path name="low-latency-playback resume">
+ <ctl name="MultiMedia5_RX QOS Vote" value="Enable" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-bt-sco">
+ <path name="low-latency-playback bt-sco" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-bt-sco-wb">
+ <path name="low-latency-playback bt-sco-wb" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-bt-sco-swb">
+ <path name="low-latency-playback bt-sco-swb" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="audio-ull-playback">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback speaker-protected">
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="audio-ull-playback headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback headset">
+ <path name="audio-ull-playback headphones" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-headphones">
+ <path name="audio-ull-playback" />
+ <path name="audio-ull-playback headphones" />
+ </path>
+
+ <path name="audio-ull-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-ull-playback bt-sco" />
+ </path>
+
+ <path name="audio-ull-playback bt-sco-swb">
+ <path name="audio-ull-playback bt-sco" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-display-port">
+ <path name="audio-ull-playback display-port" />
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="audio-ull-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-bt-sco">
+ <path name="audio-ull-playback bt-sco" />
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-bt-sco-wb">
+ <path name="audio-ull-playback bt-sco-wb" />
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-bt-sco-swb">
+ <path name="audio-ull-playback bt-sco-swb" />
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="multi-channel-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="multi-channel-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="compress-offload-playback">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback speaker-protected">
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="silence-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia9" value="1" />
+ </path>
+
+ <path name="compress-offload-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback bt-sco-swb">
+ <path name="compress-offload-playback bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-display-port">
+ <path name="compress-offload-playback display-port" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-usb-headphones">
+ <path name="compress-offload-playback usb-headphones" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback headset">
+ <path name="compress-offload-playback headphones" />
+ </path>
+
+ <path name="compress-offload-playback headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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-playback speaker-and-bt-sco">
+ <path name="compress-offload-playback bt-sco" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback bt-sco-wb" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback bt-sco-swb" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="compress-offload-playback2">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback2 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-sco-swb">
+ <path name="compress-offload-playback2 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-display-port">
+ <path name="compress-offload-playback2 display-port" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback2 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-usb-headphones">
+ <path name="compress-offload-playback2 usb-headphones" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback2 headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 headset">
+ <path name="compress-offload-playback2 headphones" />
+ </path>
+
+ <path name="compress-offload-playback2 headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
+
+ </path>
+
+ <path name="compress-offload-playback2 headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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-playback2 speaker-and-bt-sco">
+ <path name="compress-offload-playback2 bt-sco" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback2 bt-sco-wb" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback2 bt-sco-swb" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback3">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback3 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-sco-swb">
+ <path name="compress-offload-playback3 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-display-port">
+ <path name="compress-offload-playback3 display-port" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback3 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-usb-headphones">
+ <path name="compress-offload-playback3 usb-headphones" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback3 headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 headset">
+ <path name="compress-offload-playback3 headphones" />
+ </path>
+
+ <path name="compress-offload-playback3 headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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-playback3 speaker-and-bt-sco">
+ <path name="compress-offload-playback3 bt-sco" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback3 bt-sco-wb" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback3 bt-sco-swb" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback4">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback4 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-sco-swb">
+ <path name="compress-offload-playback4 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-display-port">
+ <path name="compress-offload-playback4 display-port" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+
+ <path name="compress-offload-playback4 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-usb-headphones">
+ <path name="compress-offload-playback4 usb-headphones" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback4 headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 headset">
+ <path name="compress-offload-playback4 headphones" />
+ </path>
+
+ <path name="compress-offload-playback4 headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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-playback4 speaker-and-bt-sco">
+ <path name="compress-offload-playback4 bt-sco" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback4 bt-sco-wb" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback4 bt-sco-swb" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback5">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback5 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-sco-swb">
+ <path name="compress-offload-playback5 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-display-port">
+ <path name="compress-offload-playback5 display-port" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback5 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-usb-headphones">
+ <path name="compress-offload-playback5 usb-headphones" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback5 headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 headset">
+ <path name="compress-offload-playback5 headphones" />
+ </path>
+
+ <path name="compress-offload-playback5 headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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-playback5 speaker-and-bt-sco">
+ <path name="compress-offload-playback5 bt-sco" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback5 bt-sco-wb" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback5 bt-sco-swb" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback6">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback6 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-sco-swb">
+ <path name="compress-offload-playback6 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-display-port">
+ <path name="compress-offload-playback6 display-port" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback6 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-usb-headphones">
+ <path name="compress-offload-playback6 usb-headphones" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback6 headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 headset">
+ <path name="compress-offload-playback6 headphones" />
+ </path>
+
+ <path name="compress-offload-playback6 headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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-playback6 speaker-and-bt-sco">
+ <path name="compress-offload-playback6 bt-sco" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback6 bt-sco-wb" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback6 bt-sco-swb" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback7">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback7 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-sco-swb">
+ <path name="compress-offload-playback7 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-display-port">
+ <path name="compress-offload-playback7 display-port" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback7 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-usb-headphones">
+ <path name="compress-offload-playback7 usb-headphones" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback7 headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 headset">
+ <path name="compress-offload-playback7 headphones" />
+ </path>
+
+ <path name="compress-offload-playback7 headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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-playback7 speaker-and-bt-sco">
+ <path name="compress-offload-playback7 bt-sco" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback7 bt-sco-wb" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback7 bt-sco-swb" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback8">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback8 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-sco-swb">
+ <path name="compress-offload-playback8 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-display-port">
+ <path name="compress-offload-playback8 display-port" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback8 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-usb-headphones">
+ <path name="compress-offload-playback8 usb-headphones" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback8 headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 headset">
+ <path name="compress-offload-playback8 headphones" />
+ </path>
+
+ <path name="compress-offload-playback8 headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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-playback8 speaker-and-bt-sco">
+ <path name="compress-offload-playback8 bt-sco" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback8 bt-sco-wb" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback8 bt-sco-swb" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback9">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-offload-playback9 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-sco-swb">
+ <path name="compress-offload-playback9 bt-sco" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-display-port">
+ <path name="compress-offload-playback9 display-port" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="compress-offload-playback9 afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-usb-headphones">
+ <path name="compress-offload-playback9 usb-headphones" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="compress-offload-playback9 headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 headset">
+ <path name="compress-offload-playback9 headphones" />
+ </path>
+
+ <path name="compress-offload-playback9 headphones-44.1">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 headphones-dsd">
+ <ctl name="RX_CDC_DMA_RX_0 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="compress-offload-playback9 speaker-and-bt-sco">
+ <path name="compress-offload-playback9 bt-sco" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-bt-sco-wb">
+ <path name="compress-offload-playback9 bt-sco-wb" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-bt-sco-swb">
+ <path name="compress-offload-playback9 bt-sco-swb" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="audio-record">
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="audio-record usb-headset-mic">
+ <ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="audio-record bt-sco">
+ <ctl name="MultiMedia1 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="audio-record bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-record bt-sco" />
+ </path>
+
+ <path name="audio-record bt-sco-swb">
+ <path name="audio-record bt-sco" />
+ </path>
+
+ <path name="audio-record headset-mic">
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
+ <path name="audio-record capture-fm">
+ <ctl name="MultiMedia1 Mixer TERT_MI2S_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress">
+ <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="audio-record-compress bt-sco">
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-record-compress bt-sco" />
+ </path>
+
+ <path name="audio-record-compress bt-sco-swb">
+ <path name="audio-record-compress bt-sco" />
+ </path>
+
+ <path name="audio-record-compress usb-headset-mic">
+ <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress2">
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="audio-record-compress2 bt-sco">
+ <ctl name="MultiMedia17 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-record-compress2 bt-sco" />
+ </path>
+
+ <path name="audio-record-compress2 bt-sco-swb">
+ <path name="audio-record-compress2 bt-sco" />
+ </path>
+
+ <path name="audio-record-compress2 usb-headset-mic">
+ <ctl name="MultiMedia17 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress2 headset-mic">
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
+ <path name="low-latency-record">
+ <ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="low-latency-record bt-sco">
+ <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="low-latency-record bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="low-latency-record bt-sco" />
+ </path>
+
+ <path name="low-latency-record bt-sco-swb">
+ <path name="low-latency-record bt-sco" />
+ </path>
+
+ <path name="low-latency-record usb-headset-mic">
+ <ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="low-latency-record capture-fm">
+ <ctl name="MultiMedia8 Mixer TERT_MI2S_TX" value="1" />
+ </path>
+
+ <path name="fm-virtual-record capture-fm">
+ <ctl name="MultiMedia2 Mixer TERT_MI2S_TX" value="1" />
+ </path>
+
+ <path name="play-fm">
+ <ctl name="Tert MI2S LOOPBACK Volume" value="1" />
+ <ctl name="WSA_CDC_DMA_RX_0 Port Mixer TERT_MI2S_TX" value="1" />
+ <ctl name="WSA_CDC_DMA_RX_0_DL_HL Switch" value="1" />
+ </path>
+
+ <path name="play-fm headphones">
+ <ctl name="Tert MI2S LOOPBACK Volume" value="1" />
+ <ctl name="RX_CDC_DMA_RX_0 Port Mixer TERT_MI2S_TX" value="1" />
+ <ctl name="RX_CDC_DMA_RX_0_DL_HL Switch" value="1" />
+ </path>
+
+ <path name="play-fm headset">
+ <path name="play-fm headphones" />
+ </path>
+
+ <path name="incall-rec-uplink">
+ <ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
+ </path>
+
+ <path name="incall-rec-uplink bt-sco">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink bt-sco-wb">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink bt-sco-swb">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink usb-headset-mic">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink afe-proxy">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink headset-mic">
+ <path name="incall-rec-uplink" />
+ </path>
+
+ <path name="incall-rec-uplink-compress">
+ <ctl name="MultiMedia8 Mixer VOC_REC_UL" value="1" />
+ </path>
+
+ <path name="incall-rec-uplink-compress bt-sco">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-compress bt-sco-wb">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-compress bt-sco-swb">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-compress usb-headset-mic">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-compress afe-proxy">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-compress headset-mic">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink">
+ <ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
+ </path>
+
+ <path name="incall-rec-downlink bt-sco">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink bt-sco-wb">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink bt-sco-swb">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink usb-headset-mic">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink afe-proxy">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink headset-mic">
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-downlink-compress">
+ <ctl name="MultiMedia8 Mixer VOC_REC_DL" value="1" />
+ </path>
+
+ <path name="incall-rec-downlink-compress bt-sco">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink-compress bt-sco-wb">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink-compress bt-sco-swb">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink-compress usb-headset-mic">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink-compress afe-proxy">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-downlink-compress headset-mic">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink">
+ <path name="incall-rec-uplink" />
+ <path name="incall-rec-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink bt-sco">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink bt-sco-wb">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink bt-sco-swb">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink usb-headset-mic">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink afe-proxy">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink headset-mic">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress">
+ <path name="incall-rec-uplink-compress" />
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress bt-sco">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress bt-sco-wb">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress bt-sco-swb">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress usb-headset-mic">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress afe-proxy">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="incall-rec-uplink-and-downlink-compress headset-mic">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
+ <path name="hfp-sco">
+ </path>
+
+ <path name="hfp-sco headphones">
+ </path>
+
+ <path name="hfp-sco-wb">
+ <path name="hfp-sco" />
+ </path>
+
+ <path name="hfp-sco-wb headphones">
+ <path name="hfp-sco headphones" />
+ </path>
+
+ <path name="compress-voip-call">
+ <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call bt-a2dp">
+ <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call headphones">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
+ </path>
+
+
+ <path name="compress-voip-call bt-sco">
+ <ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="compress-voip-call bt-sco" />
+ </path>
+
+ <path name="compress-voip-call bt-sco-swb">
+ <path name="compress-voip-call bt-sco" />
+ </path>
+
+ <path name="compress-voip-call afe-proxy">
+ <ctl name="AFE_PCM_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer AFE_PCM_TX_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call usb-headphones">
+ <ctl name="USB_AUDIO_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
+ </path>
+
+ <path name="compress-voip-call usb-headset">
+ <ctl name="USB_AUDIO_RX_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer USB_AUDIO_TX_Voip" value="1" />
+ </path>
+
+ <path name="voicemmode1-call">
+ <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call headphones">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call bt-sco">
+ <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="voicemmode1-call bt-sco" />
+ </path>
+
+ <path name="voicemmode1-call bt-sco-swb">
+ <path name="voicemmode1-call bt-sco" />
+ </path>
+
+ <path name="voicemmode1-call afe-proxy">
+ <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer AFE_PCM_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call usb-headphones">
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call usb-headset">
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call display-port-and-usb-headset-mic">
+ <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer USB_AUDIO_TX_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode1-call display-port">
+ <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="1" />
+ </path>
+
+ <path name="voicemmode2-call">
+ <ctl name="WSA_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call headphones">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_4_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call bt-sco">
+ <ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="voicemmode2-call bt-sco" />
+ </path>
+
+ <path name="voicemmode2-call bt-sco-swb">
+ <path name="voicemmode2-call bt-sco" />
+ </path>
+
+ <path name="voicemmode2-call afe-proxy">
+ <ctl name="AFE_PCM_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer AFE_PCM_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call usb-headphones">
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call usb-headset">
+ <ctl name="USB_AUDIO_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call display-port-and-usb-headset-mic">
+ <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer USB_AUDIO_TX_MMode2" value="1" />
+ </path>
+
+ <path name="voicemmode2-call display-port">
+ <ctl name="DISPLAY_PORT_RX_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="1" />
+ </path>
+
+ <!-- VoIP Rx settings -->
+ <path name="audio-playback-voip">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="audio-playback-voip headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="audio-playback-voip bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="audio-playback-voip bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-playback-voip bt-sco" />
+ </path>
+
+ <path name="audio-playback-voip bt-sco-swb">
+ <path name="audio-playback-voip bt-sco" />
+ </path>
+
+ <path name="audio-playback-voip bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="audio-playback-voip afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="audio-playback-voip usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="audio-playback-voip usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="audio-playback-voip display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="audio-playback-voip speaker-and-display-port">
+ <path name="audio-playback-voip display-port" />
+ <path name="audio-playback-voip" />
+ </path>
+
+ <!-- VoIP Tx settings -->
+ <path name="audio-record-voip">
+ <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="audio-record-voip usb-headset-mic">
+ <ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="audio-record-voip bt-sco">
+ <ctl name="MultiMedia10 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="audio-record-voip bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-record-voip bt-sco" />
+ </path>
+
+ <path name="audio-record-voip bt-sco-swb">
+ <path name="audio-record-voip bt-sco" />
+ </path>
+
+ <path name="audio-record-voip headset-mic">
+ <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
+ <path name="spkr-rx-calib">
+ <ctl name="WSA_CDC_DMA_RX_0_DL_HL Switch" value="1" />
+ </path>
+
+ <path name="spkr-vi-record">
+ </path>
+
+ <!-- These are actual sound device specific mixer settings -->
+ <path name="amic1">
+ <ctl name="TX DEC1 MUX" value="SWR_MIC" />
+ <ctl name="TX SMIC MUX1" value="ADC0" />
+ <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
+ <ctl name="ADC1_MIXER Switch" value="1" />
+ </path>
+
+ <path name="amic2">
+ <ctl name="TX DEC5 MUX" value="SWR_MIC" />
+ <ctl name="TX SMIC MUX5" value="ADC1" />
+ <ctl name="TX_CDC_DMA_TX_4 Channels" value="One" />
+ <ctl name="TX_AIF2_CAP Mixer DEC5" value="1" />
+ <ctl name="ADC2_MIXER Switch" value="1" />
+ <ctl name="ADC2 MUX" value="INP2" />
+ </path>
+
+ <path name="amic3">
+ <ctl name="TX DEC0 MUX" value="SWR_MIC" />
+ <ctl name="TX SMIC MUX0" value="ADC1" />
+ <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="ADC2_MIXER Switch" value="1" />
+ <ctl name="ADC2 MUX" value="INP3" />
+ </path>
+
+ <path name="amic4">
+ <ctl name="TX DEC2 MUX" value="SWR_MIC" />
+ <ctl name="TX SMIC MUX2" value="ADC2" />
+ <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
+ <ctl name="ADC3_MIXER Switch" value="1" />
+ <ctl name="ADC3 MUX" value="INP4" />
+ </path>
+
+ <path name="amic5">
+ <ctl name="TX DEC3 MUX" value="SWR_MIC" />
+ <ctl name="TX SMIC MUX3" value="ADC3" />
+ <ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
+ <ctl name="ADC4_MIXER Switch" value="1" />
+ <ctl name="ADC4 MUX" value="INP5" />
+ </path>
+
+ <path name="dmic1">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="TX DMIC MUX0" value="DMIC0" />
+ </path>
+
+ <path name="dmic2">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="TX DMIC MUX0" value="DMIC1" />
+ </path>
+
+ <path name="dmic3">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="TX DMIC MUX0" value="DMIC2" />
+ </path>
+
+ <path name="dmic4">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="TX DMIC MUX0" value="DMIC3" />
+ </path>
+
+ <path name="dmic5">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="TX DMIC MUX0" value="DMIC4" />
+ </path>
+
+ <path name="dmic6">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="TX DMIC MUX0" value="DMIC5" />
+ </path>
+
+ <path name="speaker">
+ <ctl name="WSA_CDC_DMA_RX_0 Channels" value="Two" />
+ <ctl name="WSA RX0 MUX" value="AIF1_PB" />
+ <ctl name="WSA RX1 MUX" value="AIF1_PB" />
+ <ctl name="WSA_RX0 INP0" value="RX0" />
+ <ctl name="WSA_RX1 INP0" value="RX1" />
+ <ctl name="WSA_COMP1 Switch" value="1" />
+ <ctl name="WSA_COMP2 Switch" value="1" />
+ <ctl name="SpkrLeft COMP Switch" value="1" />
+ <ctl name="SpkrLeft BOOST Switch" value="1" />
+ <ctl name="SpkrLeft VISENSE Switch" value="1" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ <ctl name="SpkrRight COMP Switch" value="1" />
+ <ctl name="SpkrRight BOOST Switch" value="1" />
+ <ctl name="SpkrRight VISENSE Switch" value="1" />
+ <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+ </path>
+
+ <path name="speaker-mono">
+ <ctl name="WSA_CDC_DMA_RX_0 Channels" value="One" />
+ <ctl name="WSA RX0 MUX" value="AIF1_PB" />
+ <ctl name="WSA_RX0 INP0" value="RX0" />
+ <ctl name="WSA_COMP1 Switch" value="1" />
+ <ctl name="SpkrLeft COMP Switch" value="1" />
+ <ctl name="SpkrLeft BOOST Switch" value="1" />
+ <ctl name="SpkrLeft VISENSE Switch" value="1" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ </path>
+
+ <path name="speaker-mono-2">
+ <ctl name="WSA_CDC_DMA_RX_0 Channels" value="One" />
+ <ctl name="WSA RX1 MUX" value="AIF1_PB" />
+ <ctl name="WSA_RX1 INP0" value="RX1" />
+ <ctl name="WSA_COMP2 Switch" value="1" />
+ <ctl name="SpkrRight COMP Switch" value="1" />
+ <ctl name="SpkrRight BOOST Switch" value="1" />
+ <ctl name="SpkrRight VISENSE Switch" value="1" />
+ <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
+ </path>
+
+
+ <path name="speaker-hdk">
+ <path name="speaker-mono" />
+ </path>
+
+ <path name="speaker-fluid">
+ <path name="speaker-mono" />
+ </path>
+
+ <path name="speaker-liquid">
+ <path name="speaker" />
+ </path>
+
+ <path name="sidetone-iir">
+ <ctl name="IIR0 Enable Band1" value="1" />
+ <ctl name="IIR0 Enable Band2" value="1" />
+ <ctl name="IIR0 Enable Band3" value="1" />
+ <ctl name="IIR0 Enable Band4" value="1" />
+ <ctl name="IIR0 Enable Band5" value="1" />
+ </path>
+
+ <path name="sidetone-headphones">
+ <path name="sidetone-iir" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="IIR0 INP0 MUX" value="DEC0" />
+ <ctl name="RX INT0 MIX2 INP" value="SRC0" />
+ <ctl name="RX INT1 MIX2 INP" value="SRC0" />
+ </path>
+
+ <path name="sidetone-handset">
+ <path name="sidetone-iir" />
+ <ctl name="IIR0 INP0 Volume" value="54" />
+ <ctl name="IIR0 INP0 MUX" value="DEC0" />
+ <ctl name="RX INT2 MIX2 INP" value="SRC0" />
+ <ctl name="WSA_RX0 INT0 SIDETONE MIX" value="SRC0" />
+ </path>
+
+ <path name="afe-sidetone">
+ <ctl name="Voice Sidetone Enable" value="1" />
+ </path>
+
+ <path name="speaker-mic">
+ <path name="dmic5" />
+ </path>
+
+ <path name="speaker-mic-liquid">
+ <path name="dmic5" />
+ </path>
+
+ <path name="speaker-mic-sbc">
+ </path>
+
+ <path name="speaker-protected">
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="1" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
+ <ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
+ <path name="speaker" />
+ <ctl name="VI_FEED_TX Channels" value="Two" />
+ <ctl name="WSA_RX_0_VI_FB_LCH_MUX" value="WSA_CDC_DMA_TX_0" />
+ <ctl name="WSA_RX_0_VI_FB_RCH_MUX" value="WSA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="voice-speaker-protected">
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="1" />
+ <ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
+ <path name="speaker-mono" />
+ <ctl name="VI_FEED_TX Channels" value="One" />
+ <ctl name="WSA_RX_0_VI_FB_LCH_MUX" value="WSA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="voice-speaker-2-protected">
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
+ <ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
+ <path name="speaker-mono-2" />
+ <ctl name="VI_FEED_TX Channels" value="One" />
+ <ctl name="WSA_RX_0_VI_FB_RCH_MUX" value="WSA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="voice-speaker-stereo-protected">
+ <path name="speaker-protected" />
+ </path>
+
+ <path name="vi-feedback">
+ </path>
+
+ <path name="vi-feedback-mono-1">
+ </path>
+
+ <path name="vi-feedback-mono-2">
+ </path>
+
+ <path name="handset">
+ <ctl name="WSA_CDC_DMA_RX_0 Channels" value="One" />
+ <ctl name="WSA RX0 MUX" value="AIF1_PB" />
+ <ctl name="WSA_RX0 INP0" value="RX0" />
+ <ctl name="WSA_COMP1 Switch" value="1" />
+ <ctl name="SpkrLeft COMP Switch" value="1" />
+ <ctl name="SpkrLeft BOOST Switch" value="1" />
+ <ctl name="SpkrLeft VISENSE Switch" value="1" />
+ <ctl name="SpkrLeft SWR DAC_Port Switch" value="1" />
+ <ctl name="EAR SPKR PA Gain" value="G_6_DB" />
+ </path>
+
+ <path name="handset-mic">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
+ <path name="amic5" />
+ </path>
+
+ <path name="headphones">
+ <ctl name="RX_MACRO RX0 MUX" value="AIF1_PB" />
+ <ctl name="RX_MACRO RX1 MUX" value="AIF1_PB" />
+ <ctl name="RX_CDC_DMA_RX_0 Channels" value="Two" />
+ <ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX1" />
+ <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
+ <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
+ <ctl name="RX_COMP1 Switch" value="1" />
+ <ctl name="RX_COMP2 Switch" value="1" />
+ <ctl name="HPHL_COMP Switch" value="1" />
+ <ctl name="HPHR_COMP Switch" value="1" />
+ <ctl name="HPHL_RDAC Switch" value="1" />
+ <ctl name="HPHR_RDAC Switch" value="1" />
+ </path>
+
+ <path name="headphones-44.1">
+ <path name="headphones" />
+ </path>
+
+ <path name="hph-highquality-mode">
+ </path>
+
+ <path name="hph-lowpower-mode">
+ </path>
+
+ <path name="true-native-mode">
+ </path>
+
+ <path name="line">
+ <path name="headphones" />
+ </path>
+
+ <path name="headset-mic">
+ <path name="amic2" />
+ <ctl name="TX_DEC0 Volume" value="84" />
+ </path>
+
+ <path name="headset-mic-liquid">
+ <path name="amic2" />
+ </path>
+ <path name="voice-handset">
+ <path name="handset" />
+ </path>
+
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
+ <path name="voice-handset-hdk">
+ <path name="handset" />
+ </path>
+
+ <path name="voice-handset-tmus-hdk">
+ <path name="handset" />
+ </path>
+ <path name="voice-speaker">
+ <path name="speaker-mono" />
+ </path>
+
+ <path name="voice-speaker-stereo">
+ <path name="speaker" />
+ </path>
+
+ <path name="voice-speaker-2">
+ <path name="speaker-mono-2" />
+ </path>
+
+ <path name="voice-speaker-hdk">
+ <path name="speaker-mono" />
+ </path>
+
+ <path name="voice-speaker-fluid">
+ <path name="speaker-fluid" />
+ </path>
+
+ <path name="voice-speaker-vbat">
+ <path name="speaker-vbat-mono" />
+ </path>
+
+ <path name="voice-speaker-2-vbat">
+ <path name="speaker-vbat-mono-2" />
+ </path>
+
+ <path name="voice-speaker-mic">
+ <path name="speaker-mic" />
+ </path>
+
+ <path name="voice-headphones">
+ <path name="headphones" />
+ </path>
+
+ <path name="voice-headset">
+ <path name="headphones" />
+ </path>
+
+ <path name="voice-line">
+ <path name="voice-headphones" />
+ </path>
+
+ <path name="voice-headset-mic">
+ <path name="headset-mic" />
+ </path>
+
+ <path name="speaker-and-headphones">
+ <path name="headphones" />
+ <path name="speaker" />
+ </path>
+
+ <path name="speaker-and-line">
+ <path name="speaker-and-headphones" />
+ </path>
+
+ <path name="speaker-and-headphones-liquid">
+ <path name="headphones" />
+ <path name="speaker" />
+ </path>
+
+ <path name="speaker-and-line-liquid">
+ <path name="speaker-and-headphones-liquid" />
+ </path>
+
+ <path name="usb-headphones">
+ </path>
+
+ <path name="usb-headset">
+ </path>
+
+ <path name="afe-proxy">
+ </path>
+
+ <path name="display-port">
+ </path>
+
+ <path name="speaker-and-usb-headphones">
+ <path name="speaker" />
+ <path name="usb-headphones" />
+ </path>
+
+ <path name="speaker-and-display-port">
+ <path name="speaker" />
+ <path name="display-port" />
+ </path>
+
+ <path name="voice-rec-mic">
+ <path name="handset-mic" />
+ </path>
+
+ <path name="camcorder-mic">
+ <path name="handset-mic" />
+ </path>
+
+ <path name="bt-sco-headset">
+ </path>
+
+ <path name="bt-sco-mic">
+ </path>
+
+ <path name="bt-sco-headset-wb">
+ </path>
+
+ <path name="bt-sco-mic-wb">
+ </path>
+
+ <path name="bt-sco-headset-swb">
+ </path>
+
+ <path name="bt-sco-mic-swb">
+ </path>
+
+ <path name="usb-headset-mic">
+ </path>
+
+ <path name="capture-fm">
+ </path>
+
+ <!-- Dual MIC devices -->
+ <path name="handset-dmic-endfire">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="Two" />
+ <path name="amic1" />
+ <path name="amic3" />
+ </path>
+
+ <path name="speaker-dmic-endfire">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="Two" />
+ <path name="amic1" />
+ <path name="amic3" />
+ </path>
+
+ <path name="dmic-endfire">
+ <path name="handset-dmic-endfire" />
+ <ctl name="IIR0 INP0 MUX" value="DEC0" />
+ </path>
+
+ <path name="dmic-endfire-liquid">
+ <path name="handset-dmic-endfire" />
+ <ctl name="IIR0 INP0 MUX" value="DEC7" />
+ </path>
+
+ <path name="handset-stereo-dmic-ef">
+ <path name="handset-dmic-endfire" />
+ </path>
+
+ <path name="speaker-stereo-dmic-ef">
+ <path name="speaker-dmic-endfire" />
+ </path>
+
+ <path name="voice-dmic-ef-tmus">
+ <path name="dmic-endfire" />
+ </path>
+
+ <path name="voice-dmic-ef">
+ <path name="dmic-endfire" />
+ </path>
+
+ <path name="voice-speaker-dmic-ef">
+ <path name="speaker-dmic-endfire" />
+ </path>
+
+ <path name="voice-rec-dmic-ef">
+ <path name="dmic-endfire" />
+ </path>
+
+ <path name="voice-rec-dmic-ef-fluence">
+ <path name="dmic-endfire" />
+ </path>
+
+ <path name="handset-stereo-dmic-ef-liquid">
+ <path name="handset-dmic-endfire" />
+ </path>
+
+ <path name="speaker-stereo-dmic-ef-liquid">
+ <path name="speaker-dmic-endfire" />
+ </path>
+
+ <path name="voice-dmic-ef-liquid-liquid">
+ <path name="dmic-endfire-liquid" />
+ </path>
+
+ <path name="voice-speaker-dmic-ef-liquid">
+ <path name="dmic-endfire-liquid" />
+ </path>
+
+ <path name="voice-rec-dmic-ef-liquid">
+ <path name="dmic-endfire-liquid" />
+ </path>
+
+ <path name="voice-rec-dmic-ef-fluence-liquid">
+ <path name="dmic-endfire-liquid" />
+ </path>
+ <path name="speaker-dmic-broadside">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="Two" />
+ <path name="amic1" />
+ <path name="amic3" />
+ </path>
+
+ <path name="dmic-broadside">
+ <path name="speaker-dmic-broadside" />
+ <ctl name="IIR0 INP0 MUX" value="DEC0" />
+ </path>
+
+ <path name="voice-speaker-dmic-broadside">
+ <path name="dmic-broadside" />
+ </path>
+
+ <!-- Tri MIC devices -->
+ <path name="three-mic">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
+ <path name="amic1" />
+ <path name="amic3" />
+ <path name="amic4" />
+ </path>
+
+ <path name="speaker-tmic">
+ <path name="three-mic" />
+ </path>
+
+ <path name="voice-speaker-tmic">
+ <path name="speaker-tmic" />
+ </path>
+
+ <!-- Quad MIC devices -->
+ <path name="speaker-qmic">
+ <ctl name="TX_CDC_DMA_TX_3 Channels" value="Four" />
+ <path name="amic1" />
+ <path name="amic3" />
+ <path name="amic4" />
+ <path name="amic5" />
+ </path>
+
+ <path name="speaker-qmic-liquid">
+ </path>
+
+ <path name="voice-speaker-qmic">
+ <path name="speaker-qmic" />
+ </path>
+
+ <path name="quad-mic">
+ <path name="speaker-qmic" />
+ </path>
+
+ <path name="voice-speaker-qmic-liquid">
+ <path name="speaker-qmic-liquid" />
+ </path>
+
+ <path name="quad-mic-liquid">
+ <path name="speaker-qmic-liquid" />
+ </path>
+
+ <!-- TTY devices -->
+
+ <path name="tty-headphones">
+ <ctl name="RX_MACRO RX0 MUX" value="AIF1_PB" />
+ <ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
+ <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX_COMP1 Switch" value="1" />
+ <ctl name="HPHL_COMP Switch" value="1" />
+ <ctl name="HPHL_RDAC Switch" value="1" />
+ </path>
+
+ <path name="voice-tty-full-headphones">
+ <ctl name="TTY Mode" value="FULL" />
+ <path name="tty-headphones" />
+ </path>
+
+ <path name="voice-tty-vco-headphones">
+ <ctl name="TTY Mode" value="VCO" />
+ <path name="tty-headphones" />
+ </path>
+
+ <path name="voice-tty-hco-handset">
+ <ctl name="TTY Mode" value="HCO" />
+ <path name="handset" />
+ </path>
+
+
+ <path name="voice-tty-hco-handset-hdk">
+ <ctl name="TTY Mode" value="HCO" />
+ <path name="handset" />
+ </path>
+
+ <path name="voice-tty-full-headset-mic">
+ <path name="amic2" />
+ <ctl name="ADC2 Volume" value="0" />
+ <ctl name="TX_DEC0 Volume" value="84" />
+ </path>
+
+ <path name="voice-tty-hco-headset-mic">
+ <path name="voice-tty-full-headset-mic" />
+ </path>
+
+ <path name="voice-tty-vco-handset-mic">
+ <path name="dmic3" />
+ </path>
+
+ <path name="unprocessed-handset-mic">
+ <path name="handset-mic" />
+ </path>
+
+ <path name="unprocessed-mic">
+ <path name="unprocessed-handset-mic" />
+ </path>
+
+ <path name="unprocessed-stereo-mic">
+ <path name="voice-rec-dmic-ef" />
+ </path>
+
+ <path name="unprocessed-three-mic">
+ <path name="three-mic" />
+ </path>
+
+ <path name="unprocessed-quad-mic">
+ <path name="quad-mic" />
+ </path>
+
+ <path name="unprocessed-headset-mic">
+ <path name="headset-mic" />
+ </path>
+
+ <!-- USB TTY start -->
+
+ <!-- full: both end tty -->
+ <path name="voice-tty-full-usb">
+ <ctl name="TTY Mode" value="FULL" />
+ <path name="usb-headphones" />
+ </path>
+
+ <path name="voice-tty-full-usb-mic">
+ <path name="usb-headset-mic" />
+ </path>
+
+ <!-- vco, in: handset mic use existing, out: tty -->
+ <path name="voice-tty-vco-usb">
+ <ctl name="TTY Mode" value="VCO" />
+ <path name="usb-headphones" />
+ </path>
+
+ <!-- hco, in: tty, out: speaker, use existing handset -->
+ <path name="voice-tty-hco-usb-mic">
+ <path name="voice-tty-full-usb-mic" />
+ </path>
+
+ <!-- USB TTY end -->
+
+ <!-- Added for ADSP testfwk -->
+ <path name="ADSP testfwk">
+ <ctl name="WSA_CDC_DMA_RX_0_DL_HL Switch" value="1" />
+ </path>
+
+ <path name="bt-a2dp">
+ <ctl name="SLIM7_RX ADM Channels" value="Two" />
+ </path>
+
+ <path name="speaker-and-bt-a2dp">
+ <path name="speaker" />
+ <path name="bt-a2dp" />
+ </path>
+
+ <path name="deep-buffer-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="low-latency-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="compress-offload-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia4" value="1" />
+ </path>
+
+ <path name="compress-offload-playback2 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia7" value="1" />
+ </path>
+
+ <path name="compress-offload-playback3 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
+ </path>
+
+ <path name="compress-offload-playback4 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia11" value="1" />
+ </path>
+
+ <path name="compress-offload-playback5 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia12" value="1" />
+ </path>
+
+ <path name="compress-offload-playback6 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia13" value="1" />
+ </path>
+
+ <path name="compress-offload-playback7 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia14" value="1" />
+ </path>
+
+ <path name="compress-offload-playback8 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia15" value="1" />
+ </path>
+
+ <path name="compress-offload-playback9 bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="audio-ull-playback bt-a2dp">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia8" value="1" />
+ </path>
+
+ <path name="deep-buffer-playback speaker-and-bt-a2dp">
+ <path name="deep-buffer-playback bt-a2dp" />
+ <path name="deep-buffer-playback" />
+ </path>
+
+ <path name="compress-offload-playback speaker-and-bt-a2dp">
+ <path name="compress-offload-playback bt-a2dp" />
+ <path name="compress-offload-playback" />
+ </path>
+
+ <path name="low-latency-playback speaker-and-bt-a2dp">
+ <path name="low-latency-playback bt-a2dp" />
+ <path name="low-latency-playback" />
+ </path>
+
+ <path name="compress-offload-playback2 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback2 bt-a2dp" />
+ <path name="compress-offload-playback2" />
+ </path>
+
+ <path name="compress-offload-playback3 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback3 bt-a2dp" />
+ <path name="compress-offload-playback3" />
+ </path>
+
+ <path name="compress-offload-playback4 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback4 bt-a2dp" />
+ <path name="compress-offload-playback4" />
+ </path>
+
+ <path name="compress-offload-playback5 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback5 bt-a2dp" />
+ <path name="compress-offload-playback5" />
+ </path>
+
+ <path name="compress-offload-playback6 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback6 bt-a2dp" />
+ <path name="compress-offload-playback6" />
+ </path>
+
+ <path name="compress-offload-playback7 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback7 bt-a2dp" />
+ <path name="compress-offload-playback7" />
+ </path>
+
+ <path name="compress-offload-playback8 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback8 bt-a2dp" />
+ <path name="compress-offload-playback8" />
+ </path>
+
+ <path name="compress-offload-playback9 speaker-and-bt-a2dp">
+ <path name="compress-offload-playback9 bt-a2dp" />
+ <path name="compress-offload-playback9" />
+ </path>
+
+ <path name="audio-ull-playback speaker-and-bt-a2dp">
+ <path name="audio-ull-playback bt-a2dp" />
+ <path name="audio-ull-playback" />
+ </path>
+
+ <path name="mmap-playback">
+ <ctl name="WSA_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="mmap-playback headphones">
+ <ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="mmap-playback speaker-and-headphones">
+ <path name="mmap-playback" />
+ <path name="mmap-playback headphones" />
+ </path>
+
+ <path name="mmap-playback bt-sco">
+ <ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="mmap-playback bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="mmap-playback bt-sco" />
+ </path>
+
+ <path name="mmap-playback bt-sco-swb">
+ <path name="mmap-playback bt-sco" />
+ </path>
+
+ <path name="mmap-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="mmap-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="mmap-playback usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="mmap-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="mmap-playback hdmi">
+ <ctl name="HDMI Mixer MultiMedia16" value="1" />
+ </path>
+
+ <path name="mmap-playback speaker-and-hdmi">
+ <path name="mmap-playback hdmi" />
+ <path name="mmap-playback" />
+ </path>
+
+ <path name="mmap-playback speaker-and-display-port">
+ <path name="mmap-playback display-port" />
+ <path name="mmap-playback" />
+ </path>
+
+ <path name="mmap-playback speaker-and-usb-headphones">
+ <path name="mmap-playback usb-headphones" />
+ <path name="mmap-playback" />
+ </path>
+
+ <path name="mmap-record">
+ <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="mmap-record bt-sco">
+ <ctl name="MultiMedia16 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="mmap-record bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="mmap-record bt-sco" />
+ </path>
+
+ <path name="mmap-record bt-sco-swb">
+ <path name="mmap-record bt-sco" />
+ </path>
+
+ <path name="mmap-record capture-fm">
+ <ctl name="MultiMedia16 Mixer TERT_MI2S_TX" value="1" />
+ </path>
+
+ <path name="mmap-record usb-headset-mic">
+ <ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="mmap-record headset-mic">
+ <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
+ <path name="hifi-playback display-port">
+ <ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="hifi-playback afe-proxy">
+ <ctl name="AFE_PCM_RX Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="hifi-playback usb-headset">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="hifi-playback usb-headphones">
+ <ctl name="USB_AUDIO_RX Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="hifi-record">
+ <ctl name="MultiMedia2 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="hifi-record usb-headset-mic">
+ <ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="incall_music_uplink">
+ <ctl name="Incall_Music Audio Mixer MultiMedia9" value="1" />
+ </path>
+
+ <path name="incall_music_uplink speaker">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink handset">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink handset-hac">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink display-port">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink bt-sco">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink bt-sco-wb">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink bt-sco-swb">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink speaker-and-display-port">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink afe-proxy">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink usb-headphones">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink usb-headset">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink speaker-and-usb-headphones">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink headphones">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink headset">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink speaker-and-headphones">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink speaker-and-bt-sco">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink voice-tty-hco-handset">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink speaker-and-bt-a2dp">
+ <path name="incall_music_uplink" />
+ </path>
+
+ <path name="incall_music_uplink bt-a2dp">
+ <path name="incall_music_uplink" />
+ </path>
+ <path name="incall_music_uplink afe-proxy">
+ <path name="incall_music_uplink" />
+ </path>
+</mixer>
diff --git a/configs/kona/mixer_paths_qrd.xml b/configs/kona/mixer_paths_qrd.xml
index 4a305ca..08176d5 100644
--- a/configs/kona/mixer_paths_qrd.xml
+++ b/configs/kona/mixer_paths_qrd.xml
@@ -61,11 +61,16 @@
<ctl name="MultiMedia5 Mixer AFE_PCM_TX" value="0" />
<ctl name="MultiMedia5 Mixer TX_CDC_DMA_TX_3" value="0" />
<ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_4" value="0" />
<ctl name="MultiMedia1 Mixer SLIM_7_TX" value="0" />
<ctl name="MultiMedia8 Mixer SLIM_7_TX" value="0" />
<ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_4" value="0" />
<ctl name="MultiMedia10 Mixer SLIM_7_TX" value="0" />
<ctl name="MultiMedia10 Mixer AFE_PCM_TX" value="0" />
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_4" value="0" />
+ <ctl name="MultiMedia17 Mixer SLIM_7_TX" value="0" />
<ctl name="DISPLAY_PORT Mixer MultiMedia1" value="0" />
<ctl name="DISPLAY_PORT Mixer MultiMedia2" value="0" />
<ctl name="DISPLAY_PORT Mixer MultiMedia3" value="0" />
@@ -129,6 +134,7 @@
<ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia10 Mixer USB_AUDIO_TX" value="0" />
+ <ctl name="MultiMedia17 Mixer USB_AUDIO_TX" value="0" />
<ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
<ctl name="WSA_CDC_DMA_RX_0 Channels" value="One" />
<ctl name="RX_CDC_DMA_RX_0 Channels" value="One" />
@@ -175,6 +181,7 @@
<ctl name="WSA_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="0" />
<ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="0" />
<ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode1" value="0" />
<!-- Multimode Voice1 BTSCO -->
<ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="0" />
<ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="0" />
@@ -191,6 +198,7 @@
<ctl name="WSA_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="0" />
<ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="0" />
<ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="0" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode2" value="0" />
<!-- Multimode Voice2 BTSCO -->
<ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="0" />
<ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="0" />
@@ -352,10 +360,10 @@
<ctl name="HPHR Volume" value="20" />
<ctl name="EAR SPKR PA Gain" value="G_DEFAULT" />
- <ctl name="TX_DEC0 Volume" value="102" />
- <ctl name="TX_DEC1 Volume" value="102" />
- <ctl name="TX_DEC2 Volume" value="102" />
- <ctl name="TX_DEC3 Volume" value="102" />
+ <ctl name="TX_DEC0 Volume" value="96" />
+ <ctl name="TX_DEC1 Volume" value="96" />
+ <ctl name="TX_DEC2 Volume" value="96" />
+ <ctl name="TX_DEC3 Volume" value="96" />
<ctl name="TX_DEC4 Volume" value="84" />
<ctl name="TX_DEC5 Volume" value="84" />
<ctl name="TX_DEC6 Volume" value="84" />
@@ -423,6 +431,7 @@
<!-- defaults for mmap record -->
<ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_3" value="0" />
+ <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_4" value="0" />
<ctl name="MultiMedia16 Mixer SLIM_7_TX" value="0" />
<ctl name="MultiMedia16 Mixer TERT_MI2S_TX" value="0" />
<ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="0" />
@@ -440,6 +449,10 @@
<ctl name="AUDIO_REF_EC_UL1 MUX" value="RX_CDC_DMA_RX_0" />
</path>
+ <path name="echo-reference headset">
+ <path name="echo-reference headphones" />
+ </path>
+
<path name="echo-reference display-port">
<ctl name="AUDIO_REF_EC_UL1 MUX" value="DISPLAY_PORT" />
</path>
@@ -515,6 +528,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia1" value="1" />
</path>
+ <path name="deep-buffer-playback headset">
+ <path name="deep-buffer-playback headphones" />
+ </path>
+
<path name="deep-buffer-playback speaker-and-headphones">
<path name="deep-buffer-playback headphones" />
<path name="deep-buffer-playback" />
@@ -590,6 +607,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia5" value="1" />
</path>
+ <path name="low-latency-playback headset">
+ <path name="low-latency-playback headphones" />
+ </path>
+
<path name="low-latency-playback speaker-and-headphones">
<path name="low-latency-playback headphones" />
<path name="low-latency-playback" />
@@ -626,6 +647,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia8" value="1" />
</path>
+ <path name="audio-ull-playback headset">
+ <path name="audio-ull-playback headphones" />
+ </path>
+
<path name="audio-ull-playback speaker-and-headphones">
<path name="audio-ull-playback" />
<path name="audio-ull-playback headphones" />
@@ -747,6 +772,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
</path>
+ <path name="compress-offload-playback headset">
+ <path name="compress-offload-playback headphones" />
+ </path>
+
<path name="compress-offload-playback headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia4" value="1" />
</path>
@@ -822,6 +851,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
</path>
+ <path name="compress-offload-playback2 headset">
+ <path name="compress-offload-playback2 headphones" />
+ </path>
+
<path name="compress-offload-playback2 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia7" value="1" />
</path>
@@ -897,6 +930,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
</path>
+ <path name="compress-offload-playback3 headset">
+ <path name="compress-offload-playback3 headphones" />
+ </path>
+
<path name="compress-offload-playback3 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
</path>
@@ -973,6 +1010,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
</path>
+ <path name="compress-offload-playback4 headset">
+ <path name="compress-offload-playback4 headphones" />
+ </path>
+
<path name="compress-offload-playback4 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia11" value="1" />
</path>
@@ -1048,6 +1089,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
</path>
+ <path name="compress-offload-playback5 headset">
+ <path name="compress-offload-playback5 headphones" />
+ </path>
+
<path name="compress-offload-playback5 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia12" value="1" />
</path>
@@ -1123,6 +1168,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
</path>
+ <path name="compress-offload-playback6 headset">
+ <path name="compress-offload-playback6 headphones" />
+ </path>
+
<path name="compress-offload-playback6 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia13" value="1" />
</path>
@@ -1198,6 +1247,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
</path>
+ <path name="compress-offload-playback7 headset">
+ <path name="compress-offload-playback7 headphones" />
+ </path>
+
<path name="compress-offload-playback7 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia14" value="1" />
</path>
@@ -1273,6 +1326,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
</path>
+ <path name="compress-offload-playback8 headset">
+ <path name="compress-offload-playback8 headphones" />
+ </path>
+
<path name="compress-offload-playback8 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia15" value="1" />
</path>
@@ -1348,6 +1405,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
</path>
+ <path name="compress-offload-playback9 headset">
+ <path name="compress-offload-playback9 headphones" />
+ </path>
+
<path name="compress-offload-playback9 headphones-44.1">
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
</path>
@@ -1393,6 +1454,10 @@
<path name="audio-record bt-sco" />
</path>
+ <path name="audio-record headset-mic">
+ <ctl name="MultiMedia1 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
<path name="audio-record capture-fm">
<ctl name="MultiMedia1 Mixer TERT_MI2S_TX" value="1" />
</path>
@@ -1418,6 +1483,31 @@
<ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="1" />
</path>
+ <path name="audio-record-compress2">
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_3" value="1" />
+ </path>
+
+ <path name="audio-record-compress2 bt-sco">
+ <ctl name="MultiMedia17 Mixer SLIM_7_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress bt-sco-wb">
+ <ctl name="BT SampleRate" value="KHZ_16" />
+ <path name="audio-record-compress2 bt-sco" />
+ </path>
+
+ <path name="audio-record-compress2 bt-sco-swb">
+ <path name="audio-record-compress2 bt-sco" />
+ </path>
+
+ <path name="audio-record-compress2 usb-headset-mic">
+ <ctl name="MultiMedia17 Mixer USB_AUDIO_TX" value="1" />
+ </path>
+
+ <path name="audio-record-compress2 headset-mic">
+ <ctl name="MultiMedia17 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
<path name="low-latency-record">
<ctl name="MultiMedia8 Mixer TX_CDC_DMA_TX_3" value="1" />
</path>
@@ -1459,6 +1549,10 @@
<ctl name="RX_CDC_DMA_RX_0_DL_HL Switch" value="1" />
</path>
+ <path name="play-fm headset">
+ <path name="play-fm headphones" />
+ </path>
+
<path name="incall-rec-uplink">
<ctl name="MultiMedia1 Mixer VOC_REC_UL" value="1" />
</path>
@@ -1483,6 +1577,10 @@
<path name="incall-rec-uplink" />
</path>
+ <path name="incall-rec-uplink headset-mic">
+ <path name="incall-rec-uplink" />
+ </path>
+
<path name="incall-rec-uplink-compress">
<ctl name="MultiMedia8 Mixer VOC_REC_UL" value="1" />
</path>
@@ -1507,6 +1605,10 @@
<path name="incall-rec-uplink-compress" />
</path>
+ <path name="incall-rec-uplink-compress headset-mic">
+ <path name="incall-rec-uplink-compress" />
+ </path>
+
<path name="incall-rec-downlink">
<ctl name="MultiMedia1 Mixer VOC_REC_DL" value="1" />
</path>
@@ -1531,6 +1633,10 @@
<path name="incall-rec-downlink" />
</path>
+ <path name="incall-rec-downlink headset-mic">
+ <path name="incall-rec-downlink" />
+ </path>
+
<path name="incall-rec-downlink-compress">
<ctl name="MultiMedia8 Mixer VOC_REC_DL" value="1" />
</path>
@@ -1555,6 +1661,10 @@
<path name="incall-rec-downlink-compress" />
</path>
+ <path name="incall-rec-downlink-compress headset-mic">
+ <path name="incall-rec-downlink-compress" />
+ </path>
+
<path name="incall-rec-uplink-and-downlink">
<path name="incall-rec-uplink" />
<path name="incall-rec-downlink" />
@@ -1580,6 +1690,10 @@
<path name="incall-rec-uplink-and-downlink" />
</path>
+ <path name="incall-rec-uplink-and-downlink headset-mic">
+ <path name="incall-rec-uplink-and-downlink" />
+ </path>
+
<path name="incall-rec-uplink-and-downlink-compress">
<path name="incall-rec-uplink-compress" />
<path name="incall-rec-downlink-compress" />
@@ -1605,6 +1719,10 @@
<path name="incall-rec-uplink-and-downlink-compress" />
</path>
+ <path name="incall-rec-uplink-and-downlink-compress headset-mic">
+ <path name="incall-rec-uplink-and-downlink-compress" />
+ </path>
+
<path name="hfp-sco">
</path>
@@ -1683,6 +1801,11 @@
<ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_3_MMode1" value="1" />
</path>
+ <path name="voicemmode1-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode1" value="1" />
+ <ctl name="VoiceMMode1_Tx Mixer TX_CDC_DMA_TX_4_MMode1" value="1" />
+ </path>
+
<path name="voicemmode1-call bt-sco">
<ctl name="SLIM_7_RX_Voice Mixer VoiceMMode1" value="1" />
<ctl name="VoiceMMode1_Tx Mixer SLIM_7_TX_MMode1" value="1" />
@@ -1737,6 +1860,11 @@
<ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_3_MMode2" value="1" />
</path>
+ <path name="voicemmode2-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer VoiceMMode2" value="1" />
+ <ctl name="VoiceMMode2_Tx Mixer TX_CDC_DMA_TX_4_MMode2" value="1" />
+ </path>
+
<path name="voicemmode2-call bt-sco">
<ctl name="SLIM_7_RX_Voice Mixer VoiceMMode2" value="1" />
<ctl name="VoiceMMode2_Tx Mixer SLIM_7_TX_MMode2" value="1" />
@@ -1849,6 +1977,10 @@
<path name="audio-record-voip bt-sco" />
</path>
+ <path name="audio-record-voip headset-mic">
+ <ctl name="MultiMedia10 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
<path name="bt-a2dp">
<ctl name="SLIM7_RX ADM Channels" value="Two" />
</path>
@@ -2056,6 +2188,10 @@
<ctl name="MultiMedia16 Mixer USB_AUDIO_TX" value="1" />
</path>
+ <path name="mmap-record headset-mic">
+ <ctl name="MultiMedia16 Mixer TX_CDC_DMA_TX_4" value="1" />
+ </path>
+
<path name="hifi-playback display-port">
<ctl name="DISPLAY_PORT Mixer MultiMedia2" value="1" />
</path>
@@ -2136,6 +2272,10 @@
<path name="incall_music_uplink" />
</path>
+ <path name="incall_music_uplink headset">
+ <path name="incall_music_uplink" />
+ </path>
+
<path name="incall_music_uplink speaker-and-headphones">
<path name="incall_music_uplink" />
</path>
@@ -2176,10 +2316,10 @@
</path>
<path name="amic2">
- <ctl name="TX DEC0 MUX" value="SWR_MIC" />
- <ctl name="TX SMIC MUX0" value="ADC1" />
- <ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
- <ctl name="TX_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="TX DEC5 MUX" value="SWR_MIC" />
+ <ctl name="TX SMIC MUX5" value="ADC1" />
+ <ctl name="TX_CDC_DMA_TX_4 Channels" value="One" />
+ <ctl name="TX_AIF2_CAP Mixer DEC5" value="1" />
<ctl name="ADC2_MIXER Switch" value="1" />
<ctl name="ADC2 MUX" value="INP2" />
</path>
@@ -2346,8 +2486,6 @@
<ctl name="RX INT1_1 MIX1 INP0" value="RX1" />
<ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
- <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
- <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
<ctl name="RX_COMP1 Switch" value="1" />
<ctl name="RX_COMP2 Switch" value="1" />
<ctl name="HPHL_COMP Switch" value="1" />
@@ -2379,13 +2517,16 @@
<path name="headset-mic">
<path name="amic2" />
- <ctl name="TX_DEC0 Volume" value="84" />
</path>
<path name="voice-handset">
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
@@ -2406,6 +2547,10 @@
<path name="headphones" />
</path>
+ <path name="voice-headset">
+ <path name="headphones" />
+ </path>
+
<path name="voice-line">
<path name="voice-headphones" />
</path>
diff --git a/configs/kona/sound_trigger_mixer_paths_cdp.xml b/configs/kona/sound_trigger_mixer_paths_cdp.xml
new file mode 100644
index 0000000..2f75edc
--- /dev/null
+++ b/configs/kona/sound_trigger_mixer_paths_cdp.xml
@@ -0,0 +1,329 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!--- Copyright (c) 2014-2019, The Linux Foundation. All rights reserved. -->
+<!--- -->
+<!--- Redistribution and use in source and binary forms, with or without -->
+<!--- modification, are permitted provided that the following conditions are -->
+<!--- met: -->
+<!--- * Redistributions of source code must retain the above copyright -->
+<!--- notice, this list of conditions and the following disclaimer. -->
+<!--- * Redistributions in binary form must reproduce the above -->
+<!--- copyright notice, this list of conditions and the following -->
+<!--- disclaimer in the documentation and/or other materials provided -->
+<!--- with the distribution. -->
+<!--- * Neither the name of The Linux Foundation nor the names of its -->
+<!--- contributors may be used to endorse or promote products derived -->
+<!--- from this software without specific prior written permission. -->
+<!--- -->
+<!--- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED -->
+<!--- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF -->
+<!--- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT -->
+<!--- ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS -->
+<!--- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR -->
+<!--- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF -->
+<!--- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR -->
+<!--- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, -->
+<!--- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE -->
+<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
+<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -->
+
+<mixer>
+ <!-- These are the initial mixer settings -->
+ <ctl name="LSM1 Mixer VA_CDC_DMA_TX_0" value="0" />
+ <ctl name="LSM2 Mixer VA_CDC_DMA_TX_0" value="0" />
+ <ctl name="LSM3 Mixer VA_CDC_DMA_TX_0" value="0" />
+ <ctl name="LSM4 Mixer VA_CDC_DMA_TX_0" value="0" />
+ <ctl name="LSM5 Mixer VA_CDC_DMA_TX_0" value="0" />
+ <ctl name="LSM6 Mixer VA_CDC_DMA_TX_0" value="0" />
+ <ctl name="LSM7 Mixer VA_CDC_DMA_TX_0" value="0" />
+ <ctl name="LSM8 Mixer VA_CDC_DMA_TX_0" 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="VA_CDC_DMA_TX_0 Channels" value="One" />
+ <ctl name="VA_AIF1_CAP Mixer DEC0" value="0" />
+ <ctl name="VA_AIF1_CAP Mixer DEC1" value="0" />
+ <ctl name="VA_AIF1_CAP Mixer DEC2" value="0" />
+ <ctl name="VA_AIF1_CAP Mixer DEC3" value="0" />
+ <ctl name="VA_AIF1_CAP Mixer DEC4" value="0" />
+ <ctl name="VA_AIF1_CAP Mixer DEC5" value="0" />
+ <ctl name="VA_AIF1_CAP Mixer DEC6" value="0" />
+ <ctl name="VA_AIF1_CAP Mixer DEC7" value="0" />
+ <ctl name="VA DEC0 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC1 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC2 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC3 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC4 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC5 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC6 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC7 MUX" value="MSM_DMIC" />
+ <ctl name="VA DMIC MUX0" value="ZERO" />
+ <ctl name="VA DMIC MUX1" value="ZERO" />
+ <ctl name="VA DMIC MUX2" value="ZERO" />
+ <ctl name="VA DMIC MUX3" value="ZERO" />
+ <ctl name="VA DMIC MUX4" value="ZERO" />
+ <ctl name="VA DMIC MUX5" value="ZERO" />
+ <ctl name="VA DMIC MUX6" value="ZERO" />
+ <ctl name="VA DMIC MUX7" value="ZERO" />
+ <ctl name="VA SMIC MUX0" value="ZERO" />
+ <ctl name="VA SMIC MUX1" value="ZERO" />
+ <ctl name="VA SMIC MUX2" value="ZERO" />
+ <ctl name="VA SMIC MUX3" value="ZERO" />
+ <ctl name="VA SMIC MUX4" value="ZERO" />
+ <ctl name="VA SMIC MUX5" value="ZERO" />
+ <ctl name="VA SMIC MUX6" value="ZERO" />
+ <ctl name="VA SMIC MUX7" value="ZERO" />
+ <ctl name="ADC1_MIXER Switch" value="0" />
+ <ctl name="ADC2_MIXER Switch" value="0" />
+ <ctl name="ADC2 MUX" value="ZERO" />
+ <ctl name="ADC3_MIXER Switch" value="0" />
+ <ctl name="ADC3 MUX" value="ZERO" />
+ <ctl name="ADC4_MIXER Switch" value="0" />
+ <ctl name="ADC4 MUX" value="ZERO" />
+ <ctl name="ADC1 Volume" value="0" />
+ <ctl name="ADC2 Volume" value="0" />
+ <ctl name="ADC3 Volume" value="0" />
+ <ctl name="ADC4 Volume" value="0" />
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="None"/>
+ <ctl name="EC Reference Channels" value="Zero"/>
+ <ctl name="EC Reference Bit Format" value="0"/>
+ <ctl name="EC Reference SampleRate" value="0"/>
+
+ <path name="listen-voice-wakeup-1">
+ <ctl name="LSM1 Mixer VA_CDC_DMA_TX_0" value="1" />
+ </path>
+
+ <path name="listen-voice-wakeup-2">
+ <ctl name="LSM2 Mixer VA_CDC_DMA_TX_0" value="1" />
+ </path>
+
+ <path name="listen-voice-wakeup-3">
+ <ctl name="LSM3 Mixer VA_CDC_DMA_TX_0" value="1" />
+ </path>
+
+ <path name="listen-voice-wakeup-4">
+ <ctl name="LSM4 Mixer VA_CDC_DMA_TX_0" value="1" />
+ </path>
+
+ <path name="listen-voice-wakeup-5">
+ <ctl name="LSM5 Mixer VA_CDC_DMA_TX_0" value="1" />
+ </path>
+
+ <path name="listen-voice-wakeup-6">
+ <ctl name="LSM6 Mixer VA_CDC_DMA_TX_0" value="1" />
+ </path>
+
+ <path name="listen-voice-wakeup-7">
+ <ctl name="LSM7 Mixer VA_CDC_DMA_TX_0" value="1" />
+ </path>
+
+ <path name="listen-voice-wakeup-8">
+ <ctl name="LSM8 Mixer VA_CDC_DMA_TX_0" value="1" />
+ </path>
+
+ <path name="listen-voice-wakeup-1 preproc">
+ <path name="listen-voice-wakeup-1" />
+ </path>
+
+ <path name="listen-voice-wakeup-2 preproc">
+ <path name="listen-voice-wakeup-2" />
+ </path>
+
+ <path name="listen-voice-wakeup-3 preproc">
+ <path name="listen-voice-wakeup-3" />
+ </path>
+
+ <path name="listen-voice-wakeup-4 preproc">
+ <path name="listen-voice-wakeup-4" />
+ </path>
+
+ <path name="listen-voice-wakeup-5 preproc">
+ <path name="listen-voice-wakeup-5" />
+ </path>
+
+ <path name="listen-voice-wakeup-6 preproc">
+ <path name="listen-voice-wakeup-6" />
+ </path>
+
+ <path name="listen-voice-wakeup-7 preproc">
+ <path name="listen-voice-wakeup-7" />
+ </path>
+
+ <path name="listen-voice-wakeup-8 preproc">
+ <path name="listen-voice-wakeup-8" />
+ </path>
+
+ <path name="listen-voice-wakeup-1 port">
+ <ctl name="LSM1 Port" value="VA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="listen-voice-wakeup-2 port">
+ <ctl name="LSM2 Port" value="VA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="listen-voice-wakeup-3 port">
+ <ctl name="LSM3 Port" value="VA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="listen-voice-wakeup-4 port">
+ <ctl name="LSM4 Port" value="VA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="listen-voice-wakeup-5 port">
+ <ctl name="LSM5 Port" value="VA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="listen-voice-wakeup-6 port">
+ <ctl name="LSM6 Port" value="VA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="listen-voice-wakeup-7 port">
+ <ctl name="LSM7 Port" value="VA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="listen-voice-wakeup-8 port">
+ <ctl name="LSM8 Port" value="VA_CDC_DMA_TX_0" />
+ </path>
+
+ <path name="listen-voice-wakeup-1 preproc port">
+ <ctl name="LSM1 Port" value="ADM_LSM_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-2 preproc port">
+ <ctl name="LSM2 Port" value="ADM_LSM_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-3 preproc port">
+ <ctl name="LSM3 Port" value="ADM_LSM_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-4 preproc port">
+ <ctl name="LSM4 Port" value="ADM_LSM_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-5 preproc port">
+ <ctl name="LSM5 Port" value="ADM_LSM_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-6 preproc port">
+ <ctl name="LSM6 Port" value="ADM_LSM_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-7 preproc port">
+ <ctl name="LSM7 Port" value="ADM_LSM_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-8 preproc port">
+ <ctl name="LSM8 Port" value="ADM_LSM_TX" />
+ </path>
+
+ <path name="listen-ape-handset-mic">
+ <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="VA DEC0 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX0" value="ADC0" />
+ <ctl name="ADC1 Volume" value="62" />
+ <ctl name="ADC1_MIXER Switch" value="1" />
+ </path>
+
+ <path name="listen-ape-handset-mic-preproc">
+ <path name="listen-ape-handset-mic" />
+ </path>
+
+ <path name="listen-ape-handset-dmic">
+ <ctl name="VA_CDC_DMA_TX_0 Channels" value="Two" />
+ <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
+ <ctl name="VA DEC0 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX0" value="ADC0" />
+ <ctl name="ADC1 Volume" value="62" />
+ <ctl name="ADC1_MIXER Switch" value="1" />
+ <ctl name="VA DEC1 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX1" value="ADC1" />
+ <ctl name="ADC2 Volume" value="62" />
+ <ctl name="ADC2_MIXER Switch" value="1" />
+ <ctl name="ADC2 MUX" value="INP3" />
+ </path>
+
+ <path name="listen-ape-handset-tmic">
+ <ctl name="VA_CDC_DMA_TX_0 Channels" value="Three" />
+ <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
+ <ctl name="VA_AIF1_CAP Mixer DEC2" value="1" />
+ <ctl name="VA DEC0 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX0" value="ADC0" />
+ <ctl name="ADC1 Volume" value="62" />
+ <ctl name="ADC1_MIXER Switch" value="1" />
+ <ctl name="VA DEC1 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX1" value="ADC1" />
+ <ctl name="ADC2 Volume" value="62" />
+ <ctl name="ADC2_MIXER Switch" value="1" />
+ <ctl name="ADC2 MUX" value="INP3" />
+ <ctl name="VA DEC2 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX2" value="ADC2" />
+ <ctl name="ADC3 Volume" value="62" />
+ <ctl name="ADC3_MIXER Switch" value="1" />
+ <ctl name="ADC3 MUX" value="INP4" />
+ </path>
+
+ <path name="listen-ape-handset-qmic">
+ <ctl name="VA_CDC_DMA_TX_0 Channels" value="Four" />
+ <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
+ <ctl name="VA_AIF1_CAP Mixer DEC2" value="1" />
+ <ctl name="VA_AIF1_CAP Mixer DEC3" value="1" />
+ <ctl name="VA DEC0 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX0" value="ADC0" />
+ <ctl name="ADC1 Volume" value="62" />
+ <ctl name="ADC1_MIXER Switch" value="1" />
+ <ctl name="VA DEC1 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX1" value="ADC1" />
+ <ctl name="ADC2 Volume" value="62" />
+ <ctl name="ADC2_MIXER Switch" value="1" />
+ <ctl name="ADC2 MUX" value="INP3" />
+ <ctl name="VA DEC3 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX3" value="ADC2" />
+ <ctl name="ADC3 Volume" value="62" />
+ <ctl name="ADC3_MIXER Switch" value="1" />
+ <ctl name="ADC3 MUX" value="INP4" />
+ <ctl name="VA DEC2 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX2" value="ADC3" />
+ <ctl name="ADC4 Volume" value="62" />
+ <ctl name="ADC4_MIXER Switch" value="1" />
+ <ctl name="ADC4 MUX" value="INP5" />
+ </path>
+
+ <path name="listen-ape-headset-mic">
+ <ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
+ <ctl name="VA DEC0 MUX" value="SWR_MIC" />
+ <ctl name="VA SMIC MUX0" value="ADC1" />
+ <ctl name="ADC2 Volume" value="62" />
+ <ctl name="ADC2_MIXER Switch" value="1" />
+ <ctl name="ADC2 MUX" value="INP2" />
+ </path>
+
+ <path name="echo-reference">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="WSA_CDC_DMA_RX_0"/>
+ <ctl name="EC Reference Channels" value="Two"/>
+ <ctl name="EC Reference Bit Format" value="S16_LE"/>
+ <ctl name="EC Reference SampleRate" value="48000"/>
+ </path>
+
+ <path name="echo-reference headset">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="WSA_CDC_DMA_RX_0"/>
+ <ctl name="EC Reference Channels" value="One"/>
+ <ctl name="EC Reference Bit Format" value="S16_LE"/>
+ <ctl name="EC Reference SampleRate" value="48000"/>
+ </path>
+
+ <path name="echo-reference a2dp">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_7_RX"/>
+ <ctl name="EC Reference Channels" value="Two"/>
+ <ctl name="EC Reference Bit Format" value="S16_LE"/>
+ <ctl name="EC Reference SampleRate" value="48000"/>
+ </path>
+
+</mixer>
diff --git a/configs/kona/sound_trigger_platform_info.xml b/configs/kona/sound_trigger_platform_info.xml
index c5b9676..7c8c25f 100644
--- a/configs/kona/sound_trigger_platform_info.xml
+++ b/configs/kona/sound_trigger_platform_info.xml
@@ -48,6 +48,7 @@
<param backend_dai_name="VA_CDC_DMA_TX_0" />
<!-- Param used to indicate if SVA has dedicated SLIM ports -->
<param dedicated_sva_path="true" />
+ <param dedicated_headset_path="false" />
<param platform_lpi_enable="true" />
</common_config>
<acdb_ids>
@@ -71,7 +72,7 @@
<sound_model_config>
<param vendor_uuid="68ab2d40-e860-11e3-95ef-0002a5d5c51b" />
<param execution_type="ADSP" />
- <param library="libsmwrapper.so" />
+ <param merge_first_stage_sound_models="false"/>
<param max_ape_phrases="20" />
<param max_ape_users="10" />
<!-- Profile specific data which the algorithm can support -->
@@ -152,7 +153,6 @@
<sound_model_config>
<param vendor_uuid="876c1b46-9d4d-40cc-a4fd-4d5ec7a80e47" />
<param execution_type="ADSP" />
- <param library="libsmwrapper.so" />
<param max_ape_phrases="1" />
<param max_ape_users="1" />
<!-- Profile specific data which the algorithm can support -->
@@ -207,7 +207,6 @@
<sound_model_config>
<param vendor_uuid="7038ddc8-30f2-11e6-b0ac-40a8f03d3f15" />
<param execution_type="ADSP" />
- <param library="none" />
<param max_ape_phrases="1" />
<param max_ape_users="1" />
<!-- Profile specific data which the algorithm can support -->
@@ -260,7 +259,6 @@
<sound_model_config>
<param vendor_uuid="9f6ad62a-1f0b-11e7-87c5-40a8f03d3f15" />
<param execution_type="ADSP" />
- <param library="none" />
<param max_ape_phrases="1" />
<param max_ape_users="1" />
<!-- Profile specific data which the algorithm can support -->
diff --git a/configs/lito/audio_effects.xml b/configs/lito/audio_effects.xml
index b6e318e..add0925 100644
--- a/configs/lito/audio_effects.xml
+++ b/configs/lito/audio_effects.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="UTF-8"?>
-<!--- Copyright (c) 2018-2019, The Linux Foundation. All rights reserved. -->
+<!--- Copyright (c) 2018-2019, 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 -->
@@ -30,9 +30,6 @@
<libraries>
<library name="bundle" path="libbundlewrapper.so"/>
<library name="reverb" path="libreverbwrapper.so"/>
- <library name="qcbassboost" path="libqcbassboost.so"/>
- <library name="qcvirt" path="libqcvirt.so"/>
- <library name="qcreverb" path="libqcreverb.so"/>
<library name="visualizer_sw" path="libvisualizer.so"/>
<library name="visualizer_hw" path="libqcomvisualizer.so"/>
<library name="downmix" path="libdownmix.so"/>
@@ -47,11 +44,11 @@
</libraries>
<effects>
<effectProxy name="bassboost" library="proxy" uuid="14804144-a5ee-4d24-aa88-0002a5d5c51b">
- <libsw library="qcbassboost" uuid="23aca180-44bd-11e2-bcfd-0800200c9a66"/>
+ <libsw library="bundle" uuid="8631f300-72e2-11df-b57e-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="2c4a8c24-1581-487f-94f6-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="virtualizer" library="proxy" uuid="d3467faa-acc7-4d34-acaf-0002a5d5c51b">
- <libsw library="qcvirt" uuid="e6c98a16-22a3-11e2-b87b-f23c91aec05e"/>
+ <libsw library="bundle" uuid="1d4033c0-8557-11df-9f2d-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="509a4498-561a-4bea-b3b1-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="equalizer" library="proxy" uuid="c8e70ecd-48ca-456e-8a4f-0002a5d5c51b">
@@ -60,19 +57,19 @@
</effectProxy>
<effect name="volume" library="bundle" uuid="119341a0-8469-11df-81f9-0002a5d5c51b"/>
<effectProxy name="reverb_env_aux" library="proxy" uuid="48404ac9-d202-4ccc-bf84-0002a5d5c51b">
- <libsw library="qcreverb" uuid="a8c1e5f3-293d-43cd-95ec-d5e26c02e217"/>
+ <libsw library="reverb" uuid="4a387fc0-8ab3-11df-8bad-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="79a18026-18fd-4185-8233-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_env_ins" library="proxy" uuid="b707403a-a1c1-4291-9573-0002a5d5c51b">
- <libsw library="qcreverb" uuid="791fff8b-8129-4655-83a4-59bc61034c3a"/>
+ <libsw library="reverb" uuid="c7a511a0-a3bb-11df-860e-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="eb64ea04-973b-43d2-8f5e-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_pre_aux" library="proxy" uuid="1b78f587-6d1c-422e-8b84-0002a5d5c51b">
- <libsw library="qcreverb" uuid="53ef1db5-c0c0-445b-b060-e34d20ebb70a"/>
+ <libsw library="reverb" uuid="f29a1400-a3bb-11df-8ddc-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="6987be09-b142-4b41-9056-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_pre_ins" library="proxy" uuid="f3e178d2-ebcb-408e-8357-0002a5d5c51b">
- <libsw library="qcreverb" uuid="b08a0e38-22a5-11e2-b87b-f23c91aec05e"/>
+ <libsw library="reverb" uuid="172cdf00-a3bc-11df-a72f-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="aa2bebf6-47cf-4613-9bca-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="visualizer" library="proxy" uuid="1d0a1a53-7d5d-48f2-8e71-27fbd10d842c">
diff --git a/configs/lito/audio_platform_info.xml b/configs/lito/audio_platform_info.xml
index 21714f4..6d14b50 100644
--- a/configs/lito/audio_platform_info.xml
+++ b/configs/lito/audio_platform_info.xml
@@ -152,6 +152,7 @@
<device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="WSA_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT" interface="WSA_CDC_DMA_RX_0"/>
diff --git a/configs/lito/audio_platform_info_intcodec.xml b/configs/lito/audio_platform_info_intcodec.xml
index 7c57bf4..7f2a4a0 100644
--- a/configs/lito/audio_platform_info_intcodec.xml
+++ b/configs/lito/audio_platform_info_intcodec.xml
@@ -129,6 +129,7 @@
<device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="WSA_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT" interface="WSA_CDC_DMA_RX_0"/>
diff --git a/configs/lito/audio_platform_info_qrd.xml b/configs/lito/audio_platform_info_qrd.xml
index e7d2662..635f321 100644
--- a/configs/lito/audio_platform_info_qrd.xml
+++ b/configs/lito/audio_platform_info_qrd.xml
@@ -128,6 +128,7 @@
<device name="SND_DEVICE_OUT_VOICE_HANDSET" backend="handset" interface="RX_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" interface="TX_CDC_DMA_TX_3"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="WSA_CDC_DMA_RX_0"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_STEREO" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2" interface="WSA_CDC_DMA_RX_0"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT" interface="WSA_CDC_DMA_RX_0"/>
diff --git a/configs/lito/audio_policy_configuration.xml b/configs/lito/audio_policy_configuration.xml
index 50920b3..a33356b 100644
--- a/configs/lito/audio_policy_configuration.xml
+++ b/configs/lito/audio_policy_configuration.xml
@@ -263,17 +263,20 @@
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
@@ -316,27 +319,8 @@
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
</devicePort>
<devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
<devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
</devicePorts>
diff --git a/configs/lito/lito.mk b/configs/lito/lito.mk
index 45258f0..00876db 100644
--- a/configs/lito/lito.mk
+++ b/configs/lito/lito.mk
@@ -81,6 +81,8 @@
AUDIO_FEATURE_ENABLED_BATTERY_LISTENER := true
##AUDIO_FEATURE_FLAGS
+BOARD_SUPPORTS_OPENSOURCE_STHAL := true
+
AUDIO_HARDWARE := audio.a2dp.default
AUDIO_HARDWARE += audio.usb.default
AUDIO_HARDWARE += audio.r_submix.default
@@ -177,7 +179,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/lito/mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_qrd.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/lito/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/lito/audio_configs_stock.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs_stock.xml \
- frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -282,7 +285,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -382,7 +385,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=true \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=true \
@@ -393,7 +395,7 @@
vendor.audio.feature.display_port.enable=true \
vendor.audio.feature.dsm_feedback.enable=false \
vendor.audio.feature.dynamic_ecns.enable=true \
-vendor.audio.feature.ext_hw_plugin.enable=true \
+vendor.audio.feature.ext_hw_plugin.enable=false \
vendor.audio.feature.external_dsp.enable=false \
vendor.audio.feature.external_speaker.enable=false \
vendor.audio.feature.external_speaker_tfa.enable=false \
diff --git a/configs/lito/mixer_paths.xml b/configs/lito/mixer_paths.xml
index 8688745..ec6be2e 100644
--- a/configs/lito/mixer_paths.xml
+++ b/configs/lito/mixer_paths.xml
@@ -2027,8 +2027,6 @@
<ctl name="RX INT1_1 MIX1 INP0" value="RX1" />
<ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
- <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
- <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
<ctl name="RX_COMP1 Switch" value="1" />
<ctl name="RX_COMP2 Switch" value="1" />
<ctl name="HPHL_COMP Switch" value="1" />
diff --git a/configs/lito/mixer_paths_qrd.xml b/configs/lito/mixer_paths_qrd.xml
index a6bdeae..b246c5a 100644
--- a/configs/lito/mixer_paths_qrd.xml
+++ b/configs/lito/mixer_paths_qrd.xml
@@ -2284,7 +2284,7 @@
</path>
<path name="speaker-protected">
- <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="1" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
<ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
<path name="speaker" />
<ctl name="VI_FEED_TX Channels" value="One" />
@@ -2292,7 +2292,7 @@
</path>
<path name="voice-speaker-protected">
- <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="1" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
<ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
<path name="speaker-mono" />
<ctl name="VI_FEED_TX Channels" value="One" />
@@ -2342,8 +2342,6 @@
<ctl name="RX INT1_1 MIX1 INP0" value="RX1" />
<ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
- <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
- <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
<ctl name="RX_COMP1 Switch" value="1" />
<ctl name="RX_COMP2 Switch" value="1" />
<ctl name="HPHL_COMP Switch" value="1" />
@@ -2381,6 +2379,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
@@ -2607,8 +2609,8 @@
<ctl name="TX DMIC MUX1" value="DMIC1" />
<ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
<ctl name="TX DMIC MUX2" value="DMIC3" />
- <ctl name="TX_AIF1_CAP Mixer DEC4" value="1" />
- <ctl name="TX DMIC MUX2" value="DMIC3" />
+ <ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
+ <ctl name="TX DMIC MUX3" value="DMIC4" />
</path>
<path name="voice-speaker-qmic">
diff --git a/configs/lito/sound_trigger_mixer_paths_qrd.xml b/configs/lito/sound_trigger_mixer_paths_qrd.xml
index ccbdd94..8e6513c 100644
--- a/configs/lito/sound_trigger_mixer_paths_qrd.xml
+++ b/configs/lito/sound_trigger_mixer_paths_qrd.xml
@@ -234,13 +234,13 @@
<ctl name="VA_CDC_DMA_TX_0 Channels" value="Three" />
<ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
<ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
- <ctl name="VA_AIF1_CAP Mixer DEC5" value="1" />
+ <ctl name="VA_AIF1_CAP Mixer DEC2" value="1" />
<ctl name="VA DEC0 MUX" value="MSM_DMIC" />
<ctl name="VA DEC1 MUX" value="MSM_DMIC" />
- <ctl name="VA DEC5 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC2 MUX" value="MSM_DMIC" />
<ctl name="VA DMIC MUX0" value="DMIC1" />
<ctl name="VA DMIC MUX1" value="DMIC2" />
- <ctl name="VA DMIC MUX5" value="DMIC5" />
+ <ctl name="VA DMIC MUX2" value="DMIC4" />
</path>
<path name="listen-ape-handset-qmic">
@@ -248,15 +248,15 @@
<ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
<ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
<ctl name="VA_AIF1_CAP Mixer DEC2" value="1" />
- <ctl name="VA_AIF1_CAP Mixer DEC5" value="1" />
+ <ctl name="VA_AIF1_CAP Mixer DEC3" value="1" />
<ctl name="VA DEC0 MUX" value="MSM_DMIC" />
<ctl name="VA DEC1 MUX" value="MSM_DMIC" />
<ctl name="VA DEC2 MUX" value="MSM_DMIC" />
- <ctl name="VA DEC5 MUX" value="MSM_DMIC" />
+ <ctl name="VA DEC3 MUX" value="MSM_DMIC" />
<ctl name="VA DMIC MUX0" value="DMIC1" />
<ctl name="VA DMIC MUX1" value="DMIC2" />
<ctl name="VA DMIC MUX2" value="DMIC3" />
- <ctl name="VA DMIC MUX5" value="DMIC5" />
+ <ctl name="VA DMIC MUX3" value="DMIC4" />
</path>
<path name="listen-ape-headset-mic">
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index 72fa6f3..a41740f 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -199,10 +199,6 @@
vendor.audio.use.sw.alac.decoder=true\
vendor.audio.use.sw.ape.decoder=true
-#property for AudioSphere Post processing
-PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.pp.asphere.enabled=false
-
#Audio voice concurrency related flags
PRODUCT_PROPERTY_OVERRIDES += \
vendor.voice.playback.conc.disabled=true\
@@ -245,7 +241,6 @@
vendor.audio.feature.a2dp_offload.enable=false \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index 25d42cf..0b0e6be 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -212,10 +212,6 @@
vendor.audio.use.sw.alac.decoder=true\
vendor.audio.use.sw.ape.decoder=true
-#property for AudioSphere Post processing
-PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.pp.asphere.enabled=false
-
#Audio voice concurrency related flags
PRODUCT_PROPERTY_OVERRIDES += \
vendor.voice.playback.conc.disabled=true\
diff --git a/configs/msm8998/msm8998.mk b/configs/msm8998/msm8998.mk
index bee32c8..6b77f69 100644
--- a/configs/msm8998/msm8998.mk
+++ b/configs/msm8998/msm8998.mk
@@ -191,7 +191,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -255,7 +255,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/msmnile/audio_platform_info.xml b/configs/msmnile/audio_platform_info.xml
index 6bfadc8..80924e2 100644
--- a/configs/msmnile/audio_platform_info.xml
+++ b/configs/msmnile/audio_platform_info.xml
@@ -143,7 +143,9 @@
<device name="SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET" backend="headphones" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_VOICE_LINE" backend="headphones" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_TTY_FULL_HEADSET" backend="headset" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_TTY_VCO_HEADSET" backend="headset" interface="SLIMBUS_6_RX"/>
<device name="SND_DEVICE_OUT_SPEAKER_AND_BT_SCO" backend="speaker-and-bt-sco" interface="SLIMBUS_0_RX-and-SLIMBUS_7_RX"/>
<device name="SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB" backend="speaker-and-bt-sco-wb" interface="SLIMBUS_0_RX-and-SLIMBUS_7_RX"/>
<device name="SND_DEVICE_IN_HEADSET_MIC" backend="headset-mic" interface="SLIMBUS_1_TX"/>
diff --git a/configs/msmnile/audio_policy_configuration.xml b/configs/msmnile/audio_policy_configuration.xml
index 657b5d1..5c05206 100644
--- a/configs/msmnile/audio_policy_configuration.xml
+++ b/configs/msmnile/audio_policy_configuration.xml
@@ -311,27 +311,8 @@
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
</devicePort>
<devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
<devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
</devicePorts>
diff --git a/configs/msmnile/mixer_paths_tavil.xml b/configs/msmnile/mixer_paths_tavil.xml
index f2e4842..fb315bf 100644
--- a/configs/msmnile/mixer_paths_tavil.xml
+++ b/configs/msmnile/mixer_paths_tavil.xml
@@ -2988,11 +2988,21 @@
<path name="tty-headphones" />
</path>
+ <path name="voice-tty-full-headset">
+ <ctl name="TTY Mode" value="FULL" />
+ <path name="tty-headphones" />
+ </path>
+
<path name="voice-tty-vco-headphones">
<ctl name="TTY Mode" value="VCO" />
<path name="tty-headphones" />
</path>
+ <path name="voice-tty-vco-headset">
+ <ctl name="TTY Mode" value="VCO" />
+ <path name="tty-headphones" />
+ </path>
+
<path name="voice-tty-hco-handset">
<ctl name="TTY Mode" value="HCO" />
<path name="handset" />
@@ -3011,7 +3021,7 @@
<path name="voice-tty-full-headset-mic">
<path name="amic2" />
<ctl name="ADC2 Volume" value="0" />
- <ctl name="DEC0 Volume" value="84" />
+ <ctl name="DEC1 Volume" value="84" />
</path>
<path name="voice-tty-hco-headset-mic">
diff --git a/configs/msmnile/msmnile.mk b/configs/msmnile/msmnile.mk
index 1764204..3315b11 100644
--- a/configs/msmnile/msmnile.mk
+++ b/configs/msmnile/msmnile.mk
@@ -176,7 +176,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile/sound_trigger_mixer_paths.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile/audio_configs_stock.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs_stock.xml \
- frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -280,7 +281,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -379,7 +380,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=false \
vendor.audio.feature.anc_headset.enable=false \
-vendor.audio.feature.audio_sphere.enable=false \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
@@ -424,7 +424,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=true \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=true \
@@ -481,6 +480,15 @@
vendor.qti.hardware.audiohalext@1.0-impl \
vendor.qti.hardware.audiohalext-utils
+# enable audio hidl hal 5.0
+PRODUCT_PACKAGES += \
+ android.hardware.audio@5.0 \
+ android.hardware.audio.common@5.0 \
+ android.hardware.audio.common@5.0-util \
+ android.hardware.audio@5.0-impl \
+ android.hardware.audio.effect@5.0 \
+ android.hardware.audio.effect@5.0-impl
+
PRODUCT_PACKAGES_ENG += \
VoicePrintTest \
VoicePrintDemo
diff --git a/configs/msmnile/sound_trigger_platform_info.xml b/configs/msmnile/sound_trigger_platform_info.xml
index 8468452..7d5e81f 100644
--- a/configs/msmnile/sound_trigger_platform_info.xml
+++ b/configs/msmnile/sound_trigger_platform_info.xml
@@ -35,6 +35,7 @@
<!--- added to <adm_config> -->
<common_config>
+ <param implementer_version="0x0100" />
<param max_cpe_sessions="1" />
<param max_wdsp_sessions="2" />
<param max_ape_sessions="8" />
diff --git a/configs/msmnile_au/audio_platform_info.xml b/configs/msmnile_au/audio_platform_info.xml
index 1dbaac1..e02397c 100644
--- a/configs/msmnile_au/audio_platform_info.xml
+++ b/configs/msmnile_au/audio_platform_info.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!-- Copyright (c) 2014, 2016-2018, The Linux Foundation. All rights reserved. -->
+<!-- Copyright (c) 2014, 2016-2019, 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 -->
@@ -29,6 +29,10 @@
<device name="SND_DEVICE_OUT_HANDSET" acdb_id="78"/>
<device name="SND_DEVICE_OUT_SPEAKER" acdb_id="78"/>
<device name="SND_DEVICE_OUT_HEADPHONES" acdb_id="78"/>
+ <device name="SND_DEVICE_OUT_BUS_MEDIA" acdb_id="78"/>
+ <device name="SND_DEVICE_OUT_BUS_SYS" acdb_id="78"/>
+ <device name="SND_DEVICE_OUT_BUS_NAV" acdb_id="14"/>
+ <device name="SND_DEVICE_OUT_BUS_PHN" acdb_id="94"/>
<device name="SND_DEVICE_OUT_BT_SCO" acdb_id="94"/>
<device name="SND_DEVICE_OUT_BT_SCO_WB" acdb_id="94"/>
<device name="SND_DEVICE_OUT_BT_A2DP" acdb_id="78"/>
@@ -36,6 +40,7 @@
<device name="SND_DEVICE_OUT_VOICE_SPEAKER" acdb_id="94"/>
<device name="SND_DEVICE_IN_HANDSET_MIC" acdb_id="11"/>
<device name="SND_DEVICE_IN_SPEAKER_MIC" acdb_id="11"/>
+ <device name="SND_DEVICE_IN_BUS" acdb_id="11"/>
<device name="SND_DEVICE_IN_HEADSET_MIC" acdb_id="11"/>
<device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" acdb_id="95"/>
<device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" acdb_id="95"/>
diff --git a/configs/msmnile_au/audio_policy_configuration.xml b/configs/msmnile_au/audio_policy_configuration.xml
index e4aec16..b00e62f 100644
--- a/configs/msmnile_au/audio_policy_configuration.xml
+++ b/configs/msmnile_au/audio_policy_configuration.xml
@@ -23,7 +23,6 @@
<!-- Global configuration Decalaration -->
<globalConfiguration speaker_drc_enabled="true"/>
-
<!-- Modules section:
There is one section per audio HW module present on the platform.
Each module section will contains two mandatory tags for audio HAL “halVersion” and “name”.
@@ -33,6 +32,11 @@
“devicePorts”: a list of device descriptors for all input and output devices accessible via this
module.
This contains both permanently attached devices and removable devices.
+ "gain": constraints applied to the millibel values:
+ - maxValueMB >= minValueMB
+ - defaultValueMB >= minValueMB && defaultValueMB <= maxValueMB
+ - (maxValueMB - minValueMB) % stepValueMB == 0
+ - (defaultValueMB - minValueMB) % stepValueMB == 0
“mixPorts”: listing all output and input streams exposed by the audio HAL
“routes”: list of possible connections between input and output devices or between stream and
devices.
@@ -47,18 +51,37 @@
-->
<modules>
<!-- Primary Audio HAL -->
- <module name="primary" halVersion="2.0">
+ <module name="primary" halVersion="3.0">
<attachedDevices>
- <item>Earpiece</item>
- <item>Speaker</item>
+ <item>Media Bus</item>
+ <item>Sys Notification Bus</item>
+ <item>Nav Guidance Bus</item>
+ <item>Phone Bus</item>
<item>Telephony Tx</item>
<item>Built-In Mic</item>
<item>Built-In Back Mic</item>
<item>FM Tuner</item>
<item>Telephony Rx</item>
</attachedDevices>
- <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <defaultOutputDevice>Media Bus</defaultOutputDevice>
<mixPorts>
+ <mixPort name="media" role="source"
+ flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="sys_notification" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="nav_guidance" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="phone" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
<mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
@@ -145,6 +168,12 @@
<profile name="" format="AUDIO_FORMAT_AAC_ADTS_HE_V2"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="dsd_compress_passthrough" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
@@ -197,6 +226,42 @@
<devicePorts>
<!-- Output devices declaration, i.e. Sink DEVICE PORT -->
+ <devicePort tagName="Media Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="BUS00_MEDIA">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Sys Notification Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="BUS01_SYS_NOTIFICATION">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Nav Guidance Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="BUS02_NAV_GUIDANCE">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Phone Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="BUS03_PHONE">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+ </gains>
+ </devicePort>
<devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
@@ -295,6 +360,14 @@
</devicePorts>
<!-- route declaration, i.e. list all available sources for a given sink -->
<routes>
+ <route type="mix" sink="Media Bus"
+ sources="media,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
+ <route type="mix" sink="Sys Notification Bus"
+ sources="sys_notification"/>
+ <route type="mix" sink="Nav Guidance Bus"
+ sources="nav_guidance"/>
+ <route type="mix" sink="Phone Bus"
+ sources="phone"/>
<route type="mix" sink="Earpiece"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
<route type="mix" sink="Speaker"
diff --git a/configs/msmnile_au/mixer_paths_adp.xml b/configs/msmnile_au/mixer_paths_adp.xml
index b07e235..63012be 100644
--- a/configs/msmnile_au/mixer_paths_adp.xml
+++ b/configs/msmnile_au/mixer_paths_adp.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!-- Copyright (c) 2016-2018, The Linux Foundation. All rights reserved. -->
+<!-- Copyright (c) 2016-2019, 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 -->
@@ -329,6 +329,8 @@
<path name="deep-buffer-playback">
<ctl name="TERT_TDM_RX_0 Channels" value="Six" />
<ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+ <ctl name="QUAT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="QUAT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
</path>
<path name="deep-buffer-playback speaker-protected">
@@ -525,6 +527,8 @@
<path name="compress-offload-playback">
<ctl name="TERT_TDM_RX_0 Channels" value="Six" />
<ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia4" value="1" />
+ <ctl name="QUAT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="QUAT_TDM_RX_0 Audio Mixer MultiMedia4" value="1" />
</path>
<path name="compress-offload-playback speaker-protected">
@@ -600,6 +604,8 @@
<path name="compress-offload-playback2">
<ctl name="TERT_TDM_RX_0 Channels" value="Six" />
<ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia7" value="1" />
+ <ctl name="QUAT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="QUAT_TDM_RX_0 Audio Mixer MultiMedia7" value="1" />
</path>
<path name="compress-offload-playback2 display-port">
@@ -1134,6 +1140,27 @@
<path name="compress-offload-playback9" />
</path>
+ <!-- The following use cases are used for car streams -->
+ <path name="media-playback">
+ <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="sys-notification-playback">
+ <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="nav-guidance-playback">
+ <ctl name="TERT_TDM_RX_1 Channels" value="One" />
+ <ctl name="TERT_TDM_RX_1 Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="phone-playback">
+ <ctl name="TERT_TDM_RX_2 Channels" value="One" />
+ <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
<path name="audio-record">
<ctl name="TERT_TDM_TX_0 Channels" value="One" />
<ctl name="MultiMedia1 Mixer TERT_TDM_TX_0" value="1" />
@@ -1621,9 +1648,15 @@
</path>
<path name="speaker-adp">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
</path>
<path name="speaker-custom">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
+ </path>
+
+ <path name="bus-speaker">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
</path>
<path name="sidetone-iir">
diff --git a/configs/msmnile_au/mixer_paths_custom.xml b/configs/msmnile_au/mixer_paths_custom.xml
index d8e45cd..9d8507a 100644
--- a/configs/msmnile_au/mixer_paths_custom.xml
+++ b/configs/msmnile_au/mixer_paths_custom.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!-- Copyright (c) 2016-2018, The Linux Foundation. All rights reserved. -->
+<!-- Copyright (c) 2016-2019, 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 -->
@@ -1130,6 +1130,27 @@
<path name="compress-offload-playback9" />
</path>
+ <!-- The following use cases are used for car streams -->
+ <path name="media-playback">
+ <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="sys-notification-playback">
+ <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="nav-guidance-playback">
+ <ctl name="TERT_TDM_RX_1 Channels" value="One" />
+ <ctl name="TERT_TDM_RX_1 Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="phone-playback">
+ <ctl name="TERT_TDM_RX_2 Channels" value="One" />
+ <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
<path name="audio-record">
<ctl name="TERT_TDM_TX_0 Channels" value="One" />
<ctl name="MultiMedia1 Mixer TERT_TDM_TX_0" value="1" />
@@ -1617,9 +1638,15 @@
</path>
<path name="speaker-adp">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
</path>
<path name="speaker-custom">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
+ </path>
+
+ <path name="bus-speaker">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
</path>
<path name="sidetone-iir">
diff --git a/configs/msmnile_au/msmnile_au.mk b/configs/msmnile_au/msmnile_au.mk
index c1b8630..394dfea 100644
--- a/configs/msmnile_au/msmnile_au.mk
+++ b/configs/msmnile_au/msmnile_au.mk
@@ -69,8 +69,8 @@
AUDIO_FEATURE_ENABLED_BATTERY_LISTENER := false
##AUDIO_FEATURE_FLAGS
+AUDIO_FEATURE_ENABLED_AUTO_HAL := true
AUDIO_FEATURE_ENABLED_EXT_HW_PLUGIN := true
-AUDIO_FEATURE_ENABLED_BUS_ADDRESS := true
AUDIO_FEATURE_ENABLED_AUDIO_CONTROL_HAL := true
##AUTOMOTIVE_AUDIO_FEATURE_FLAGS
@@ -79,6 +79,9 @@
DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
endif
+#Automotive audio specific device overlays
+DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/msmnile_au/overlay
+
PRODUCT_COPY_FILES += \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/audio_output_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_output_policy.conf \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/audio_effects.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.conf \
@@ -111,6 +114,10 @@
PRODUCT_COPY_FILES += \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/listen_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/listen_platform_info.xml
+#Audio HAL version
+PRODUCT_PROPERTY_OVERRIDES += \
+vendor.audio.hal.maj.version=3
+
# Reduce client buffer size for fast audio output tracks
PRODUCT_PROPERTY_OVERRIDES += \
af.fast_track_multiplier=1
@@ -188,6 +195,14 @@
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.parser.ip.buffer.size=262144
+#Enable 16 bit PCM offload by default
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.offload.pcm.16bit.enable=true
+
+#Enable 24 bit PCM offload by default
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.offload.pcm.24bit.enable=true
+
#flac sw decoder 24 bit decode capability
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.flac.sw.decoder.24bit=true
@@ -274,3 +289,21 @@
android.hardware.audio@4.0-impl \
android.hardware.audio.effect@4.0 \
android.hardware.audio.effect@4.0-impl
+
+# for HIDL related audiocontrol packages
+PRODUCT_PACKAGES += \
+ vendor.qti.hardware.automotive.audiocontrol@1.0-service \
+ android.hardware.automotive.audiocontrol@1.0
+
+ifeq ($(ENABLE_HYP),true)
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.audio.calfile0=/vendor/etc/acdbdata/adsp_avs_config.acdb\
+persist.vendor.audio.calfile1=/vendor/etc/acdbdata/ADP/Bluetooth_cal.acdb\
+persist.vendor.audio.calfile2=/vendor/etc/acdbdata/ADP/Codec_cal.acdb\
+persist.vendor.audio.calfile3=/vendor/etc/acdbdata/ADP/General_cal.acdb\
+persist.vendor.audio.calfile4=/vendor/etc/acdbdata/ADP/Global_cal.acdb\
+persist.vendor.audio.calfile5=/vendor/etc/acdbdata/ADP/Handset_cal.acdb\
+persist.vendor.audio.calfile6=/vendor/etc/acdbdata/ADP/Hdmi_cal.acdb\
+persist.vendor.audio.calfile7=/vendor/etc/acdbdata/ADP/Headset_cal.acdb\
+persist.vendor.audio.calfile8=/vendor/etc/acdbdata/ADP/Speaker_cal.acdb
+endif
diff --git a/configs/msmnile_au/overlay/frameworks/base/core/res/res/values/config.xml b/configs/msmnile_au/overlay/frameworks/base/core/res/res/values/config.xml
new file mode 100644
index 0000000..0274f9e
--- /dev/null
+++ b/configs/msmnile_au/overlay/frameworks/base/core/res/res/values/config.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+/*
+** Copyright (c) 2018-2019, The Linux Foundation. All rights reserved.
+** Not a Contribution.
+*/
+/*
+** Copyright 2009, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+-->
+
+<resources>
+ <!-- Car uses hardware amplifier for volume. -->
+ <bool name="config_useFixedVolume">false</bool>
+ <!--
+ Handle volume keys directly in CarAudioService without passing them to the foreground app
+ -->
+ <bool name="config_handleVolumeKeysInWindowManager">true</bool>
+</resources>
diff --git a/configs/msmnile_au/overlay/packages/services/Car/service/res/values/config.xml b/configs/msmnile_au/overlay/packages/services/Car/service/res/values/config.xml
new file mode 100644
index 0000000..ac5f818
--- /dev/null
+++ b/configs/msmnile_au/overlay/packages/services/Car/service/res/values/config.xml
@@ -0,0 +1,33 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+/*
+** Copyright (c) 2018-2019, The Linux Foundation. All rights reserved.
+** Not a Contribution.
+*/
+/*
+** Copyright 2015, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+-->
+
+<!--
+ Overlay resources to configure car service based on each OEM's preference.
+ See also packages/services/Car/service/res/values/config.xml
+-->
+<resources>
+ <!-- Configuration to enable usage of dynamic audio routing. If this is set to false,
+ dynamic audio routing is disabled and audio works in legacy mode. It may be useful
+ during initial development where audio hal does not support bus based addressing yet. -->
+ <bool name="audioUseDynamicRouting">true</bool>
+</resources>
diff --git a/configs/msmnile_au/overlay/packages/services/Car/service/res/xml/car_volume_groups.xml b/configs/msmnile_au/overlay/packages/services/Car/service/res/xml/car_volume_groups.xml
new file mode 100644
index 0000000..850d4d4
--- /dev/null
+++ b/configs/msmnile_au/overlay/packages/services/Car/service/res/xml/car_volume_groups.xml
@@ -0,0 +1,57 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+ Copyright (c) 2018-2019, The Linux Foundation. All rights reserved.
+ Not a Contribution.
+
+ Copyright (C) 2018 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<!--
+ Defines the all available volume groups for volume control in a car.
+ One can overlay this configuration to customize the groups.
+
+ This configuration will be populated by CarAudioService and
+ surfaced to Car Settings App and/or other volume control interfaces.
+
+ Certain constraints applied to this configuration
+ - One context should not appear in two groups
+ - All contexts are assigned
+ - One bus should not appear in two groups
+ - All gain controllers (set on each bus) in one group have same step value
+
+ It is fine that there are buses that do not appear in any group, those buses
+ may be reserved for other usages.
+
+ Important note: when overlaying this configuration,
+ make sure the resources are in the same package as CarAudioService.
+-->
+<volumeGroups xmlns:car="http://schemas.android.com/apk/res-auto">
+ <group>
+ <context car:context="music"/>
+ </group>
+ <group>
+ <context car:context="call"/>
+ <context car:context="call_ring"/>
+ </group>
+ <group>
+ <context car:context="alarm"/>
+ <context car:context="notification"/>
+ <context car:context="system_sound"/>
+ </group>
+ <group>
+ <context car:context="navigation"/>
+ <context car:context="voice_command"/>
+ </group>
+</volumeGroups>
diff --git a/configs/msmsteppe/audio_effects.xml b/configs/msmsteppe/audio_effects.xml
index 7c0cd22..add0925 100644
--- a/configs/msmsteppe/audio_effects.xml
+++ b/configs/msmsteppe/audio_effects.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="UTF-8"?>
-<!--- Copyright (c) 2018, The Linux Foundation. All rights reserved. -->
+<!--- Copyright (c) 2018-2019, 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 -->
@@ -30,9 +30,6 @@
<libraries>
<library name="bundle" path="libbundlewrapper.so"/>
<library name="reverb" path="libreverbwrapper.so"/>
- <library name="qcbassboost" path="libqcbassboost.so"/>
- <library name="qcvirt" path="libqcvirt.so"/>
- <library name="qcreverb" path="libqcreverb.so"/>
<library name="visualizer_sw" path="libvisualizer.so"/>
<library name="visualizer_hw" path="libqcomvisualizer.so"/>
<library name="downmix" path="libdownmix.so"/>
@@ -47,11 +44,11 @@
</libraries>
<effects>
<effectProxy name="bassboost" library="proxy" uuid="14804144-a5ee-4d24-aa88-0002a5d5c51b">
- <libsw library="qcbassboost" uuid="23aca180-44bd-11e2-bcfd-0800200c9a66"/>
+ <libsw library="bundle" uuid="8631f300-72e2-11df-b57e-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="2c4a8c24-1581-487f-94f6-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="virtualizer" library="proxy" uuid="d3467faa-acc7-4d34-acaf-0002a5d5c51b">
- <libsw library="qcvirt" uuid="e6c98a16-22a3-11e2-b87b-f23c91aec05e"/>
+ <libsw library="bundle" uuid="1d4033c0-8557-11df-9f2d-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="509a4498-561a-4bea-b3b1-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="equalizer" library="proxy" uuid="c8e70ecd-48ca-456e-8a4f-0002a5d5c51b">
@@ -60,19 +57,19 @@
</effectProxy>
<effect name="volume" library="bundle" uuid="119341a0-8469-11df-81f9-0002a5d5c51b"/>
<effectProxy name="reverb_env_aux" library="proxy" uuid="48404ac9-d202-4ccc-bf84-0002a5d5c51b">
- <libsw library="qcreverb" uuid="a8c1e5f3-293d-43cd-95ec-d5e26c02e217"/>
+ <libsw library="reverb" uuid="4a387fc0-8ab3-11df-8bad-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="79a18026-18fd-4185-8233-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_env_ins" library="proxy" uuid="b707403a-a1c1-4291-9573-0002a5d5c51b">
- <libsw library="qcreverb" uuid="791fff8b-8129-4655-83a4-59bc61034c3a"/>
+ <libsw library="reverb" uuid="c7a511a0-a3bb-11df-860e-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="eb64ea04-973b-43d2-8f5e-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_pre_aux" library="proxy" uuid="1b78f587-6d1c-422e-8b84-0002a5d5c51b">
- <libsw library="qcreverb" uuid="53ef1db5-c0c0-445b-b060-e34d20ebb70a"/>
+ <libsw library="reverb" uuid="f29a1400-a3bb-11df-8ddc-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="6987be09-b142-4b41-9056-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_pre_ins" library="proxy" uuid="f3e178d2-ebcb-408e-8357-0002a5d5c51b">
- <libsw library="qcreverb" uuid="b08a0e38-22a5-11e2-b87b-f23c91aec05e"/>
+ <libsw library="reverb" uuid="172cdf00-a3bc-11df-a72f-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="aa2bebf6-47cf-4613-9bca-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="visualizer" library="proxy" uuid="1d0a1a53-7d5d-48f2-8e71-27fbd10d842c">
diff --git a/configs/msmsteppe/audio_platform_info.xml b/configs/msmsteppe/audio_platform_info.xml
index 09b2d9b..30fbf1e 100644
--- a/configs/msmsteppe/audio_platform_info.xml
+++ b/configs/msmsteppe/audio_platform_info.xml
@@ -102,6 +102,7 @@
<usecase name="USECASE_AUDIO_PLAYBACK_MMAP" type="out" id="33" />
<usecase name="USECASE_AUDIO_RECORD_MMAP" type="in" id="33" />
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="in" id="12" />
+ <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="27" />
</pcm_ids>
<config_params>
<param key="spkr_1_tz_name" value="wsatz.13"/>
diff --git a/configs/msmsteppe/audio_platform_info_intcodec.xml b/configs/msmsteppe/audio_platform_info_intcodec.xml
index ac6c4cc..6273fb8 100644
--- a/configs/msmsteppe/audio_platform_info_intcodec.xml
+++ b/configs/msmsteppe/audio_platform_info_intcodec.xml
@@ -64,6 +64,7 @@
<usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="16" />
<usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="16" />
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="in" id="12" />
+ <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="27" />
</pcm_ids>
<config_params>
<!-- In the below value string, the value indicates default mono -->
diff --git a/configs/msmsteppe/audio_platform_info_qrd.xml b/configs/msmsteppe/audio_platform_info_qrd.xml
index bcda82f..c49fcb5 100644
--- a/configs/msmsteppe/audio_platform_info_qrd.xml
+++ b/configs/msmsteppe/audio_platform_info_qrd.xml
@@ -64,6 +64,7 @@
<usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="16" />
<usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="16" />
<usecase name="USECASE_AUDIO_A2DP_ABR_FEEDBACK" type="in" id="12" />
+ <usecase name="USECASE_INCALL_MUSIC_UPLINK" type="out" id="27" />
</pcm_ids>
<config_params>
<!-- In the below value string, the value indicates default mono -->
diff --git a/configs/msmsteppe/audio_policy_configuration.xml b/configs/msmsteppe/audio_policy_configuration.xml
index f86a518..b092687 100644
--- a/configs/msmsteppe/audio_policy_configuration.xml
+++ b/configs/msmsteppe/audio_policy_configuration.xml
@@ -167,6 +167,12 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="incall_music_uplink" role="source"
+ flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
<mixPort name="usb_surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
@@ -305,27 +311,8 @@
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
</devicePort>
<devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
<devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
</devicePorts>
@@ -358,7 +345,7 @@
<route type="mix" sink="USB Headset Out"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out,hifi_playback"/>
<route type="mix" sink="Telephony Tx"
- sources="voice_tx"/>
+ sources="voice_tx,incall_music_uplink"/>
<route type="mix" sink="voice_rx"
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
diff --git a/configs/msmsteppe/mixer_paths_idp.xml b/configs/msmsteppe/mixer_paths_idp.xml
index a48defd..2ce8d12 100644
--- a/configs/msmsteppe/mixer_paths_idp.xml
+++ b/configs/msmsteppe/mixer_paths_idp.xml
@@ -1833,76 +1833,79 @@
<ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
</path>
- <path name="incall-music-uplink">
+ <path name="incall_music_uplink">
<ctl name="Incall_Music Audio Mixer MultiMedia9" value="1" />
</path>
- <path name="incall-music-uplink speaker">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink handset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink handset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink handset-hac">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink handset-hac">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink display-port">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink display-port">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-sco">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-sco">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-sco-wb">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-sco-wb">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-display-port">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-display-port">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink afe-proxy">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink afe-proxy">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink usb-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink usb-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink usb-headset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink usb-headset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-usb-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-usb-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-bt-sco">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-bt-sco">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink voice-tty-hco-handset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink voice-tty-hco-handset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-bt-a2dp">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-bt-a2dp">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-a2dp">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-a2dp">
+ <path name="incall_music_uplink" />
+ </path>
+ <path name="incall_music_uplink afe-proxy">
+ <path name="incall_music_uplink" />
</path>
<path name="spkr-rx-calib">
@@ -2147,6 +2150,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/msmsteppe/mixer_paths_qrd.xml b/configs/msmsteppe/mixer_paths_qrd.xml
index 53c91c3..5665322 100644
--- a/configs/msmsteppe/mixer_paths_qrd.xml
+++ b/configs/msmsteppe/mixer_paths_qrd.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!-- Copyright (c) 2015-2018, The Linux Foundation. All rights reserved. -->
+<!-- Copyright (c) 2015-2019, 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 -->
@@ -1895,76 +1895,79 @@
<ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
</path>
- <path name="incall-music-uplink">
+ <path name="incall_music_uplink">
<ctl name="Incall_Music Audio Mixer MultiMedia9" value="1" />
</path>
- <path name="incall-music-uplink speaker">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink handset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink handset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink handset-hac">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink handset-hac">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink display-port">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink display-port">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-sco">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-sco">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-sco-wb">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-sco-wb">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-display-port">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-display-port">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink afe-proxy">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink afe-proxy">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink usb-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink usb-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink usb-headset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink usb-headset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-usb-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-usb-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-bt-sco">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-bt-sco">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink voice-tty-hco-handset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink voice-tty-hco-handset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-bt-a2dp">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-bt-a2dp">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-a2dp">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-a2dp">
+ <path name="incall_music_uplink" />
+ </path>
+ <path name="incall_music_uplink afe-proxy">
+ <path name="incall_music_uplink" />
</path>
<path name="spkr-rx-calib">
@@ -2172,6 +2175,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
@@ -2189,7 +2196,15 @@
</path>
<path name="voice-headphones">
- <path name="headphones" />
+ <ctl name="RX_MACRO RX0 MUX" value="AIF1_PB" />
+ <ctl name="RX_MACRO RX1 MUX" value="AIF1_PB" />
+ <ctl name="RX_CDC_DMA_RX_0 Channels" value="Two" />
+ <ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX1" />
+ <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="HPHL_RDAC Switch" value="1" />
+ <ctl name="HPHR_RDAC Switch" value="1" />
</path>
<path name="voice-line">
diff --git a/configs/msmsteppe/mixer_paths_tavil.xml b/configs/msmsteppe/mixer_paths_tavil.xml
index 1a2bf55..d0db2fb 100644
--- a/configs/msmsteppe/mixer_paths_tavil.xml
+++ b/configs/msmsteppe/mixer_paths_tavil.xml
@@ -2938,76 +2938,79 @@
<ctl name="MultiMedia2 Mixer USB_AUDIO_TX" value="1" />
</path>
- <path name="incall-music-uplink">
+ <path name="incall_music_uplink">
<ctl name="Incall_Music Audio Mixer MultiMedia9" value="1" />
</path>
- <path name="incall-music-uplink speaker">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink handset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink handset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink handset-hac">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink handset-hac">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink display-port">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink display-port">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-sco">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-sco">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-sco-wb">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-sco-wb">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-display-port">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-display-port">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink afe-proxy">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink afe-proxy">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink usb-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink usb-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink usb-headset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink usb-headset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-usb-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-usb-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-headphones">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-headphones">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-bt-sco">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-bt-sco">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink voice-tty-hco-handset">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink voice-tty-hco-handset">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink speaker-and-bt-a2dp">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink speaker-and-bt-a2dp">
+ <path name="incall_music_uplink" />
</path>
- <path name="incall-music-uplink bt-a2dp">
- <path name="incall-music-uplink" />
+ <path name="incall_music_uplink bt-a2dp">
+ <path name="incall_music_uplink" />
+ </path>
+ <path name="incall_music_uplink afe-proxy">
+ <path name="incall_music_uplink" />
</path>
</mixer>
diff --git a/configs/msmsteppe/mixer_paths_wcd9375.xml b/configs/msmsteppe/mixer_paths_wcd9375.xml
index 680f445..c4d2af7 100644
--- a/configs/msmsteppe/mixer_paths_wcd9375.xml
+++ b/configs/msmsteppe/mixer_paths_wcd9375.xml
@@ -2155,6 +2155,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/msmsteppe/mixer_paths_wcd9375qrd.xml b/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
index 758a1d3..aee360c 100644
--- a/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
+++ b/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
@@ -2237,6 +2237,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/msmsteppe/msmsteppe.mk b/configs/msmsteppe/msmsteppe.mk
index 753a674..ec546ac 100644
--- a/configs/msmsteppe/msmsteppe.mk
+++ b/configs/msmsteppe/msmsteppe.mk
@@ -176,6 +176,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe/mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_qrd.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe/mixer_paths_wcd9375qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_wcd9375qrd.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe/mixer_paths_tavil.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tavil.xml \
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -240,6 +242,10 @@
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.offload.buffer.size.kb=32
+#Minimum duration for offload playback in secs
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.offload.min.duration.secs=30
+
#Enable offload audio video playback by default
PRODUCT_PROPERTY_OVERRIDES += \
audio.offload.video=true
@@ -283,7 +289,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -356,7 +362,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
@@ -378,7 +383,7 @@
vendor.audio.feature.hfp.enable=true \
vendor.audio.feature.hifi_audio.enable=false \
vendor.audio.feature.hwdep_cal.enable=false \
-vendor.audio.feature.incall_music.enable=false \
+vendor.audio.feature.incall_music.enable=true \
vendor.audio.feature.keep_alive.enable=false \
vendor.audio.feature.kpi_optimize.enable=true \
vendor.audio.feature.maxx_audio.enable=false \
@@ -388,7 +393,7 @@
vendor.audio.feature.spkr_prot.enable=true \
vendor.audio.feature.ssrec.enable=true \
vendor.audio.feature.usb_offload.enable=true \
-vendor.audio.feature.usb_offload_burst_mode.enable=false \
+vendor.audio.feature.usb_offload_burst_mode.enable=true \
vendor.audio.feature.usb_offload_sidetone_volume.enable=false \
vendor.audio.feature.deepbuffer_as_primary.enable=false \
vendor.audio.feature.vbat.enable=true \
@@ -396,6 +401,10 @@
vendor.audio.feature.audiozoom.enable=false \
vendor.audio.feature.snd_mon.enable=true
+#enable AAC frame ctl for A2DP sinks
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.aac_frm_ctl.enabled=true
+
# for HIDL related packages
PRODUCT_PACKAGES += \
android.hardware.audio@2.0-service \
@@ -409,6 +418,15 @@
android.hardware.audio.effect@4.0 \
android.hardware.audio.effect@4.0-impl
+# enable audio hidl hal 5.0
+PRODUCT_PACKAGES += \
+ android.hardware.audio@5.0 \
+ android.hardware.audio.common@5.0 \
+ android.hardware.audio.common@5.0-util \
+ android.hardware.audio@5.0-impl \
+ android.hardware.audio.effect@5.0 \
+ android.hardware.audio.effect@5.0-impl
+
PRODUCT_PACKAGES_ENG += \
VoicePrintTest \
VoicePrintDemo
diff --git a/configs/msmsteppe/sound_trigger_mixer_paths.xml b/configs/msmsteppe/sound_trigger_mixer_paths.xml
index a489e7f..90de0d3 100644
--- a/configs/msmsteppe/sound_trigger_mixer_paths.xml
+++ b/configs/msmsteppe/sound_trigger_mixer_paths.xml
@@ -206,11 +206,11 @@
<ctl name="TX_DEC3 Volume" value="102" />
<ctl name="TX DMIC MUX0" value="DMIC2" />
<ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
- <ctl name="TX DMIC MUX1" value="DMIC1" />
+ <ctl name="TX DMIC MUX1" value="DMIC0" />
<ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
<ctl name="TX DMIC MUX2" value="DMIC3" />
<ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
- <ctl name="TX DMIC MUX3" value="DMIC0" />
+ <ctl name="TX DMIC MUX3" value="DMIC1" />
</path>
<path name="echo-reference">
diff --git a/configs/msmsteppe/sound_trigger_mixer_paths_wcd9340.xml b/configs/msmsteppe/sound_trigger_mixer_paths_wcd9340.xml
index 55dd42f..f74c4fe 100644
--- a/configs/msmsteppe/sound_trigger_mixer_paths_wcd9340.xml
+++ b/configs/msmsteppe/sound_trigger_mixer_paths_wcd9340.xml
@@ -199,7 +199,7 @@
<ctl name= "DMIC MUX0" value="DMIC2" />
<ctl name= "DEC0 Volume" value="84" />
<ctl name= "ADC MUX1" value="DMIC" />
- <ctl name= "DMIC MUX1" value="DMIC0" />
+ <ctl name= "DMIC MUX1" value="DMIC5" />
<ctl name= "DEC1 Volume" value="84" />
<ctl name= "WDMA3 PORT0 MUX" value="DEC0" />
<ctl name= "WDMA3 PORT1 MUX" value="DEC1" />
@@ -217,7 +217,7 @@
<ctl name= "DMIC MUX1" value="DMIC0" />
<ctl name= "DEC1 Volume" value="84" />
<ctl name= "ADC MUX2" value="DMIC" />
- <ctl name= "DMIC MUX2" value="DMIC1" />
+ <ctl name= "DMIC MUX2" value="DMIC5" />
<ctl name= "DEC2 Volume" value="84" />
<ctl name= "WDMA3 PORT0 MUX" value="DEC0" />
<ctl name= "WDMA3 PORT1 MUX" value="DEC1" />
@@ -237,10 +237,10 @@
<ctl name= "DMIC MUX1" value="DMIC0" />
<ctl name= "DEC1 Volume" value="84" />
<ctl name= "ADC MUX2" value="DMIC" />
- <ctl name= "DMIC MUX2" value="DMIC1" />
+ <ctl name= "DMIC MUX2" value="DMIC5" />
<ctl name= "DEC2 Volume" value="84" />
<ctl name= "ADC MUX3" value="DMIC" />
- <ctl name= "DMIC MUX3" value="DMIC3" />
+ <ctl name= "DMIC MUX3" value="DMIC1" />
<ctl name= "DEC3 Volume" value="84" />
<ctl name= "WDMA3 PORT0 MUX" value="DEC0" />
<ctl name= "WDMA3 PORT1 MUX" value="DEC1" />
@@ -298,7 +298,7 @@
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
<ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
<ctl name="CDC_IF TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
<ctl name="DMIC MUX8" value="DMIC5" />
@@ -312,13 +312,13 @@
<ctl name="SLIM_0_TX Channels" value="Three" />
<ctl name="CDC_IF TX5 MUX" value="DEC5" />
<ctl name="ADC MUX5" value="DMIC" />
- <ctl name="DMIC MUX5" value="DMIC1" />
+ <ctl name="DMIC MUX5" value="DMIC2" />
<ctl name="CDC_IF TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
- <ctl name="DMIC MUX6" value="DMIC5" />
+ <ctl name="DMIC MUX6" value="DMIC0" />
<ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="DMIC MUX7" value="DMIC5" />
</path>
<path name="listen-ape-handset-qmic">
@@ -329,16 +329,16 @@
<ctl name="SLIM_0_TX Channels" value="Four" />
<ctl name="CDC_IF TX5 MUX" value="DEC5" />
<ctl name="ADC MUX5" value="DMIC" />
- <ctl name="DMIC MUX5" value="DMIC1" />
+ <ctl name="DMIC MUX5" value="DMIC2" />
<ctl name="CDC_IF TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
- <ctl name="DMIC MUX6" value="DMIC5" />
+ <ctl name="DMIC MUX6" value="DMIC0" />
<ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="DMIC MUX7" value="DMIC5" />
<ctl name="CDC_IF TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC0" />
+ <ctl name="DMIC MUX8" value="DMIC1" />
</path>
<path name="echo-reference">
diff --git a/configs/msmsteppe/sound_trigger_platform_info.xml b/configs/msmsteppe/sound_trigger_platform_info.xml
index 413f4c6..a85a180 100644
--- a/configs/msmsteppe/sound_trigger_platform_info.xml
+++ b/configs/msmsteppe/sound_trigger_platform_info.xml
@@ -54,6 +54,8 @@
</common_config>
<acdb_ids>
+ <!--For internal codec please enable below device-->
+ <!--param DEVICE_HANDSET_MIC_APE="130" /-->
<param DEVICE_HANDSET_MIC_APE="100" />
<param DEVICE_HANDSET_MIC_CPE="128" />
<param DEVICE_HANDSET_MIC_ECPP_CPE="128" />
@@ -127,6 +129,28 @@
<param read_rsp_ids="0x00020013, 0x3, 0x00020016" />
<param custom_config_ids="0x00012C0D, 0x3, 0x00012C20" />
</gcs_usecase>
+ <gcs_usecase>
+ <param uid="0x7" />
+ <param acdb_devices="DEVICE_HANDSET_DMIC_CPE" />
+ <param load_sound_model_ids="0x00012C0D, 0x7, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x7, 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x7, 0x00012B05" />
+ <param read_cmd_ids="0x00020013, 0x7, 0x00020015" />
+ <param read_rsp_ids="0x00020013, 0x7, 0x00020016" />
+ <param custom_config_ids="0x00012C0D, 0x7, 0x00012C20" />
+ <param det_event_type_ids="0x00012C0D, 0x7, 0x00012C2A" />
+ </gcs_usecase>
+ <gcs_usecase>
+ <param uid="0x8" />
+ <param acdb_devices="DEVICE_HANDSET_DMIC_CPE" />
+ <param load_sound_model_ids="0x00012C0D, 0x8, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x8, 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x8, 0x00012B05" />
+ <param read_cmd_ids="0x00020013, 0x8, 0x00020015" />
+ <param read_rsp_ids="0x00020013, 0x8, 0x00020016" />
+ <param custom_config_ids="0x00012C0D, 0x8, 0x00012C20" />
+ <param det_event_type_ids="0x00012C0D, 0x8, 0x00012C2A" />
+ </gcs_usecase>
<!-- Module and param ids with which the algorithm is integrated
in non-graphite firmware (note these must come after gcs params)
Extends flexibility to have different ids based on execution type.
diff --git a/configs/msmsteppe_au/audio_platform_info.xml b/configs/msmsteppe_au/audio_platform_info.xml
index 1b49031..a33ae3f 100644
--- a/configs/msmsteppe_au/audio_platform_info.xml
+++ b/configs/msmsteppe_au/audio_platform_info.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!-- Copyright (c) 2014, 2016-2018, The Linux Foundation. All rights reserved. -->
+<!-- Copyright (c) 2014, 2016-2019, 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 -->
@@ -97,10 +97,10 @@
<usecase name="USECASE_AUDIO_PLAYBACK_MMAP" type="out" id="28" />
<usecase name="USECASE_AUDIO_RECORD_MMAP" type="in" id="28" />
<usecase name="USECASE_AUDIO_RECORD" type="in" id="0" />
- <usecase name="USECASE_AUDIO_HFP_SCO" type="in" id="29" />
- <usecase name="USECASE_AUDIO_HFP_SCO" type="out" id="29" />
- <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="29" />
- <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="out" id="29" />
+ <usecase name="USECASE_AUDIO_HFP_SCO" type="in" id="36" />
+ <usecase name="USECASE_AUDIO_HFP_SCO" type="out" id="36" />
+ <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="36" />
+ <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="out" id="36" />
</pcm_ids>
<config_params>
diff --git a/configs/msmsteppe_au/audio_policy_configuration.xml b/configs/msmsteppe_au/audio_policy_configuration.xml
index e4aec16..4d9340d 100644
--- a/configs/msmsteppe_au/audio_policy_configuration.xml
+++ b/configs/msmsteppe_au/audio_policy_configuration.xml
@@ -145,6 +145,12 @@
<profile name="" format="AUDIO_FORMAT_AAC_ADTS_HE_V2"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000"
channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="dsd_compress_passthrough" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
diff --git a/configs/msmsteppe_au/msmsteppe_au.mk b/configs/msmsteppe_au/msmsteppe_au.mk
index 87893a5..51829bd 100644
--- a/configs/msmsteppe_au/msmsteppe_au.mk
+++ b/configs/msmsteppe_au/msmsteppe_au.mk
@@ -2,7 +2,6 @@
#
#AUDIO_FEATURE_FLAGS
BOARD_USES_ALSA_AUDIO := true
-TARGET_USES_AOSP_FOR_AUDIO := false
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
USE_CUSTOM_AUDIO_POLICY := 1
@@ -75,6 +74,9 @@
DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
endif
+#Automotive audio specific device overlays
+DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/msmsteppe_au/overlay
+
PRODUCT_COPY_FILES += \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/audio_output_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_output_policy.conf \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/audio_effects.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.conf \
@@ -184,6 +186,14 @@
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.parser.ip.buffer.size=262144
+#Enable 16 bit PCM offload by default
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.offload.pcm.16bit.enable=true
+
+#Enable 24 bit PCM offload by default
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.offload.pcm.24bit.enable=true
+
#flac sw decoder 24 bit decode capability
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.flac.sw.decoder.24bit=true
diff --git a/configs/msmsteppe_au/overlay/frameworks/base/core/res/res/values/config.xml b/configs/msmsteppe_au/overlay/frameworks/base/core/res/res/values/config.xml
new file mode 100644
index 0000000..0274f9e
--- /dev/null
+++ b/configs/msmsteppe_au/overlay/frameworks/base/core/res/res/values/config.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+/*
+** Copyright (c) 2018-2019, The Linux Foundation. All rights reserved.
+** Not a Contribution.
+*/
+/*
+** Copyright 2009, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+-->
+
+<resources>
+ <!-- Car uses hardware amplifier for volume. -->
+ <bool name="config_useFixedVolume">false</bool>
+ <!--
+ Handle volume keys directly in CarAudioService without passing them to the foreground app
+ -->
+ <bool name="config_handleVolumeKeysInWindowManager">true</bool>
+</resources>
diff --git a/configs/msmsteppe_au/overlay/packages/services/Car/service/res/values/config.xml b/configs/msmsteppe_au/overlay/packages/services/Car/service/res/values/config.xml
new file mode 100644
index 0000000..ac5f818
--- /dev/null
+++ b/configs/msmsteppe_au/overlay/packages/services/Car/service/res/values/config.xml
@@ -0,0 +1,33 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+/*
+** Copyright (c) 2018-2019, The Linux Foundation. All rights reserved.
+** Not a Contribution.
+*/
+/*
+** Copyright 2015, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+-->
+
+<!--
+ Overlay resources to configure car service based on each OEM's preference.
+ See also packages/services/Car/service/res/values/config.xml
+-->
+<resources>
+ <!-- Configuration to enable usage of dynamic audio routing. If this is set to false,
+ dynamic audio routing is disabled and audio works in legacy mode. It may be useful
+ during initial development where audio hal does not support bus based addressing yet. -->
+ <bool name="audioUseDynamicRouting">true</bool>
+</resources>
diff --git a/configs/msmsteppe_au/overlay/packages/services/Car/service/res/xml/car_volume_groups.xml b/configs/msmsteppe_au/overlay/packages/services/Car/service/res/xml/car_volume_groups.xml
new file mode 100644
index 0000000..850d4d4
--- /dev/null
+++ b/configs/msmsteppe_au/overlay/packages/services/Car/service/res/xml/car_volume_groups.xml
@@ -0,0 +1,57 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+ Copyright (c) 2018-2019, The Linux Foundation. All rights reserved.
+ Not a Contribution.
+
+ Copyright (C) 2018 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<!--
+ Defines the all available volume groups for volume control in a car.
+ One can overlay this configuration to customize the groups.
+
+ This configuration will be populated by CarAudioService and
+ surfaced to Car Settings App and/or other volume control interfaces.
+
+ Certain constraints applied to this configuration
+ - One context should not appear in two groups
+ - All contexts are assigned
+ - One bus should not appear in two groups
+ - All gain controllers (set on each bus) in one group have same step value
+
+ It is fine that there are buses that do not appear in any group, those buses
+ may be reserved for other usages.
+
+ Important note: when overlaying this configuration,
+ make sure the resources are in the same package as CarAudioService.
+-->
+<volumeGroups xmlns:car="http://schemas.android.com/apk/res-auto">
+ <group>
+ <context car:context="music"/>
+ </group>
+ <group>
+ <context car:context="call"/>
+ <context car:context="call_ring"/>
+ </group>
+ <group>
+ <context car:context="alarm"/>
+ <context car:context="notification"/>
+ <context car:context="system_sound"/>
+ </group>
+ <group>
+ <context car:context="navigation"/>
+ <context car:context="voice_command"/>
+ </group>
+</volumeGroups>
diff --git a/configs/sdm660/sdm660.mk b/configs/sdm660/sdm660.mk
index 84f0f1e..5695851 100644
--- a/configs/sdm660/sdm660.mk
+++ b/configs/sdm660/sdm660.mk
@@ -202,7 +202,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -266,7 +266,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/sdm710/sdm710.mk b/configs/sdm710/sdm710.mk
index c47a146..fb01728 100644
--- a/configs/sdm710/sdm710.mk
+++ b/configs/sdm710/sdm710.mk
@@ -296,7 +296,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -365,7 +365,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
@@ -418,6 +417,15 @@
android.hardware.audio.effect@4.0 \
android.hardware.audio.effect@4.0-impl
+# enable audio hidl hal 5.0
+PRODUCT_PACKAGES += \
+ android.hardware.audio@5.0 \
+ android.hardware.audio.common@5.0 \
+ android.hardware.audio.common@5.0-util \
+ android.hardware.audio@5.0-impl \
+ android.hardware.audio.effect@5.0 \
+ android.hardware.audio.effect@5.0-impl
+
PRODUCT_PACKAGES_ENG += \
VoicePrintTest \
VoicePrintDemo
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index 4fb1485..c3c3578 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -186,6 +186,10 @@
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.offload.buffer.size.kb=32
+#Minimum duration for offload playback in secs
+PRODUCT_PROPERTY_OVERRIDES += \
+audio.offload.min.duration.secs=30
+
#Enable offload audio video playback by default
PRODUCT_PROPERTY_OVERRIDES += \
audio.offload.video=true
@@ -229,7 +233,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -310,7 +314,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
@@ -363,6 +366,15 @@
android.hardware.audio.effect@4.0 \
android.hardware.audio.effect@4.0-impl
+# enable audio hidl hal 5.0
+PRODUCT_PACKAGES += \
+ android.hardware.audio@5.0 \
+ android.hardware.audio.common@5.0 \
+ android.hardware.audio.common@5.0-util \
+ android.hardware.audio@5.0-impl \
+ android.hardware.audio.effect@5.0 \
+ android.hardware.audio.effect@5.0-impl
+
PRODUCT_PACKAGES_ENG += \
VoicePrintTest \
VoicePrintDemo
diff --git a/configs/sdm845/sound_trigger_platform_info.xml b/configs/sdm845/sound_trigger_platform_info.xml
index 0942fab..a80765b 100644
--- a/configs/sdm845/sound_trigger_platform_info.xml
+++ b/configs/sdm845/sound_trigger_platform_info.xml
@@ -32,6 +32,7 @@
<!--- 0x0102: Includes acdb_ids param with the gcs_usecase tag. This matches -->
<!--- the gcs_usecase with the acdb device that uses it. -->
<common_config>
+ <param implementer_version="0x0100" />
<param max_cpe_sessions="1" />
<param max_wdsp_sessions="2" />
<param max_ape_sessions="8" />
diff --git a/configs/trinket/audio_effects.xml b/configs/trinket/audio_effects.xml
index a1cc069..add0925 100644
--- a/configs/trinket/audio_effects.xml
+++ b/configs/trinket/audio_effects.xml
@@ -30,9 +30,6 @@
<libraries>
<library name="bundle" path="libbundlewrapper.so"/>
<library name="reverb" path="libreverbwrapper.so"/>
- <library name="qcbassboost" path="libqcbassboost.so"/>
- <library name="qcvirt" path="libqcvirt.so"/>
- <library name="qcreverb" path="libqcreverb.so"/>
<library name="visualizer_sw" path="libvisualizer.so"/>
<library name="visualizer_hw" path="libqcomvisualizer.so"/>
<library name="downmix" path="libdownmix.so"/>
@@ -47,11 +44,11 @@
</libraries>
<effects>
<effectProxy name="bassboost" library="proxy" uuid="14804144-a5ee-4d24-aa88-0002a5d5c51b">
- <libsw library="qcbassboost" uuid="23aca180-44bd-11e2-bcfd-0800200c9a66"/>
+ <libsw library="bundle" uuid="8631f300-72e2-11df-b57e-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="2c4a8c24-1581-487f-94f6-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="virtualizer" library="proxy" uuid="d3467faa-acc7-4d34-acaf-0002a5d5c51b">
- <libsw library="qcvirt" uuid="e6c98a16-22a3-11e2-b87b-f23c91aec05e"/>
+ <libsw library="bundle" uuid="1d4033c0-8557-11df-9f2d-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="509a4498-561a-4bea-b3b1-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="equalizer" library="proxy" uuid="c8e70ecd-48ca-456e-8a4f-0002a5d5c51b">
@@ -60,19 +57,19 @@
</effectProxy>
<effect name="volume" library="bundle" uuid="119341a0-8469-11df-81f9-0002a5d5c51b"/>
<effectProxy name="reverb_env_aux" library="proxy" uuid="48404ac9-d202-4ccc-bf84-0002a5d5c51b">
- <libsw library="qcreverb" uuid="a8c1e5f3-293d-43cd-95ec-d5e26c02e217"/>
+ <libsw library="reverb" uuid="4a387fc0-8ab3-11df-8bad-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="79a18026-18fd-4185-8233-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_env_ins" library="proxy" uuid="b707403a-a1c1-4291-9573-0002a5d5c51b">
- <libsw library="qcreverb" uuid="791fff8b-8129-4655-83a4-59bc61034c3a"/>
+ <libsw library="reverb" uuid="c7a511a0-a3bb-11df-860e-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="eb64ea04-973b-43d2-8f5e-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_pre_aux" library="proxy" uuid="1b78f587-6d1c-422e-8b84-0002a5d5c51b">
- <libsw library="qcreverb" uuid="53ef1db5-c0c0-445b-b060-e34d20ebb70a"/>
+ <libsw library="reverb" uuid="f29a1400-a3bb-11df-8ddc-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="6987be09-b142-4b41-9056-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="reverb_pre_ins" library="proxy" uuid="f3e178d2-ebcb-408e-8357-0002a5d5c51b">
- <libsw library="qcreverb" uuid="b08a0e38-22a5-11e2-b87b-f23c91aec05e"/>
+ <libsw library="reverb" uuid="172cdf00-a3bc-11df-a72f-0002a5d5c51b"/>
<libhw library="offload_bundle" uuid="aa2bebf6-47cf-4613-9bca-0002a5d5c51b"/>
</effectProxy>
<effectProxy name="visualizer" library="proxy" uuid="1d0a1a53-7d5d-48f2-8e71-27fbd10d842c">
diff --git a/configs/trinket/audio_policy_configuration.xml b/configs/trinket/audio_policy_configuration.xml
index 0939e3b..8015afa 100644
--- a/configs/trinket/audio_policy_configuration.xml
+++ b/configs/trinket/audio_policy_configuration.xml
@@ -263,17 +263,20 @@
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
<devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
- encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC">
+ encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE AUDIO_FORMAT_APTX_TW
+SP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
@@ -316,27 +319,8 @@
samplingRates="8000,16000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
</devicePort>
<devicePort tagName="USB Device In" type="AUDIO_DEVICE_IN_USB_DEVICE" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
<devicePort tagName="USB Headset In" type="AUDIO_DEVICE_IN_USB_HEADSET" role="source">
- <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
- <!-- edit as needed -->
- <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
- samplingRates="44100,48000,64000,88200,96000,128000,176400,192000"
- channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_INDEX_MASK_6,AUDIO_CHANNEL_IN_7POINT1,AUDIO_CHANNEL_INDEX_MASK_8"/>
</devicePort>
</devicePorts>
diff --git a/configs/trinket/mixer_paths_idp.xml b/configs/trinket/mixer_paths_idp.xml
index b341a3c..5e769db 100644
--- a/configs/trinket/mixer_paths_idp.xml
+++ b/configs/trinket/mixer_paths_idp.xml
@@ -2185,6 +2185,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/trinket/mixer_paths_qrd.xml b/configs/trinket/mixer_paths_qrd.xml
index 7039dbb..9fbc525 100644
--- a/configs/trinket/mixer_paths_qrd.xml
+++ b/configs/trinket/mixer_paths_qrd.xml
@@ -2180,6 +2180,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
diff --git a/configs/trinket/trinket.mk b/configs/trinket/trinket.mk
index 44babfa..5176889 100644
--- a/configs/trinket/trinket.mk
+++ b/configs/trinket/trinket.mk
@@ -92,6 +92,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/trinket/mixer_paths_tavil.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tavil.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/trinket/mixer_paths_tasha.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tasha.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/trinket/mixer_paths_tashalite.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tashalite.xml \
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -185,7 +187,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -233,12 +235,15 @@
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.adm.buffering.ms=2
+#enable AAC frame ctl for A2DP sinks
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.aac_frm_ctl.enabled=true
+
#add dynamic feature flags here
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/hal/Android.mk b/hal/Android.mk
index 0ce2d6e..a671373 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -210,12 +210,14 @@
endif
# Hardware specific feature
-ifeq ($(strip $(BOARD_SUPPORTS_QAHW)),true)
- LOCAL_CFLAGS += -DAUDIO_HW_EXTN_API_ENABLED
- LOCAL_SRC_FILES += audio_hw_extn_api.c
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_QAP)),true)
+LOCAL_CFLAGS += -DQAP_EXTN_ENABLED -Wno-tautological-pointer-compare
+LOCAL_SRC_FILES += audio_extn/qap.c
+LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/qap_wrapper/
+LOCAL_HEADER_LIBRARIES += audio_qaf_headers
+LOCAL_SHARED_LIBRARIES += libqap_wrapper liblog
endif
-# Hardware specific feature
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_LISTEN)),true)
LOCAL_CFLAGS += -DAUDIO_LISTEN_ENABLED
LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/audio-listen
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index f00f74d..f9f33d1 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -98,12 +98,13 @@
bool cin_attached_usecase(audio_usecase_t uc_id);
bool cin_format_supported(audio_format_t format);
size_t cin_get_buffer_size(struct stream_in *in);
-int cin_start_input_stream(struct stream_in *in);
+int cin_open_input_stream(struct stream_in *in);
void cin_stop_input_stream(struct stream_in *in);
void cin_close_input_stream(struct stream_in *in);
+void cin_free_input_stream_resources(struct stream_in *in);
int cin_read(struct stream_in *in, void *buffer,
size_t bytes, size_t *bytes_read);
-int cin_configure_input_stream(struct stream_in *in);
+int cin_configure_input_stream(struct stream_in *in, struct audio_config *in_config);
void audio_extn_set_snd_card_split(const char* in_snd_card_name)
{
@@ -366,9 +367,9 @@
}
}
-static int update_custom_mtmx_coefficients(struct audio_device *adev,
- struct audio_custom_mtmx_params *params,
- int pcm_device_id)
+static int update_custom_mtmx_coefficients_v2(struct audio_device *adev,
+ struct audio_custom_mtmx_params *params,
+ int pcm_device_id)
{
struct mixer_ctl *ctl = NULL;
char *mixer_name_prefix = "AudStr";
@@ -430,9 +431,9 @@
return 0;
}
-static void set_custom_mtmx_params(struct audio_device *adev,
- struct audio_custom_mtmx_params_info *pinfo,
- int pcm_device_id, bool enable)
+static void set_custom_mtmx_params_v2(struct audio_device *adev,
+ struct audio_custom_mtmx_params_info *pinfo,
+ int pcm_device_id, bool enable)
{
struct mixer_ctl *ctl = NULL;
char *mixer_name_prefix = "AudStr";
@@ -465,7 +466,7 @@
ALOGE("%s: ERROR. Mixer ctl set failed", __func__);
}
-void audio_extn_set_custom_mtmx_params(struct audio_device *adev,
+void audio_extn_set_custom_mtmx_params_v2(struct audio_device *adev,
struct audio_usecase *usecase,
bool enable)
{
@@ -535,16 +536,402 @@
params = platform_get_custom_mtmx_params(adev->platform, &info);
if (params) {
if (enable)
- ret = update_custom_mtmx_coefficients(adev, params,
+ ret = update_custom_mtmx_coefficients_v2(adev, params,
pcm_device_id);
if (ret < 0)
ALOGE("%s: error updating mtmx coeffs err:%d", __func__, ret);
else
- set_custom_mtmx_params(adev, &info, pcm_device_id, enable);
+ set_custom_mtmx_params_v2(adev, &info, pcm_device_id, enable);
}
}
}
+static int set_custom_mtmx_output_channel_map(struct audio_device *adev,
+ char *mixer_name_prefix,
+ uint32_t ch_count,
+ bool enable)
+{
+ struct mixer_ctl *ctl = NULL;
+ char mixer_ctl_name[128] = {0};
+ int ret = 0;
+ int channel_map[AUDIO_MAX_DSP_CHANNELS] = {0};
+
+ ALOGV("%s channel_count %d", __func__, ch_count);
+
+ if (!enable) {
+ ALOGV("%s: reset output channel map", __func__);
+ goto exit;
+ }
+
+ switch (ch_count) {
+ case 2:
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ break;
+ case 4:
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_LS;
+ channel_map[3] = PCM_CHANNEL_RS;
+ break;
+ case 6:
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_FC;
+ channel_map[3] = PCM_CHANNEL_LFE;
+ channel_map[4] = PCM_CHANNEL_LS;
+ channel_map[5] = PCM_CHANNEL_RS;
+ break;
+ case 8:
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_FC;
+ channel_map[3] = PCM_CHANNEL_LFE;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ break;
+ case 10:
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_LFE;
+ channel_map[3] = PCM_CHANNEL_FC;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ channel_map[8] = PCM_CHANNEL_TFL;
+ channel_map[9] = PCM_CHANNEL_TFR;
+ break;
+ case 12:
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_FC;
+ channel_map[3] = PCM_CHANNEL_LFE;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ channel_map[8] = PCM_CHANNEL_TFL;
+ channel_map[9] = PCM_CHANNEL_TFR;
+ channel_map[10] = PCM_CHANNEL_TSL;
+ channel_map[11] = PCM_CHANNEL_TSR;
+ break;
+ case 14:
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_LFE;
+ channel_map[3] = PCM_CHANNEL_FC;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ channel_map[8] = PCM_CHANNEL_TFL;
+ channel_map[9] = PCM_CHANNEL_TFR;
+ channel_map[10] = PCM_CHANNEL_TSL;
+ channel_map[11] = PCM_CHANNEL_TSR;
+ channel_map[12] = PCM_CHANNEL_FLC;
+ channel_map[13] = PCM_CHANNEL_FRC;
+ break;
+ case 16:
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_FC;
+ channel_map[3] = PCM_CHANNEL_LFE;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ channel_map[8] = PCM_CHANNEL_TFL;
+ channel_map[9] = PCM_CHANNEL_TFR;
+ channel_map[10] = PCM_CHANNEL_TSL;
+ channel_map[11] = PCM_CHANNEL_TSR;
+ channel_map[12] = PCM_CHANNEL_FLC;
+ channel_map[13] = PCM_CHANNEL_FRC;
+ channel_map[14] = PCM_CHANNEL_RLC;
+ channel_map[15] = PCM_CHANNEL_RRC;
+ break;
+ default:
+ ALOGE("%s: unsupported channels(%d) for setting channel map",
+ __func__, ch_count);
+ return -EINVAL;
+ }
+
+exit:
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "%s %s",
+ mixer_name_prefix, "Output Channel Map");
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: ERROR. Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ ret = mixer_ctl_set_array(ctl, channel_map, ch_count);
+ return ret;
+}
+
+static int update_custom_mtmx_coefficients_v1(struct audio_device *adev,
+ struct audio_custom_mtmx_params *params,
+ struct audio_custom_mtmx_in_params *in_params,
+ int pcm_device_id,
+ usecase_type_t type,
+ bool enable)
+{
+ struct mixer_ctl *ctl = NULL;
+ char mixer_ctl_name[128] = {0};
+ struct audio_custom_mtmx_params_info *pinfo = ¶ms->info;
+ char mixer_name_prefix[100];
+ int i = 0, err = 0, rule = 0;
+ uint32_t mtrx_row_cnt = 0, mtrx_column_cnt = 0;
+ int reset_coeffs[AUDIO_MAX_DSP_CHANNELS] = {0};
+
+ ALOGI("%s: ip_channels %d, op_channels %d, pcm_device_id %d, usecase type %d, enable %d",
+ __func__, pinfo->ip_channels, pinfo->op_channels, pcm_device_id,
+ type, enable);
+
+ if (!strcmp(pinfo->fe_name, "")) {
+ ALOGE("%s: Error. no front end defined", __func__);
+ return -EINVAL;
+ }
+
+ strlcpy(mixer_name_prefix, pinfo->fe_name, sizeof(mixer_name_prefix));
+
+ /*
+ * Enable/Disable channel mixer.
+ * If enable, use params and in_params to configure mixer.
+ * If disable, reset previously configured mixer.
+ */
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "%s %s",
+ mixer_name_prefix, "Channel Mixer");
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: ERROR. Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ if (enable)
+ err = mixer_ctl_set_enum_by_string(ctl, "Enable");
+ else
+ err = mixer_ctl_set_enum_by_string(ctl, "Disable");
+
+ if (err) {
+ ALOGE("%s: ERROR. %s channel mixer failed", __func__,
+ enable ? "Enable" : "Disable");
+ return -EINVAL;
+ }
+
+ /* Configure output channels of channel mixer */
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "%s %s",
+ mixer_name_prefix, "Channels");
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: ERROR. Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ mtrx_row_cnt = pinfo->op_channels;
+ mtrx_column_cnt = pinfo->ip_channels;
+
+ if (enable)
+ err = mixer_ctl_set_value(ctl, 0, mtrx_row_cnt);
+ else
+ err = mixer_ctl_set_value(ctl, 0, 0);
+
+ if (err) {
+ ALOGE("%s: ERROR. %s mixer output channels failed", __func__,
+ enable ? "Set" : "Reset");
+ return -EINVAL;
+ }
+
+
+ /* To keep output channel map in sync with asm driver channel mapping */
+ err = set_custom_mtmx_output_channel_map(adev, mixer_name_prefix, mtrx_row_cnt,
+ enable);
+ if (err) {
+ ALOGE("%s: ERROR. %s mtmx output channel map failed", __func__,
+ enable ? "Set" : "Reset");
+ return -EINVAL;
+ }
+
+ /* Send channel mixer rule */
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "%s %s",
+ mixer_name_prefix, "Channel Rule");
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: ERROR. Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ mixer_ctl_set_value(ctl, 0, rule);
+
+ /* Send channel coefficients for each output channel */
+ for (i = 0; i < mtrx_row_cnt; i++) {
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "%s %s%d",
+ mixer_name_prefix, "Output Channel", i+1);
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: ERROR. Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ if (enable)
+ err = mixer_ctl_set_array(ctl,
+ ¶ms->coeffs[mtrx_column_cnt * i],
+ mtrx_column_cnt);
+ else
+ err = mixer_ctl_set_array(ctl,
+ reset_coeffs,
+ mtrx_column_cnt);
+ if (err) {
+ ALOGE("%s: ERROR. %s coefficients failed for output channel %d",
+ __func__, enable ? "Set" : "Reset", i);
+ return -EINVAL;
+ }
+ }
+
+ /* Configure backend interfaces with information provided in xml */
+ i = 0;
+ while (in_params->in_ch_info[i].ch_count != 0) {
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "%s %s%d",
+ mixer_name_prefix, "Channel", i+1);
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: ERROR. Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+ if (enable) {
+ ALOGD("%s: mixer %s, interface %s", __func__, mixer_ctl_name,
+ in_params->in_ch_info[i].hw_interface);
+ err = mixer_ctl_set_enum_by_string(ctl,
+ in_params->in_ch_info[i].hw_interface);
+ } else {
+ err = mixer_ctl_set_enum_by_string(ctl, "ZERO");
+ }
+
+ if (err) {
+ ALOGE("%s: ERROR. %s channel backend interface failed", __func__,
+ enable ? "Set" : "Reset");
+ return -EINVAL;
+ }
+ i++;
+ }
+
+ return 0;
+}
+
+
+void audio_extn_set_custom_mtmx_params_v1(struct audio_device *adev,
+ struct audio_usecase *usecase,
+ bool enable)
+{
+ struct audio_custom_mtmx_params_info info = {0};
+ struct audio_custom_mtmx_params *params = NULL;
+ struct audio_custom_mtmx_in_params_info in_info = {0};
+ struct audio_custom_mtmx_in_params *in_params = NULL;
+ int pcm_device_id = -1, ret = 0;
+ uint32_t feature_id = 0;
+
+ switch(usecase->type) {
+ case PCM_CAPTURE:
+ if (usecase->stream.in) {
+ pcm_device_id =
+ platform_get_pcm_device_id(usecase->id, PCM_CAPTURE);
+ info.snd_device = usecase->in_snd_device;
+ } else {
+ ALOGE("%s: invalid input stream for capture usecase id:%d",
+ __func__, usecase->id);
+ return;
+ }
+ break;
+ case PCM_PLAYBACK:
+ default:
+ ALOGV("%s: unsupported usecase id:%d", __func__, usecase->id);
+ return;
+ }
+
+ ALOGD("%s: snd device %d", __func__, info.snd_device);
+ info.id = feature_id;
+ info.usecase_id = usecase->id;
+ info.op_channels = audio_channel_count_from_in_mask(
+ usecase->stream.in->channel_mask);
+
+ in_info.usecase_id = info.usecase_id;
+ in_info.op_channels = info.op_channels;
+ in_params = platform_get_custom_mtmx_in_params(adev->platform, &in_info);
+ if (!in_params) {
+ ALOGE("%s: Could not get in params for usecase %d, channels %d",
+ __func__, in_info.usecase_id, in_info.op_channels);
+ return;
+ }
+
+ info.ip_channels = in_params->ip_channels;
+ ALOGD("%s: ip channels %d, op channels %d", __func__, info.ip_channels, info.op_channels);
+
+ params = platform_get_custom_mtmx_params(adev->platform, &info);
+ if (params) {
+ ret = update_custom_mtmx_coefficients_v1(adev, params, in_params,
+ pcm_device_id, usecase->type, enable);
+ if (ret < 0)
+ ALOGE("%s: error updating mtmx coeffs err:%d", __func__, ret);
+ }
+}
+
+snd_device_t audio_extn_get_loopback_snd_device(struct audio_device *adev,
+ struct audio_usecase *usecase,
+ int channel_count)
+{
+ snd_device_t snd_device = SND_DEVICE_NONE;
+ struct audio_custom_mtmx_in_params_info in_info = {0};
+ struct audio_custom_mtmx_in_params *in_params = NULL;
+
+ if (!adev || !usecase) {
+ ALOGE("%s: Invalid params", __func__);
+ return snd_device;
+ }
+
+ in_info.usecase_id = usecase->id;
+ in_info.op_channels = channel_count;
+ in_params = platform_get_custom_mtmx_in_params(adev->platform, &in_info);
+ if (!in_params) {
+ ALOGE("%s: Could not get in params for usecase %d, channels %d",
+ __func__, in_info.usecase_id, in_info.op_channels);
+ return snd_device;
+ }
+
+ switch(in_params->mic_ch) {
+ case 2:
+ snd_device = SND_DEVICE_IN_HANDSET_DMIC_AND_EC_REF_LOOPBACK;
+ break;
+ case 4:
+ snd_device = SND_DEVICE_IN_HANDSET_QMIC_AND_EC_REF_LOOPBACK;
+ break;
+ case 6:
+ snd_device = SND_DEVICE_IN_HANDSET_6MIC_AND_EC_REF_LOOPBACK;
+ break;
+ case 8:
+ snd_device = SND_DEVICE_IN_HANDSET_8MIC_AND_EC_REF_LOOPBACK;
+ break;
+ default:
+ ALOGE("%s: Unsupported mic channels %d",
+ __func__, in_params->mic_ch);
+ break;
+ }
+
+ ALOGD("%s: return snd device %d", __func__, snd_device);
+ return snd_device;
+}
+
#ifndef DTS_EAGLE
#define audio_extn_hpx_set_parameters(adev, parms) (0)
#define audio_extn_hpx_get_parameters(query, reply) (0)
@@ -2893,10 +3280,10 @@
*channel_mask_updated = false;
int max_mic_count = platform_get_max_mic_count(adev->platform);
- /* validate input params*/
+ /* validate input params. Avoid updated channel mask if loopback device */
if ((channel_count == 6) &&
- (in->format == AUDIO_FORMAT_PCM_16_BIT)) {
-
+ (in->format == AUDIO_FORMAT_PCM_16_BIT) &&
+ (!is_loopback_input_device(in->device))) {
switch (max_mic_count) {
case 4:
config->channel_mask = AUDIO_CHANNEL_INDEX_MASK_4;
@@ -4262,52 +4649,52 @@
int audio_extn_ext_hw_plugin_deinit(void *plugin)
{
return ((ext_hw_plugin_deinit) ?
- ext_hw_plugin_deinit(plugin): -1);
+ ext_hw_plugin_deinit(plugin): 0);
}
int audio_extn_ext_hw_plugin_usecase_start(void *plugin, struct audio_usecase *usecase)
{
return ((ext_hw_plugin_usecase_start) ?
- ext_hw_plugin_usecase_start(plugin, usecase): -1);
+ ext_hw_plugin_usecase_start(plugin, usecase): 0);
}
int audio_extn_ext_hw_plugin_usecase_stop(void *plugin, struct audio_usecase *usecase)
{
return ((ext_hw_plugin_usecase_stop) ?
- ext_hw_plugin_usecase_stop(plugin, usecase): -1);
+ ext_hw_plugin_usecase_stop(plugin, usecase): 0);
}
int audio_extn_ext_hw_plugin_set_parameters(void *plugin,
struct str_parms *parms)
{
return ((ext_hw_plugin_set_parameters) ?
- ext_hw_plugin_set_parameters(plugin, parms): -1);
+ ext_hw_plugin_set_parameters(plugin, parms): 0);
}
int audio_extn_ext_hw_plugin_get_parameters(void *plugin,
struct str_parms *query, struct str_parms *reply)
{
return ((ext_hw_plugin_get_parameters) ?
- ext_hw_plugin_get_parameters(plugin, query, reply): -1);
+ ext_hw_plugin_get_parameters(plugin, query, reply): 0);
}
int audio_extn_ext_hw_plugin_set_mic_mute(void *plugin, bool mute)
{
return ((ext_hw_plugin_set_mic_mute) ?
- ext_hw_plugin_set_mic_mute(plugin, mute): -1);
+ ext_hw_plugin_set_mic_mute(plugin, mute): 0);
}
int audio_extn_ext_hw_plugin_get_mic_mute(void *plugin, bool *mute)
{
return ((ext_hw_plugin_get_mic_mute) ?
- ext_hw_plugin_get_mic_mute(plugin, mute): -1);
+ ext_hw_plugin_get_mic_mute(plugin, mute): 0);
}
int audio_extn_ext_hw_plugin_set_audio_gain(void *plugin,
struct audio_usecase *usecase, uint32_t gain)
{
return ((ext_hw_plugin_set_audio_gain) ?
- ext_hw_plugin_set_audio_gain(plugin, usecase, gain): -1);
+ ext_hw_plugin_set_audio_gain(plugin, usecase, gain): 0);
}
// END: EXT_HW_PLUGIN ===================================================================
@@ -4608,6 +4995,7 @@
audio_extn_utils_is_dolby_format;
passthru_init(init_config);
ALOGD("%s:: ---- Feature HDMI_PASSTHROUGH is Enabled ----", __func__);
+ return;
}
feature_disabled:
@@ -4676,9 +5064,9 @@
{
return (audio_extn_compress_in_enabled? cin_get_buffer_size(in): 0);
}
-int audio_extn_cin_start_input_stream(struct stream_in *in)
+int audio_extn_cin_open_input_stream(struct stream_in *in)
{
- return (audio_extn_compress_in_enabled? cin_start_input_stream(in): -1);
+ return (audio_extn_compress_in_enabled? cin_open_input_stream(in): -1);
}
void audio_extn_cin_stop_input_stream(struct stream_in *in)
{
@@ -4688,15 +5076,19 @@
{
(audio_extn_compress_in_enabled? cin_close_input_stream(in): NULL);
}
+void audio_extn_cin_free_input_stream_resources(struct stream_in *in)
+{
+ return (audio_extn_compress_in_enabled? cin_free_input_stream_resources(in): NULL);
+}
int audio_extn_cin_read(struct stream_in *in, void *buffer,
size_t bytes, size_t *bytes_read)
{
return (audio_extn_compress_in_enabled?
cin_read(in, buffer, bytes, bytes_read): -1);
}
-int audio_extn_cin_configure_input_stream(struct stream_in *in)
+int audio_extn_cin_configure_input_stream(struct stream_in *in, struct audio_config *in_config)
{
- return (audio_extn_compress_in_enabled? cin_configure_input_stream(in): -1);
+ return (audio_extn_compress_in_enabled? cin_configure_input_stream(in, in_config): -1);
}
// END: COMPRESS_IN ====================================================
@@ -4721,7 +5113,6 @@
void battery_listener_feature_init(bool is_feature_enabled)
{
audio_extn_battery_listener_enabled = is_feature_enabled;
- ALOGD("%s: ---- Feature BATTERY_LISTENER is %s----", __func__, is_feature_enabled? "ENABLED": "NOT ENABLED");
if (is_feature_enabled) {
// dlopen lib
batt_listener_lib_handle = dlopen(BATTERY_LISTENER_LIB_PATH, RTLD_NOW);
@@ -4741,6 +5132,8 @@
ALOGE("%s: dlsym failed", __func__);
goto feature_disabled;
}
+ ALOGD("%s: ---- Feature BATTERY_LISTENER is enabled ----", __func__);
+ return;
}
feature_disabled:
@@ -5141,6 +5534,8 @@
audio_extn_passthru_set_parameters(adev, parms);
audio_extn_ext_disp_set_parameters(adev, parms);
audio_extn_qaf_set_parameters(adev, parms);
+ if (audio_extn_qap_is_enabled())
+ audio_extn_qap_set_parameters(adev, parms);
if (adev->offload_effects_set_parameters != NULL)
adev->offload_effects_set_parameters(parms);
audio_extn_set_aptx_dec_bt_addr(adev, parms);
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 5e2e643..7364d76 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -216,6 +216,9 @@
//END: EXTN_QDSP_PLUGIN ===========================================
+#define MIN_OFFLOAD_BUFFER_DURATION_MS 5 /* 5ms */
+#define MAX_OFFLOAD_BUFFER_DURATION_MS (100 * 1000) /* 100s */
+
void audio_extn_set_parameters(struct audio_device *adev,
struct str_parms *parms);
@@ -434,6 +437,7 @@
#define audio_extn_sound_trigger_update_device_status(snd_dev, event) (0)
#define audio_extn_sound_trigger_update_stream_status(uc_info, event) (0)
#define audio_extn_sound_trigger_update_battery_status(charging) (0)
+#define audio_extn_sound_trigger_update_screen_status(screen_off) (0)
#define audio_extn_sound_trigger_set_parameters(adev, parms) (0)
#define audio_extn_sound_trigger_get_parameters(adev, query, reply) (0)
#define audio_extn_sound_trigger_check_and_get_session(in) (0)
@@ -458,6 +462,7 @@
void audio_extn_sound_trigger_update_stream_status(struct audio_usecase *uc_info,
st_event_type_t event);
void audio_extn_sound_trigger_update_battery_status(bool charging);
+void audio_extn_sound_trigger_update_screen_status(bool screen_off);
void audio_extn_sound_trigger_set_parameters(struct audio_device *adev,
struct str_parms *parms);
void audio_extn_sound_trigger_check_and_get_session(struct stream_in *in);
@@ -910,6 +915,59 @@
#define audio_extn_is_qaf_stream(out) (0)
#endif
+
+#ifdef QAP_EXTN_ENABLED
+/*
+ * Helper funtion to know if HAL QAP extention is enabled or not.
+ */
+bool audio_extn_qap_is_enabled();
+/*
+ * QAP HAL extention init, called during bootup/HAL device open.
+ * QAP library will be loaded in this funtion.
+ */
+int audio_extn_qap_init(struct audio_device *adev);
+void audio_extn_qap_deinit();
+/*
+ * if HAL QAP is enabled and inited succesfully then all then this funtion
+ * gets called for all the open_output_stream requests, in other words
+ * the core audio_hw->open_output_stream is overridden by this funtion
+ */
+int audio_extn_qap_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address __unused);
+void audio_extn_qap_close_output_stream(struct audio_hw_device *dev __unused,
+ struct audio_stream_out *stream);
+/*
+ * this funtion is how HAL QAP extention gets to know the device connection/disconnection
+ */
+int audio_extn_qap_set_parameters(struct audio_device *adev, struct str_parms *parms);
+int audio_extn_qap_out_set_param_data(struct stream_out *out,
+ audio_extn_param_id param_id,
+ audio_extn_param_payload *payload);
+int audio_extn_qap_out_get_param_data(struct stream_out *out,
+ audio_extn_param_id param_id,
+ audio_extn_param_payload *payload);
+/*
+ * helper funtion.
+ */
+bool audio_extn_is_qap_stream(struct stream_out *out);
+#else
+#define audio_extn_qap_is_enabled() (0)
+#define audio_extn_qap_deinit() (0)
+#define audio_extn_qap_close_output_stream adev_close_output_stream
+#define audio_extn_qap_open_output_stream adev_open_output_stream
+#define audio_extn_qap_init(adev) (0)
+#define audio_extn_qap_set_parameters(adev, parms) (0)
+#define audio_extn_qap_out_set_param_data(out, param_id, payload) (0)
+#define audio_extn_qap_out_get_param_data(out, param_id, payload) (0)
+#define audio_extn_is_qap_stream(out) (0)
+#endif
+
+
#ifdef AUDIO_EXTN_BT_HAL_ENABLED
int audio_extn_bt_hal_load(void **handle);
int audio_extn_bt_hal_open_output_stream(void *handle, int in_rate, audio_channel_mask_t channel_mask, int bit_width);
@@ -1011,12 +1069,13 @@
bool audio_extn_cin_attached_usecase(audio_usecase_t uc_id);
bool audio_extn_cin_format_supported(audio_format_t format);
size_t audio_extn_cin_get_buffer_size(struct stream_in *in);
-int audio_extn_cin_start_input_stream(struct stream_in *in);
+int audio_extn_cin_open_input_stream(struct stream_in *in);
void audio_extn_cin_stop_input_stream(struct stream_in *in);
void audio_extn_cin_close_input_stream(struct stream_in *in);
+void audio_extn_cin_free_input_stream_resources(struct stream_in *in);
int audio_extn_cin_read(struct stream_in *in, void *buffer,
size_t bytes, size_t *bytes_read);
-int audio_extn_cin_configure_input_stream(struct stream_in *in);
+int audio_extn_cin_configure_input_stream(struct stream_in *in, struct audio_config *in_config);
// END: COMPRESS_INPUT_ENABLED ===============================
//START: SOURCE_TRACKING_FEATURE ==============================================
@@ -1091,6 +1150,8 @@
uint64_t *frames, struct timespec *timestamp, int32_t clock_id);
int audio_extn_utils_pcm_get_dsp_presentation_pos(struct stream_out *out,
uint64_t *frames, struct timespec *timestamp, int32_t clock_id);
+size_t audio_extn_utils_get_input_buffer_size(uint32_t, audio_format_t, int, int64_t, bool);
+int audio_extn_utils_get_perf_mode_flag(void);
#ifdef AUDIO_HW_LOOPBACK_ENABLED
/* API to create audio patch */
int audio_extn_hw_loopback_create_audio_patch(struct audio_hw_device *dev,
@@ -1218,6 +1279,13 @@
#define audio_extn_auto_hal_create_audio_patch(dev, num_sources,\
sources, num_sinks, sinks, handle) (0)
#define audio_extn_auto_hal_release_audio_patch(dev, handle) (0)
+#define audio_extn_auto_hal_get_car_audio_stream_from_address(address) (-1)
+#define audio_extn_auto_hal_open_output_stream(out) (0)
+#define audio_extn_auto_hal_is_bus_device_usecase(uc_id) (0)
+#define audio_extn_auto_hal_get_snd_device_for_car_audio_stream(out) (0)
+#define audio_extn_auto_hal_get_audio_port(dev, config) (0)
+#define audio_extn_auto_hal_set_audio_port_config(dev, config) (0)
+#define audio_extn_auto_hal_set_parameters(adev, parms) (0)
#else
int32_t audio_extn_auto_hal_init(struct audio_device *adev);
void audio_extn_auto_hal_deinit(void);
@@ -1231,6 +1299,16 @@
audio_patch_handle_t *handle);
int audio_extn_auto_hal_release_audio_patch(struct audio_hw_device *dev,
audio_patch_handle_t handle);
+int32_t audio_extn_auto_hal_get_car_audio_stream_from_address(const char *address);
+int32_t audio_extn_auto_hal_open_output_stream(struct stream_out *out);
+bool audio_extn_auto_hal_is_bus_device_usecase(audio_usecase_t uc_id);
+snd_device_t audio_extn_auto_hal_get_snd_device_for_car_audio_stream(struct stream_out *out);
+int audio_extn_auto_hal_get_audio_port(struct audio_hw_device *dev,
+ struct audio_port *config);
+int audio_extn_auto_hal_set_audio_port_config(struct audio_hw_device *dev,
+ const struct audio_port_config *config);
+void audio_extn_auto_hal_set_parameters(struct audio_device *adev,
+ struct str_parms *parms);
#endif
bool audio_extn_edid_is_supported_sr(edid_audio_info* info, int sr);
@@ -1252,7 +1330,13 @@
void audio_extn_set_cpu_affinity();
bool audio_extn_is_record_play_concurrency_enabled();
bool audio_extn_is_concurrent_capture_enabled();
-void audio_extn_set_custom_mtmx_params(struct audio_device *adev,
+void audio_extn_set_custom_mtmx_params_v2(struct audio_device *adev,
struct audio_usecase *usecase,
bool enable);
+void audio_extn_set_custom_mtmx_params_v1(struct audio_device *adev,
+ struct audio_usecase *usecase,
+ bool enable);
+snd_device_t audio_extn_get_loopback_snd_device(struct audio_device *adev,
+ struct audio_usecase *usecase,
+ int channel_count);
#endif /* AUDIO_EXTN_H */
diff --git a/hal/audio_extn/auto_hal.c b/hal/audio_extn/auto_hal.c
index c70dc17..ad5e331 100644
--- a/hal/audio_extn/auto_hal.c
+++ b/hal/audio_extn/auto_hal.c
@@ -54,12 +54,23 @@
typedef struct auto_hal_module {
struct audio_device *adev;
+ card_status_t card_status;
struct hostless_config hostless;
} auto_hal_module_t;
/* Auto hal module struct */
static struct auto_hal_module *auto_hal = NULL;
+extern struct pcm_config pcm_config_deep_buffer;
+extern struct pcm_config pcm_config_low_latency;
+
+static const audio_usecase_t bus_device_usecases[] = {
+ USECASE_AUDIO_PLAYBACK_MEDIA,
+ USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION,
+ USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE,
+ USECASE_AUDIO_PLAYBACK_PHONE,
+};
+
/* Note: Due to ADP H/W design, SoC TERT/SEC TDM CLK and FSYNC lines are
* both connected with CODEC and a single master is needed to provide
* consistent CLK and FSYNC to slaves, hence configuring SoC TERT TDM as
@@ -310,6 +321,274 @@
return ret;
}
+int32_t audio_extn_auto_hal_get_car_audio_stream_from_address(const char *address)
+{
+ int32_t bus_num = -1;
+ char *str = NULL;
+ char *last_r = NULL;
+ char local_address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
+
+ /* bus device with null address error out */
+ if (address == NULL) {
+ ALOGE("%s: null address for car stream", __func__);
+ return -1;
+ }
+
+ /* strtok will modify the original string. make a copy first */
+ strlcpy(local_address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN);
+
+ /* extract bus number from address */
+ str = strtok_r(local_address, "BUS_",&last_r);
+ if (str != NULL)
+ bus_num = (int32_t)strtol(str, (char **)NULL, 10);
+
+ /* validate bus number */
+ if ((bus_num < 0) || (bus_num >= MAX_CAR_AUDIO_STREAMS)) {
+ ALOGE("%s: invalid bus number %d", __func__, bus_num);
+ return -1;
+ }
+
+ return (0x1 << bus_num);
+}
+
+int32_t audio_extn_auto_hal_open_output_stream(struct stream_out *out)
+{
+ int ret = 0;
+ unsigned int channels = audio_channel_count_from_out_mask(out->channel_mask);
+
+ switch(out->car_audio_stream) {
+ case CAR_AUDIO_STREAM_MEDIA:
+ /* media bus stream shares pcm device with deep-buffer */
+ out->usecase = USECASE_AUDIO_PLAYBACK_MEDIA;
+ out->config = pcm_config_deep_buffer;
+ out->config.period_size = get_output_period_size(out->sample_rate, out->format,
+ channels, DEEP_BUFFER_OUTPUT_PERIOD_DURATION);
+ if (out->config.period_size <= 0) {
+ ALOGE("Invalid configuration period size is not valid");
+ ret = -EINVAL;
+ goto error;
+ }
+ break;
+ case CAR_AUDIO_STREAM_SYS_NOTIFICATION:
+ /* sys notification bus stream shares pcm device with low-latency */
+ out->usecase = USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION;
+ out->config = pcm_config_low_latency;
+ break;
+ case CAR_AUDIO_STREAM_NAV_GUIDANCE:
+ out->usecase = USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE;
+ out->config = pcm_config_deep_buffer;
+ out->config.period_size = get_output_period_size(out->sample_rate, out->format,
+ channels, DEEP_BUFFER_OUTPUT_PERIOD_DURATION);
+ if (out->config.period_size <= 0) {
+ ALOGE("Invalid configuration period size is not valid");
+ ret = -EINVAL;
+ goto error;
+ }
+ break;
+ case CAR_AUDIO_STREAM_PHONE:
+ out->usecase = USECASE_AUDIO_PLAYBACK_PHONE;
+ out->config = pcm_config_low_latency;
+ break;
+ default:
+ ALOGE("%s: Car audio stream %x not supported", __func__,
+ out->car_audio_stream);
+ ret = -EINVAL;
+ goto error;
+ }
+
+error:
+ return ret;
+}
+
+bool audio_extn_auto_hal_is_bus_device_usecase(audio_usecase_t uc_id)
+{
+ unsigned int i;
+ for (i = 0; i < sizeof(bus_device_usecases)/sizeof(bus_device_usecases[0]); i++) {
+ if (uc_id == bus_device_usecases[i])
+ return true;
+ }
+ return false;
+}
+
+snd_device_t audio_extn_auto_hal_get_snd_device_for_car_audio_stream(struct stream_out *out)
+{
+ snd_device_t snd_device = SND_DEVICE_NONE;
+
+ switch(out->car_audio_stream) {
+ case CAR_AUDIO_STREAM_MEDIA:
+ snd_device = SND_DEVICE_OUT_BUS_MEDIA;
+ break;
+ case CAR_AUDIO_STREAM_SYS_NOTIFICATION:
+ snd_device = SND_DEVICE_OUT_BUS_SYS;
+ break;
+ case CAR_AUDIO_STREAM_NAV_GUIDANCE:
+ snd_device = SND_DEVICE_OUT_BUS_NAV;
+ break;
+ case CAR_AUDIO_STREAM_PHONE:
+ snd_device = SND_DEVICE_OUT_BUS_PHN;
+ break;
+ default:
+ ALOGE("%s: Unknown car audio stream (%x)",
+ __func__, out->car_audio_stream);
+ }
+ return snd_device;
+}
+
+int audio_extn_auto_hal_get_audio_port(struct audio_hw_device *dev __unused,
+ struct audio_port *config __unused)
+{
+ return -ENOSYS;
+}
+
+/* Volume min/max defined by audio policy configuration in millibel.
+ * Support a range of -60dB to 6dB.
+ */
+#define MIN_VOLUME_VALUE_MB -6000
+#define MAX_VOLUME_VALUE_MB 600
+#define STEP_VALUE_MB 100
+int audio_extn_auto_hal_set_audio_port_config(struct audio_hw_device *dev,
+ const struct audio_port_config *config)
+{
+ struct audio_device *adev = (struct audio_device *)dev;
+ int ret = 0;
+ struct listnode *node = NULL;
+ float volume = 0.0;
+
+ ALOGV("%s: enter", __func__);
+
+ if (!config) {
+ ALOGE("%s: invalid input parameters", __func__);
+ return -EINVAL;
+ }
+
+ /* For Android automotive, audio port config from car framework
+ * allows volume gain to be set to device at audio HAL level, where
+ * the gain can be applied in DSP mixer or CODEC amplifier.
+ *
+ * Following routing should be considered:
+ * MIX -> DEVICE
+ * DEVICE -> MIX
+ * DEVICE -> DEVICE
+ *
+ * For BUS devices routed to/from mixer, gain will be applied to DSP
+ * mixer via kernel control which audio HAL stream is associated with.
+ *
+ * For external (source) device (FM TUNER/AUX), routing is typically
+ * done with AudioPatch to (sink) device (SPKR), thus gain should be
+ * applied to CODEC amplifier via codec plugin extention as audio HAL
+ * stream may not be available for external audio routing.
+ */
+ if (config->type == AUDIO_PORT_TYPE_DEVICE) {
+ ALOGI("%s: device port: type %x, address %s, gain %d mB", __func__,
+ config->ext.device.type,
+ config->ext.device.address,
+ config->gain.values[0]);
+ if (config->role == AUDIO_PORT_ROLE_SINK) {
+ /* handle output devices */
+ pthread_mutex_lock(&adev->lock);
+ list_for_each(node, &adev->active_outputs_list) {
+ streams_output_ctxt_t *out_ctxt = node_to_item(node,
+ streams_output_ctxt_t,
+ list);
+ /* limit audio gain support for bus device only */
+ if (out_ctxt->output->devices == AUDIO_DEVICE_OUT_BUS &&
+ out_ctxt->output->devices == config->ext.device.type &&
+ strcmp(out_ctxt->output->address,
+ config->ext.device.address) == 0) {
+ /* millibel = 1/100 dB = 1/1000 bel
+ * q13 = (10^(mdb/100/20))*(2^13)
+ */
+ if(config->gain.values[0] <= (MIN_VOLUME_VALUE_MB + STEP_VALUE_MB))
+ volume = 0.0 ;
+ else
+ volume = powf(10.0, ((float)config->gain.values[0] / 2000));
+ ALOGV("%s: set volume to stream: %p", __func__,
+ &out_ctxt->output->stream);
+ /* set gain if output stream is active */
+ out_ctxt->output->stream.set_volume(
+ &out_ctxt->output->stream,
+ volume, volume);
+ }
+ }
+ /* NOTE: Ideally audio patch list is a superset of output stream list above.
+ * However, audio HAL does not maintain patches for mix -> device or
+ * device -> mix currently. Thus doing separate lookups for device ->
+ * device in audio patch list.
+ * FIXME: Cannot cache the gain if audio patch is not created. Expected gain
+ * to be part of port config upon audio patch creation. If not, need
+ * to create a list of audio port configs in adev context.
+ */
+#if 0
+ list_for_each(node, &adev->audio_patch_record_list) {
+ struct audio_patch_record *patch_record = node_to_item(node,
+ struct audio_patch_record,
+ list);
+ /* limit audio gain support for bus device only */
+ if (patch_record->sink.type == AUDIO_PORT_TYPE_DEVICE &&
+ patch_record->sink.role == AUDIO_PORT_ROLE_SINK &&
+ patch_record->sink.ext.device.type == AUDIO_DEVICE_OUT_BUS &&
+ patch_record->sink.ext.device.type == config->ext.device.type &&
+ strcmp(patch_record->sink.ext.device.address,
+ config->ext.device.address) == 0) {
+ /* cache / update gain per audio patch sink */
+ patch_record->sink.gain = config->gain;
+
+ struct audio_usecase *uc_info = get_usecase_from_list(adev,
+ patch_record->usecase);
+ if (!uc_info) {
+ ALOGE("%s: failed to find the usecase %d",
+ __func__, patch_record->usecase);
+ ret = -EINVAL;
+ } else {
+ volume = config->gain->values[0];
+ /* linear interpolation from millibel to level */
+ int vol_level = lrint(((volume + (0 - MIN_VOLUME_VALUE_MB)) /
+ (MAX_VOLUME_VALUE_MB - MIN_VOLUME_VALUE_MB)) * 40);
+ ALOGV("%s: set volume to patch: %p", __func__,
+ patch_record->handle);
+ ret = audio_extn_ext_hw_plugin_set_audio_gain(adev,
+ uc_info, vol_level);
+ }
+ }
+ }
+#endif
+ pthread_mutex_unlock(&adev->lock);
+ } else if (config->role == AUDIO_PORT_ROLE_SOURCE) {
+ // FIXME: handle input devices.
+ }
+ }
+
+ /* Only handle device port currently. */
+
+ ALOGV("%s: exit", __func__);
+ return ret;
+}
+
+void audio_extn_auto_hal_set_parameters(struct audio_device *adev __unused,
+ struct str_parms *parms)
+{
+ int ret = 0;
+ char value[32]={0};
+
+ ALOGV("%s: enter", __func__);
+
+ ret = str_parms_get_str(parms, "SND_CARD_STATUS", value, sizeof(value));
+ if (ret >= 0) {
+ char *snd_card_status = value+2;
+ ALOGV("%s: snd card status %s", __func__, snd_card_status);
+ if (strstr(snd_card_status, "OFFLINE")) {
+ auto_hal->card_status = CARD_STATUS_OFFLINE;
+ audio_extn_auto_hal_disable_hostless();
+ }
+ else if (strstr(snd_card_status, "ONLINE")) {
+ auto_hal->card_status = CARD_STATUS_ONLINE;
+ audio_extn_auto_hal_enable_hostless();
+ }
+ }
+
+ ALOGV("%s: exit", __func__);
+}
+
int32_t audio_extn_auto_hal_init(struct audio_device *adev)
{
int32_t ret = 0;
diff --git a/hal/audio_extn/compress_in.c b/hal/audio_extn/compress_in.c
index 6cf6b81..6b525b0 100644
--- a/hal/audio_extn/compress_in.c
+++ b/hal/audio_extn/compress_in.c
@@ -100,7 +100,7 @@
* only after validating that input against cin_attached_usecase
* except below calls
* 1. cin_applicable_stream(in)
- * 2. cin_configure_input_stream(in)
+ * 2. cin_configure_input_stream(in, in_config)
*/
bool cin_attached_usecase(audio_usecase_t uc_id)
@@ -179,7 +179,7 @@
return sz;
}
-int cin_start_input_stream(struct stream_in *in)
+int cin_open_input_stream(struct stream_in *in)
{
int ret = -EINVAL;
struct audio_device *adev = in->dev;
@@ -208,12 +208,23 @@
ALOGV("%s: in %p, cin_data %p", __func__, in, cin_data);
if (cin_data->compr) {
+ compress_stop(cin_data->compr);
+ }
+}
+
+
+void cin_close_input_stream(struct stream_in *in)
+{
+ cin_private_data_t *cin_data = (cin_private_data_t *) in->cin_extn;
+
+ ALOGV("%s: in %p, cin_data %p", __func__, in, cin_data);
+ if (cin_data->compr) {
compress_close(cin_data->compr);
cin_data->compr = NULL;
}
}
-void cin_close_input_stream(struct stream_in *in)
+void cin_free_input_stream_resources(struct stream_in *in)
{
cin_private_data_t *cin_data = (cin_private_data_t *) in->cin_extn;
@@ -265,9 +276,8 @@
return ret;
}
-int cin_configure_input_stream(struct stream_in *in)
+int cin_configure_input_stream(struct stream_in *in, struct audio_config *in_config)
{
- struct audio_device *adev = in->dev;
struct audio_config config = {.format = 0};
int ret = 0, buffer_size = 0, meta_size = sizeof(struct snd_codec_metadata);
cin_private_data_t *cin_data = NULL;
@@ -304,7 +314,8 @@
config.channel_mask = in->channel_mask;
config.format = in->format;
in->config.channels = audio_channel_count_from_in_mask(in->channel_mask);
- buffer_size = adev->device.get_input_buffer_size(&adev->device, &config);
+ buffer_size = audio_extn_utils_get_input_buffer_size(config.sample_rate, config.format,
+ in->config.channels, in_config->offload_info.duration_us / 1000, false);
cin_data->compr_config.fragment_size = buffer_size;
cin_data->compr_config.codec->id = get_snd_codec_id(in->format);
@@ -321,6 +332,11 @@
else
cin_data->compr_config.codec->compr_passthr = PASSTHROUGH_GEN;
+ if (in->flags & AUDIO_INPUT_FLAG_FAST) {
+ ALOGD("%s: Setting latency mode to true", __func__);
+ cin_data->compr_config.codec->flags |= audio_extn_utils_get_perf_mode_flag();
+ }
+
if ((in->flags & AUDIO_INPUT_FLAG_TIMESTAMP) ||
(in->flags & AUDIO_INPUT_FLAG_PASSTHROUGH)) {
compress_config_set_timstamp_flag(&cin_data->compr_config);
@@ -332,6 +348,6 @@
return ret;
err_config:
- cin_close_input_stream(in);
+ cin_free_input_stream_resources(in);
return ret;
}
diff --git a/hal/audio_extn/ext_hw_plugin.c b/hal/audio_extn/ext_hw_plugin.c
index 45ba1d7..6b4a718 100644
--- a/hal/audio_extn/ext_hw_plugin.c
+++ b/hal/audio_extn/ext_hw_plugin.c
@@ -265,6 +265,8 @@
case USECASE_AUDIO_PLAYBACK_OFFLOAD8:
case USECASE_AUDIO_PLAYBACK_OFFLOAD9:
case USECASE_AUDIO_PLAYBACK_ULL:
+ case USECASE_AUDIO_PLAYBACK_MEDIA:
+ case USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION:
*plugin_usecase = AUDIO_HAL_PLUGIN_USECASE_DEFAULT_PLAYBACK;
break;
case USECASE_AUDIO_RECORD:
@@ -281,6 +283,12 @@
case USECASE_VOICEMMODE1_CALL:
*plugin_usecase = AUDIO_HAL_PLUGIN_USECASE_CS_VOICE_CALL;
break;
+ case USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE:
+ *plugin_usecase = AUDIO_HAL_PLUGIN_USECASE_DRIVER_SIDE_PLAYBACK;
+ break;
+ case USECASE_AUDIO_PLAYBACK_PHONE:
+ *plugin_usecase = AUDIO_HAL_PLUGIN_USECASE_PHONE_PLAYBACK;
+ break;
default:
ret = -EINVAL;
}
diff --git a/hal/audio_extn/hw_loopback.c b/hal/audio_extn/hw_loopback.c
index 76c8873..afc029b 100644
--- a/hal/audio_extn/hw_loopback.c
+++ b/hal/audio_extn/hw_loopback.c
@@ -329,8 +329,6 @@
list_remove(&uc_info_rx->list);
free(uc_info_rx);
- adev->active_input = get_next_active_input(adev);
-
if (inout->ip_hdlr_handle) {
ret = audio_extn_ip_hdlr_intf_close(inout->ip_hdlr_handle, true, inout);
if (ret < 0)
@@ -522,7 +520,6 @@
memcpy(&loopback_source_stream.usecase, uc_info_rx,
sizeof(struct audio_usecase));
- adev->active_input = &loopback_source_stream;
select_devices(adev, uc_info_rx->id);
select_devices(adev, uc_info_tx->id);
diff --git a/hal/audio_extn/maxxaudio.c b/hal/audio_extn/maxxaudio.c
index 31feb02..a8f09fc 100644
--- a/hal/audio_extn/maxxaudio.c
+++ b/hal/audio_extn/maxxaudio.c
@@ -352,7 +352,7 @@
static bool find_sup_dev(char *name)
{
- char *token;
+ char *token, *saveptr = NULL;
const char s[2] = ",";
bool ret = false;
char sup_devs[128];
@@ -362,7 +362,7 @@
// 2. Both string content are equal
strncpy(sup_devs, SUPPORT_DEV, sizeof(sup_devs));
- token = strtok(sup_devs, s);
+ token = strtok_r(sup_devs, s, &saveptr);
while (token != NULL) {
if (strncmp(token, name, strlen(token)) == 0 &&
strlen(token) == strlen(name)) {
@@ -370,7 +370,7 @@
ret = true;
break;
}
- token = strtok(NULL, s);
+ token = strtok_r(NULL, s, &saveptr);
}
return ret;
@@ -399,6 +399,7 @@
int ret = 0;
int32_t fd = -1;
char *idd;
+ char *saveptr = NULL;
if (enable) {
ret = snprintf(path, sizeof(path), "/proc/asound/card%u/usbid", card);
@@ -418,7 +419,7 @@
goto done;
}
//replace '\n' to '\0'
- idd = strtok(id, "\n");
+ idd = strtok_r(id, "\n", &saveptr);
if (find_sup_dev(idd)) {
ALOGV("%s: support usbid is %s", __func__, id);
diff --git a/hal/audio_extn/qap.c b/hal/audio_extn/qap.c
new file mode 100644
index 0000000..0625737
--- /dev/null
+++ b/hal/audio_extn/qap.c
@@ -0,0 +1,3137 @@
+/*
+ * Copyright (c) 2016-2019, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "audio_hw_qap"
+#define LOG_NDEBUG 0
+#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define DEBUG_MSG_VV DEBUG_MSG
+#else
+#define DEBUG_MSG_VV(a...) do { } while(0)
+#endif
+
+#define DEBUG_MSG(arg,...) ALOGE("%s: %d: " arg, __func__, __LINE__, ##__VA_ARGS__)
+#define ERROR_MSG(arg,...) ALOGE("%s: %d: " arg, __func__, __LINE__, ##__VA_ARGS__)
+
+#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 2
+#define COMPRESS_PASSTHROUGH_DDP_FRAGMENT_SIZE 4608
+
+#define QAP_DEFAULT_COMPR_AUDIO_HANDLE 1001
+#define QAP_DEFAULT_COMPR_PASSTHROUGH_HANDLE 1002
+#define QAP_DEFAULT_PASSTHROUGH_HANDLE 1003
+
+#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 300
+
+#define MIN_PCM_OFFLOAD_FRAGMENT_SIZE 512
+#define MAX_PCM_OFFLOAD_FRAGMENT_SIZE (240 * 1024)
+
+#define DIV_ROUND_UP(x, y) (((x) + (y) - 1)/(y))
+#define ALIGN(x, y) ((y) * DIV_ROUND_UP((x), (y)))
+
+/* Pcm input node buffer size is 6144 bytes, i.e, 32msec for 48000 samplerate */
+#define QAP_MODULE_PCM_INPUT_BUFFER_LATENCY 32
+
+#define MS12_PCM_OUT_FRAGMENT_SIZE 1536 //samples
+#define MS12_PCM_IN_FRAGMENT_SIZE 1536 //samples
+
+#define DD_FRAME_SIZE 1536
+#define DDP_FRAME_SIZE DD_FRAME_SIZE
+/*
+ * DD encoder output size for one frame.
+ */
+#define DD_ENCODER_OUTPUT_SIZE 2560
+/*
+ * DDP encoder output size for one frame.
+ */
+#define DDP_ENCODER_OUTPUT_SIZE 4608
+
+/*********TODO Need to get correct values.*************************/
+
+#define DTS_PCM_OUT_FRAGMENT_SIZE 1024 //samples
+
+#define DTS_FRAME_SIZE 1536
+#define DTSHD_FRAME_SIZE DTS_FRAME_SIZE
+/*
+ * DTS encoder output size for one frame.
+ */
+#define DTS_ENCODER_OUTPUT_SIZE 2560
+/*
+ * DTSHD encoder output size for one frame.
+ */
+#define DTSHD_ENCODER_OUTPUT_SIZE 4608
+/******************************************************************/
+
+/*
+ * QAP Latency to process buffers since out_write from primary HAL
+ */
+#define QAP_COMPRESS_OFFLOAD_PROCESSING_LATENCY 18
+#define QAP_PCM_OFFLOAD_PROCESSING_LATENCY 48
+
+//TODO: Need to handle for DTS
+#define QAP_DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1536
+
+#include <stdlib.h>
+#include <pthread.h>
+#include <errno.h>
+#include <dlfcn.h>
+#include <unistd.h>
+#include <sys/resource.h>
+#include <sys/prctl.h>
+#include <cutils/properties.h>
+#include <cutils/str_parms.h>
+#include <cutils/log.h>
+#include <cutils/atomic.h>
+#include "audio_utils/primitives.h"
+#include "audio_hw.h"
+#include "platform_api.h"
+#include <platform.h>
+#include <system/thread_defs.h>
+#include <cutils/sched_policy.h>
+#include "audio_extn.h"
+#include <qti_audio.h>
+#include <qap_api.h>
+#include "sound/compress_params.h"
+#include "ip_hdlr_intf.h"
+#include "dolby_ms12.h"
+
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_QAF
+#include <log_utils.h>
+#endif
+
+//TODO: Need to remove this.
+#define QAP_OUTPUT_SAMPLING_RATE 48000
+
+#ifdef QAP_DUMP_ENABLED
+FILE *fp_output_writer_hdmi = NULL;
+#endif
+
+//Types of MM module, currently supported by QAP.
+typedef enum {
+ MS12,
+ DTS_M8,
+ MAX_MM_MODULE_TYPE,
+ INVALID_MM_MODULE
+} mm_module_type;
+
+typedef enum {
+ QAP_OUT_TRANSCODE_PASSTHROUGH = 0, /* Transcode passthrough via MM module*/
+ QAP_OUT_OFFLOAD_MCH, /* Multi-channel PCM offload*/
+ QAP_OUT_OFFLOAD, /* PCM offload */
+
+ MAX_QAP_MODULE_OUT
+} mm_module_output_type;
+
+typedef enum {
+ QAP_IN_MAIN = 0, /* Single PID Main/Primary or Dual-PID stream */
+ QAP_IN_ASSOC, /* Associated/Secondary stream */
+ QAP_IN_PCM, /* PCM stream. */
+ QAP_IN_MAIN_2, /* Single PID Main2 stream */
+ MAX_QAP_MODULE_IN
+} mm_module_input_type;
+
+typedef enum {
+ STOPPED, /*Stream is in stop state. */
+ STOPPING, /*Stream is stopping, waiting for EOS. */
+ RUN, /*Stream is in run state. */
+ MAX_STATES
+} qap_stream_state;
+
+struct qap_module {
+ audio_session_handle_t session_handle;
+ void *qap_lib;
+ void *qap_handle;
+
+ /*Input stream of MM module */
+ struct stream_out *stream_in[MAX_QAP_MODULE_IN];
+ /*Output Stream from MM module */
+ struct stream_out *stream_out[MAX_QAP_MODULE_OUT];
+
+ /*Media format associated with each output id raised by mm module. */
+ qap_session_outputs_config_t session_outputs_config;
+ /*Flag is set if media format is changed for an mm module output. */
+ bool is_media_fmt_changed[MAX_QAP_MODULE_OUT];
+ /*Index to be updated in session_outputs_config array for a new mm module output. */
+ int new_out_format_index;
+
+ //BT session handle.
+ void *bt_hdl;
+
+ float vol_left;
+ float vol_right;
+ bool is_vol_set;
+ qap_stream_state stream_state[MAX_QAP_MODULE_IN];
+ bool is_session_closing;
+ bool is_session_output_active;
+ pthread_cond_t session_output_cond;
+ pthread_mutex_t session_output_lock;
+
+};
+
+struct qap {
+ struct audio_device *adev;
+
+ pthread_mutex_t lock;
+
+ bool bt_connect;
+ bool hdmi_connect;
+ int hdmi_sink_channels;
+
+ //Flag to indicate if QAP transcode output stream is enabled from any mm module.
+ bool passthrough_enabled;
+ //Flag to indicate if QAP mch pcm output stream is enabled from any mm module.
+ bool mch_pcm_hdmi_enabled;
+
+ //Flag to indicate if msmd is supported.
+ bool qap_msmd_enabled;
+
+ bool qap_output_block_handling;
+ //Handle of QAP input stream, which is routed as QAP passthrough.
+ struct stream_out *passthrough_in;
+ //Handle of QAP passthrough stream.
+ struct stream_out *passthrough_out;
+
+ struct qap_module qap_mod[MAX_MM_MODULE_TYPE];
+};
+
+//Global handle of QAP. Access to this should be protected by mutex lock.
+static struct qap *p_qap = NULL;
+
+/* Gets the pointer to qap module for the qap input stream. */
+static struct qap_module* get_qap_module_for_input_stream_l(struct stream_out *out)
+{
+ struct qap_module *qap_mod = NULL;
+ int i, j;
+ if (!p_qap) return NULL;
+
+ for (i = 0; i < MAX_MM_MODULE_TYPE; i++) {
+ for (j = 0; j < MAX_QAP_MODULE_IN; j++) {
+ if (p_qap->qap_mod[i].stream_in[j] == out) {
+ qap_mod = &(p_qap->qap_mod[i]);
+ break;
+ }
+ }
+ }
+
+ return qap_mod;
+}
+
+/* Finds the mm module input stream index for the QAP input stream. */
+static int get_input_stream_index_l(struct stream_out *out)
+{
+ int index = -1, j;
+ struct qap_module* qap_mod = NULL;
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if (!qap_mod) return index;
+
+ for (j = 0; j < MAX_QAP_MODULE_IN; j++) {
+ if (qap_mod->stream_in[j] == out) {
+ index = j;
+ break;
+ }
+ }
+
+ return index;
+}
+
+static void set_stream_state_l(struct stream_out *out, int state)
+{
+ struct qap_module *qap_mod = get_qap_module_for_input_stream_l(out);
+ int index = get_input_stream_index_l(out);
+ if (qap_mod && index >= 0) qap_mod->stream_state[index] = state;
+}
+
+static bool check_stream_state_l(struct stream_out *out, int state)
+{
+ struct qap_module *qap_mod = get_qap_module_for_input_stream_l(out);
+ int index = get_input_stream_index_l(out);
+ if (qap_mod && index >= 0) return ((int)qap_mod->stream_state[index] == state);
+ return false;
+}
+
+/* Finds the right mm module for the QAP input stream format. */
+static mm_module_type get_mm_module_for_format_l(audio_format_t format)
+{
+ int j;
+
+ DEBUG_MSG("Format 0x%x", format);
+
+ if (format == AUDIO_FORMAT_PCM_16_BIT) {
+ //If dts is not supported then alway support pcm with MS12
+ if (!property_get_bool("vendor.audio.qap.dts_m8", false)) { //TODO: Need to add this property for DTS.
+ return MS12;
+ }
+
+ //If QAP passthrough is active then send the PCM stream to primary HAL.
+ if (!p_qap->passthrough_out) {
+ /* Iff any stream is active in MS12 module then route PCM stream to it. */
+ for (j = 0; j < MAX_QAP_MODULE_IN; j++) {
+ if (p_qap->qap_mod[MS12].stream_in[j]) {
+ return MS12;
+ }
+ }
+ }
+ return INVALID_MM_MODULE;
+ }
+
+ switch (format & AUDIO_FORMAT_MAIN_MASK) {
+ case AUDIO_FORMAT_AC3:
+ case AUDIO_FORMAT_E_AC3:
+ case AUDIO_FORMAT_AAC:
+ case AUDIO_FORMAT_AAC_ADTS:
+ case AUDIO_FORMAT_AC4:
+ return MS12;
+ case AUDIO_FORMAT_DTS:
+ case AUDIO_FORMAT_DTS_HD:
+ return DTS_M8;
+ default:
+ return INVALID_MM_MODULE;
+ }
+}
+
+static bool is_main_active_l(struct qap_module* qap_mod)
+{
+ return (qap_mod->stream_in[QAP_IN_MAIN] || qap_mod->stream_in[QAP_IN_MAIN_2]);
+}
+
+static bool is_dual_main_active_l(struct qap_module* qap_mod)
+{
+ return (qap_mod->stream_in[QAP_IN_MAIN] && qap_mod->stream_in[QAP_IN_MAIN_2]);
+}
+
+//Checks if any main or pcm stream is running in the session.
+static bool is_any_stream_running_l(struct qap_module* qap_mod)
+{
+ //Not checking associated stream.
+ struct stream_out *out = qap_mod->stream_in[QAP_IN_MAIN];
+ struct stream_out *out_pcm = qap_mod->stream_in[QAP_IN_PCM];
+ struct stream_out *out_main2 = qap_mod->stream_in[QAP_IN_MAIN_2];
+
+ if ((out == NULL || (out != NULL && check_stream_state_l(out, STOPPED)))
+ && (out_main2 == NULL || (out_main2 != NULL && check_stream_state_l(out_main2, STOPPED)))
+ && (out_pcm == NULL || (out_pcm != NULL && check_stream_state_l(out_pcm, STOPPED)))) {
+ return false;
+ }
+ return true;
+}
+
+/* Gets the pcm output buffer size(in samples) for the mm module. */
+static uint32_t get_pcm_output_buffer_size_samples_l(struct qap_module *qap_mod)
+{
+ uint32_t pcm_output_buffer_size = 0;
+
+ if (qap_mod == &p_qap->qap_mod[MS12]) {
+ pcm_output_buffer_size = MS12_PCM_OUT_FRAGMENT_SIZE;
+ } else if (qap_mod == &p_qap->qap_mod[DTS_M8]) {
+ pcm_output_buffer_size = DTS_PCM_OUT_FRAGMENT_SIZE;
+ }
+
+ return pcm_output_buffer_size;
+}
+
+static int get_media_fmt_array_index_for_output_id_l(
+ struct qap_module* qap_mod,
+ uint32_t output_id)
+{
+ int i;
+ for (i = 0; i < MAX_SUPPORTED_OUTPUTS; i++) {
+ if (qap_mod->session_outputs_config.output_config[i].id == output_id) {
+ return i;
+ }
+ }
+ return -1;
+}
+
+/* Acquire Mutex lock on output stream */
+static void lock_output_stream_l(struct stream_out *out)
+{
+ pthread_mutex_lock(&out->pre_lock);
+ pthread_mutex_lock(&out->lock);
+ pthread_mutex_unlock(&out->pre_lock);
+}
+
+/* Release Mutex lock on output stream */
+static void unlock_output_stream_l(struct stream_out *out)
+{
+ pthread_mutex_unlock(&out->lock);
+}
+
+/* Checks if stream can be routed as QAP passthrough or not. */
+static bool audio_extn_qap_passthrough_enabled(struct stream_out *out)
+{
+ DEBUG_MSG("Format 0x%x", out->format);
+ bool is_enabled = false;
+
+ if (!p_qap) return false;
+
+ if ((!property_get_bool("vendor.audio.qap.reencode", false))
+ && property_get_bool("vendor.audio.qap.passthrough", false)) {
+
+ if ((out->format == AUDIO_FORMAT_PCM_16_BIT) && (popcount(out->channel_mask) > 2)) {
+ is_enabled = true;
+ } else if (property_get_bool("vendor.audio.offload.passthrough", false)) {
+ switch (out->format) {
+ case AUDIO_FORMAT_AC3:
+ case AUDIO_FORMAT_E_AC3:
+ case AUDIO_FORMAT_DTS:
+ case AUDIO_FORMAT_DTS_HD:
+ case AUDIO_FORMAT_DOLBY_TRUEHD:
+ case AUDIO_FORMAT_IEC61937: {
+ is_enabled = true;
+ break;
+ }
+ default:
+ is_enabled = false;
+ break;
+ }
+ }
+ }
+
+ return is_enabled;
+}
+
+/*Closes all pcm hdmi output from QAP. */
+static void close_all_pcm_hdmi_output_l()
+{
+ int i;
+ //Closing all the PCM HDMI output stream from QAP.
+ for (i = 0; i < MAX_MM_MODULE_TYPE; i++) {
+ if (p_qap->qap_mod[i].stream_out[QAP_OUT_OFFLOAD_MCH]) {
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(p_qap->qap_mod[i].stream_out[QAP_OUT_OFFLOAD_MCH]));
+ p_qap->qap_mod[i].stream_out[QAP_OUT_OFFLOAD_MCH] = NULL;
+ }
+
+ if ((p_qap->qap_mod[i].stream_out[QAP_OUT_OFFLOAD])
+ && (p_qap->qap_mod[i].stream_out[QAP_OUT_OFFLOAD]->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(p_qap->qap_mod[i].stream_out[QAP_OUT_OFFLOAD]));
+ p_qap->qap_mod[i].stream_out[QAP_OUT_OFFLOAD] = NULL;
+ }
+ }
+
+ p_qap->mch_pcm_hdmi_enabled = 0;
+}
+
+static void close_all_hdmi_output_l()
+{
+ int k;
+ for (k = 0; k < MAX_MM_MODULE_TYPE; k++) {
+ if (p_qap->qap_mod[k].stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]) {
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(p_qap->qap_mod[k].stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]));
+ p_qap->qap_mod[k].stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH] = NULL;
+ }
+ }
+ p_qap->passthrough_enabled = 0;
+
+ close_all_pcm_hdmi_output_l();
+}
+
+static int qap_out_callback(stream_callback_event_t event, void *param __unused, void *cookie)
+{
+ struct stream_out *out = (struct stream_out *)cookie;
+
+ out->client_callback(event, NULL, out->client_cookie);
+ return 0;
+}
+
+/* Creates the QAP passthrough output stream. */
+static int create_qap_passthrough_stream_l()
+{
+ DEBUG_MSG("Entry");
+
+ int ret = 0;
+ struct stream_out *out = p_qap->passthrough_in;
+
+ if (!out) return -EINVAL;
+
+ pthread_mutex_lock(&p_qap->lock);
+ lock_output_stream_l(out);
+
+ //Creating QAP passthrough output stream.
+ if (NULL == p_qap->passthrough_out) {
+ audio_output_flags_t flags;
+ struct audio_config config;
+ audio_devices_t devices;
+
+ config.sample_rate = config.offload_info.sample_rate = out->sample_rate;
+ config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+ config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+ config.offload_info.format = out->format;
+ config.offload_info.bit_width = out->bit_width;
+ config.format = out->format;
+ config.offload_info.channel_mask = config.channel_mask = out->channel_mask;
+
+ //Device is copied from the QAP passthrough input stream.
+ devices = out->devices;
+ flags = out->flags;
+
+ ret = adev_open_output_stream((struct audio_hw_device *)p_qap->adev,
+ QAP_DEFAULT_PASSTHROUGH_HANDLE,
+ devices,
+ flags,
+ &config,
+ (struct audio_stream_out **)&(p_qap->passthrough_out),
+ NULL);
+ if (ret < 0) {
+ ERROR_MSG("adev_open_output_stream failed with ret = %d!", ret);
+ unlock_output_stream_l(out);
+ return ret;
+ }
+ p_qap->passthrough_in = out;
+ p_qap->passthrough_out->stream.set_callback((struct audio_stream_out *)p_qap->passthrough_out,
+ (stream_callback_t) qap_out_callback, out);
+ }
+
+ unlock_output_stream_l(out);
+
+ //Since QAP-Passthrough is created, close other HDMI outputs.
+ close_all_hdmi_output_l();
+
+ pthread_mutex_unlock(&p_qap->lock);
+ return ret;
+}
+
+
+/* Stops a QAP module stream.*/
+static int audio_extn_qap_stream_stop(struct stream_out *out)
+{
+ int ret = 0;
+ DEBUG_MSG("Output Stream 0x%x", (int)out);
+
+ if (!check_stream_state_l(out, RUN)) return ret;
+
+ struct qap_module *qap_mod = get_qap_module_for_input_stream_l(out);
+
+ if (!qap_mod || !qap_mod->session_handle|| !out->qap_stream_handle) {
+ ERROR_MSG("Wrong state to process qap_mod(%p) sess_hadl(%p) strm hndl(%p)",
+ qap_mod, qap_mod->session_handle, out->qap_stream_handle);
+ return -EINVAL;
+ }
+
+ ret = qap_module_cmd(out->qap_stream_handle,
+ QAP_MODULE_CMD_STOP,
+ sizeof(QAP_MODULE_CMD_STOP),
+ NULL,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("stop failed %d", ret);
+ return -EINVAL;
+ }
+
+ return ret;
+}
+
+static int qap_out_drain(struct audio_stream_out* stream, audio_drain_type_t type)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = 0;
+ struct qap_module *qap_mod = NULL;
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ DEBUG_MSG("Output Stream %p", out);
+
+ lock_output_stream_l(out);
+
+ //If QAP passthrough is enabled then block the drain on module stream.
+ if (p_qap->passthrough_out) {
+ pthread_mutex_lock(&p_qap->lock);
+ //If drain is received for QAP passthorugh stream then call the primary HAL api.
+ if (p_qap->passthrough_in == out) {
+ status = p_qap->passthrough_out->stream.drain(
+ (struct audio_stream_out *)p_qap->passthrough_out, type);
+ }
+ pthread_mutex_unlock(&p_qap->lock);
+ } else if (!is_any_stream_running_l(qap_mod)) {
+ //If stream is already stopped then send the drain ready.
+ out->client_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out->client_cookie);
+ set_stream_state_l(out, STOPPED);
+ } else {
+ qap_audio_buffer_t *buffer;
+ buffer = (qap_audio_buffer_t *) calloc(1, sizeof(qap_audio_buffer_t));
+ buffer->common_params.offset = 0;
+ buffer->common_params.data = buffer;
+ buffer->common_params.size = 0;
+ buffer->buffer_parms.input_buf_params.flags = QAP_BUFFER_EOS;
+ DEBUG_MSG("Queing EOS buffer %p flags %d size %d", buffer, buffer->buffer_parms.input_buf_params.flags, buffer->common_params.size);
+ status = qap_module_process(out->qap_stream_handle, buffer);
+ if (QAP_STATUS_OK != status) {
+ ERROR_MSG("EOS buffer queing failed%d", status);
+ return -EINVAL;
+ }
+
+ //Drain the module input stream.
+ /* Stream stop will trigger EOS and on EOS_EVENT received
+ from callback DRAIN_READY command is sent */
+ status = audio_extn_qap_stream_stop(out);
+
+ if (status == 0) {
+ //Setting state to stopping as client is expecting drain_ready event.
+ set_stream_state_l(out, STOPPING);
+ }
+ }
+
+ unlock_output_stream_l(out);
+ return status;
+}
+
+
+/* Flush the QAP module input stream. */
+static int audio_extn_qap_stream_flush(struct stream_out *out)
+{
+ DEBUG_MSG("Output Stream %p", out);
+ int ret = -EINVAL;
+ struct qap_module *qap_mod = NULL;
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if (!qap_mod || !qap_mod->session_handle|| !out->qap_stream_handle) {
+ ERROR_MSG("Wrong state to process qap_mod(%p) sess_hadl(%p) strm hndl(%p)",
+ qap_mod, qap_mod->session_handle, out->qap_stream_handle);
+ return -EINVAL;
+ }
+
+ ret = qap_module_cmd(out->qap_stream_handle,
+ QAP_MODULE_CMD_FLUSH,
+ sizeof(QAP_MODULE_CMD_FLUSH),
+ NULL,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("flush failed %d", ret);
+ return -EINVAL;
+ }
+
+ return ret;
+}
+
+
+/* Pause the QAP module input stream. */
+static int qap_stream_pause_l(struct stream_out *out)
+{
+ struct qap_module *qap_mod = NULL;
+ int ret = -EINVAL;
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if (!qap_mod || !qap_mod->session_handle|| !out->qap_stream_handle) {
+ ERROR_MSG("Wrong state to process qap_mod(%p) sess_hadl(%p) strm hndl(%p)",
+ qap_mod, qap_mod->session_handle, out->qap_stream_handle);
+ return -EINVAL;
+ }
+
+ ret = qap_module_cmd(out->qap_stream_handle,
+ QAP_MODULE_CMD_PAUSE,
+ sizeof(QAP_MODULE_CMD_PAUSE),
+ NULL,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("pause failed %d", ret);
+ return -EINVAL;
+ }
+
+ return ret;
+}
+
+
+/* Flush the QAP input stream. */
+static int qap_out_flush(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = 0;
+
+ DEBUG_MSG("Output Stream %p", out);
+ lock_output_stream_l(out);
+
+ if (!out->standby) {
+ //If QAP passthrough is active then block the flush on module input streams.
+ if (p_qap->passthrough_out) {
+ pthread_mutex_lock(&p_qap->lock);
+ //If flush is received for the QAP passthrough stream then call the primary HAL api.
+ if (p_qap->passthrough_in == out) {
+ status = p_qap->passthrough_out->stream.flush(
+ (struct audio_stream_out *)p_qap->passthrough_out);
+ out->offload_state = OFFLOAD_STATE_IDLE;
+ }
+ pthread_mutex_unlock(&p_qap->lock);
+ } else {
+ //Flush the module input stream.
+ status = audio_extn_qap_stream_flush(out);
+ }
+ }
+ unlock_output_stream_l(out);
+ DEBUG_MSG("Exit");
+ return status;
+}
+
+
+/* Pause a QAP input stream. */
+static int qap_out_pause(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = 0;
+ DEBUG_MSG("Output Stream %p", out);
+
+ lock_output_stream_l(out);
+
+ //If QAP passthrough is enabled then block the pause on module stream.
+ if (p_qap->passthrough_out) {
+ pthread_mutex_lock(&p_qap->lock);
+ //If pause is received for QAP passthorugh stream then call the primary HAL api.
+ if (p_qap->passthrough_in == out) {
+ status = p_qap->passthrough_out->stream.pause(
+ (struct audio_stream_out *)p_qap->passthrough_out);
+ out->offload_state = OFFLOAD_STATE_PAUSED;
+ }
+ pthread_mutex_unlock(&p_qap->lock);
+ } else {
+ //Pause the module input stream.
+ status = qap_stream_pause_l(out);
+ }
+
+ unlock_output_stream_l(out);
+ return status;
+}
+
+static void close_qap_passthrough_stream_l()
+{
+ if (p_qap->passthrough_out != NULL) { //QAP pasthroug is enabled. Close it.
+ pthread_mutex_lock(&p_qap->lock);
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(p_qap->passthrough_out));
+ p_qap->passthrough_out = NULL;
+ pthread_mutex_unlock(&p_qap->lock);
+
+ if (p_qap->passthrough_in->qap_stream_handle) {
+ qap_out_pause((struct audio_stream_out*)p_qap->passthrough_in);
+ qap_out_flush((struct audio_stream_out*)p_qap->passthrough_in);
+ qap_out_drain((struct audio_stream_out*)p_qap->passthrough_in,
+ (audio_drain_type_t)STREAM_CBK_EVENT_DRAIN_READY);
+ }
+ }
+}
+
+static int qap_out_standby(struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct qap_module *qap_mod = NULL;
+ int status = 0;
+ int i;
+
+ ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
+ stream, out->usecase, use_case_table[out->usecase]);
+
+ lock_output_stream_l(out);
+
+ //If QAP passthrough is active then block standby on all the input streams of QAP mm modules.
+ if (p_qap->passthrough_out) {
+ //If standby is received on QAP passthrough stream then forward it to primary HAL.
+ if (p_qap->passthrough_in == out) {
+ status = p_qap->passthrough_out->stream.common.standby(
+ (struct audio_stream *)p_qap->passthrough_out);
+ }
+ } else if (check_stream_state_l(out, RUN)) {
+ //If QAP passthrough stream is not active then stop the QAP module stream.
+ status = audio_extn_qap_stream_stop(out);
+
+ if (status == 0) {
+ //Setting state to stopped as client not expecting drain_ready event.
+ set_stream_state_l(out, STOPPED);
+ }
+ if(p_qap->qap_output_block_handling) {
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ for (i = 0; i < MAX_QAP_MODULE_IN; i++) {
+ if (qap_mod->stream_in[i] != NULL &&
+ check_stream_state_l(qap_mod->stream_in[i], RUN)) {
+ break;
+ }
+ }
+
+ if (i != MAX_QAP_MODULE_IN) {
+ DEBUG_MSG("[%s] stream is still active.", use_case_table[qap_mod->stream_in[i]->usecase]);
+ } else {
+ pthread_mutex_lock(&qap_mod->session_output_lock);
+ qap_mod->is_session_output_active = false;
+ pthread_mutex_unlock(&qap_mod->session_output_lock);
+ DEBUG_MSG(" all the input streams are either closed or stopped(standby) block the MM module output");
+ }
+ }
+ }
+
+ if (!out->standby) {
+ out->standby = true;
+ }
+
+ unlock_output_stream_l(out);
+ return status;
+}
+
+/* Sets the volume to PCM output stream. */
+static int qap_out_set_volume(struct audio_stream_out *stream, float left, float right)
+{
+ int ret = 0;
+ struct stream_out *out = (struct stream_out *)stream;
+ struct qap_module *qap_mod = NULL;
+
+ DEBUG_MSG("Left %f, Right %f", left, right);
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if (!qap_mod) {
+ return -EINVAL;
+ }
+
+ pthread_mutex_lock(&p_qap->lock);
+ qap_mod->vol_left = left;
+ qap_mod->vol_right = right;
+ qap_mod->is_vol_set = true;
+ pthread_mutex_unlock(&p_qap->lock);
+
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD] != NULL) {
+ ret = qap_mod->stream_out[QAP_OUT_OFFLOAD]->stream.set_volume(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD], left, right);
+ }
+
+ return ret;
+}
+
+/* Starts a QAP module stream. */
+static int qap_stream_start_l(struct stream_out *out)
+{
+ int ret = 0;
+ struct qap_module *qap_mod = NULL;
+
+ DEBUG_MSG("Output Stream = %p", out);
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if ((!qap_mod) || (!qap_mod->session_handle)) {
+ ERROR_MSG("QAP mod is not inited (%p) or session is not yet opened (%p) ",
+ qap_mod, qap_mod->session_handle);
+ return -EINVAL;
+ }
+ if (out->qap_stream_handle) {
+ ret = qap_module_cmd(out->qap_stream_handle,
+ QAP_MODULE_CMD_START,
+ sizeof(QAP_MODULE_CMD_START),
+ NULL,
+ NULL,
+ NULL);
+ if (ret != QAP_STATUS_OK) {
+ ERROR_MSG("start failed");
+ ret = -EINVAL;
+ }
+ } else
+ ERROR_MSG("QAP stream not yet opened, drop this cmd");
+
+ DEBUG_MSG("exit");
+ return ret;
+
+}
+
+static int qap_start_output_stream(struct stream_out *out)
+{
+ int ret = 0;
+ struct audio_device *adev = out->dev;
+
+ if ((out->usecase < 0) || (out->usecase >= AUDIO_USECASE_MAX)) {
+ ret = -EINVAL;
+ DEBUG_MSG("Use case out of bounds sleeping for 500ms");
+ usleep(50000);
+ return ret;
+ }
+
+ ALOGD("%s: enter: stream(%p)usecase(%d: %s) devices(%#x)",
+ __func__, &out->stream, out->usecase, use_case_table[out->usecase],
+ out->devices);
+
+ if (CARD_STATUS_OFFLINE == out->card_status ||
+ CARD_STATUS_OFFLINE == adev->card_status) {
+ ALOGE("%s: sound card is not active/SSR returning error", __func__);
+ ret = -EIO;
+ usleep(50000);
+ return ret;
+ }
+
+ return qap_stream_start_l(out);
+}
+
+/* Sends input buffer to the QAP MM module. */
+static int qap_module_write_input_buffer(struct stream_out *out, const void *buffer, int bytes)
+{
+ int ret = -EINVAL;
+ struct qap_module *qap_mod = NULL;
+ qap_audio_buffer_t buff;
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if ((!qap_mod) || (!qap_mod->session_handle) || (!out->qap_stream_handle)) {
+ return ret;
+ }
+
+ //If data received on associated stream when all other stream are stopped then drop the data.
+ if (out == qap_mod->stream_in[QAP_IN_ASSOC] && !is_any_stream_running_l(qap_mod))
+ return bytes;
+
+ memset(&buff, 0, sizeof(buff));
+ buff.common_params.offset = 0;
+ buff.common_params.size = bytes;
+ buff.common_params.data = (void *) buffer;
+ buff.common_params.timestamp = QAP_BUFFER_NO_TSTAMP;
+ buff.buffer_parms.input_buf_params.flags = QAP_BUFFER_NO_TSTAMP;
+ DEBUG_MSG("calling module process with bytes %d %p", bytes, buffer);
+ ret = qap_module_process(out->qap_stream_handle, &buff);
+
+ if(ret > 0) set_stream_state_l(out, RUN);
+
+ return ret;
+}
+
+static ssize_t qap_out_write(struct audio_stream_out *stream, const void *buffer, size_t bytes)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct audio_device *adev = out->dev;
+ ssize_t ret = 0;
+ struct qap_module *qap_mod = NULL;
+
+ DEBUG_MSG_VV("bytes = %d, usecase[%s] and flags[%x] for handle[%p]",
+ (int)bytes, use_case_table[out->usecase], out->flags, out);
+
+ lock_output_stream_l(out);
+
+ // If QAP passthrough is active then block writing data to QAP mm module.
+ if (p_qap->passthrough_out) {
+ //If write is received for the QAP passthrough stream then send the buffer to primary HAL.
+ if (p_qap->passthrough_in == out) {
+ ret = p_qap->passthrough_out->stream.write(
+ (struct audio_stream_out *)(p_qap->passthrough_out),
+ buffer,
+ bytes);
+ if (ret > 0) out->standby = false;
+ }
+ unlock_output_stream_l(out);
+ return ret;
+ } else if (out->standby) {
+ pthread_mutex_lock(&adev->lock);
+ ret = qap_start_output_stream(out);
+ pthread_mutex_unlock(&adev->lock);
+ if (ret == 0) {
+ out->standby = false;
+ if(p_qap->qap_output_block_handling) {
+ qap_mod = get_qap_module_for_input_stream_l(out);
+
+ pthread_mutex_lock(&qap_mod->session_output_lock);
+ if (qap_mod->is_session_output_active == false) {
+ qap_mod->is_session_output_active = true;
+ pthread_cond_signal(&qap_mod->session_output_cond);
+ DEBUG_MSG("Wake up MM module output thread");
+ }
+ pthread_mutex_unlock(&qap_mod->session_output_lock);
+ }
+ } else {
+ goto exit;
+ }
+ }
+
+ if ((adev->is_channel_status_set == false) && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ audio_utils_set_hdmi_channel_status(out, (char *)buffer, bytes);
+ adev->is_channel_status_set = true;
+ }
+
+ ret = qap_module_write_input_buffer(out, buffer, bytes);
+ DEBUG_MSG_VV("Bytes consumed [%d] by MM Module", (int)ret);
+
+ if (ret >= 0) {
+ out->written += ret / ((popcount(out->channel_mask) * sizeof(short)));
+ }
+
+
+exit:
+ unlock_output_stream_l(out);
+
+ if (ret < 0) {
+ if (ret == -EAGAIN) {
+ DEBUG_MSG_VV("No space available to consume bytes, post msg to cb thread");
+ bytes = 0;
+ } else if (ret == -ENOMEM || ret == -EPERM) {
+ if (out->pcm)
+ ERROR_MSG("error %d, %s", (int)ret, pcm_get_error(out->pcm));
+ qap_out_standby(&out->stream.common);
+ DEBUG_MSG("SLEEP for 100sec");
+ usleep(bytes * 1000000
+ / audio_stream_out_frame_size(stream)
+ / out->stream.common.get_sample_rate(&out->stream.common));
+ }
+ } else if (ret < (ssize_t)bytes) {
+ //partial buffer copied to the module.
+ DEBUG_MSG_VV("Not enough space available to consume all the bytes");
+ bytes = ret;
+ }
+ return bytes;
+}
+
+/* Gets PCM offload buffer size for a given config. */
+static uint32_t qap_get_pcm_offload_buffer_size(audio_offload_info_t* info,
+ uint32_t samples_per_frame)
+{
+ uint32_t fragment_size = 0;
+
+ fragment_size = (samples_per_frame * (info->bit_width >> 3) * popcount(info->channel_mask));
+
+ 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,
+ ((info->bit_width >> 3) * popcount(info->channel_mask) * 32));
+
+ ALOGI("Qap PCM offload Fragment size is %d bytes", fragment_size);
+
+ return fragment_size;
+}
+
+static uint32_t qap_get_pcm_offload_input_buffer_size(audio_offload_info_t* info)
+{
+ return qap_get_pcm_offload_buffer_size(info, MS12_PCM_IN_FRAGMENT_SIZE);
+}
+
+static uint32_t qap_get_pcm_offload_output_buffer_size(struct qap_module *qap_mod,
+ audio_offload_info_t* info)
+{
+ return qap_get_pcm_offload_buffer_size(info, get_pcm_output_buffer_size_samples_l(qap_mod));
+}
+
+/* Gets buffer latency in samples. */
+static int get_buffer_latency(struct stream_out *out, uint32_t buffer_size, uint32_t *latency)
+{
+ unsigned long int samples_in_one_encoded_frame;
+ unsigned long int size_of_one_encoded_frame;
+
+ switch (out->format) {
+ case AUDIO_FORMAT_AC3:
+ samples_in_one_encoded_frame = DD_FRAME_SIZE;
+ size_of_one_encoded_frame = DD_ENCODER_OUTPUT_SIZE;
+ break;
+ case AUDIO_FORMAT_E_AC3:
+ samples_in_one_encoded_frame = DDP_FRAME_SIZE;
+ size_of_one_encoded_frame = DDP_ENCODER_OUTPUT_SIZE;
+ break;
+ case AUDIO_FORMAT_DTS:
+ samples_in_one_encoded_frame = DTS_FRAME_SIZE;
+ size_of_one_encoded_frame = DTS_ENCODER_OUTPUT_SIZE;
+ break;
+ case AUDIO_FORMAT_DTS_HD:
+ samples_in_one_encoded_frame = DTSHD_FRAME_SIZE;
+ size_of_one_encoded_frame = DTSHD_ENCODER_OUTPUT_SIZE;
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ samples_in_one_encoded_frame = 1;
+ size_of_one_encoded_frame = ((out->bit_width) >> 3) * popcount(out->channel_mask);
+ break;
+ default:
+ *latency = 0;
+ return (-EINVAL);
+ }
+
+ *latency = ((buffer_size * samples_in_one_encoded_frame) / size_of_one_encoded_frame);
+ return 0;
+}
+
+/* Returns the number of frames rendered to outside observer. */
+static int qap_get_rendered_frames(struct stream_out *out, uint64_t *frames)
+{
+ int ret = 0, i;
+ struct str_parms *parms;
+// int value = 0;
+ int module_latency = 0;
+ uint32_t kernel_latency = 0;
+ uint32_t dsp_latency = 0;
+ int signed_frames = 0;
+ char* kvpairs = NULL;
+ struct qap_module *qap_mod = NULL;
+
+ DEBUG_MSG("Output Format %d", out->format);
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if (!qap_mod || !qap_mod->session_handle|| !out->qap_stream_handle) {
+ ERROR_MSG("Wrong state to process qap_mod(%p) sess_hadl(%p) strm hndl(%p)",
+ qap_mod, qap_mod->session_handle, out->qap_stream_handle);
+ return -EINVAL;
+ }
+
+ //Get MM module latency.
+/* Tobeported
+ kvpairs = qap_mod->qap_audio_stream_get_param(out->qap_stream_handle, "get_latency");
+*/
+ if (kvpairs) {
+ parms = str_parms_create_str(kvpairs);
+ ret = str_parms_get_int(parms, "get_latency", &module_latency);
+ if (ret >= 0) {
+ str_parms_destroy(parms);
+ parms = NULL;
+ }
+ free(kvpairs);
+ kvpairs = NULL;
+ }
+
+ //Get kernel Latency
+ for (i = MAX_QAP_MODULE_OUT - 1; i >= 0; i--) {
+ if (qap_mod->stream_out[i] == NULL) {
+ continue;
+ } else {
+ unsigned int num_fragments = qap_mod->stream_out[i]->compr_config.fragments;
+ uint32_t fragment_size = qap_mod->stream_out[i]->compr_config.fragment_size;
+ uint32_t kernel_buffer_size = num_fragments * fragment_size;
+ get_buffer_latency(qap_mod->stream_out[i], kernel_buffer_size, &kernel_latency);
+ break;
+ }
+ }
+
+ //Get DSP latency
+ if ((qap_mod->stream_out[QAP_OUT_OFFLOAD] != NULL)
+ || (qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH] != NULL)) {
+ unsigned int sample_rate = 0;
+ audio_usecase_t platform_latency = 0;
+
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD])
+ sample_rate = qap_mod->stream_out[QAP_OUT_OFFLOAD]->sample_rate;
+ else if (qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH])
+ sample_rate = qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->sample_rate;
+
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD])
+ platform_latency =
+ platform_render_latency(qap_mod->stream_out[QAP_OUT_OFFLOAD]->usecase);
+ else
+ platform_latency =
+ platform_render_latency(qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->usecase);
+
+ dsp_latency = (platform_latency * sample_rate) / 1000000LL;
+ } else if (qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH] != NULL) {
+ unsigned int sample_rate = 0;
+
+ sample_rate = qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]->sample_rate; //TODO: How this sample rate can be used?
+ dsp_latency = (COMPRESS_OFFLOAD_PLAYBACK_LATENCY * sample_rate) / 1000;
+ }
+
+ // MM Module Latency + Kernel Latency + DSP Latency
+ if ( audio_extn_bt_hal_get_output_stream(qap_mod->bt_hdl) != NULL) {
+ out->platform_latency = module_latency + audio_extn_bt_hal_get_latency(qap_mod->bt_hdl);
+ } else {
+ out->platform_latency = (uint32_t)module_latency + kernel_latency + dsp_latency;
+ }
+
+ if (out->format & AUDIO_FORMAT_PCM_16_BIT) {
+ *frames = 0;
+ signed_frames = out->written - out->platform_latency;
+ // It would be unusual for this value to be negative, but check just in case ...
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ }
+/* Tobeported
+ }
+ else {
+
+ kvpairs = qap_mod->qap_audio_stream_get_param(out->qap_stream_handle, "position");
+ if (kvpairs) {
+ parms = str_parms_create_str(kvpairs);
+ ret = str_parms_get_int(parms, "position", &value);
+ if (ret >= 0) {
+ *frames = value;
+ signed_frames = value - out->platform_latency;
+ // It would be unusual for this value to be negative, but check just in case ...
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ }
+ }
+ str_parms_destroy(parms);
+ }
+*/
+ } else {
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static int qap_out_get_render_position(const struct audio_stream_out *stream,
+ uint32_t *dsp_frames)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int ret = 0;
+ uint64_t frames=0;
+ struct qap_module* qap_mod = NULL;
+ ALOGV("%s, Output Stream %p,dsp frames %d",__func__, stream, (int)dsp_frames);
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if (!qap_mod) {
+ ret = out->stream.get_render_position(stream, dsp_frames);
+ ALOGV("%s, non qap_MOD DSP FRAMES %d",__func__, (int)dsp_frames);
+ return ret;
+ }
+
+ if (p_qap->passthrough_out) {
+ pthread_mutex_lock(&p_qap->lock);
+ ret = p_qap->passthrough_out->stream.get_render_position((struct audio_stream_out *)p_qap->passthrough_out, dsp_frames);
+ pthread_mutex_unlock(&p_qap->lock);
+ ALOGV("%s, PASS THROUGH DSP FRAMES %p",__func__, dsp_frames);
+ return ret;
+ }
+ frames=*dsp_frames;
+ ret = qap_get_rendered_frames(out, &frames);
+ *dsp_frames = (uint32_t)frames;
+ ALOGV("%s, DSP FRAMES %d",__func__, (int)dsp_frames);
+ return ret;
+}
+
+static int qap_out_get_presentation_position(const struct audio_stream_out *stream,
+ uint64_t *frames,
+ struct timespec *timestamp)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int ret = 0;
+
+// DEBUG_MSG_VV("Output Stream %p", stream);
+
+ //If QAP passthorugh output stream is active.
+ if (p_qap->passthrough_out) {
+ if (p_qap->passthrough_in == out) {
+ //If api is called for QAP passthorugh stream then call the primary HAL api to get the position.
+ pthread_mutex_lock(&p_qap->lock);
+ ret = p_qap->passthrough_out->stream.get_presentation_position(
+ (struct audio_stream_out *)p_qap->passthrough_out,
+ frames,
+ timestamp);
+ pthread_mutex_unlock(&p_qap->lock);
+ } else {
+ //If api is called for other stream then return zero frames.
+ *frames = 0;
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ }
+ return ret;
+ }
+
+ ret = qap_get_rendered_frames(out, frames);
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+
+ return ret;
+}
+
+static uint32_t qap_out_get_latency(const struct audio_stream_out *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ uint32_t latency = 0;
+ struct qap_module *qap_mod = NULL;
+ DEBUG_MSG_VV("Output Stream %p", out);
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if (!qap_mod) {
+ return 0;
+ }
+
+ //If QAP passthrough is active then block the get latency on module input streams.
+ if (p_qap->passthrough_out) {
+ pthread_mutex_lock(&p_qap->lock);
+ //If get latency is called for the QAP passthrough stream then call the primary HAL api.
+ if (p_qap->passthrough_in == out) {
+ latency = p_qap->passthrough_out->stream.get_latency(
+ (struct audio_stream_out *)p_qap->passthrough_out);
+ }
+ pthread_mutex_unlock(&p_qap->lock);
+ } else {
+ if (is_offload_usecase(out->usecase)) {
+ latency = COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
+ } else {
+ uint32_t sample_rate = 0;
+ latency = QAP_MODULE_PCM_INPUT_BUFFER_LATENCY; //Input latency
+
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD])
+ sample_rate = qap_mod->stream_out[QAP_OUT_OFFLOAD]->sample_rate;
+ else if (qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH])
+ sample_rate = qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->sample_rate;
+
+ if (sample_rate) {
+ latency += (get_pcm_output_buffer_size_samples_l(qap_mod) * 1000) / out->sample_rate;
+ }
+ }
+
+ if ( audio_extn_bt_hal_get_output_stream(qap_mod->bt_hdl) != NULL) {
+ if (is_offload_usecase(out->usecase)) {
+ latency = audio_extn_bt_hal_get_latency(qap_mod->bt_hdl) +
+ QAP_COMPRESS_OFFLOAD_PROCESSING_LATENCY;
+ } else {
+ latency = audio_extn_bt_hal_get_latency(qap_mod->bt_hdl) +
+ QAP_PCM_OFFLOAD_PROCESSING_LATENCY;
+ }
+ }
+ }
+
+ DEBUG_MSG_VV("Latency %d", latency);
+ return latency;
+}
+
+static bool qap_check_and_get_compressed_device_format(int device, int *format)
+{
+ switch (device) {
+ case (AUDIO_DEVICE_OUT_AUX_DIGITAL | QAP_AUDIO_FORMAT_AC3):
+ *format = AUDIO_FORMAT_AC3;
+ return true;
+ case (AUDIO_DEVICE_OUT_AUX_DIGITAL | QAP_AUDIO_FORMAT_EAC3):
+ *format = AUDIO_FORMAT_E_AC3;
+ return true;
+ case (AUDIO_DEVICE_OUT_AUX_DIGITAL | QAP_AUDIO_FORMAT_DTS):
+ *format = AUDIO_FORMAT_DTS;
+ return true;
+ default:
+ return false;
+ }
+}
+
+static void set_out_stream_channel_map(struct stream_out *out, qap_output_config_t * media_fmt)
+{
+ if (media_fmt == NULL || out == NULL) {
+ return;
+ }
+ struct audio_out_channel_map_param chmap = {0,{0}};
+ int i = 0;
+ chmap.channels = media_fmt->channels;
+ for (i = 0; i < chmap.channels && i < AUDIO_CHANNEL_COUNT_MAX && i < AUDIO_QAF_MAX_CHANNELS;
+ i++) {
+ chmap.channel_map[i] = media_fmt->ch_map[i];
+ }
+ audio_extn_utils_set_channel_map(out, &chmap);
+}
+
+bool audio_extn_is_qap_enabled()
+{
+ bool prop_enabled = false;
+ char value[PROPERTY_VALUE_MAX] = {0};
+ property_get("vendor.audio.qap.enabled", value, NULL);
+ prop_enabled = atoi(value) || !strncmp("true", value, 4);
+ DEBUG_MSG("%d", prop_enabled);
+ return (prop_enabled);
+}
+
+void static qap_close_all_output_streams(struct qap_module *qap_mod)
+{
+ int i =0;
+ struct stream_out *stream_out = NULL;
+ DEBUG_MSG("Entry");
+
+ for (i = 0; i < MAX_QAP_MODULE_OUT; i++) {
+ stream_out = qap_mod->stream_out[i];
+ if (stream_out != NULL) {
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev, (struct audio_stream_out *)stream_out);
+ DEBUG_MSG("Closed outputenum=%d session 0x%x %s",
+ i, (int)stream_out, use_case_table[stream_out->usecase]);
+ qap_mod->stream_out[i] = NULL;
+ }
+ memset(&qap_mod->session_outputs_config.output_config[i], 0, sizeof(qap_session_outputs_config_t));
+ qap_mod->is_media_fmt_changed[i] = false;
+ }
+ DEBUG_MSG("exit");
+}
+
+/* Call back function for mm module. */
+static void qap_session_callback(qap_session_handle_t session_handle __unused,
+ void *prv_data,
+ qap_callback_event_t event_id,
+ int size,
+ void *data)
+{
+
+ /*
+ For SPKR:
+ 1. Open pcm device if device_id passed to it SPKR and write the data to
+ pcm device
+
+ For HDMI
+ 1.Open compress device for HDMI(PCM or AC3) based on current hdmi o/p format and write
+ data to the HDMI device.
+ */
+ int ret;
+ audio_output_flags_t flags;
+ struct qap_module* qap_mod = (struct qap_module*)prv_data;
+ struct audio_stream_out *bt_stream = NULL;
+ int format;
+ int8_t *data_buffer_p = NULL;
+ uint32_t buffer_size = 0;
+ bool need_to_recreate_stream = false;
+ struct audio_config config;
+ qap_output_config_t *new_conf = NULL;
+ qap_audio_buffer_t *buffer = (qap_audio_buffer_t *) data;
+ uint32_t device = 0;
+
+ if (qap_mod->is_session_closing) {
+ DEBUG_MSG("Dropping event as session is closing."
+ "Event = 0x%X, Bytes to write %d", event_id, size);
+ return;
+ }
+
+ if(p_qap->qap_output_block_handling) {
+ pthread_mutex_lock(&qap_mod->session_output_lock);
+ if (!qap_mod->is_session_output_active) {
+ qap_close_all_output_streams(qap_mod);
+ DEBUG_MSG("disabling MM module output by blocking the output thread");
+ pthread_cond_wait(&qap_mod->session_output_cond, &qap_mod->session_output_lock);
+ DEBUG_MSG("MM module output Enabled, output thread active");
+ }
+ pthread_mutex_unlock(&qap_mod->session_output_lock);
+ }
+
+ /* Default config initialization. */
+ config.sample_rate = config.offload_info.sample_rate = QAP_OUTPUT_SAMPLING_RATE;
+ config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+ config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+ config.format = config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT;
+ config.offload_info.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+
+ pthread_mutex_lock(&p_qap->lock);
+
+ if (event_id == QAP_CALLBACK_EVENT_OUTPUT_CFG_CHANGE) {
+ new_conf = &buffer->buffer_parms.output_buf_params.output_config;
+ qap_output_config_t *cached_conf = NULL;
+ int index = -1;
+
+ DEBUG_MSG("Received QAP_CALLBACK_EVENT_OUTPUT_CFG_CHANGE event for output id=0x%x",
+ buffer->buffer_parms.output_buf_params.output_id);
+
+ DEBUG_MSG("sample rate=%d bitwidth=%d format = %d channels =0x%x",
+ new_conf->sample_rate,
+ new_conf->bit_width,
+ new_conf->format,
+ new_conf->channels);
+
+ if ( (uint32_t)size < sizeof(qap_output_config_t)) {
+ ERROR_MSG("Size is not proper for the event AUDIO_OUTPUT_MEDIA_FORMAT_EVENT.");
+ return ;
+ }
+
+ index = get_media_fmt_array_index_for_output_id_l(qap_mod, buffer->buffer_parms.output_buf_params.output_id);
+
+ DEBUG_MSG("index = %d", index);
+
+ if (index >= 0) {
+ cached_conf = &qap_mod->session_outputs_config.output_config[index];
+ } else if (index < 0 && qap_mod->new_out_format_index < MAX_QAP_MODULE_OUT) {
+ index = qap_mod->new_out_format_index;
+ cached_conf = &qap_mod->session_outputs_config.output_config[index];
+ qap_mod->new_out_format_index++;
+ }
+
+ if (cached_conf == NULL) {
+ ERROR_MSG("Maximum output from a QAP module is reached. Can not process new output.");
+ return ;
+ }
+
+ if (memcmp(cached_conf, new_conf, sizeof(qap_output_config_t)) != 0) {
+ memcpy(cached_conf, new_conf, sizeof(qap_output_config_t));
+ qap_mod->is_media_fmt_changed[index] = true;
+ }
+ } else if (event_id == QAP_CALLBACK_EVENT_DATA) {
+ data_buffer_p = (int8_t*)buffer->common_params.data+buffer->common_params.offset;
+ buffer_size = buffer->common_params.size;
+ device = buffer->buffer_parms.output_buf_params.output_id;
+
+ DEBUG_MSG_VV("Received QAP_CALLBACK_EVENT_DATA event buff size(%d) for outputid=0x%x",
+ buffer_size, buffer->buffer_parms.output_buf_params.output_id);
+
+ if (buffer && buffer->common_params.data) {
+ int index = -1;
+
+ index = get_media_fmt_array_index_for_output_id_l(qap_mod, buffer->buffer_parms.output_buf_params.output_id);
+ DEBUG_MSG("index = %d", index);
+ if (index > -1 && qap_mod->is_media_fmt_changed[index]) {
+ DEBUG_MSG("FORMAT changed, recreate stream");
+ need_to_recreate_stream = true;
+ qap_mod->is_media_fmt_changed[index] = false;
+
+ qap_output_config_t *new_config = &qap_mod->session_outputs_config.output_config[index];
+
+ config.sample_rate = config.offload_info.sample_rate = new_config->sample_rate;
+ config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+ config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+ config.offload_info.bit_width = new_config->bit_width;
+
+ if (new_config->format == QAP_AUDIO_FORMAT_PCM_16_BIT) {
+ if (new_config->bit_width == 16)
+ config.format = config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT;
+ else if (new_config->bit_width == 24)
+ config.format = config.offload_info.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ else
+ config.format = config.offload_info.format = AUDIO_FORMAT_PCM_32_BIT;
+ } else if (new_config->format == QAP_AUDIO_FORMAT_AC3)
+ config.format = config.offload_info.format = AUDIO_FORMAT_AC3;
+ else if (new_config->format == QAP_AUDIO_FORMAT_EAC3)
+ config.format = config.offload_info.format = AUDIO_FORMAT_E_AC3;
+ else if (new_config->format == QAP_AUDIO_FORMAT_DTS)
+ config.format = config.offload_info.format = AUDIO_FORMAT_DTS;
+
+ device |= (new_config->format & AUDIO_FORMAT_MAIN_MASK);
+
+ config.channel_mask = audio_channel_out_mask_from_count(new_config->channels);
+ config.offload_info.channel_mask = config.channel_mask;
+ DEBUG_MSG("sample rate=%d bitwidth=%d format = %d channels=%d channel_mask=%d device =0x%x",
+ config.sample_rate,
+ config.offload_info.bit_width,
+ config.offload_info.format,
+ new_config->channels,
+ config.channel_mask,
+ device);
+ }
+ }
+
+ if (p_qap->passthrough_out != NULL) {
+ //If QAP passthrough is active then all the module output will be dropped.
+ pthread_mutex_unlock(&p_qap->lock);
+ DEBUG_MSG("QAP-PSTH is active, DROPPING DATA!");
+ return;
+ }
+
+ if (qap_check_and_get_compressed_device_format(device, &format)) {
+ /*
+ * CASE 1: Transcoded output of mm module.
+ * If HDMI is not connected then drop the data.
+ * Only one HDMI output can be supported from all the mm modules of QAP.
+ * Multi-Channel PCM HDMI output streams will be closed from all the mm modules.
+ * If transcoded output of other module is already enabled then this data will be dropped.
+ */
+
+ if (!p_qap->hdmi_connect) {
+ DEBUG_MSG("HDMI not connected, DROPPING DATA!");
+ pthread_mutex_unlock(&p_qap->lock);
+ return;
+ }
+
+ //Closing all the PCM HDMI output stream from QAP.
+ close_all_pcm_hdmi_output_l();
+
+ /* If Media format was changed for this stream then need to re-create the stream. */
+ if (need_to_recreate_stream && qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]) {
+ DEBUG_MSG("closing Transcode Passthrough session ox%x",
+ (int)qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]);
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]));
+ qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH] = NULL;
+ p_qap->passthrough_enabled = false;
+ }
+
+ if (!p_qap->passthrough_enabled
+ && !(qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH])) {
+
+ audio_devices_t devices;
+
+ config.format = config.offload_info.format = format;
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+
+ flags = (AUDIO_OUTPUT_FLAG_NON_BLOCKING
+ | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
+ | AUDIO_OUTPUT_FLAG_DIRECT
+ | AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH);
+ devices = AUDIO_DEVICE_OUT_AUX_DIGITAL;
+
+ DEBUG_MSG("Opening Transcode Passthrough out(outputenum=%d) session 0x%x with below params",
+ QAP_OUT_TRANSCODE_PASSTHROUGH,
+ (int)qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]);
+
+ DEBUG_MSG("sample rate=%d bitwidth=%d format = 0x%x channel mask=0x%x flags=0x%x device =0x%x",
+ config.sample_rate,
+ config.offload_info.bit_width,
+ config.offload_info.format,
+ config.offload_info.channel_mask,
+ flags,
+ devices);
+
+ ret = adev_open_output_stream((struct audio_hw_device *)p_qap->adev,
+ QAP_DEFAULT_COMPR_PASSTHROUGH_HANDLE,
+ devices,
+ flags,
+ &config,
+ (struct audio_stream_out **)&(qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]),
+ NULL);
+ if (ret < 0) {
+ ERROR_MSG("Failed opening Transcode Passthrough out(outputenum=%d) session 0x%x",
+ QAP_OUT_TRANSCODE_PASSTHROUGH,
+ (int)qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]);
+ pthread_mutex_unlock(&p_qap->lock);
+ return;
+ } else
+ DEBUG_MSG("Opened Transcode Passthrough out(outputenum=%d) session 0x%x",
+ QAP_OUT_TRANSCODE_PASSTHROUGH,
+ (int)qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]);
+
+
+ if (format == AUDIO_FORMAT_E_AC3) {
+ qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]->compr_config.fragment_size =
+ COMPRESS_PASSTHROUGH_DDP_FRAGMENT_SIZE;
+ }
+ qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]->compr_config.fragments =
+ COMPRESS_OFFLOAD_NUM_FRAGMENTS;
+
+ p_qap->passthrough_enabled = true;
+ }
+
+ if (qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]) {
+ DEBUG_MSG_VV("Writing Bytes(%d) to QAP_OUT_TRANSCODE_PASSTHROUGH output(%p) buff ptr(%p)",
+ buffer_size, qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH],
+ data_buffer_p);
+ ret = qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH]->stream.write(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_TRANSCODE_PASSTHROUGH],
+ data_buffer_p,
+ buffer_size);
+ }
+ }
+ else if ((device & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+ && (p_qap->hdmi_connect)
+ && (p_qap->hdmi_sink_channels > 2)) {
+
+ /* CASE 2: Multi-Channel PCM output to HDMI.
+ * If any other HDMI output is already enabled then this has to be dropped.
+ */
+
+ if (p_qap->passthrough_enabled) {
+ //Closing all the multi-Channel PCM HDMI output stream from QAP.
+ close_all_pcm_hdmi_output_l();
+
+ //If passthrough is active then pcm hdmi output has to be dropped.
+ pthread_mutex_unlock(&p_qap->lock);
+ DEBUG_MSG("Compressed passthrough enabled, DROPPING DATA!");
+ return;
+ }
+
+ /* If Media format was changed for this stream then need to re-create the stream. */
+ if (need_to_recreate_stream && qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]) {
+ DEBUG_MSG("closing MCH PCM session ox%x", (int)qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]);
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]));
+ qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH] = NULL;
+ p_qap->mch_pcm_hdmi_enabled = false;
+ }
+
+ if (!p_qap->mch_pcm_hdmi_enabled && !(qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH])) {
+ audio_devices_t devices;
+
+ if (event_id == AUDIO_DATA_EVENT) {
+ config.offload_info.format = config.format = AUDIO_FORMAT_PCM_16_BIT;
+
+ if (p_qap->hdmi_sink_channels == 8) {
+ config.offload_info.channel_mask = config.channel_mask =
+ AUDIO_CHANNEL_OUT_7POINT1;
+ } else if (p_qap->hdmi_sink_channels == 6) {
+ config.offload_info.channel_mask = config.channel_mask =
+ AUDIO_CHANNEL_OUT_5POINT1;
+ } else {
+ config.offload_info.channel_mask = config.channel_mask =
+ AUDIO_CHANNEL_OUT_STEREO;
+ }
+ }
+
+ devices = AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ flags = AUDIO_OUTPUT_FLAG_DIRECT;
+
+ DEBUG_MSG("Opening MCH PCM out(outputenum=%d) session ox%x with below params",
+ QAP_OUT_OFFLOAD_MCH,
+ (int)qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]);
+
+ DEBUG_MSG("sample rate=%d bitwidth=%d format = 0x%x channel mask=0x%x flags=0x%x device =0x%x",
+ config.sample_rate,
+ config.offload_info.bit_width,
+ config.offload_info.format,
+ config.offload_info.channel_mask,
+ flags,
+ devices);
+
+ ret = adev_open_output_stream((struct audio_hw_device *)p_qap->adev,
+ QAP_DEFAULT_COMPR_AUDIO_HANDLE,
+ devices,
+ flags,
+ &config,
+ (struct audio_stream_out **)&(qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]),
+ NULL);
+ if (ret < 0) {
+ ERROR_MSG("Failed opening MCH PCM out(outputenum=%d) session ox%x",
+ QAP_OUT_OFFLOAD_MCH,
+ (int)qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]);
+ pthread_mutex_unlock(&p_qap->lock);
+ return;
+ } else
+ DEBUG_MSG("Opened MCH PCM out(outputenum=%d) session ox%x",
+ QAP_OUT_OFFLOAD_MCH,
+ (int)qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]);
+
+ set_out_stream_channel_map(qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH], new_conf);
+
+ qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->compr_config.fragments =
+ COMPRESS_OFFLOAD_NUM_FRAGMENTS;
+ qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->compr_config.fragment_size =
+ qap_get_pcm_offload_output_buffer_size(qap_mod, &config.offload_info);
+
+ p_qap->mch_pcm_hdmi_enabled = true;
+
+ if ((qap_mod->stream_in[QAP_IN_MAIN]
+ && qap_mod->stream_in[QAP_IN_MAIN]->client_callback != NULL) ||
+ (qap_mod->stream_in[QAP_IN_MAIN_2]
+ && qap_mod->stream_in[QAP_IN_MAIN_2]->client_callback != NULL)) {
+
+ if (qap_mod->stream_in[QAP_IN_MAIN]) {
+ qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->stream.set_callback(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH],
+ qap_mod->stream_in[QAP_IN_MAIN]->client_callback,
+ qap_mod->stream_in[QAP_IN_MAIN]->client_cookie);
+ }
+ if (qap_mod->stream_in[QAP_IN_MAIN_2]) {
+ qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->stream.set_callback(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH],
+ qap_mod->stream_in[QAP_IN_MAIN_2]->client_callback,
+ qap_mod->stream_in[QAP_IN_MAIN_2]->client_cookie);
+ }
+ } else if (qap_mod->stream_in[QAP_IN_PCM]
+ && qap_mod->stream_in[QAP_IN_PCM]->client_callback != NULL) {
+
+ qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->stream.set_callback(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH],
+ qap_mod->stream_in[QAP_IN_PCM]->client_callback,
+ qap_mod->stream_in[QAP_IN_PCM]->client_cookie);
+ }
+ }
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]) {
+ DEBUG_MSG_VV("Writing Bytes(%d) to QAP_OUT_OFFLOAD_MCH output(%p) buff ptr(%p)",
+ buffer_size, qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH],
+ data_buffer_p);
+ ret = qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH]->stream.write(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD_MCH],
+ data_buffer_p,
+ buffer_size);
+ }
+ }
+ else {
+ /* CASE 3: PCM output.
+ */
+
+ /* If Media format was changed for this stream then need to re-create the stream. */
+ if (need_to_recreate_stream && qap_mod->stream_out[QAP_OUT_OFFLOAD]) {
+ DEBUG_MSG("closing PCM session ox%x", (int)qap_mod->stream_out[QAP_OUT_OFFLOAD]);
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(qap_mod->stream_out[QAP_OUT_OFFLOAD]));
+ qap_mod->stream_out[QAP_OUT_OFFLOAD] = NULL;
+ }
+
+ bt_stream = audio_extn_bt_hal_get_output_stream(qap_mod->bt_hdl);
+ if (bt_stream != NULL) {
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD]) {
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(qap_mod->stream_out[QAP_OUT_OFFLOAD]));
+ qap_mod->stream_out[QAP_OUT_OFFLOAD] = NULL;
+ }
+
+ audio_extn_bt_hal_out_write(p_qap->bt_hdl, data_buffer_p, buffer_size);
+ } else if (NULL == qap_mod->stream_out[QAP_OUT_OFFLOAD]) {
+ audio_devices_t devices;
+
+ if (qap_mod->stream_in[QAP_IN_MAIN])
+ devices = qap_mod->stream_in[QAP_IN_MAIN]->devices;
+ else
+ devices = qap_mod->stream_in[QAP_IN_PCM]->devices;
+
+ //If multi channel pcm or passthrough is already enabled then remove the hdmi flag from device.
+ if (p_qap->mch_pcm_hdmi_enabled || p_qap->passthrough_enabled) {
+ if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+ devices ^= AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ }
+ if (devices == 0) {
+ devices = device;
+ }
+
+ flags = AUDIO_OUTPUT_FLAG_DIRECT;
+
+
+ DEBUG_MSG("Opening Stereo PCM out(outputenum=%d) session ox%x with below params",
+ QAP_OUT_OFFLOAD,
+ (int)qap_mod->stream_out[QAP_OUT_OFFLOAD]);
+
+
+ DEBUG_MSG("sample rate=%d bitwidth=%d format = 0x%x channel mask=0x%x flags=0x%x device =0x%x",
+ config.sample_rate,
+ config.offload_info.bit_width,
+ config.offload_info.format,
+ config.offload_info.channel_mask,
+ flags,
+ devices);
+
+
+ /* TODO:: Need to Propagate errors to framework */
+ ret = adev_open_output_stream((struct audio_hw_device *)p_qap->adev,
+ QAP_DEFAULT_COMPR_AUDIO_HANDLE,
+ devices,
+ flags,
+ &config,
+ (struct audio_stream_out **)&(qap_mod->stream_out[QAP_OUT_OFFLOAD]),
+ NULL);
+ if (ret < 0) {
+ ERROR_MSG("Failed opening Stereo PCM out(outputenum=%d) session ox%x",
+ QAP_OUT_OFFLOAD,
+ (int)qap_mod->stream_out[QAP_OUT_OFFLOAD]);
+ pthread_mutex_unlock(&p_qap->lock);
+ return;
+ } else
+ DEBUG_MSG("Opened Stereo PCM out(outputenum=%d) session ox%x",
+ QAP_OUT_OFFLOAD,
+ (int)qap_mod->stream_out[QAP_OUT_OFFLOAD]);
+
+ set_out_stream_channel_map(qap_mod->stream_out[QAP_OUT_OFFLOAD], new_conf);
+
+ if ((qap_mod->stream_in[QAP_IN_MAIN]
+ && qap_mod->stream_in[QAP_IN_MAIN]->client_callback != NULL) ||
+ (qap_mod->stream_in[QAP_IN_MAIN_2]
+ && qap_mod->stream_in[QAP_IN_MAIN_2]->client_callback != NULL)) {
+
+ if (qap_mod->stream_in[QAP_IN_MAIN]) {
+ qap_mod->stream_out[QAP_OUT_OFFLOAD]->stream.set_callback(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD],
+ qap_mod->stream_in[QAP_IN_MAIN]->client_callback,
+ qap_mod->stream_in[QAP_IN_MAIN]->client_cookie);
+ }
+ if (qap_mod->stream_in[QAP_IN_MAIN_2]) {
+ qap_mod->stream_out[QAP_OUT_OFFLOAD]->stream.set_callback(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD],
+ qap_mod->stream_in[QAP_IN_MAIN_2]->client_callback,
+ qap_mod->stream_in[QAP_IN_MAIN_2]->client_cookie);
+ }
+ } else if (qap_mod->stream_in[QAP_IN_PCM]
+ && qap_mod->stream_in[QAP_IN_PCM]->client_callback != NULL) {
+
+ qap_mod->stream_out[QAP_OUT_OFFLOAD]->stream.set_callback(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD],
+ qap_mod->stream_in[QAP_IN_PCM]->client_callback,
+ qap_mod->stream_in[QAP_IN_PCM]->client_cookie);
+ }
+
+ qap_mod->stream_out[QAP_OUT_OFFLOAD]->compr_config.fragments =
+ COMPRESS_OFFLOAD_NUM_FRAGMENTS;
+ qap_mod->stream_out[QAP_OUT_OFFLOAD]->compr_config.fragment_size =
+ qap_get_pcm_offload_output_buffer_size(qap_mod, &config.offload_info);
+
+ if (qap_mod->is_vol_set) {
+ DEBUG_MSG("Setting Volume Left[%f], Right[%f]", qap_mod->vol_left, qap_mod->vol_right);
+ qap_mod->stream_out[QAP_OUT_OFFLOAD]->stream.set_volume(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD],
+ qap_mod->vol_left,
+ qap_mod->vol_right);
+ }
+ }
+
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD]) {
+ DEBUG_MSG_VV("Writing Bytes(%d) to QAP_OUT_OFFLOAD output(%p) buff ptr(%p)",
+ buffer_size, qap_mod->stream_out[QAP_OUT_OFFLOAD],
+ data_buffer_p);
+ ret = qap_mod->stream_out[QAP_OUT_OFFLOAD]->stream.write(
+ (struct audio_stream_out *)qap_mod->stream_out[QAP_OUT_OFFLOAD],
+ data_buffer_p,
+ buffer_size);
+ }
+ }
+ DEBUG_MSG_VV("Bytes consumed [%d] by Audio HAL", ret);
+ }
+ else if (event_id == QAP_CALLBACK_EVENT_EOS
+ || event_id == QAP_CALLBACK_EVENT_MAIN_2_EOS
+ || event_id == QAP_CALLBACK_EVENT_EOS_ASSOC) {
+ struct stream_out *out = qap_mod->stream_in[QAP_IN_MAIN];
+ struct stream_out *out_pcm = qap_mod->stream_in[QAP_IN_PCM];
+ struct stream_out *out_main2 = qap_mod->stream_in[QAP_IN_MAIN_2];
+ struct stream_out *out_assoc = qap_mod->stream_in[QAP_IN_ASSOC];
+
+ /**
+ * TODO:: Only DD/DDP Associate Eos is handled, need to add support
+ * for other formats.
+ */
+ if (event_id == QAP_CALLBACK_EVENT_EOS
+ && (out_pcm != NULL)
+ && (check_stream_state_l(out_pcm, STOPPING))) {
+
+ lock_output_stream_l(out_pcm);
+ out_pcm->client_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out_pcm->client_cookie);
+ set_stream_state_l(out_pcm, STOPPED);
+ unlock_output_stream_l(out_pcm);
+ DEBUG_MSG("sent pcm DRAIN_READY");
+ } else if ( event_id == QAP_CALLBACK_EVENT_EOS_ASSOC
+ && (out_assoc != NULL)
+ && (check_stream_state_l(out_assoc, STOPPING))) {
+
+ lock_output_stream_l(out_assoc);
+ out_assoc->client_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out_assoc->client_cookie);
+ set_stream_state_l(out_assoc, STOPPED);
+ unlock_output_stream_l(out_assoc);
+ DEBUG_MSG("sent associated DRAIN_READY");
+ } else if (event_id == QAP_CALLBACK_EVENT_MAIN_2_EOS
+ && (out_main2 != NULL)
+ && (check_stream_state_l(out_main2, STOPPING))) {
+
+ lock_output_stream_l(out_main2);
+ out_main2->client_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out_main2->client_cookie);
+ set_stream_state_l(out_main2, STOPPED);
+ unlock_output_stream_l(out_main2);
+ DEBUG_MSG("sent main2 DRAIN_READY");
+ } else if ((out != NULL) && (check_stream_state_l(out, STOPPING))) {
+ lock_output_stream_l(out);
+ out->client_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out->client_cookie);
+ set_stream_state_l(out, STOPPED);
+ unlock_output_stream_l(out);
+ DEBUG_MSG("sent main DRAIN_READY");
+ }
+ }
+ else if (event_id == QAP_CALLBACK_EVENT_EOS || event_id == QAP_CALLBACK_EVENT_EOS_ASSOC) {
+ struct stream_out *out = NULL;
+
+ if (event_id == QAP_CALLBACK_EVENT_EOS) {
+ out = qap_mod->stream_in[QAP_IN_MAIN];
+ } else {
+ out = qap_mod->stream_in[QAP_IN_ASSOC];
+ }
+
+ if ((out != NULL) && (check_stream_state_l(out, STOPPING))) {
+ lock_output_stream_l(out);
+ out->client_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out->client_cookie);
+ set_stream_state_l(out, STOPPED);
+ unlock_output_stream_l(out);
+ DEBUG_MSG("sent DRAIN_READY");
+ }
+ }
+
+ pthread_mutex_unlock(&p_qap->lock);
+ return;
+}
+
+static int qap_sess_close(struct qap_module* qap_mod)
+{
+ int j;
+ int ret = -EINVAL;
+
+ DEBUG_MSG("Closing Session.");
+
+ //Check if all streams are closed or not.
+ for (j = 0; j < MAX_QAP_MODULE_IN; j++) {
+ if (qap_mod->stream_in[j] != NULL) {
+ break;
+ }
+ }
+ if (j != MAX_QAP_MODULE_IN) {
+ DEBUG_MSG("Some stream is still active, Can not close session.");
+ return 0;
+ }
+
+ qap_mod->is_session_closing = true;
+ if(p_qap->qap_output_block_handling) {
+ pthread_mutex_lock(&qap_mod->session_output_lock);
+ if (qap_mod->is_session_output_active == false) {
+ pthread_cond_signal(&qap_mod->session_output_cond);
+ DEBUG_MSG("Wake up MM module output thread");
+ }
+ pthread_mutex_unlock(&qap_mod->session_output_lock);
+ }
+ pthread_mutex_lock(&p_qap->lock);
+
+ if (!qap_mod || !qap_mod->session_handle) {
+ ERROR_MSG("Wrong state to process qap_mod(%p) sess_hadl(%p)",
+ qap_mod, qap_mod->session_handle);
+ return -EINVAL;
+ }
+
+ ret = qap_session_close(qap_mod->session_handle);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("close session failed %d", ret);
+ return -EINVAL;
+ } else
+ DEBUG_MSG("Closed QAP session 0x%x", (int)qap_mod->session_handle);
+
+ qap_mod->session_handle = NULL;
+ qap_mod->is_vol_set = false;
+ memset(qap_mod->stream_state, 0, sizeof(qap_mod->stream_state));
+
+ qap_close_all_output_streams(qap_mod);
+
+ qap_mod->new_out_format_index = 0;
+
+ pthread_mutex_unlock(&p_qap->lock);
+ qap_mod->is_session_closing = false;
+ DEBUG_MSG("Exit.");
+
+ return 0;
+}
+
+static int qap_stream_close(struct stream_out *out)
+{
+ int ret = -EINVAL;
+ struct qap_module *qap_mod = NULL;
+ int index = -1;
+ DEBUG_MSG("Flag [0x%x], Stream handle [%p]", out->flags, out->qap_stream_handle);
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ index = get_input_stream_index_l(out);
+
+ if (!qap_mod || !qap_mod->session_handle || (index < 0) || !out->qap_stream_handle) {
+ ERROR_MSG("Wrong state to process qap_mod(%p) sess_hadl(%p) strm hndl(%p), index %d",
+ qap_mod, qap_mod->session_handle, out->qap_stream_handle, index);
+ return -EINVAL;
+ }
+
+ pthread_mutex_lock(&p_qap->lock);
+
+ set_stream_state_l(out,STOPPED);
+ qap_mod->stream_in[index] = NULL;
+
+ lock_output_stream_l(out);
+
+ ret = qap_module_deinit(out->qap_stream_handle);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("deinit failed %d", ret);
+ return -EINVAL;
+ } else
+ DEBUG_MSG("module(ox%x) closed successfully", (int)out->qap_stream_handle);
+
+
+ out->qap_stream_handle = NULL;
+ unlock_output_stream_l(out);
+
+ pthread_mutex_unlock(&p_qap->lock);
+
+ //If all streams are closed then close the session.
+ qap_sess_close(qap_mod);
+
+ DEBUG_MSG("Exit");
+ return ret;
+}
+
+#define MAX_INIT_PARAMS 6
+
+static void update_qap_session_init_params(audio_session_handle_t session_handle) {
+ DEBUG_MSG("Entry");
+ qap_status_t ret = QAP_STATUS_OK;
+ uint32_t cmd_data[MAX_INIT_PARAMS] = {0};
+
+ /* all init params should be sent
+ * together so gang them up.
+ */
+ cmd_data[0] = MS12_SESSION_CFG_MAX_CHS;
+ cmd_data[1] = 6;/*5.1 channels*/
+
+ cmd_data[2] = MS12_SESSION_CFG_BS_OUTPUT_MODE;
+ cmd_data[3] = 3;/*DDP Re-encoding and DDP to DD Transcoding*/
+
+ cmd_data[4] = MS12_SESSION_CFG_CHMOD_LOCKING;
+ cmd_data[MAX_INIT_PARAMS - 1] = 1;/*Lock to 6 channel*/
+
+ ret = qap_session_cmd(session_handle,
+ QAP_SESSION_CMD_SET_PARAM,
+ MAX_INIT_PARAMS * sizeof(uint32_t),
+ &cmd_data[0],
+ NULL,
+ NULL);
+ if (ret != QAP_STATUS_OK) {
+ ERROR_MSG("session init params config failed");
+ }
+ DEBUG_MSG("Exit");
+ return;
+}
+
+/* Query HDMI EDID and sets module output accordingly.*/
+static void qap_set_hdmi_configuration_to_module()
+{
+ int ret = 0;
+ int channels = 0;
+ char prop_value[PROPERTY_VALUE_MAX] = {0};
+ bool passth_support = false;
+ qap_session_outputs_config_t *session_outputs_config = NULL;
+
+
+ DEBUG_MSG("Entry");
+
+ if (!p_qap) {
+ return;
+ }
+
+ if (!p_qap->hdmi_connect) {
+ return;
+ }
+
+ p_qap->hdmi_sink_channels = 0;
+
+ if (p_qap->qap_mod[MS12].session_handle)
+ session_outputs_config = &p_qap->qap_mod[MS12].session_outputs_config;
+ else if (p_qap->qap_mod[DTS_M8].session_handle)
+ session_outputs_config = &p_qap->qap_mod[DTS_M8].session_outputs_config;
+ else {
+ DEBUG_MSG("HDMI connection comes even before session is setup");
+ return;
+ }
+
+ session_outputs_config->num_output = 1;
+ //QAP re-encoding and DSP offload passthrough is supported.
+ if (property_get_bool("vendor.audio.offload.passthrough", false)
+ && property_get_bool("vendor.audio.qap.reencode", false)) {
+
+ if (p_qap->qap_mod[MS12].session_handle) {
+
+ bool do_setparam = false;
+ property_get("vendor.audio.qap.hdmi.out", prop_value, NULL);
+
+ if (platform_is_edid_supported_format(p_qap->adev->platform, AUDIO_FORMAT_E_AC3)
+ && (strncmp(prop_value, "ddp", 3) == 0)) {
+ do_setparam = true;
+ session_outputs_config->output_config[0].format = QAP_AUDIO_FORMAT_EAC3;
+ session_outputs_config->output_config[0].id = AUDIO_DEVICE_OUT_HDMI|QAP_AUDIO_FORMAT_EAC3;
+ } else if (platform_is_edid_supported_format(p_qap->adev->platform, AUDIO_FORMAT_AC3)) {
+ do_setparam = true;
+ session_outputs_config->output_config[0].format = QAP_AUDIO_FORMAT_AC3;
+ session_outputs_config->output_config[0].id = AUDIO_DEVICE_OUT_HDMI|QAP_AUDIO_FORMAT_AC3;
+ }
+ if (do_setparam) {
+ DEBUG_MSG(" Enabling HDMI(Passthrough out) from MS12 wrapper outputid=0x%x",
+ session_outputs_config->output_config[0].id);
+ ret = qap_session_cmd(p_qap->qap_mod[MS12].session_handle,
+ QAP_SESSION_CMD_SET_OUTPUTS,
+ sizeof(qap_session_outputs_config_t),
+ session_outputs_config,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("Unable to register AUDIO_DEVICE_OUT_HDMI device with QAP %d", ret);
+ return;
+ }
+ passth_support = true;
+ }
+ }
+
+ if (p_qap->qap_mod[DTS_M8].session_handle) {
+
+ bool do_setparam = false;
+ if (platform_is_edid_supported_format(p_qap->adev->platform, AUDIO_FORMAT_DTS)) {
+ do_setparam = true;
+ session_outputs_config->output_config[0].format = QAP_AUDIO_FORMAT_DTS;
+ session_outputs_config->output_config[0].id = AUDIO_DEVICE_OUT_HDMI|QAP_AUDIO_FORMAT_DTS;
+ }
+
+ if (do_setparam) {
+ ret = qap_session_cmd(p_qap->qap_mod[DTS_M8].session_handle,
+ QAP_SESSION_CMD_SET_OUTPUTS,
+ sizeof(qap_session_outputs_config_t),
+ session_outputs_config,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("Unable to register AUDIO_DEVICE_OUT_HDMI device with QAP %d", ret);
+ return;
+ }
+ passth_support = true;
+ }
+ }
+ }
+ //Compressed passthrough is not enabled.
+ if (!passth_support) {
+
+ channels = platform_edid_get_max_channels(p_qap->adev->platform);
+ session_outputs_config->output_config[0].format = QAP_AUDIO_FORMAT_PCM_16_BIT;
+
+ switch (channels) {
+ case 8:
+ DEBUG_MSG("Switching Qap output to 7.1 channels");
+ session_outputs_config->output_config[0].channels = 8;
+ if (!p_qap->qap_msmd_enabled)
+ session_outputs_config->output_config[0].id = AUDIO_DEVICE_OUT_HDMI|QAP_AUDIO_FORMAT_PCM_16_BIT;
+ p_qap->hdmi_sink_channels = channels;
+ break;
+ case 6:
+ DEBUG_MSG("Switching Qap output to 5.1 channels");
+ session_outputs_config->output_config[0].channels = 6;
+ if (!p_qap->qap_msmd_enabled)
+ session_outputs_config->output_config[0].id = AUDIO_DEVICE_OUT_HDMI|QAP_AUDIO_FORMAT_PCM_16_BIT;
+ p_qap->hdmi_sink_channels = channels;
+ break;
+ default:
+ DEBUG_MSG("Switching Qap output to default channels");
+ session_outputs_config->output_config[0].channels = 2;
+ if (!p_qap->qap_msmd_enabled)
+ session_outputs_config->output_config[0].id = AUDIO_DEVICE_OUT_HDMI|QAP_AUDIO_FORMAT_PCM_16_BIT;
+ p_qap->hdmi_sink_channels = 2;
+ break;
+ }
+
+ if (p_qap->qap_mod[MS12].session_handle) {
+ DEBUG_MSG(" Enabling HDMI(MCH PCM out) from MS12 wrapper outputid = %x", session_outputs_config->output_config[0].id);
+ ret = qap_session_cmd(p_qap->qap_mod[MS12].session_handle,
+ QAP_SESSION_CMD_SET_OUTPUTS,
+ sizeof(qap_session_outputs_config_t),
+ session_outputs_config,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("Unable to register AUDIO_DEVICE_OUT_HDMI device with QAP %d", ret);
+ return;
+ }
+ }
+ if (p_qap->qap_mod[DTS_M8].session_handle) {
+ ret = qap_session_cmd(p_qap->qap_mod[MS12].session_handle,
+ QAP_SESSION_CMD_SET_OUTPUTS,
+ sizeof(qap_session_outputs_config_t),
+ session_outputs_config,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("Unable to register AUDIO_DEVICE_OUT_HDMI device with QAP %d", ret);
+ return;
+ }
+ }
+
+ }
+ DEBUG_MSG("Exit");
+}
+
+
+static void qap_set_default_configuration_to_module()
+{
+ qap_session_outputs_config_t *session_outputs_config = NULL;
+ int ret = 0;
+
+ DEBUG_MSG("Entry");
+
+ if (!p_qap) {
+ return;
+ }
+
+ if (!p_qap->bt_connect) {
+ DEBUG_MSG("BT is not connected.");
+ }
+
+ //ms12 wrapper don't support bt, treat this as speaker and routign to bt
+ //will take care as a part of data callback notifier
+
+
+ if (p_qap->qap_mod[MS12].session_handle)
+ session_outputs_config = &p_qap->qap_mod[MS12].session_outputs_config;
+ else if (p_qap->qap_mod[DTS_M8].session_handle)
+ session_outputs_config = &p_qap->qap_mod[DTS_M8].session_outputs_config;
+
+ session_outputs_config->num_output = 1;
+ session_outputs_config->output_config[0].id = AUDIO_DEVICE_OUT_SPEAKER;
+ session_outputs_config->output_config[0].format = QAP_AUDIO_FORMAT_PCM_16_BIT;
+
+
+ if (p_qap->qap_mod[MS12].session_handle) {
+ DEBUG_MSG(" Enabling speaker(PCM out) from MS12 wrapper outputid = %x", session_outputs_config->output_config[0].id);
+ ret = qap_session_cmd(p_qap->qap_mod[MS12].session_handle,
+ QAP_SESSION_CMD_SET_OUTPUTS,
+ sizeof(qap_session_outputs_config_t),
+ session_outputs_config,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("Unable to register AUDIO_DEVICE_OUT_SPEAKER device with QAP %d", ret);
+ return;
+ }
+ }
+ if (p_qap->qap_mod[DTS_M8].session_handle) {
+ ret = qap_session_cmd(p_qap->qap_mod[DTS_M8].session_handle,
+ QAP_SESSION_CMD_SET_OUTPUTS,
+ sizeof(qap_session_outputs_config_t),
+ session_outputs_config,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != ret) {
+ ERROR_MSG("Unable to register AUDIO_DEVICE_OUT_SPEAKER device with QAP %d", ret);
+ return;
+ }
+ }
+}
+
+
+/* Open a MM module session with QAP. */
+static int audio_extn_qap_session_open(mm_module_type mod_type, __unused struct stream_out *out)
+{
+ DEBUG_MSG("%s %d", __func__, __LINE__);
+ int ret = 0;
+
+ struct qap_module *qap_mod = NULL;
+
+ if (mod_type >= MAX_MM_MODULE_TYPE)
+ return -ENOTSUP; //Not supported by QAP module.
+
+ pthread_mutex_lock(&p_qap->lock);
+
+ qap_mod = &(p_qap->qap_mod[mod_type]);
+
+ //If session is already opened then return.
+ if (qap_mod->session_handle) {
+ DEBUG_MSG("QAP Session is already opened.");
+ pthread_mutex_unlock(&p_qap->lock);
+ return 0;
+ }
+
+ if (MS12 == mod_type) {
+ if (NULL == (qap_mod->session_handle = (void *)qap_session_open(QAP_SESSION_MS12_OTT, qap_mod->qap_lib))) {
+ ERROR_MSG("Failed to open QAP session, lib_handle 0x%x", (int)qap_mod->qap_lib);
+ ret = -EINVAL;
+ goto exit;
+ } else
+ DEBUG_MSG("Opened QAP session 0x%x", (int)qap_mod->session_handle);
+
+ update_qap_session_init_params(qap_mod->session_handle);
+ }
+
+ if (QAP_STATUS_OK != (qap_session_set_callback (qap_mod->session_handle, &qap_session_callback, (void *)qap_mod))) {
+ ERROR_MSG("Failed to register QAP session callback");
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ qap_mod->is_session_output_active = true;
+
+ if(p_qap->hdmi_connect)
+ qap_set_hdmi_configuration_to_module();
+ else
+ qap_set_default_configuration_to_module();
+
+exit:
+ pthread_mutex_unlock(&p_qap->lock);
+ return ret;
+}
+
+
+
+static int qap_map_input_format(audio_format_t audio_format, qap_audio_format_t *format)
+{
+ if (audio_format == AUDIO_FORMAT_AC3) {
+ *format = QAP_AUDIO_FORMAT_AC3;
+ DEBUG_MSG( "File Format is AC3!");
+ } else if (audio_format == AUDIO_FORMAT_E_AC3) {
+ *format = QAP_AUDIO_FORMAT_EAC3;
+ DEBUG_MSG( "File Format is E_AC3!");
+ } else if ((audio_format == AUDIO_FORMAT_AAC_ADTS_LC) ||
+ (audio_format == AUDIO_FORMAT_AAC_ADTS_HE_V1) ||
+ (audio_format == AUDIO_FORMAT_AAC_ADTS_HE_V2) ||
+ (audio_format == AUDIO_FORMAT_AAC_LC) ||
+ (audio_format == AUDIO_FORMAT_AAC_HE_V1) ||
+ (audio_format == AUDIO_FORMAT_AAC_HE_V2) ||
+ (audio_format == AUDIO_FORMAT_AAC_LATM_LC) ||
+ (audio_format == AUDIO_FORMAT_AAC_LATM_HE_V1) ||
+ (audio_format == AUDIO_FORMAT_AAC_LATM_HE_V2)) {
+ *format = QAP_AUDIO_FORMAT_AAC_ADTS;
+ DEBUG_MSG( "File Format is AAC!");
+ } else if (audio_format == AUDIO_FORMAT_DTS) {
+ *format = QAP_AUDIO_FORMAT_DTS;
+ DEBUG_MSG( "File Format is DTS!");
+ } else if (audio_format == AUDIO_FORMAT_DTS_HD) {
+ *format = QAP_AUDIO_FORMAT_DTS_HD;
+ DEBUG_MSG( "File Format is DTS_HD!");
+ } else if (audio_format == AUDIO_FORMAT_PCM_16_BIT) {
+ *format = QAP_AUDIO_FORMAT_PCM_16_BIT;
+ DEBUG_MSG( "File Format is PCM_16!");
+ } else if (audio_format == AUDIO_FORMAT_PCM_32_BIT) {
+ *format = QAP_AUDIO_FORMAT_PCM_32_BIT;
+ DEBUG_MSG( "File Format is PCM_32!");
+ } else if (audio_format == AUDIO_FORMAT_PCM_24_BIT_PACKED) {
+ *format = QAP_AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ DEBUG_MSG( "File Format is PCM_24!");
+ } else if ((audio_format == AUDIO_FORMAT_PCM_8_BIT) ||
+ (audio_format == AUDIO_FORMAT_PCM_8_24_BIT)) {
+ *format = QAP_AUDIO_FORMAT_PCM_8_24_BIT;
+ DEBUG_MSG( "File Format is PCM_8_24!");
+ } else {
+ ERROR_MSG( "File Format not supported!");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+void qap_module_callback(__unused qap_module_handle_t module_handle,
+ void* priv_data,
+ qap_module_callback_event_t event_id,
+ __unused int size,
+ __unused void *data)
+{
+ struct stream_out *out=(struct stream_out *)priv_data;
+
+ DEBUG_MSG("Entry");
+ if (QAP_MODULE_CALLBACK_EVENT_SEND_INPUT_BUFFER == event_id) {
+ DEBUG_MSG("QAP_MODULE_CALLBACK_EVENT_SEND_INPUT_BUFFER for (%p)", out);
+ if (out->client_callback) {
+ out->client_callback(STREAM_CBK_EVENT_WRITE_READY, NULL, out->client_cookie);
+ }
+ else
+ DEBUG_MSG("client has no callback registered, no action needed for this event %d",
+ event_id);
+ }
+ else
+ DEBUG_MSG("Un Recognized event %d", event_id);
+
+ DEBUG_MSG("exit");
+ return;
+}
+
+
+/* opens a stream in QAP module. */
+static int qap_stream_open(struct stream_out *out,
+ struct audio_config *config,
+ audio_output_flags_t flags,
+ audio_devices_t devices)
+{
+ int status = -EINVAL;
+ mm_module_type mmtype = get_mm_module_for_format_l(config->format);
+ struct qap_module* qap_mod = NULL;
+ qap_module_config_t input_config = {0};
+
+ DEBUG_MSG("Flags 0x%x, Device 0x%x for use case %s out 0x%x", flags, devices, use_case_table[out->usecase], (int)out);
+
+ if (mmtype >= MAX_MM_MODULE_TYPE) {
+ ERROR_MSG("Unsupported Stream");
+ return -ENOTSUP;
+ }
+
+ //Open the module session, if not opened already.
+ status = audio_extn_qap_session_open(mmtype, out);
+ qap_mod = &(p_qap->qap_mod[mmtype]);
+
+ if ((status != 0) || (!qap_mod->session_handle ))
+ return status;
+
+ input_config.sample_rate = config->sample_rate;
+ input_config.channels = popcount(config->channel_mask);
+ if (input_config.format != AUDIO_FORMAT_PCM_16_BIT) {
+ input_config.format &= AUDIO_FORMAT_MAIN_MASK;
+ }
+ input_config.module_type = QAP_MODULE_DECODER;
+ status = qap_map_input_format(config->format, &input_config.format);
+ if (status == -EINVAL)
+ return -EINVAL;
+
+ DEBUG_MSG("qap_stream_open sample_rate(%d) channels(%d) devices(%#x) flags(%#x) format(%#x)",
+ input_config.sample_rate, input_config.channels, devices, flags, input_config.format);
+
+ if (input_config.format == QAP_AUDIO_FORMAT_PCM_16_BIT) {
+ //If PCM stream is already opened then fail this stream open.
+ if (qap_mod->stream_in[QAP_IN_PCM]) {
+ ERROR_MSG("PCM input is already active.");
+ return -ENOTSUP;
+ }
+ input_config.flags = QAP_MODULE_FLAG_SYSTEM_SOUND;
+ status = qap_module_init(qap_mod->session_handle, &input_config, &out->qap_stream_handle);
+ if (QAP_STATUS_OK != status) {
+ ERROR_MSG("Unable to open PCM(QAP_MODULE_FLAG_SYSTEM_SOUND) QAP module %d", status);
+ return -EINVAL;
+ } else
+ DEBUG_MSG("QAP_MODULE_FLAG_SYSTEM_SOUND, module(ox%x) opened successfully", (int)out->qap_stream_handle);
+
+ qap_mod->stream_in[QAP_IN_PCM] = out;
+ } else if ((flags & AUDIO_OUTPUT_FLAG_MAIN) && (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)) {
+ if (is_main_active_l(qap_mod) || is_dual_main_active_l(qap_mod)) {
+ ERROR_MSG("Dual Main or Main already active. So, Cannot open main and associated stream");
+ return -EINVAL;
+ } else {
+ input_config.flags = QAP_MODULE_FLAG_PRIMARY;
+ status = qap_module_init(qap_mod->session_handle, &input_config, &out->qap_stream_handle);
+ if (QAP_STATUS_OK != status) {
+ ERROR_MSG("Unable to open QAP stream/module with QAP_MODULE_FLAG_PRIMARY flag %d", status);
+ return -EINVAL;
+ } else
+ DEBUG_MSG("QAP_MODULE_FLAG_PRIMARY, module opened successfully 0x%x", (int)out->qap_stream_handle);;
+
+ qap_mod->stream_in[QAP_IN_MAIN] = out;
+ }
+ } else if ((flags & AUDIO_OUTPUT_FLAG_MAIN) || ((!(flags & AUDIO_OUTPUT_FLAG_MAIN)) && (!(flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)))) {
+ /* Assume Main if no flag is set */
+ if (is_dual_main_active_l(qap_mod)) {
+ ERROR_MSG("Dual Main already active. So, Cannot open main stream");
+ return -EINVAL;
+ } else if (is_main_active_l(qap_mod) && qap_mod->stream_in[QAP_IN_ASSOC]) {
+ ERROR_MSG("Main and Associated already active. So, Cannot open main stream");
+ return -EINVAL;
+ } else if (is_main_active_l(qap_mod) && (mmtype != MS12)) {
+ ERROR_MSG("Main already active and Not an MS12 format. So, Cannot open another main stream");
+ return -EINVAL;
+ } else {
+ input_config.flags = QAP_MODULE_FLAG_PRIMARY;
+ status = qap_module_init(qap_mod->session_handle, &input_config, &out->qap_stream_handle);
+ if (QAP_STATUS_OK != status) {
+ ERROR_MSG("Unable to open QAP stream/module with QAP_MODULE_FLAG_PRIMARY flag %d", status);
+ return -EINVAL;
+ } else
+ DEBUG_MSG("QAP_MODULE_FLAG_PRIMARY, module opened successfully 0x%x", (int)out->qap_stream_handle);
+
+ if(qap_mod->stream_in[QAP_IN_MAIN]) {
+ qap_mod->stream_in[QAP_IN_MAIN_2] = out;
+ } else {
+ qap_mod->stream_in[QAP_IN_MAIN] = out;
+ }
+ }
+ } else if ((flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)) {
+ if (is_dual_main_active_l(qap_mod)) {
+ ERROR_MSG("Dual Main already active. So, Cannot open associated stream");
+ return -EINVAL;
+ } else if (!is_main_active_l(qap_mod)) {
+ ERROR_MSG("Main not active. So, Cannot open associated stream");
+ return -EINVAL;
+ } else if (qap_mod->stream_in[QAP_IN_ASSOC]) {
+ ERROR_MSG("Associated already active. So, Cannot open associated stream");
+ return -EINVAL;
+ }
+ input_config.flags = QAP_MODULE_FLAG_SECONDARY;
+ status = qap_module_init(qap_mod->session_handle, &input_config, &out->qap_stream_handle);
+ if (QAP_STATUS_OK != status) {
+ ERROR_MSG("Unable to open QAP stream/module with QAP_MODULE_FLAG_SECONDARY flag %d", status);
+ return -EINVAL;
+ } else
+ DEBUG_MSG("QAP_MODULE_FLAG_SECONDARY, module opened successfully 0x%x", (int)out->qap_stream_handle);
+
+ qap_mod->stream_in[QAP_IN_ASSOC] = out;
+ }
+
+ if (out->qap_stream_handle) {
+ status = qap_module_set_callback(out->qap_stream_handle, &qap_module_callback, out);
+ if (QAP_STATUS_OK != status) {
+ ERROR_MSG("Unable to register module callback %d", status);
+ return -EINVAL;
+ } else
+ DEBUG_MSG("Module call back registered 0x%x cookie 0x%x", (int)out->qap_stream_handle, (int)out);
+ }
+
+ if (status != 0) {
+ //If no stream is active then close the session.
+ qap_sess_close(qap_mod);
+ return 0;
+ }
+
+ //If Device is HDMI, QAP passthrough is enabled and there is no previous QAP passthrough input stream.
+ if ((!p_qap->passthrough_in)
+ && (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+ && audio_extn_qap_passthrough_enabled(out)) {
+ //Assign the QAP passthrough input stream.
+ p_qap->passthrough_in = out;
+
+ //If HDMI is connected and format is supported by HDMI then create QAP passthrough output stream.
+ if (p_qap->hdmi_connect
+ && platform_is_edid_supported_format(p_qap->adev->platform, out->format)) {
+ status = create_qap_passthrough_stream_l();
+ if (status < 0) {
+ qap_stream_close(out);
+ ERROR_MSG("QAP passthrough stream creation failed with error %d", status);
+ return status;
+ }
+ }
+ /*Else: since QAP passthrough input stream is already initialized,
+ * when hdmi is connected
+ * then qap passthrough output stream will be created.
+ */
+ }
+
+ DEBUG_MSG();
+ return status;
+}
+
+static int qap_out_resume(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = 0;
+ DEBUG_MSG("Output Stream %p", out);
+
+
+ lock_output_stream_l(out);
+
+ //If QAP passthrough is active then block the resume on module input streams.
+ if (p_qap->passthrough_out) {
+ //If resume is received for the QAP passthrough stream then call the primary HAL api.
+ pthread_mutex_lock(&p_qap->lock);
+ if (p_qap->passthrough_in == out) {
+ status = p_qap->passthrough_out->stream.resume(
+ (struct audio_stream_out*)p_qap->passthrough_out);
+ if (!status) out->offload_state = OFFLOAD_STATE_PLAYING;
+ }
+ pthread_mutex_unlock(&p_qap->lock);
+ } else {
+ //Flush the module input stream.
+ status = qap_stream_start_l(out);
+ }
+
+ unlock_output_stream_l(out);
+
+ DEBUG_MSG();
+ return status;
+}
+
+static int qap_out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ struct str_parms *parms;
+ char value[32];
+ int val = 0;
+ struct stream_out *out = (struct stream_out *)stream;
+ int ret = 0;
+ int err = 0;
+ struct qap_module *qap_mod = NULL;
+
+ DEBUG_MSG("usecase(%d: %s) kvpairs: %s", out->usecase, use_case_table[out->usecase], kvpairs);
+
+ parms = str_parms_create_str(kvpairs);
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+ if (err < 0)
+ return err;
+ val = atoi(value);
+
+ qap_mod = get_qap_module_for_input_stream_l(out);
+ if (!qap_mod) return (-EINVAL);
+
+ //TODO: HDMI is connected but user doesn't want HDMI output, close both HDMI outputs.
+
+ /* Setting new device information to the mm module input streams.
+ * This is needed if QAP module output streams are not created yet.
+ */
+ out->devices = val;
+
+#ifndef SPLIT_A2DP_ENABLED
+ if (val == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) {
+ //If device is BT then open the BT stream if not already opened.
+ if ( audio_extn_bt_hal_get_output_stream(qap_mod->bt_hdl) == NULL
+ && audio_extn_bt_hal_get_device(qap_mod->bt_hdl) != NULL) {
+ ret = audio_extn_bt_hal_open_output_stream(qap_mod->bt_hdl,
+ QAP_OUTPUT_SAMPLING_RATE,
+ AUDIO_CHANNEL_OUT_STEREO,
+ CODEC_BACKEND_DEFAULT_BIT_WIDTH);
+ if (ret != 0) {
+ ERROR_MSG("BT Output stream open failure!");
+ }
+ }
+ } else if (val != 0) {
+ //If device is not BT then close the BT stream if already opened.
+ if ( audio_extn_bt_hal_get_output_stream(qap_mod->bt_hdl) != NULL) {
+ audio_extn_bt_hal_close_output_stream(qap_mod->bt_hdl);
+ }
+ }
+#endif
+
+ if (p_qap->passthrough_in == out) { //Device routing is received for QAP passthrough stream.
+
+ if (!(val & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { //HDMI route is disabled.
+
+ //If QAP pasthrough output is enabled. Close it.
+ close_qap_passthrough_stream_l();
+
+ //Send the routing information to mm module pcm output.
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD]) {
+ ret = qap_mod->stream_out[QAP_OUT_OFFLOAD]->stream.common.set_parameters(
+ (struct audio_stream *)qap_mod->stream_out[QAP_OUT_OFFLOAD], kvpairs);
+ }
+ //else: device info is updated in the input streams.
+ } else { //HDMI route is enabled.
+
+ //create the QAf passthrough stream, if not created already.
+ ret = create_qap_passthrough_stream_l();
+
+ if (p_qap->passthrough_out != NULL) { //If QAP passthrough out is enabled then send routing information.
+ ret = p_qap->passthrough_out->stream.common.set_parameters(
+ (struct audio_stream *)p_qap->passthrough_out, kvpairs);
+ }
+ }
+ } else {
+ //Send the routing information to mm module pcm output.
+ if (qap_mod->stream_out[QAP_OUT_OFFLOAD]) {
+ ret = qap_mod->stream_out[QAP_OUT_OFFLOAD]->stream.common.set_parameters(
+ (struct audio_stream *)qap_mod->stream_out[QAP_OUT_OFFLOAD], kvpairs);
+ }
+ //else: device info is updated in the input streams.
+ }
+ str_parms_destroy(parms);
+
+ return ret;
+}
+
+/* Checks if a stream is QAP stream or not. */
+bool audio_extn_is_qap_stream(struct stream_out *out)
+{
+ struct qap_module *qap_mod = get_qap_module_for_input_stream_l(out);
+
+ if (qap_mod) {
+ return true;
+ }
+ return false;
+}
+
+#if 0
+/* API to send playback stream specific config parameters */
+int audio_extn_qap_out_set_param_data(struct stream_out *out,
+ audio_extn_param_id param_id,
+ audio_extn_param_payload *payload)
+{
+ int ret = -EINVAL;
+ int index;
+ struct stream_out *new_out = NULL;
+ struct audio_adsp_event *adsp_event;
+ struct qap_module *qap_mod = get_qap_module_for_input_stream_l(out);
+
+ if (!out || !qap_mod || !payload) {
+ ERROR_MSG("Invalid Param");
+ return ret;
+ }
+
+ /* apply param for all active out sessions */
+ for (index = 0; index < MAX_QAP_MODULE_OUT; index++) {
+ new_out = qap_mod->stream_out[index];
+ if (!new_out) continue;
+
+ /*ADSP event is not supported for passthrough*/
+ if ((param_id == AUDIO_EXTN_PARAM_ADSP_STREAM_CMD)
+ && !(new_out->flags == AUDIO_OUTPUT_FLAG_DIRECT)) continue;
+ if (new_out->standby)
+ new_out->stream.write((struct audio_stream_out *)new_out, NULL, 0);
+ lock_output_stream_l(new_out);
+ ret = audio_extn_out_set_param_data(new_out, param_id, payload);
+ if (ret)
+ ERROR_MSG("audio_extn_out_set_param_data error %d", ret);
+ unlock_output_stream_l(new_out);
+ }
+ return ret;
+}
+
+int audio_extn_qap_out_get_param_data(struct stream_out *out,
+ audio_extn_param_id param_id,
+ audio_extn_param_payload *payload)
+{
+ int ret = -EINVAL, i;
+ struct stream_out *new_out = NULL;
+ struct qap_module *qap_mod = get_qap_module_for_input_stream_l(out);
+
+ if (!out || !qap_mod || !payload) {
+ ERROR_MSG("Invalid Param");
+ return ret;
+ }
+
+ if (!p_qap->hdmi_connect) {
+ ERROR_MSG("hdmi not connected");
+ return ret;
+ }
+
+ /* get session which is routed to hdmi*/
+ if (p_qap->passthrough_out)
+ new_out = p_qap->passthrough_out;
+ else {
+ for (i = 0; i < MAX_QAP_MODULE_OUT; i++) {
+ if (qap_mod->stream_out[i]) {
+ new_out = qap_mod->stream_out[i];
+ break;
+ }
+ }
+ }
+
+ if (!new_out) {
+ ERROR_MSG("No stream active.");
+ return ret;
+ }
+
+ if (new_out->standby)
+ new_out->stream.write((struct audio_stream_out *)new_out, NULL, 0);
+
+ lock_output_stream_l(new_out);
+ ret = audio_extn_out_get_param_data(new_out, param_id, payload);
+ if (ret)
+ ERROR_MSG("audio_extn_out_get_param_data error %d", ret);
+ unlock_output_stream_l(new_out);
+
+ return ret;
+}
+#endif
+
+int audio_extn_qap_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address)
+{
+ int ret = 0;
+ struct stream_out *out;
+
+ DEBUG_MSG("Entry");
+ ret = adev_open_output_stream(dev, handle, devices, flags, config, stream_out, address);
+ if (*stream_out == NULL) {
+ ERROR_MSG("Stream open failed %d", ret);
+ return ret;
+ }
+
+#ifndef LINUX_ENABLED
+//Bypass QAP for dummy PCM session opened by APM during boot time
+ if(flags == 0) {
+ ALOGD("bypassing QAP for flags is equal to none");
+ return ret;
+ }
+#endif
+
+ out = (struct stream_out *)*stream_out;
+
+ DEBUG_MSG("%s 0x%x", use_case_table[out->usecase], (int)out);
+
+ ret = qap_stream_open(out, config, flags, devices);
+ if (ret < 0) {
+ ERROR_MSG("Error opening QAP stream err[%d]", ret);
+ //Stream not supported by QAP, Bypass QAP.
+ return 0;
+ }
+
+ /* Override function pointers based on qap definitions */
+ out->stream.set_volume = qap_out_set_volume;
+ out->stream.pause = qap_out_pause;
+ out->stream.resume = qap_out_resume;
+ out->stream.drain = qap_out_drain;
+ out->stream.flush = qap_out_flush;
+
+ out->stream.common.standby = qap_out_standby;
+ out->stream.common.set_parameters = qap_out_set_parameters;
+ out->stream.get_latency = qap_out_get_latency;
+ out->stream.get_render_position = qap_out_get_render_position;
+ out->stream.write = qap_out_write;
+ out->stream.get_presentation_position = qap_out_get_presentation_position;
+ out->platform_latency = 0;
+
+ /*TODO: Need to handle this for DTS*/
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_LOW_LATENCY) {
+ out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
+ out->config.period_size = QAP_DEEP_BUFFER_OUTPUT_PERIOD_SIZE;
+ out->config.period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT;
+ out->config.start_threshold = QAP_DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+ out->config.avail_min = QAP_DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+ } else if(out->flags == AUDIO_OUTPUT_FLAG_DIRECT) {
+ out->compr_config.fragment_size = qap_get_pcm_offload_input_buffer_size(&(config->offload_info));
+ }
+
+ *stream_out = &out->stream;
+
+ DEBUG_MSG("Exit");
+ return 0;
+}
+
+void audio_extn_qap_close_output_stream(struct audio_hw_device *dev,
+ struct audio_stream_out *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct qap_module* qap_mod = get_qap_module_for_input_stream_l(out);
+
+ DEBUG_MSG("%s 0x%x", use_case_table[out->usecase], (int)out);
+
+ if (!qap_mod) {
+ DEBUG_MSG("qap module is NULL, nothing to close");
+ /*closing non-MS12/default output stream opened with qap */
+ adev_close_output_stream(dev, stream);
+ return;
+ }
+
+ DEBUG_MSG("stream_handle(%p) format = %x", out, out->format);
+
+ //If close is received for QAP passthrough stream then close the QAP passthrough output.
+ if (p_qap->passthrough_in == out) {
+ if (p_qap->passthrough_out) {
+ ALOGD("%s %d closing stream handle %p", __func__, __LINE__, p_qap->passthrough_out);
+ pthread_mutex_lock(&p_qap->lock);
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(p_qap->passthrough_out));
+ pthread_mutex_unlock(&p_qap->lock);
+ p_qap->passthrough_out = NULL;
+ }
+
+ p_qap->passthrough_in = NULL;
+ }
+
+ qap_stream_close(out);
+
+ adev_close_output_stream(dev, stream);
+
+ DEBUG_MSG("Exit");
+}
+
+/* Check if QAP is supported or not. */
+bool audio_extn_qap_is_enabled()
+{
+ bool prop_enabled = false;
+ char value[PROPERTY_VALUE_MAX] = {0};
+ property_get("vendor.audio.qap.enabled", value, NULL);
+ prop_enabled = atoi(value) || !strncmp("true", value, 4);
+ return (prop_enabled);
+}
+
+/* QAP set parameter function. For Device connect and disconnect. */
+int audio_extn_qap_set_parameters(struct audio_device *adev, struct str_parms *parms)
+{
+ int status = 0, val = 0;
+ qap_session_outputs_config_t *session_outputs_config = NULL;
+
+ if (!p_qap) {
+ return -EINVAL;
+ }
+
+ DEBUG_MSG("Entry");
+
+ status = str_parms_get_int(parms, AUDIO_PARAMETER_DEVICE_CONNECT, &val);
+
+ if ((status >= 0) && audio_is_output_device(val)) {
+ if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) { //HDMI is connected.
+ DEBUG_MSG("AUDIO_DEVICE_OUT_AUX_DIGITAL connected");
+ p_qap->hdmi_connect = 1;
+ p_qap->hdmi_sink_channels = 0;
+
+ if (p_qap->passthrough_in) { //If QAP passthrough is already initialized.
+ lock_output_stream_l(p_qap->passthrough_in);
+ if (platform_is_edid_supported_format(adev->platform,
+ p_qap->passthrough_in->format)) {
+ //If passthrough format is supported by HDMI then create the QAP passthrough output if not created already.
+ create_qap_passthrough_stream_l();
+ //Ignoring the returned error, If error then QAP passthrough is disabled.
+ } else {
+ //If passthrough format is not supported by HDMI then close the QAP passthrough output if already created.
+ close_qap_passthrough_stream_l();
+ }
+ unlock_output_stream_l(p_qap->passthrough_in);
+ }
+
+ qap_set_hdmi_configuration_to_module();
+
+ } else if (val & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) {
+ DEBUG_MSG("AUDIO_DEVICE_OUT_BLUETOOTH_A2DP connected");
+ p_qap->bt_connect = 1;
+ qap_set_default_configuration_to_module();
+#ifndef SPLIT_A2DP_ENABLED
+ for (k = 0; k < MAX_MM_MODULE_TYPE; k++) {
+ if (!p_qap->qap_mod[k].bt_hdl) {
+ DEBUG_MSG("Opening a2dp output...");
+ status = audio_extn_bt_hal_load(&p_qap->qap_mod[k].bt_hdl);
+ if (status != 0) {
+ ERROR_MSG("Error opening BT module");
+ return status;
+ }
+ }
+ }
+#endif
+ }
+ //TODO else if: Need to consider other devices.
+ }
+
+ status = str_parms_get_int(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, &val);
+ if ((status >= 0) && audio_is_output_device(val)) {
+ DEBUG_MSG("AUDIO_DEVICE_OUT_AUX_DIGITAL disconnected");
+ if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+
+ p_qap->hdmi_sink_channels = 0;
+
+ p_qap->passthrough_enabled = 0;
+ p_qap->mch_pcm_hdmi_enabled = 0;
+ p_qap->hdmi_connect = 0;
+
+ if (p_qap->qap_mod[MS12].session_handle)
+ session_outputs_config = &p_qap->qap_mod[MS12].session_outputs_config;
+ else if (p_qap->qap_mod[DTS_M8].session_handle)
+ session_outputs_config = &p_qap->qap_mod[DTS_M8].session_outputs_config;
+ else {
+ DEBUG_MSG("HDMI disconnection comes even before session is setup");
+ return 0;
+ }
+
+ session_outputs_config->num_output = 1;
+
+ session_outputs_config->output_config[0].id = AUDIO_DEVICE_OUT_SPEAKER;
+ session_outputs_config->output_config[0].format = QAP_AUDIO_FORMAT_PCM_16_BIT;
+
+
+ if (p_qap->qap_mod[MS12].session_handle) {
+ DEBUG_MSG(" Enabling speaker(PCM out) from MS12 wrapper outputid = %x", session_outputs_config->output_config[0].id);
+ status = qap_session_cmd(p_qap->qap_mod[MS12].session_handle,
+ QAP_SESSION_CMD_SET_OUTPUTS,
+ sizeof(qap_session_outputs_config_t),
+ session_outputs_config,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != status) {
+ ERROR_MSG("Unable to register AUDIO_DEVICE_OUT_SPEAKER device with QAP %d",status);
+ return -EINVAL;
+ }
+ }
+ if (p_qap->qap_mod[DTS_M8].session_handle) {
+ status = qap_session_cmd(p_qap->qap_mod[MS12].session_handle,
+ QAP_SESSION_CMD_SET_OUTPUTS,
+ sizeof(qap_session_outputs_config_t),
+ session_outputs_config,
+ NULL,
+ NULL);
+ if (QAP_STATUS_OK != status) {
+ ERROR_MSG("Unable to register AUDIO_DEVICE_OUT_SPEAKER device with QAP %d", status);
+ return -EINVAL;
+ }
+ }
+
+ close_all_hdmi_output_l();
+ close_qap_passthrough_stream_l();
+ } else if (val & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) {
+ DEBUG_MSG("AUDIO_DEVICE_OUT_BLUETOOTH_A2DP disconnected");
+ p_qap->bt_connect = 0;
+ //reconfig HDMI as end device (if connected)
+ if(p_qap->hdmi_connect)
+ qap_set_hdmi_configuration_to_module();
+#ifndef SPLIT_A2DP_ENABLED
+ DEBUG_MSG("Closing a2dp output...");
+ for (k = 0; k < MAX_MM_MODULE_TYPE; k++) {
+ if (p_qap->qap_mod[k].bt_hdl) {
+ audio_extn_bt_hal_unload(p_qap->qap_mod[k].bt_hdl);
+ p_qap->qap_mod[k].bt_hdl = NULL;
+ }
+ }
+#endif
+ }
+ //TODO else if: Need to consider other devices.
+ }
+
+#if 0
+ /* does this need to be ported to QAP?*/
+ for (k = 0; k < MAX_MM_MODULE_TYPE; k++) {
+ kv_parirs = str_parms_to_str(parms);
+ if (p_qap->qap_mod[k].session_handle) {
+ p_qap->qap_mod[k].qap_audio_session_set_param(
+ p_qap->qap_mod[k].session_handle, kv_parirs);
+ }
+ }
+#endif
+
+ DEBUG_MSG("Exit");
+ return status;
+}
+
+/* Create the QAP. */
+int audio_extn_qap_init(struct audio_device *adev)
+{
+ DEBUG_MSG("Entry");
+
+ p_qap = calloc(1, sizeof(struct qap));
+ if (p_qap == NULL) {
+ ERROR_MSG("Out of memory");
+ return -ENOMEM;
+ }
+
+ p_qap->adev = adev;
+
+ if (property_get_bool("vendor.audio.qap.msmd", false)) {
+ DEBUG_MSG("MSMD enabled.");
+ p_qap->qap_msmd_enabled = 1;
+ }
+
+ if (property_get_bool("vendor.audio.qap.output.block.handling", false)) {
+ DEBUG_MSG("out put thread blocking handling enabled.");
+ p_qap->qap_output_block_handling = 1;
+ }
+ pthread_mutex_init(&p_qap->lock, (const pthread_mutexattr_t *) NULL);
+
+ int i = 0;
+
+ for (i = 0; i < MAX_MM_MODULE_TYPE; i++) {
+ char value[PROPERTY_VALUE_MAX] = {0};
+ char lib_name[PROPERTY_VALUE_MAX] = {0};
+ struct qap_module *qap_mod = &(p_qap->qap_mod[i]);
+
+ if (i == MS12) {
+ property_get("vendor.audio.qap.library", value, NULL);
+ snprintf(lib_name, PROPERTY_VALUE_MAX, "%s", value);
+
+ DEBUG_MSG("Opening Ms12 library at %s", lib_name);
+ qap_mod->qap_lib = ( void *) qap_load_library(lib_name);
+ if (qap_mod->qap_lib == NULL) {
+ ERROR_MSG("qap load lib failed for MS12 %s", lib_name);
+ continue;
+ }
+ DEBUG_MSG("Loaded QAP lib at %s", lib_name);
+ pthread_mutex_init(&qap_mod->session_output_lock, (const pthread_mutexattr_t *) NULL);
+ pthread_cond_init(&qap_mod->session_output_cond, (const pthread_condattr_t *)NULL);
+ } else if (i == DTS_M8) {
+ property_get("vendor.audio.qap.m8.library", value, NULL);
+ snprintf(lib_name, PROPERTY_VALUE_MAX, "%s", value);
+ qap_mod->qap_lib = dlopen(lib_name, RTLD_NOW);
+ if (qap_mod->qap_lib == NULL) {
+ ERROR_MSG("DLOPEN failed for DTS M8 %s", lib_name);
+ continue;
+ }
+ DEBUG_MSG("DLOPEN successful for %s", lib_name);
+ pthread_mutex_init(&qap_mod->session_output_lock, (const pthread_mutexattr_t *) NULL);
+ pthread_cond_init(&qap_mod->session_output_cond, (const pthread_condattr_t *)NULL);
+ } else {
+ continue;
+ }
+ }
+
+ DEBUG_MSG("Exit");
+ return 0;
+}
+
+/* Tear down the qap extension. */
+void audio_extn_qap_deinit()
+{
+ int i;
+ DEBUG_MSG("Entry");
+ char value[PROPERTY_VALUE_MAX] = {0};
+ char lib_name[PROPERTY_VALUE_MAX] = {0};
+
+ if (p_qap != NULL) {
+ for (i = 0; i < MAX_MM_MODULE_TYPE; i++) {
+ if (p_qap->qap_mod[i].session_handle != NULL)
+ qap_sess_close(&p_qap->qap_mod[i]);
+
+ if (p_qap->qap_mod[i].qap_lib != NULL) {
+ if (i == MS12) {
+ property_get("vendor.audio.qap.library", value, NULL);
+ snprintf(lib_name, PROPERTY_VALUE_MAX, "%s", value);
+ DEBUG_MSG("lib_name %s", lib_name);
+ if (QAP_STATUS_OK != qap_unload_library(p_qap->qap_mod[i].qap_lib))
+ ERROR_MSG("Failed to unload MS12 library lib name %s", lib_name);
+ else
+ DEBUG_MSG("closed/unloaded QAP lib at %s", lib_name);
+ p_qap->qap_mod[i].qap_lib = NULL;
+ } else {
+ dlclose(p_qap->qap_mod[i].qap_lib);
+ p_qap->qap_mod[i].qap_lib = NULL;
+ }
+ pthread_mutex_destroy(&p_qap->qap_mod[i].session_output_lock);
+ pthread_cond_destroy(&p_qap->qap_mod[i].session_output_cond);
+ }
+ }
+
+ if (p_qap->passthrough_out) {
+ adev_close_output_stream((struct audio_hw_device *)p_qap->adev,
+ (struct audio_stream_out *)(p_qap->passthrough_out));
+ p_qap->passthrough_out = NULL;
+ }
+
+ pthread_mutex_destroy(&p_qap->lock);
+ free(p_qap);
+ p_qap = NULL;
+ }
+ DEBUG_MSG("Exit");
+}
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index a07796a..aa13c2b 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -84,6 +84,7 @@
AUDIO_EVENT_CAPTURE_STREAM_INACTIVE,
AUDIO_EVENT_CAPTURE_STREAM_ACTIVE,
AUDIO_EVENT_BATTERY_STATUS_CHANGED,
+ AUDIO_EVENT_SCREEN_STATUS_CHANGED,
AUDIO_EVENT_GET_PARAM,
AUDIO_EVENT_UPDATE_ECHO_REF
} audio_event_type_t;
@@ -605,6 +606,17 @@
st_dev->st_callback(AUDIO_EVENT_BATTERY_STATUS_CHANGED, &ev_info);
}
+void audio_extn_sound_trigger_update_screen_status(bool screen_off)
+{
+ struct audio_event_info ev_info = {{0}, {0}};
+
+ if (!st_dev)
+ return;
+
+ ev_info.u.value = screen_off;
+ st_dev->st_callback(AUDIO_EVENT_SCREEN_STATUS_CHANGED, &ev_info);
+}
+
void audio_extn_sound_trigger_set_parameters(struct audio_device *adev __unused,
struct str_parms *params)
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index e6859fe..c1ee008 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -395,6 +395,7 @@
return -1;
}
memcpy(tmp, interval_str_start, eol-interval_str_start);
+ tmp[eol-interval_str_start] = '\0';
sscanf(tmp, "%lu %2s", &interval, &time_unit[0]);
if (!strcmp(time_unit, "us")) {
multiplier = 1;
@@ -628,6 +629,7 @@
int32_t fd=-1;
char path[128];
int ret = 0;
+ char *saveptr = NULL;
memset(usb_card_info->usbid, 0, sizeof(usb_card_info->usbid));
@@ -655,7 +657,7 @@
goto done;
}
- strtok(usb_card_info->usbid, "\n");
+ strtok_r(usb_card_info->usbid, "\n", &saveptr);
done:
if (fd >= 0)
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index ea0d324..30bc10d 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -37,6 +37,7 @@
#include "platform.h"
#include "platform_api.h"
#include "audio_extn.h"
+#include "voice_extn.h"
#include "voice.h"
#include <sound/compress_params.h>
#include <sound/compress_offload.h>
@@ -1140,7 +1141,8 @@
(usecase->id != USECASE_AUDIO_TRANSCODE_LOOPBACK_RX) &&
(!is_interactive_usecase(usecase->id)) &&
(!is_offload_usecase(usecase->id)) &&
- (usecase->type != PCM_CAPTURE)) {
+ (usecase->type != PCM_CAPTURE) &&
+ (!audio_extn_auto_hal_is_bus_device_usecase(usecase->id))) {
ALOGV("%s: a rx/tx/loopback path where app type cfg is not required %d", __func__, usecase->id);
rc = 0;
goto exit_send_app_type_cfg;
@@ -1248,6 +1250,88 @@
return rc;
}
+static int audio_extn_utils_check_input_parameters(uint32_t sample_rate,
+ audio_format_t format,
+ int channel_count)
+{
+ int ret = 0;
+
+ if (((format != AUDIO_FORMAT_PCM_16_BIT) && (format != AUDIO_FORMAT_PCM_8_24_BIT) &&
+ (format != AUDIO_FORMAT_PCM_24_BIT_PACKED) && (format != AUDIO_FORMAT_PCM_32_BIT) &&
+ (format != AUDIO_FORMAT_PCM_FLOAT)) &&
+ !voice_extn_compress_voip_is_format_supported(format) &&
+ !audio_extn_compr_cap_format_supported(format) &&
+ !audio_extn_cin_format_supported(format))
+ ret = -EINVAL;
+
+ switch (channel_count) {
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ case 6:
+ case 8:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ switch (sample_rate) {
+ case 8000:
+ case 11025:
+ case 12000:
+ case 16000:
+ case 22050:
+ case 24000:
+ case 32000:
+ case 44100:
+ case 48000:
+ case 88200:
+ case 96000:
+ case 176400:
+ case 192000:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static inline uint32_t audio_extn_utils_nearest_multiple(uint32_t num, uint32_t multiplier)
+{
+ uint32_t remainder = 0;
+
+ if (!multiplier)
+ return num;
+
+ remainder = num % multiplier;
+ if (remainder)
+ num += (multiplier - remainder);
+
+ return num;
+}
+
+static inline uint32_t audio_extn_utils_lcm(uint32_t num1, uint32_t num2)
+{
+ uint32_t high = num1, low = num2, temp = 0;
+
+ if (!num1 || !num2)
+ return 0;
+
+ if (num1 < num2) {
+ high = num2;
+ low = num1;
+ }
+
+ while (low != 0) {
+ temp = low;
+ low = high % low;
+ high = temp;
+ }
+ return (num1 * num2)/high;
+}
+
int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
struct audio_usecase *usecase)
{
@@ -1441,11 +1525,15 @@
uint32_t get_alsa_fragment_size(uint32_t bytes_per_sample,
uint32_t sample_rate,
- uint32_t noOfChannels)
+ uint32_t noOfChannels,
+ int64_t duration_ms)
{
uint32_t fragment_size = 0;
uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
+ if (duration_ms >= MIN_OFFLOAD_BUFFER_DURATION_MS && duration_ms <= MAX_OFFLOAD_BUFFER_DURATION_MS)
+ pcm_offload_time = duration_ms;
+
fragment_size = (pcm_offload_time
* sample_rate
* bytes_per_sample
@@ -1480,7 +1568,8 @@
out->compr_config.fragment_size =
get_alsa_fragment_size(hal_op_bytes_per_sample,
out->sample_rate,
- popcount(out->channel_mask));
+ popcount(out->channel_mask),
+ out->info.duration_us / 1000);
if ((src_format != dst_format) &&
hal_op_bytes_per_sample != hal_ip_bytes_per_sample) {
@@ -2859,3 +2948,51 @@
return is_running_with_enhanced_fwk;
}
+
+int audio_extn_utils_get_perf_mode_flag(void)
+{
+#ifdef COMPRESSED_PERF_MODE_FLAG
+ return COMPRESSED_PERF_MODE_FLAG;
+#else
+ return 0;
+#endif
+}
+
+size_t audio_extn_utils_get_input_buffer_size(uint32_t sample_rate,
+ audio_format_t format,
+ int channel_count,
+ int64_t duration_ms,
+ bool is_low_latency)
+{
+ size_t size = 0;
+ size_t capture_duration = AUDIO_CAPTURE_PERIOD_DURATION_MSEC;
+ uint32_t bytes_per_period_sample = 0;
+
+
+ if (audio_extn_utils_check_input_parameters(sample_rate, format, channel_count) != 0)
+ return 0;
+
+ if (duration_ms >= MIN_OFFLOAD_BUFFER_DURATION_MS && duration_ms <= MAX_OFFLOAD_BUFFER_DURATION_MS)
+ capture_duration = duration_ms;
+
+ size = (sample_rate * capture_duration) / 1000;
+ if (is_low_latency)
+ size = LOW_LATENCY_CAPTURE_PERIOD_SIZE;
+
+
+ bytes_per_period_sample = audio_bytes_per_sample(format) * channel_count;
+ size *= bytes_per_period_sample;
+
+ /* make sure the size is multiple of 32 bytes and additionally multiple of
+ * the frame_size (required for 24bit samples and non-power-of-2 channel counts)
+ * At 48 kHz mono 16-bit PCM:
+ * 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
+ * 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
+ *
+ * The loop reaches result within 32 iterations, as initial size is
+ * already a multiple of frame_size
+ */
+ size = audio_extn_utils_nearest_multiple(size, audio_extn_utils_lcm(32, bytes_per_period_sample));
+
+ return size;
+}
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index a5df951..71f62a7 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -400,7 +400,12 @@
[USECASE_AUDIO_EC_REF_LOOPBACK] = "ec-ref-audio-capture",
- [USECASE_AUDIO_A2DP_ABR_FEEDBACK] = "a2dp-abr-feedback"
+ [USECASE_AUDIO_A2DP_ABR_FEEDBACK] = "a2dp-abr-feedback",
+
+ [USECASE_AUDIO_PLAYBACK_MEDIA] = "media-playback",
+ [USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION] = "sys-notification-playback",
+ [USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE] = "nav-guidance-playback",
+ [USECASE_AUDIO_PLAYBACK_PHONE] = "phone-playback",
};
static const audio_usecase_t offload_usecases[] = {
@@ -483,6 +488,11 @@
STRING_TO_ENUM(192000),
};
+struct in_effect_list {
+ struct listnode list;
+ effect_handle_t handle;
+};
+
static struct audio_device *adev = NULL;
static pthread_mutex_t adev_init_lock = PTHREAD_MUTEX_INITIALIZER;
static unsigned int audio_device_ref_count;
@@ -511,7 +521,6 @@
audio_microphone_direction_t dir);
static int in_set_microphone_field_dimension(const struct audio_stream_in *stream, float zoom);
-
static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
int flags __unused)
{
@@ -923,7 +932,12 @@
char mixer_ctl_name[] = "Audio Effect";
struct mixer_ctl *ctl;
long set_values[6];
- struct stream_in *in = adev->active_input;
+ struct stream_in *in = adev_get_active_input(adev);
+
+ if (in == NULL) {
+ ALOGE("%s: active input stream is NULL", __func__);
+ return -EINVAL;
+ }
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
@@ -951,7 +965,12 @@
int ret = 0;
struct audio_effect_config other_effect_config;
struct audio_usecase *usecase = NULL;
- struct stream_in *in = adev->active_input;
+ struct stream_in *in = adev_get_active_input(adev);
+
+ if (in == NULL) {
+ ALOGE("%s: active input stream is NULL", __func__);
+ return -EINVAL;
+ }
usecase = get_usecase_from_list(adev, in->usecase);
if (!usecase)
@@ -981,7 +1000,7 @@
struct audio_usecase *usecase = NULL;
int ret = 0;
unsigned int param_value = 0;
- struct stream_in *in = adev->active_input;
+ struct stream_in *in = adev_get_active_input(adev);
if(!voice_extn_is_dynamic_ecns_enabled())
return ENOSYS;
@@ -1030,13 +1049,16 @@
if(!voice_extn_is_dynamic_ecns_enabled())
return;
- if (adev->active_input->enable_aec) {
- enable_disable_effect(adev, EFFECT_AEC, true);
- }
+ struct stream_in *in = adev_get_active_input(adev);
- if (adev->active_input->enable_ns &&
- adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
- enable_disable_effect(adev, EFFECT_NS, true);
+ if (in != NULL && !in->standby) {
+ if (in->enable_aec)
+ enable_disable_effect(adev, EFFECT_AEC, true);
+
+ if (in->enable_ns &&
+ in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+ enable_disable_effect(adev, EFFECT_NS, true);
+ }
}
}
@@ -1059,6 +1081,7 @@
snd_device_t snd_device;
char mixer_path[MIXER_PATH_MAX_LENGTH];
struct stream_out *out = NULL;
+ struct stream_in *in = NULL;
int ret = 0;
if (usecase == NULL)
@@ -1066,10 +1089,40 @@
ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
- if (usecase->type == PCM_CAPTURE || usecase->type == TRANSCODE_LOOPBACK_TX)
+ if (usecase->type == PCM_CAPTURE) {
+ struct stream_in *in = usecase->stream.in;
+ struct audio_usecase *uinfo;
snd_device = usecase->in_snd_device;
- else
+
+ if (in) {
+ if (in->enable_aec || in->enable_ec_port) {
+ audio_devices_t out_device = AUDIO_DEVICE_OUT_SPEAKER;
+ struct listnode *node;
+ struct audio_usecase *voip_usecase = get_usecase_from_list(adev,
+ USECASE_AUDIO_PLAYBACK_VOIP);
+ if (voip_usecase) {
+ out_device = voip_usecase->stream.out->devices;
+ } else if (adev->primary_output &&
+ !adev->primary_output->standby) {
+ out_device = adev->primary_output->devices;
+ } else {
+ list_for_each(node, &adev->usecase_list) {
+ uinfo = node_to_item(node, struct audio_usecase, list);
+ if (uinfo->type != PCM_CAPTURE) {
+ out_device = uinfo->stream.out->devices;
+ break;
+ }
+ }
+ }
+ platform_set_echo_reference(adev, true, out_device);
+ in->ec_opened = true;
+ }
+ }
+ } else if (usecase->type == TRANSCODE_LOOPBACK_TX) {
+ snd_device = usecase->in_snd_device;
+ } else {
snd_device = usecase->out_snd_device;
+ }
#ifdef DS1_DOLBY_DAP_ENABLED
audio_extn_dolby_set_dmid(adev);
@@ -1087,7 +1140,16 @@
if (out && out->compr)
audio_extn_utils_compress_set_clk_rec_mode(usecase);
}
- audio_extn_set_custom_mtmx_params(adev, usecase, true);
+
+ if (usecase->type == PCM_CAPTURE) {
+ in = usecase->stream.in;
+ if (in && is_loopback_input_device(in->device)) {
+ ALOGD("%s: set custom mtmx params v1", __func__);
+ audio_extn_set_custom_mtmx_params_v1(adev, usecase, true);
+ }
+ } else {
+ audio_extn_set_custom_mtmx_params_v2(adev, usecase, true);
+ }
// we shouldn't truncate mixer_path
ALOGW_IF(strlcpy(mixer_path, use_case_table[usecase->id], sizeof(mixer_path))
@@ -1112,6 +1174,7 @@
{
snd_device_t snd_device;
char mixer_path[MIXER_PATH_MAX_LENGTH];
+ struct stream_in *in = NULL;
if (usecase == NULL || usecase->id == USECASE_INVALID)
return -EINVAL;
@@ -1128,12 +1191,30 @@
platform_add_backend_name(mixer_path, snd_device, usecase);
ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path);
audio_route_reset_and_update_path(adev->audio_route, mixer_path);
+ if (usecase->type == PCM_CAPTURE) {
+ struct stream_in *in = usecase->stream.in;
+ if (in && in->ec_opened) {
+ platform_set_echo_reference(in->dev, false, AUDIO_DEVICE_NONE);
+ in->ec_opened = false;
+ }
+ }
audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_FREE);
audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_FREE);
- audio_extn_set_custom_mtmx_params(adev, usecase, false);
+
+ if (usecase->type == PCM_CAPTURE) {
+ in = usecase->stream.in;
+ if (in && is_loopback_input_device(in->device)) {
+ ALOGD("%s: reset custom mtmx params v1", __func__);
+ audio_extn_set_custom_mtmx_params_v1(adev, usecase, false);
+ }
+ } else {
+ audio_extn_set_custom_mtmx_params_v2(adev, usecase, false);
+ }
+
if ((usecase->type == PCM_PLAYBACK) &&
(usecase->stream.out != NULL))
usecase->stream.out->pspd_coeff_sent = false;
+
ALOGV("%s: exit", __func__);
return 0;
}
@@ -1243,7 +1324,7 @@
}
if (((snd_device == SND_DEVICE_IN_HANDSET_6MIC) ||
(snd_device == SND_DEVICE_IN_HANDSET_QMIC)) &&
- (audio_extn_ffv_get_stream() == adev->active_input)) {
+ (audio_extn_ffv_get_stream() == adev_get_active_input(adev))) {
ALOGD("%s: init ec ref loopback", __func__);
audio_extn_ffv_init_ec_ref_loopback(adev, snd_device);
}
@@ -1325,7 +1406,7 @@
audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
} else if (((snd_device == SND_DEVICE_IN_HANDSET_6MIC) ||
(snd_device == SND_DEVICE_IN_HANDSET_QMIC)) &&
- (audio_extn_ffv_get_stream() == adev->active_input)) {
+ (audio_extn_ffv_get_stream() == adev_get_active_input(adev))) {
ALOGD("%s: deinit ec ref loopback", __func__);
audio_extn_ffv_deinit_ec_ref_loopback(adev, snd_device);
}
@@ -1618,8 +1699,10 @@
/* Update voc calibration before enabling VoIP route */
if (usecase->type == VOIP_CALL)
status = platform_switch_voice_call_device_post(adev->platform,
- usecase->out_snd_device,
- platform_get_input_snd_device(adev->platform, uc_info->devices));
+ usecase->out_snd_device,
+ platform_get_input_snd_device(
+ adev->platform, NULL,
+ uc_info->devices));
enable_audio_route(adev, usecase);
if (usecase->stream.out && usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
out_set_voip_volume(&usecase->stream.out->stream,
@@ -1974,19 +2057,6 @@
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
*/
@@ -2220,6 +2290,97 @@
int out_standby_l(struct audio_stream *stream);
+struct stream_in *adev_get_active_input(const struct audio_device *adev)
+{
+ struct listnode *node;
+ struct stream_in *last_active_in = NULL;
+
+ /* Get last added active input.
+ * TODO: We may use a priority mechanism to pick highest priority active source */
+ list_for_each(node, &adev->usecase_list)
+ {
+ struct audio_usecase *usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == PCM_CAPTURE && usecase->stream.in != NULL)
+ last_active_in = usecase->stream.in;
+ }
+
+ return last_active_in;
+}
+
+struct stream_in *get_voice_communication_input(const struct audio_device *adev)
+{
+ struct listnode *node;
+
+ /* First check active inputs with voice communication source and then
+ * any input if audio mode is in communication */
+ list_for_each(node, &adev->usecase_list)
+ {
+ struct audio_usecase *usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == PCM_CAPTURE && usecase->stream.in != NULL &&
+ usecase->stream.in->source == AUDIO_SOURCE_VOICE_COMMUNICATION)
+ return usecase->stream.in;
+ }
+ if (adev->mode == AUDIO_MODE_IN_COMMUNICATION)
+ return adev_get_active_input(adev);
+
+ return NULL;
+}
+
+/*
+ * Aligned with policy.h
+ */
+static inline int source_priority(int inputSource)
+{
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ return 9;
+ case AUDIO_SOURCE_CAMCORDER:
+ return 8;
+ case AUDIO_SOURCE_VOICE_PERFORMANCE:
+ return 7;
+ case AUDIO_SOURCE_UNPROCESSED:
+ return 6;
+ case AUDIO_SOURCE_MIC:
+ return 5;
+ case AUDIO_SOURCE_ECHO_REFERENCE:
+ return 4;
+ case AUDIO_SOURCE_FM_TUNER:
+ return 3;
+ case AUDIO_SOURCE_VOICE_RECOGNITION:
+ return 2;
+ case AUDIO_SOURCE_HOTWORD:
+ return 1;
+ default:
+ break;
+ }
+ return 0;
+}
+
+static struct stream_in *get_priority_input(struct audio_device *adev)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+ int last_priority = 0, priority;
+ struct stream_in *priority_in = NULL;
+ struct stream_in *in;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == PCM_CAPTURE) {
+ in = usecase->stream.in;
+ if (!in)
+ continue;
+ priority = source_priority(in->source);
+
+ if (priority > last_priority) {
+ last_priority = priority;
+ priority_in = in;
+ }
+ }
+ }
+ return priority_in;
+}
+
int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
{
snd_device_t out_snd_device = SND_DEVICE_NONE;
@@ -2249,7 +2410,9 @@
}
out_snd_device = platform_get_output_snd_device(adev->platform,
usecase->stream.out);
- in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices);
+ in_snd_device = platform_get_input_snd_device(adev->platform,
+ NULL,
+ usecase->stream.out->devices);
usecase->devices = usecase->stream.out->devices;
} else if (usecase->type == TRANSCODE_LOOPBACK_RX) {
if (usecase->stream.inout == NULL) {
@@ -2268,7 +2431,7 @@
ALOGE("%s: stream.inout is NULL", __func__);
return -EINVAL;
}
- in_snd_device = platform_get_input_snd_device(adev->platform, AUDIO_DEVICE_NONE);
+ in_snd_device = platform_get_input_snd_device(adev->platform, NULL, AUDIO_DEVICE_NONE);
usecase->devices = in_snd_device;
} else {
/*
@@ -2328,21 +2491,17 @@
usecase->devices = usecase->stream.out->devices;
in_snd_device = SND_DEVICE_NONE;
if (out_snd_device == SND_DEVICE_NONE) {
+ struct stream_out *voip_out = adev->primary_output;
+ struct stream_in *voip_in = get_voice_communication_input(adev);
out_snd_device = platform_get_output_snd_device(adev->platform,
- usecase->stream.out);
+ usecase->stream.out);
voip_usecase = get_usecase_from_list(adev, USECASE_AUDIO_PLAYBACK_VOIP);
- if (voip_usecase == NULL && adev->primary_output && !adev->primary_output->standby)
- voip_usecase = get_usecase_from_list(adev, adev->primary_output->usecase);
- if ((usecase->stream.out != NULL &&
- voip_usecase != NULL &&
- usecase->stream.out->usecase == voip_usecase->id) &&
- adev->active_input &&
- (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
- adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
- out_snd_device != usecase->out_snd_device) {
- select_devices(adev, adev->active_input->usecase);
- }
+ if (voip_usecase)
+ voip_out = voip_usecase->stream.out;
+
+ if (usecase->stream.out == voip_out && voip_in != NULL)
+ select_devices(adev, voip_in->usecase);
}
} else if (usecase->type == PCM_CAPTURE) {
if (usecase->stream.in == NULL) {
@@ -2353,24 +2512,39 @@
out_snd_device = SND_DEVICE_NONE;
if (in_snd_device == SND_DEVICE_NONE) {
audio_devices_t out_device = AUDIO_DEVICE_NONE;
- if (adev->active_input &&
- (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
- (adev->mode == AUDIO_MODE_IN_COMMUNICATION &&
- adev->active_input->source == AUDIO_SOURCE_MIC))) {
- voip_usecase = get_usecase_from_list(adev, USECASE_AUDIO_PLAYBACK_VOIP);
- if (voip_usecase != NULL && voip_usecase->stream.out != NULL)
+ struct stream_in *voip_in = get_voice_communication_input(adev);
+ struct stream_in *priority_in = NULL;
+
+ if (voip_in != NULL) {
+ struct audio_usecase *voip_usecase = get_usecase_from_list(adev,
+ USECASE_AUDIO_PLAYBACK_VOIP);
+
+ usecase->stream.in->enable_ec_port = false;
+
+ if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
+ out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
+ } else if (voip_usecase) {
out_device = voip_usecase->stream.out->devices;
- else if (adev->primary_output && !adev->primary_output->standby)
+ } else if (adev->primary_output &&
+ !adev->primary_output->standby) {
out_device = adev->primary_output->devices;
- platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
- } else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
- out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
+ } else {
+ /* forcing speaker o/p device to get matching i/p pair
+ in case o/p is not routed from same primary HAL */
+ out_device = AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ priority_in = voip_in;
} else {
- /* forcing speaker o/p device to get matching i/p pair
- in case o/p is not routed from same primary HAL */
- out_device = AUDIO_DEVICE_OUT_SPEAKER;
+ /* get the input with the highest priority source*/
+ priority_in = get_priority_input(adev);
+
+ if (!priority_in)
+ priority_in = usecase->stream.in;
}
- in_snd_device = platform_get_input_snd_device(adev->platform, out_device);
+
+ in_snd_device = platform_get_input_snd_device(adev->platform,
+ priority_in,
+ out_device);
}
}
}
@@ -2524,22 +2698,23 @@
(usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
}
+ }
+ enable_audio_route(adev, usecase);
- /* Notify device change info to effect clients registered */
+ /* Notify device change info to effect clients registered */
+ if (usecase->type == PCM_PLAYBACK) {
audio_extn_gef_notify_device_config(
usecase->stream.out->devices,
usecase->stream.out->channel_mask,
usecase->stream.out->app_type_cfg.sample_rate,
platform_get_snd_device_acdb_id(usecase->out_snd_device));
}
- enable_audio_route(adev, usecase);
audio_extn_qdsp_set_device(usecase);
/* If input stream is already running then effect needs to be
applied on the new input device that's being enabled here. */
- if ((in_snd_device != SND_DEVICE_NONE) && (adev->active_input != NULL) &&
- (!adev->active_input->standby))
+ if (in_snd_device != SND_DEVICE_NONE)
check_and_enable_effect(adev);
if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
@@ -2563,14 +2738,13 @@
in_snd_device);
if (is_btsco_device(out_snd_device, in_snd_device) || is_a2dp_device(out_snd_device)) {
-
+ struct stream_in *in = adev_get_active_input(adev);
if (usecase->type == VOIP_CALL) {
- if (adev->active_input != NULL &&
- !adev->active_input->standby) {
+ if (in != NULL && !in->standby) {
if (is_bt_soc_on(adev) == false){
ALOGD("BT SCO MIC disconnected while in connection");
- if (adev->active_input->pcm != NULL)
- pcm_stop(adev->active_input->pcm);
+ if (in->pcm != NULL)
+ pcm_stop(in->pcm);
}
}
if ((usecase->stream.out != NULL) && (usecase->stream.out != adev->primary_output)
@@ -2615,6 +2789,7 @@
}
struct audio_device *adev = in->dev;
+ struct stream_in *priority_in = NULL;
ALOGV("%s: enter: usecase(%d: %s)", __func__,
in->usecase, use_case_table[in->usecase]);
@@ -2625,6 +2800,8 @@
return -EINVAL;
}
+ priority_in = get_priority_input(adev);
+
if (audio_extn_ext_hw_plugin_usecase_stop(adev->ext_hw_plugin, uc_info))
ALOGE("%s: failed to stop ext hw plugin", __func__);
@@ -2637,10 +2814,18 @@
/* 2. Disable the tx device */
disable_snd_device(adev, uc_info->in_snd_device);
+ if (is_loopback_input_device(in->device))
+ audio_extn_keep_alive_stop(KEEP_ALIVE_OUT_PRIMARY);
+
list_remove(&uc_info->list);
free(uc_info);
- adev->active_input = get_next_active_input(adev);
+ if (priority_in == in) {
+ priority_in = get_priority_input(adev);
+ if (priority_in)
+ select_devices(adev, priority_in->usecase);
+ }
+
enable_gcov();
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
@@ -2702,7 +2887,6 @@
goto error_config;
}
- adev->active_input = in;
uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
if (!uc_info) {
@@ -2727,8 +2911,10 @@
if (audio_extn_ext_hw_plugin_usecase_start(adev->ext_hw_plugin, uc_info))
ALOGE("%s: failed to start ext hw plugin", __func__);
+ android_atomic_acquire_cas(true, false, &(in->capture_stopped));
+
if (audio_extn_cin_attached_usecase(in->usecase)) {
- ret = audio_extn_cin_start_input_stream(in);
+ ret = audio_extn_cin_open_input_stream(in);
if (ret)
goto error_open;
else
@@ -2820,6 +3006,9 @@
audio_extn_audiozoom_set_microphone_direction(in, in->zoom);
audio_extn_audiozoom_set_microphone_field_dimension(in, in->direction);
+ if (is_loopback_input_device(in->device))
+ audio_extn_keep_alive_start(KEEP_ALIVE_OUT_PRIMARY);
+
done_open:
audio_streaming_hint_end();
audio_extn_perf_lock_release(&adev->perf_lock_handle);
@@ -2833,7 +3022,6 @@
stop_input_stream(in);
error_config:
- adev->active_input = get_next_active_input(adev);
/*
* sleep 50ms to allow sufficient time for kernel
* drivers to recover incases like SSR.
@@ -3509,6 +3697,8 @@
out->usecase == USECASE_AUDIO_PLAYBACK_ULL) && (out->apply_volume)) {
out_set_pcm_volume(&out->stream, out->volume_l, out->volume_r);
out->apply_volume = false;
+ } else if (audio_extn_auto_hal_is_bus_device_usecase(out->usecase)) {
+ out_set_pcm_volume(&out->stream, out->volume_l, out->volume_r);
}
} else {
platform_set_stream_channel_map(adev->platform, out->channel_mask,
@@ -3571,6 +3761,22 @@
adev->offload_effects_start_output(out->handle, out->pcm_device_id, adev->mixer);
audio_extn_check_and_set_dts_hpx_state(adev);
}
+
+ if (out->devices & AUDIO_DEVICE_OUT_BUS) {
+ /* Update cached volume from media to offload/direct stream */
+ struct listnode *node = NULL;
+ list_for_each(node, &adev->active_outputs_list) {
+ streams_output_ctxt_t *out_ctxt = node_to_item(node,
+ streams_output_ctxt_t,
+ list);
+ if (out_ctxt->output->usecase == USECASE_AUDIO_PLAYBACK_MEDIA) {
+ out->volume_l = out_ctxt->output->volume_l;
+ out->volume_r = out_ctxt->output->volume_r;
+ }
+ }
+ out_set_compr_volume(&out->stream,
+ out->volume_l, out->volume_r);
+ }
}
if (ret == 0) {
@@ -3660,6 +3866,9 @@
case 4:
case 6:
case 8:
+ case 10:
+ case 12:
+ case 14:
break;
default:
ret = -EINVAL;
@@ -3811,10 +4020,10 @@
is_low_latency);
}
-static size_t get_output_period_size(uint32_t sample_rate,
- audio_format_t format,
- int channel_count,
- int duration /*in millisecs*/)
+size_t get_output_period_size(uint32_t sample_rate,
+ audio_format_t format,
+ int channel_count,
+ int duration /*in millisecs*/)
{
size_t size = 0;
uint32_t bytes_per_sample = audio_bytes_per_sample(format);
@@ -3904,8 +4113,6 @@
return out->compr_config.fragment_size;
} else if(out->usecase == USECASE_COMPRESS_VOIP_CALL)
return voice_extn_compress_voip_out_get_buffer_size(out);
- else if(out->usecase == USECASE_AUDIO_PLAYBACK_VOIP)
- return VOIP_IO_BUF_SIZE(out->config.rate, DEFAULT_VOIP_BUF_DURATION_MS, DEFAULT_VOIP_BIT_DEPTH_BYTE);
else if (is_offload_usecase(out->usecase) &&
out->flags == AUDIO_OUTPUT_FLAG_DIRECT)
return out->hal_fragment_size;
@@ -4955,6 +5162,13 @@
out->volume_l = left;
out->volume_r = right;
return ret;
+ } else if (audio_extn_auto_hal_is_bus_device_usecase(out->usecase)) {
+ ALOGV("%s: bus device set volume called", __func__);
+ if (!out->standby)
+ ret = out_set_pcm_volume(stream, left, right);
+ out->volume_l = left;
+ out->volume_r = right;
+ return ret;
}
return -ENOSYS;
@@ -5216,7 +5430,7 @@
*/
usecase = get_usecase_from_list(adev, out->usecase);
if (usecase != NULL) {
- audio_extn_set_custom_mtmx_params(adev, usecase, true);
+ audio_extn_set_custom_mtmx_params_v2(adev, usecase, true);
out->pspd_coeff_sent = true;
}
}
@@ -5957,8 +6171,6 @@
if(in->usecase == USECASE_COMPRESS_VOIP_CALL)
return voice_extn_compress_voip_in_get_buffer_size(in);
- else if(in->usecase == USECASE_AUDIO_RECORD_VOIP)
- return VOIP_IO_BUF_SIZE(in->config.rate, DEFAULT_VOIP_BUF_DURATION_MS, DEFAULT_VOIP_BIT_DEPTH_BYTE);
else if(audio_extn_compr_cap_usecase_supported(in->usecase))
return audio_extn_compr_cap_get_buffer_size(in->config.format);
else if(audio_extn_cin_attached_usecase(in->usecase))
@@ -6020,7 +6232,7 @@
in->capture_started = false;
} else {
if (audio_extn_cin_attached_usecase(in->usecase))
- audio_extn_cin_stop_input_stream(in);
+ audio_extn_cin_close_input_stream(in);
}
if (in->pcm) {
@@ -6030,11 +6242,11 @@
in->pcm = NULL;
}
- if(do_stop) {
+ if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION)
adev->enable_voicerx = false;
- platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
+
+ if (do_stop)
status = stop_input_stream(in);
- }
if (in->source == AUDIO_SOURCE_VOICE_RECOGNITION)
adev->num_va_sessions--;
@@ -6324,6 +6536,13 @@
in->standby = 0;
}
+ /* Avoid read if capture_stopped is set */
+ if (android_atomic_acquire_load(&(in->capture_stopped)) > 0) {
+ ALOGD("%s: force stopped catpure session, ignoring read request", __func__);
+ ret = -EINVAL;
+ goto exit;
+ }
+
// what's the duration requested by the client?
long ns = 0;
@@ -6448,6 +6667,51 @@
return ret;
}
+static int in_update_effect_list(bool add, effect_handle_t effect,
+ struct listnode *head)
+{
+ struct listnode *node;
+ struct in_effect_list *elist = NULL;
+ struct in_effect_list *target = NULL;
+ int ret = 0;
+
+ if (!head)
+ return ret;
+
+ list_for_each(node, head) {
+ elist = node_to_item(node, struct in_effect_list, list);
+ if (elist->handle == effect) {
+ target = elist;
+ break;
+ }
+ }
+
+ if (add) {
+ if (target) {
+ ALOGD("effect %p already exist", effect);
+ return ret;
+ }
+
+ target = (struct in_effect_list *)
+ calloc(1, sizeof(struct in_effect_list));
+
+ if (!target) {
+ ALOGE("%s:fail to allocate memory", __func__);
+ return -ENOMEM;
+ }
+
+ target->handle = effect;
+ list_add_tail(head, &target->list);
+ } else {
+ if (target) {
+ list_remove(&target->list);
+ free(target);
+ }
+ }
+
+ return ret;
+}
+
static int add_remove_audio_effect(const struct audio_stream *stream,
effect_handle_t effect,
bool enable)
@@ -6465,13 +6729,18 @@
lock_input_stream(in);
pthread_mutex_lock(&in->dev->lock);
if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
- in->source == AUDIO_SOURCE_VOICE_RECOGNITION ||
- in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
- in->enable_aec != enable &&
+ in->source == AUDIO_SOURCE_VOICE_RECOGNITION ||
+ adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
(memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) {
+
+ in_update_effect_list(enable, effect, &in->aec_list);
+ enable = !list_empty(&in->aec_list);
+ if (enable == in->enable_aec)
+ goto exit;
+
in->enable_aec = enable;
- if (!enable)
- platform_set_echo_reference(in->dev, enable, AUDIO_DEVICE_NONE);
+ ALOGD("AEC enable %d", enable);
+
if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) {
in->dev->enable_voicerx = enable;
@@ -6489,9 +6758,15 @@
}
}
- if (in->enable_ns != enable &&
- (memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0)) {
+ if (memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0) {
+
+ in_update_effect_list(enable, effect, &in->ns_list);
+ enable = !list_empty(&in->ns_list);
+ if (enable == in->enable_ns)
+ goto exit;
+
in->enable_ns = enable;
+ ALOGD("NS enable %d", enable);
if (!in->standby) {
if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) {
@@ -6501,6 +6776,7 @@
select_devices(in->dev, in->usecase);
}
}
+exit:
pthread_mutex_unlock(&in->dev->lock);
pthread_mutex_unlock(&in->lock);
@@ -6828,7 +7104,7 @@
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
- const char *address __unused)
+ const char *address)
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_out *out;
@@ -6859,8 +7135,8 @@
out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
ALOGD("%s: enter: format(%#x) sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)\
- stream_handle(%p)", __func__, config->format, config->sample_rate, config->channel_mask,
- devices, flags, &out->stream);
+ stream_handle(%p) address(%s)", __func__, config->format, config->sample_rate, config->channel_mask,
+ devices, flags, &out->stream, address);
if (!out) {
@@ -6983,6 +7259,23 @@
out->config.format = pcm_format_from_audio_format(out->format);
}
+ /* validate bus device address */
+ if (out->devices & AUDIO_DEVICE_OUT_BUS) {
+ /* extract car audio stream index */
+ out->car_audio_stream =
+ audio_extn_auto_hal_get_car_audio_stream_from_address(address);
+ if (out->car_audio_stream < 0) {
+ ALOGE("%s: invalid car audio stream %x",
+ __func__, out->car_audio_stream);
+ ret = -EINVAL;
+ goto error_open;
+ }
+ /* save car audio stream and address for bus device */
+ strlcpy(out->address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN);
+ ALOGV("%s: address %s, car_audio_stream %x",
+ __func__, out->address, out->car_audio_stream);
+ }
+
/* Check for VOIP usecase */
if (is_voip_rx) {
if (!voice_extn_is_compress_voip_supported()) {
@@ -6996,8 +7289,17 @@
out->volume_r = INVALID_OUT_VOLUME;
out->config = default_pcm_config_voip_copp;
- out->config.period_size = VOIP_IO_BUF_SIZE(out->sample_rate, DEFAULT_VOIP_BUF_DURATION_MS, DEFAULT_VOIP_BIT_DEPTH_BYTE)/2;
out->config.rate = out->sample_rate;
+ uint32_t channel_count =
+ audio_channel_count_from_out_mask(out->channel_mask);
+ uint32_t buffer_size = get_stream_buffer_size(DEFAULT_VOIP_BUF_DURATION_MS,
+ out->sample_rate, out->format,
+ channel_count, false);
+ uint32_t frame_size = audio_bytes_per_sample(out->format) * channel_count;
+ if (frame_size != 0)
+ out->config.period_size = buffer_size / frame_size;
+ else
+ ALOGW("%s: frame size is 0 for format %#x", __func__, out->format);
}
} else {
if ((out->dev->mode == AUDIO_MODE_IN_COMMUNICATION ||
@@ -7089,6 +7391,11 @@
ALOGV("non-offload DIRECT_usecase ... usecase selected %d ", out->usecase);
}
+ if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
+ ALOGD("%s: Setting latency mode to true", __func__);
+ out->compr_config.codec->flags |= audio_extn_utils_get_perf_mode_flag();
+ }
+
if (out->usecase == USECASE_INVALID) {
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
config->format == 0 && config->sample_rate == 0 &&
@@ -7172,6 +7479,10 @@
out->compr_config.fragments = DIRECT_PCM_NUM_FRAGMENTS;
+ if ((config->offload_info.duration_us >= MIN_OFFLOAD_BUFFER_DURATION_MS * 1000) &&
+ (config->offload_info.duration_us <= MAX_OFFLOAD_BUFFER_DURATION_MS * 1000))
+ out->info.duration_us = (int64_t)config->offload_info.duration_us;
+
/* Check if alsa session is configured with the same format as HAL input format,
* if not then derive correct fragment size needed to accomodate the
* conversion of HAL input format to alsa format.
@@ -7456,6 +7767,13 @@
adev->haptics_config.channels = 1;
} else
adev->haptics_config.channels = audio_channel_count_from_out_mask(out->channel_mask & AUDIO_CHANNEL_HAPTIC_ALL);
+ } else if (out->devices & AUDIO_DEVICE_OUT_BUS) {
+ ret = audio_extn_auto_hal_open_output_stream(out);
+ if (ret) {
+ ALOGE("%s: Failed to open output stream for bus device", __func__);
+ ret = -EINVAL;
+ goto error_open;
+ }
} else {
/* primary path is the default path selected if no other outputs are available/suitable */
out->usecase = GET_USECASE_AUDIO_PLAYBACK_PRIMARY(use_db_as_primary);
@@ -7781,6 +8099,7 @@
adev->screen_off = false;
else
adev->screen_off = true;
+ audio_extn_sound_trigger_update_screen_status(adev->screen_off);
}
ret = str_parms_get_int(parms, "rotation", &val);
@@ -7883,6 +8202,21 @@
adev->allow_afe_proxy_usage = true;
}
}
+ if (audio_is_a2dp_out_device(device)) {
+ struct audio_usecase *usecase;
+ struct listnode *node;
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (PCM_PLAYBACK == usecase->type && usecase->stream.out &&
+ (usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+ usecase->stream.out->a2dp_compress_mute) {
+ struct stream_out *out = usecase->stream.out;
+ ALOGD("Unmuting the stream when Bt-A2dp disconnected and stream is mute");
+ out->a2dp_compress_mute = false;
+ out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
+ }
+ }
+ }
}
audio_extn_hfp_set_parameters(adev, parms);
@@ -7897,13 +8231,17 @@
if (usecase->stream.out && (usecase->type == PCM_PLAYBACK) &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP)){
ALOGD("reconfigure a2dp... forcing device switch");
-
pthread_mutex_unlock(&adev->lock);
lock_output_stream(usecase->stream.out);
pthread_mutex_lock(&adev->lock);
audio_extn_a2dp_set_handoff_mode(true);
+ ALOGD("Switching to speaker and muting the stream before select_devices");
+ check_a2dp_restore_l(adev, usecase->stream.out, false);
//force device switch to re configure encoder
select_devices(adev, usecase->id);
+ ALOGD("Unmuting the stream after select_devices");
+ usecase->stream.out->a2dp_compress_mute = false;
+ out_set_compr_volume(&usecase->stream.out->stream, usecase->stream.out->volume_l, usecase->stream.out->volume_r);
audio_extn_a2dp_set_handoff_mode(false);
pthread_mutex_unlock(&usecase->stream.out->lock);
break;
@@ -8324,6 +8662,8 @@
in->af_period_multiplier = 1;
in->direction = MIC_DIRECTION_UNSPECIFIED;
in->zoom = 0;
+ list_init(&in->aec_list);
+ list_init(&in->ns_list);
ALOGV("%s: source %d, config->channel_mask %#x", __func__, source, config->channel_mask);
if (source == AUDIO_SOURCE_VOICE_UPLINK ||
@@ -8409,6 +8749,8 @@
}
if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
+ (flags & AUDIO_INPUT_FLAG_TIMESTAMP) == 0 &&
+ (flags & AUDIO_INPUT_FLAG_COMPRESS) == 0 &&
(flags & AUDIO_INPUT_FLAG_FAST) != 0) {
is_low_latency = true;
#if LOW_LATENCY_CAPTURE_USE_CASE
@@ -8529,7 +8871,7 @@
(in->dev->mode != AUDIO_MODE_IN_COMMUNICATION)) {
audio_extn_compr_cap_init(in);
} else if (audio_extn_cin_applicable_stream(in)) {
- ret = audio_extn_cin_configure_input_stream(in);
+ ret = audio_extn_cin_configure_input_stream(in, config);
if (ret)
goto err_open;
} else {
@@ -8580,7 +8922,7 @@
ALOGV("%s: overriding usecase with USECASE_AUDIO_RECORD_COMPRESS2 and appending compress flag", __func__);
if (audio_extn_cin_applicable_stream(in)) {
in->sample_rate = config->sample_rate;
- ret = audio_extn_cin_configure_input_stream(in);
+ ret = audio_extn_cin_configure_input_stream(in, config);
if (ret)
goto err_open;
}
@@ -8664,7 +9006,7 @@
/* Disable echo reference if there are no active input, hfp call
* and sound trigger while closing input stream
*/
- if (!adev->active_input &&
+ if (adev_get_active_input(adev) == NULL &&
!audio_extn_hfp_is_active(adev) &&
!audio_extn_sound_trigger_check_ec_ref_enable())
platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
@@ -8707,7 +9049,7 @@
audio_extn_compr_cap_deinit();
if (audio_extn_cin_attached_usecase(in->usecase))
- audio_extn_cin_close_input_stream(in);
+ audio_extn_cin_free_input_stream_resources(in);
if (in->is_st_session) {
ALOGV("%s: sound trigger pcm stop lab", __func__);
@@ -8812,13 +9154,10 @@
uc_info.id = audio_usecase;
uc_info.type = usecase_type;
if (dir) {
- adev->active_input = ∈
memset(&in, 0, sizeof(in));
in.device = audio_device;
in.source = AUDIO_SOURCE_VOICE_COMMUNICATION;
uc_info.stream.in = ∈
- } else {
- adev->active_input = NULL;
}
memset(&out, 0, sizeof(out));
out.devices = audio_device; /* only field needed in select_devices */
@@ -8853,7 +9192,6 @@
list_remove(&uc_info.list);
}
}
- adev->active_input = NULL; /* restore adev state */
return 0;
}
@@ -8893,13 +9231,21 @@
int adev_get_audio_port(struct audio_hw_device *dev, struct audio_port *config)
{
- return audio_extn_hw_loopback_get_audio_port(dev, config);
+ int ret = 0;
+
+ ret = audio_extn_hw_loopback_get_audio_port(dev, config);
+ ret |= audio_extn_auto_hal_get_audio_port(dev, config);
+ return ret;
}
int adev_set_audio_port_config(struct audio_hw_device *dev,
const struct audio_port_config *config)
{
- return audio_extn_hw_loopback_set_audio_port_config(dev, config);
+ int ret = 0;
+
+ ret = audio_extn_hw_loopback_set_audio_port_config(dev, config);
+ ret |= audio_extn_auto_hal_set_audio_port_config(dev, config);
+ return ret;
}
static int adev_dump(const audio_hw_device_t *device __unused,
@@ -8911,9 +9257,9 @@
static int adev_close(hw_device_t *device)
{
size_t i;
- struct audio_device *adev = (struct audio_device *)device;
+ struct audio_device *adev_temp = (struct audio_device *)device;
- if (!adev)
+ if (!adev_temp)
return 0;
pthread_mutex_lock(&adev_init_lock);
@@ -8929,6 +9275,8 @@
audio_extn_utils_release_streams_cfg_lists(
&adev->streams_output_cfg_list,
&adev->streams_input_cfg_list);
+ if (audio_extn_qap_is_enabled())
+ audio_extn_qap_deinit();
if (audio_extn_qaf_is_enabled())
audio_extn_qaf_deinit();
audio_route_free(adev->audio_route);
@@ -9004,6 +9352,7 @@
adev->card_status = status;
platform_snd_card_update(adev->platform, status);
audio_extn_fm_set_parameters(adev, parms);
+ audio_extn_auto_hal_set_parameters(adev, parms);
} else if (is_ext_device_status) {
platform_set_parameters(adev->platform, parms);
}
@@ -9038,7 +9387,7 @@
select_devices(adev, uc_info->id);
pthread_mutex_lock(&out->compr_mute_lock);
if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
- (out->a2dp_compress_mute)) {
+ (out->a2dp_compress_mute) && (uc_info->out_snd_device == SND_DEVICE_OUT_BT_A2DP)) {
out->a2dp_compress_mute = false;
out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
}
@@ -9167,7 +9516,6 @@
/* Set the default route before the PCM stream is opened */
adev->mode = AUDIO_MODE_NORMAL;
- adev->active_input = NULL;
adev->primary_output = NULL;
adev->out_device = AUDIO_DEVICE_NONE;
adev->bluetooth_nrec = true;
@@ -9209,6 +9557,20 @@
}
adev->extspk = audio_extn_extspk_init(adev);
+ if (audio_extn_qap_is_enabled()) {
+ ret = audio_extn_qap_init(adev);
+ if (ret < 0) {
+ pthread_mutex_destroy(&adev->lock);
+ free(adev);
+ adev = NULL;
+ ALOGE("%s: Failed to init platform data, aborting.", __func__);
+ *device = NULL;
+ pthread_mutex_unlock(&adev_init_lock);
+ return ret;
+ }
+ adev->device.open_output_stream = audio_extn_qap_open_output_stream;
+ adev->device.close_output_stream = audio_extn_qap_close_output_stream;
+ }
if (audio_extn_qaf_is_enabled()) {
ret = audio_extn_qaf_init(adev);
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 9a93ed7..4810896 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -222,6 +222,12 @@
USECASE_AUDIO_A2DP_ABR_FEEDBACK,
+ /* car streams usecases */
+ USECASE_AUDIO_PLAYBACK_MEDIA,
+ USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION,
+ USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE,
+ USECASE_AUDIO_PLAYBACK_PHONE,
+
AUDIO_USECASE_MAX
};
@@ -290,6 +296,21 @@
RENDER_MODE_AUDIO_STC_MASTER,
} render_mode_t;
+#ifdef AUDIO_EXTN_AUTO_HAL_ENABLED
+/* This defines the physical car streams supported in audio HAL,
+ * limited by the available frontend PCM driver.
+ * Max number of physical streams supported is currently 8 and is
+ * represented by stream bit flag as indicated in vehicle HAL interface.
+ */
+#define MAX_CAR_AUDIO_STREAMS 8
+enum {
+ CAR_AUDIO_STREAM_MEDIA = 0x1,
+ CAR_AUDIO_STREAM_SYS_NOTIFICATION = 0x2,
+ CAR_AUDIO_STREAM_NAV_GUIDANCE = 0x4,
+ CAR_AUDIO_STREAM_PHONE = 0x8,
+};
+#endif
+
struct stream_app_type_cfg {
int sample_rate;
uint32_t bit_width;
@@ -378,6 +399,7 @@
card_status_t card_status;
void* qaf_stream_handle;
+ void* qap_stream_handle;
pthread_cond_t qaf_offload_cond;
pthread_t qaf_offload_thread;
struct listnode qaf_offload_cmd_list;
@@ -407,6 +429,9 @@
error_log_t *error_log;
bool pspd_coeff_sent;
+
+ char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
+ int car_audio_stream;
};
struct stream_in {
@@ -424,6 +449,10 @@
bool enable_aec;
bool enable_ns;
audio_format_t format;
+ bool enable_ec_port;
+ bool ec_opened;
+ struct listnode aec_list;
+ struct listnode ns_list;
int64_t mmap_time_offset_nanos; /* fudge factor to correct inaccuracies in DSP */
audio_io_handle_t capture_handle;
audio_input_flags_t flags;
@@ -444,6 +473,8 @@
float zoom;
audio_microphone_direction_t direction;
+ volatile int32_t capture_stopped;
+
/* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
audio_format_t supported_formats[MAX_SUPPORTED_FORMATS + 1];
@@ -541,7 +572,6 @@
struct mixer *mixer;
audio_mode_t mode;
audio_devices_t out_device;
- struct stream_in *active_input;
struct stream_out *primary_output;
struct stream_out *voice_tx_output;
struct stream_out *current_call_output;
@@ -660,8 +690,6 @@
struct audio_usecase *get_usecase_from_list(const struct audio_device *adev,
audio_usecase_t uc_id);
-struct stream_in *get_next_active_input(const struct audio_device *adev);
-
bool is_offload_usecase(audio_usecase_t uc_id);
bool audio_is_true_native_stream_active(struct audio_device *adev);
@@ -694,11 +722,24 @@
streams_output_ctxt_t *out_get_stream(struct audio_device *dev,
audio_io_handle_t output);
+size_t get_output_period_size(uint32_t sample_rate,
+ audio_format_t format,
+ int channel_count,
+ int duration /*in millisecs*/);
+
#define LITERAL_TO_STRING(x) #x
#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
__FILE__ ":" LITERAL_TO_STRING(__LINE__)\
" ASSERT_FATAL(" #condition ") failed.")
+static inline bool is_loopback_input_device(audio_devices_t device) {
+ if (!audio_is_output_device(device) &&
+ ((device & AUDIO_DEVICE_IN_LOOPBACK) == AUDIO_DEVICE_IN_LOOPBACK))
+ return true;
+ else
+ return false;
+}
+
/*
* NOTE: when multiple mutexes have to be acquired, always take the
* stream_in or stream_out mutex first, followed by the audio_device mutex.
diff --git a/hal/audio_hw_extn_api.c b/hal/audio_hw_extn_api.c
index af73375..22c8685 100644
--- a/hal/audio_hw_extn_api.c
+++ b/hal/audio_hw_extn_api.c
@@ -34,6 +34,7 @@
#include <inttypes.h>
#include <errno.h>
#include <log/log.h>
+#include <cutils/atomic.h>
#include <hardware/audio.h>
#include "sound/compress_params.h"
@@ -190,6 +191,31 @@
return ret;
}
+int qahwi_in_stop(struct audio_stream_in* stream) {
+ struct stream_in *in = (struct stream_in *)stream;
+ struct audio_device *adev = in->dev;
+
+ ALOGD("%s processing, in %p", __func__, in);
+
+ pthread_mutex_lock(&adev->lock);
+
+ if (!in->standby) {
+ if (in->pcm != NULL ) {
+ pcm_stop(in->pcm);
+ } else if (audio_extn_cin_attached_usecase(in->usecase)) {
+ audio_extn_cin_stop_input_stream(in);
+ }
+
+ /* Set the atomic variable when the session is stopped */
+ if (android_atomic_acquire_cas(false, true, &(in->capture_stopped)) == 0)
+ ALOGI("%s: capture_stopped bit set", __func__);
+ }
+
+ pthread_mutex_unlock(&adev->lock);
+
+ return 0;
+}
+
ssize_t qahwi_in_read_v2(struct audio_stream_in *stream, void* buffer,
size_t bytes, uint64_t *timestamp)
{
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index bdd10b0..8b9b53d 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -2845,6 +2845,20 @@
}
}
+struct audio_custom_mtmx_in_params *platform_get_custom_mtmx_in_params(void *platform,
+ struct audio_custom_mtmx_in_params_info *info)
+{
+ ALOGW("%s: not implemented!", __func__);
+ return -ENOSYS;
+}
+
+int platform_add_custom_mtmx_in_params(void *platform,
+ struct audio_custom_mtmx_in_params_info *info)
+{
+ ALOGW("%s: not implemented!", __func__);
+ return -ENOSYS;
+}
+
void platform_release_acdb_metainfo_key(void *platform)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -4163,6 +4177,7 @@
bool use_voip_out_devices = false;
bool prop_rec_play_enabled = false;
char recConcPropValue[PROPERTY_VALUE_MAX];
+ struct stream_in *in = adev_get_active_input(adev);
if (property_get("vendor.audio.rec.playback.conc.disabled", recConcPropValue, NULL)) {
prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
@@ -4171,8 +4186,8 @@
(my_data->rec_play_conc_set || adev->mode == AUDIO_MODE_IN_COMMUNICATION);
ALOGV("platform_get_output_snd_device use_voip_out_devices : %d",use_voip_out_devices);
- audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
- AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
+ audio_channel_mask_t channel_mask = (in == NULL) ?
+ AUDIO_CHANNEL_IN_MONO : in->channel_mask;
int channel_count = popcount(channel_mask);
ALOGV("%s: enter: output devices(%#x)", __func__, devices);
@@ -4191,7 +4206,7 @@
* enforced audible (e.g. Camera shutter sound).
*/
if ((mode == AUDIO_MODE_IN_CALL) ||
- voice_is_in_call(adev) ||
+ voice_check_voicecall_usecases_active(adev) ||
voice_extn_compress_voip_is_active(adev))
is_active_voice_call = true;
@@ -4274,7 +4289,7 @@
}
if ((mode == AUDIO_MODE_IN_CALL) ||
- voice_is_in_call(adev) ||
+ voice_check_voicecall_usecases_active(adev) ||
voice_extn_compress_voip_is_active(adev)) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
@@ -4513,9 +4528,9 @@
{
struct audio_device *adev = my_data->adev;
snd_device_t snd_device = SND_DEVICE_NONE;
+ struct stream_in *in = adev_get_active_input(adev);
- if (adev->active_input->enable_aec &&
- adev->active_input->enable_ns) {
+ if (in != NULL && in->enable_aec && in->enable_ns) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
if (my_data->fluence_in_spkr_mode) {
if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
@@ -4547,7 +4562,7 @@
snd_device = SND_DEVICE_IN_USB_HEADSET_MIC_AEC;
}
platform_set_echo_reference(adev, true, out_device);
- } else if (adev->active_input->enable_aec) {
+ } else if (in != NULL && in->enable_aec) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
if (my_data->fluence_in_spkr_mode) {
if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
@@ -4579,7 +4594,7 @@
snd_device = SND_DEVICE_IN_USB_HEADSET_MIC_AEC;
}
platform_set_echo_reference(adev, true, out_device);
- } else if (adev->active_input->enable_ns) {
+ } else if (in != NULL && in->enable_ns) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
if (my_data->fluence_in_spkr_mode) {
if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
@@ -4625,33 +4640,35 @@
return get_snd_device_for_voice_comm_ecns_disabled(my_data, out_device, in_device);
}
-snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device)
+snd_device_t platform_get_input_snd_device(void *platform,
+ struct stream_in *in,
+ audio_devices_t out_device)
{
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
- /*
- * TODO: active_input always points to last opened input. Source returned will
- * be wrong if more than one active inputs are present.
- */
- audio_source_t source = (adev->active_input == NULL) ?
- AUDIO_SOURCE_DEFAULT : adev->active_input->source;
-
- audio_mode_t mode = adev->mode;
- audio_devices_t in_device = ((adev->active_input == NULL) ?
- AUDIO_DEVICE_NONE : adev->active_input->device)
- & ~AUDIO_DEVICE_BIT_IN;
- audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
- AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
+ audio_mode_t mode = adev->mode;
snd_device_t snd_device = SND_DEVICE_NONE;
- int channel_count = popcount(channel_mask);
- int str_bitwidth = (adev->active_input == NULL) ?
- CODEC_BACKEND_DEFAULT_BIT_WIDTH : adev->active_input->bit_width;
+
+ if (in == NULL) {
+ in = adev_get_active_input(adev);
+ }
+
+ audio_source_t source = (in == NULL) ? AUDIO_SOURCE_DEFAULT : in->source;
+ audio_devices_t in_device =
+ ((in == NULL) ? AUDIO_DEVICE_NONE : in->device) & ~AUDIO_DEVICE_BIT_IN;
+ audio_channel_mask_t channel_mask = (in == NULL) ? AUDIO_CHANNEL_IN_MONO : in->channel_mask;
+ int channel_count = audio_channel_count_from_in_mask(channel_mask);
+
+ int str_bitwidth = (in == NULL) ?
+ CODEC_BACKEND_DEFAULT_BIT_WIDTH : in->bit_width;
ALOGV("%s: enter: out_device(%#x) in_device(%#x) channel_count (%d) channel_mask (0x%x)",
__func__, out_device, in_device, channel_count, channel_mask);
if (my_data->external_mic) {
- if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) || voice_is_in_call(adev) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_check_voicecall_usecases_active(adev) ||
+ voice_extn_compress_voip_is_active(adev) ||
+ audio_extn_hfp_is_active(adev))) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
out_device & AUDIO_DEVICE_OUT_EARPIECE ||
out_device & AUDIO_DEVICE_OUT_SPEAKER )
@@ -4665,8 +4682,10 @@
if (snd_device != AUDIO_DEVICE_NONE)
goto exit;
- if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) || voice_is_in_call(adev) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_check_voicecall_usecases_active(adev) ||
+ voice_extn_compress_voip_is_active(adev) ||
+ audio_extn_hfp_is_active(adev))) {
if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
!voice_extn_compress_voip_is_active(adev)) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
@@ -4834,7 +4853,7 @@
snd_device = SND_DEVICE_IN_QUAD_MIC;
}
if (snd_device == SND_DEVICE_NONE) {
- if (adev->active_input->enable_ns)
+ if (in != NULL && in->enable_ns)
snd_device = SND_DEVICE_IN_VOICE_REC_MIC_NS;
else
snd_device = SND_DEVICE_IN_VOICE_REC_MIC;
@@ -4876,16 +4895,15 @@
in_device = ((out_device == AUDIO_DEVICE_NONE) ?
AUDIO_DEVICE_IN_BUILTIN_MIC : in_device) & ~AUDIO_DEVICE_BIT_IN;
- if (adev->active_input) {
+ if (in)
snd_device = get_snd_device_for_voice_comm(my_data, out_device, in_device);
- }
} else if (source == AUDIO_SOURCE_MIC) {
if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC &&
channel_count == 1) {
if(my_data->fluence_in_audio_rec) {
if ((my_data->fluence_type & FLUENCE_HEX_MIC) &&
(my_data->source_mic_type & SOURCE_HEX_MIC) &&
- (audio_extn_ffv_get_stream() == adev->active_input)) {
+ (audio_extn_ffv_get_stream() == in)) {
snd_device = audio_extn_ffv_get_capture_snd_device();
} else if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
(my_data->source_mic_type & SOURCE_QUAD_MIC)) {
@@ -4907,7 +4925,7 @@
goto exit;
}
- if (adev->active_input && (audio_extn_ssr_get_stream() == adev->active_input))
+ if (in && (audio_extn_ssr_get_stream() == in))
snd_device = SND_DEVICE_IN_THREE_MIC;
if (snd_device != SND_DEVICE_NONE) {
@@ -4918,7 +4936,7 @@
!(in_device & AUDIO_DEVICE_IN_VOICE_CALL) &&
!(in_device & AUDIO_DEVICE_IN_COMMUNICATION)) {
if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- if (adev->active_input && (audio_extn_ssr_get_stream() == adev->active_input))
+ if (in && (audio_extn_ssr_get_stream() == in))
snd_device = SND_DEVICE_IN_QUAD_MIC;
else if ((my_data->fluence_type & (FLUENCE_DUAL_MIC | FLUENCE_TRI_MIC | FLUENCE_QUAD_MIC)) &&
(channel_count == 2) && (my_data->source_mic_type & SOURCE_DUAL_MIC))
@@ -5243,9 +5261,9 @@
goto done_key_audcal;
}
- if(cal.dev_id) {
- if(audio_is_input_device(cal.dev_id)) {
- cal.snd_dev_id = platform_get_input_snd_device(platform, cal.dev_id);
+ if (cal.dev_id) {
+ if (audio_is_input_device(cal.dev_id)) {
+ cal.snd_dev_id = platform_get_input_snd_device(platform, NULL, cal.dev_id);
} else {
out.devices = cal.dev_id;
out.sample_rate = cal.sampling_rate;
@@ -5569,7 +5587,7 @@
}
if(cal.dev_id & AUDIO_DEVICE_BIT_IN) {
- cal.snd_dev_id = platform_get_input_snd_device(platform, cal.dev_id);
+ cal.snd_dev_id = platform_get_input_snd_device(platform, NULL, cal.dev_id);
} else if(cal.dev_id) {
out.devices = cal.dev_id;
out.sample_rate = cal.sampling_rate;
@@ -8530,7 +8548,7 @@
size_t actual_mic_count = 0;
snd_device_t active_input_snd_device =
- platform_get_input_snd_device(platform, usecase->stream.in->device);
+ platform_get_input_snd_device(platform, usecase->stream.in, AUDIO_DEVICE_NONE);
if (active_input_snd_device == SND_DEVICE_NONE) {
ALOGI("%s: No active microphones found", __func__);
goto end;
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index e337870..3c94532 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2018, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2019, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -246,6 +246,11 @@
SND_DEVICE_IN_INCALL_REC_TX,
SND_DEVICE_IN_INCALL_REC_RX_TX,
SND_DEVICE_IN_LINE,
+ SND_DEVICE_IN_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_DMIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_QMIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_6MIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_8MIC_AND_EC_REF_LOOPBACK,
SND_DEVICE_IN_END,
SND_DEVICE_MAX = SND_DEVICE_IN_END,
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 0a81969..137e700 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -374,6 +374,20 @@
return -ENOSYS;
}
+struct audio_custom_mtmx_in_params *platform_get_custom_mtmx_in_params(void *platform,
+ struct audio_custom_mtmx_in_params_info *info)
+{
+ ALOGW("%s: not implemented!", __func__);
+ return -ENOSYS;
+}
+
+int platform_add_custom_mtmx_in_params(void *platform,
+ struct audio_custom_mtmx_in_params_info *info)
+{
+ ALOGW("%s: not implemented!", __func__);
+ return -ENOSYS;
+}
+
void platform_deinit(void *platform)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -536,7 +550,7 @@
voice_is_in_call_rec_stream(usecase->stream.in))
snd_device = voice_get_incall_rec_snd_device(usecase->in_snd_device);
else if ((usecase->type == PCM_HFP_CALL) || (usecase->type == PCM_CAPTURE))
- snd_device = platform_get_input_snd_device(adev->platform,
+ snd_device = platform_get_input_snd_device(adev->platform, NULL,
adev->primary_output->devices);
acdb_dev_id = acdb_device_table[snd_device];
if (acdb_dev_id < 0) {
@@ -794,21 +808,23 @@
return snd_device;
}
-snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device)
+snd_device_t platform_get_input_snd_device(void *platform,
+ struct stream_in *in,
+ audio_devices_t out_device)
{
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
- audio_source_t source = (adev->active_input == NULL) ?
- AUDIO_SOURCE_DEFAULT : adev->active_input->source;
-
- audio_mode_t mode = adev->mode;
- audio_devices_t in_device = ((adev->active_input == NULL) ?
- AUDIO_DEVICE_NONE : adev->active_input->device)
- & ~AUDIO_DEVICE_BIT_IN;
- audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
- AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
+ audio_mode_t mode = adev->mode;
snd_device_t snd_device = SND_DEVICE_NONE;
+ if (in == NULL)
+ in = adev_get_active_input(adev);
+
+ audio_source_t source = (in == NULL) ? AUDIO_SOURCE_DEFAULT : in->source;
+ audio_devices_t in_device =
+ ((in == NULL) ? AUDIO_DEVICE_NONE : in->device) & ~AUDIO_DEVICE_BIT_IN;
+ audio_channel_mask_t channel_mask = (in == NULL) ? AUDIO_CHANNEL_IN_MONO : in->channel_mask;
+
ALOGV("%s: enter: out_device(%#x) in_device(%#x)",
__func__, out_device, in_device);
if ((out_device != AUDIO_DEVICE_NONE) && (mode == AUDIO_MODE_IN_CALL)) {
@@ -882,8 +898,8 @@
} else if (source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
if (out_device & AUDIO_DEVICE_OUT_SPEAKER)
in_device = AUDIO_DEVICE_IN_BACK_MIC;
- if (adev->active_input) {
- if (adev->active_input->enable_aec) {
+ if (in) {
+ if (in->enable_aec) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
snd_device = SND_DEVICE_IN_SPEAKER_MIC_AEC;
} else if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index 727f906..2c66208 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013 - 2018 The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013 - 2019 The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -104,6 +104,11 @@
SND_DEVICE_IN_VOICE_REC_DMIC,
SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
SND_DEVICE_IN_USB_HEADSET_MIC,
+ SND_DEVICE_IN_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_DMIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_QMIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_6MIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_8MIC_AND_EC_REF_LOOPBACK,
SND_DEVICE_IN_END,
SND_DEVICE_MAX = SND_DEVICE_IN_END,
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index e77ab83..7dfa819 100755
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -688,6 +688,9 @@
} else if (!strncmp(snd_card_name, "sm6150-idp-snd-card",
sizeof("sm6150-idp-snd-card"))) {
strlcpy(hw_info->name, "sm6150", sizeof(hw_info->name));
+ } else if (!strncmp(snd_card_name, "sm6150-wcd9375-snd-card",
+ sizeof("sm6150-wcd9375-snd-card"))) {
+ strlcpy(hw_info->name, "sm6150", sizeof(hw_info->name));
} else if (!strncmp(snd_card_name, "sm6150-qrd-snd-card",
sizeof("sm6150-qrd-snd-card"))) {
hw_info->is_stereo_spkr = false;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 9ee58cb..1b14d63 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -350,6 +350,7 @@
struct spkr_device_chmap *spkr_ch_map;
bool use_sprk_default_sample_rate;
struct listnode custom_mtmx_params_list;
+ struct listnode custom_mtmx_in_params_list;
};
struct spkr_device_chmap {
@@ -469,6 +470,15 @@
[USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM8] =
{PLAYBACK_INTERACTIVE_STRM_DEVICE8, PLAYBACK_INTERACTIVE_STRM_DEVICE8},
[USECASE_AUDIO_EC_REF_LOOPBACK] = {-1, -1}, /* pcm id updated from platform info file */
+ [USECASE_AUDIO_PLAYBACK_MEDIA] = {MEDIA_PCM_DEVICE,
+ MEDIA_PCM_DEVICE},
+ [USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION] = {SYS_NOTIFICATION_PCM_DEVICE,
+ SYS_NOTIFICATION_PCM_DEVICE},
+ [USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE] = {NAV_GUIDANCE_PCM_DEVICE,
+ NAV_GUIDANCE_PCM_DEVICE},
+ [USECASE_AUDIO_PLAYBACK_PHONE] = {PHONE_PCM_DEVICE,
+ PHONE_PCM_DEVICE},
+
};
/* Array to store sound devices */
@@ -517,7 +527,9 @@
[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP] = "speaker-safe-and-bt-a2dp",
[SND_DEVICE_OUT_VOICE_HANDSET_TMUS] = "voice-handset-tmus",
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
+ [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADSET] = "voice-tty-full-headset",
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
+ [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADSET] = "voice-tty-vco-headset",
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
[SND_DEVICE_OUT_VOICE_TTY_FULL_USB] = "voice-tty-full-usb",
[SND_DEVICE_OUT_VOICE_TTY_VCO_USB] = "voice-tty-vco-usb",
@@ -565,6 +577,10 @@
[SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB] = "wsa-speaker-and-bt-sco-wb",
[SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB] = "wsa-speaker-and-bt-sco-wb",
[SND_DEVICE_OUT_VOICE_HEARING_AID] = "hearing-aid",
+ [SND_DEVICE_OUT_BUS_MEDIA] = "bus-speaker",
+ [SND_DEVICE_OUT_BUS_SYS] = "bus-speaker",
+ [SND_DEVICE_OUT_BUS_NAV] = "bus-speaker",
+ [SND_DEVICE_OUT_BUS_PHN] = "bus-speaker",
/* Capture sound devices */
[SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
@@ -697,6 +713,12 @@
[SND_DEVICE_OUT_VOIP_SPEAKER] = "voip-speaker",
[SND_DEVICE_OUT_VOIP_HEADPHONES] = "voip-headphones",
[SND_DEVICE_IN_VOICE_HEARING_AID] = "hearing-aid-mic",
+ [SND_DEVICE_IN_BUS] = "bus-mic",
+ [SND_DEVICE_IN_EC_REF_LOOPBACK] = "ec-ref-loopback",
+ [SND_DEVICE_IN_HANDSET_DMIC_AND_EC_REF_LOOPBACK] = "handset-dmic-and-ec-ref-loopback",
+ [SND_DEVICE_IN_HANDSET_QMIC_AND_EC_REF_LOOPBACK] = "handset-qmic-and-ec-ref-loopback",
+ [SND_DEVICE_IN_HANDSET_6MIC_AND_EC_REF_LOOPBACK] = "handset-6mic-and-ec-ref-loopback",
+ [SND_DEVICE_IN_HANDSET_8MIC_AND_EC_REF_LOOPBACK] = "handset-8mic-and-ec-ref-loopback",
};
// Platform specific backend bit width table
@@ -786,7 +808,9 @@
[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP] = 14,
[SND_DEVICE_OUT_VOICE_HANDSET_TMUS] = 88,
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
+ [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADSET] = 17,
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
+ [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADSET] = 17,
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
[SND_DEVICE_OUT_VOICE_TTY_FULL_USB] = 17,
[SND_DEVICE_OUT_VOICE_TTY_VCO_USB] = 17,
@@ -819,6 +843,10 @@
[SND_DEVICE_OUT_SPEAKER_PROTECTED_RAS] = 134,
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT_RAS] = 134,
[SND_DEVICE_OUT_VOICE_HEARING_AID] = 45,
+ [SND_DEVICE_OUT_BUS_MEDIA] = 78,
+ [SND_DEVICE_OUT_BUS_SYS] = 78,
+ [SND_DEVICE_OUT_BUS_NAV] = 14,
+ [SND_DEVICE_OUT_BUS_PHN] = 94,
[SND_DEVICE_IN_HANDSET_MIC] = 4,
[SND_DEVICE_IN_HANDSET_MIC_SB] = 163,
[SND_DEVICE_IN_HANDSET_MIC_EXTERNAL] = 4,
@@ -946,6 +974,7 @@
[SND_DEVICE_IN_CAMCORDER_SELFIE_INVERT_LANDSCAPE] = 4,
[SND_DEVICE_IN_CAMCORDER_SELFIE_PORTRAIT] = 4,
[SND_DEVICE_IN_VOICE_HEARING_AID] = 44,
+ [SND_DEVICE_IN_BUS] = 11,
};
struct name_to_index {
@@ -977,6 +1006,7 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_STEREO)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_WSA)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_HFP)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
@@ -1002,7 +1032,9 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET_TMUS)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HAC_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_SCO)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB)},
@@ -1042,6 +1074,10 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_SPEAKER)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEARING_AID)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BUS_MEDIA)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BUS_SYS)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BUS_NAV)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BUS_PHN)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_SB)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_EXTERNAL)},
@@ -1171,6 +1207,12 @@
{TO_NAME_INDEX(SND_DEVICE_IN_CAMCORDER_SELFIE_PORTRAIT)},
/* For legacy xml file parsing */
{TO_NAME_INDEX(SND_DEVICE_IN_CAMCORDER_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_BUS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_EC_REF_LOOPBACK)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_AND_EC_REF_LOOPBACK)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_QMIC_AND_EC_REF_LOOPBACK)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_6MIC_AND_EC_REF_LOOPBACK)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_8MIC_AND_EC_REF_LOOPBACK)},
};
static char * backend_tag_table[SND_DEVICE_MAX] = {0};
@@ -1232,7 +1274,10 @@
{TO_NAME_INDEX(USECASE_AUDIO_RECORD_VOIP)},
{TO_NAME_INDEX(USECASE_AUDIO_TRANSCODE_LOOPBACK_RX)},
{TO_NAME_INDEX(USECASE_AUDIO_TRANSCODE_LOOPBACK_TX)},
-
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_MEDIA)},
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION)},
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE)},
+ {TO_NAME_INDEX(USECASE_AUDIO_PLAYBACK_PHONE)},
};
static const struct name_to_index usecase_type_index[USECASE_TYPE_MAX] = {
@@ -1620,8 +1665,10 @@
!strncmp(snd_card_name, "sdm439-snd-card-mtp",
sizeof("sdm439-snd-card-mtp")) ||
!strncmp(snd_card_name, "sm6150-wcd9375qrd-snd-card",
- sizeof("sm6150-wcd9375qrd-snd-card"))) {
- ALOGI("%s: snd_card_name: %s", __func__, snd_card_name);
+ sizeof("sm6150-wcd9375qrd-snd-card")) ||
+ !strncmp(snd_card_name, "sm6150-wcd9375-snd-card",
+ sizeof("sm6150-wcd9375-snd-card"))) {
+ ALOGI("%s: snd_card_name: %s",__func__,snd_card_name);
my_data->is_internal_codec = true;
my_data->is_slimbus_interface = false;
}
@@ -1652,12 +1699,12 @@
{
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
- if (adev->active_input == NULL) {
+ struct stream_in *in = adev_get_active_input(adev);
+ if (in == NULL)
return base;
- }
- unsigned int sr = adev->active_input->sample_rate;
- unsigned int ch = popcount(adev->active_input->channel_mask);
- unsigned int bit_width = adev->active_input->bit_width;
+ unsigned int sr = in->sample_rate;
+ unsigned int ch = popcount(in->channel_mask);
+ unsigned int bit_width = in->bit_width;
if (audio_extn_usb_is_config_supported(&bit_width, &sr, &ch, false)
&& ((ch == 6) || (ch == 8))) {
return other;
@@ -2088,6 +2135,7 @@
hw_interface_table[SND_DEVICE_OUT_VOICE_HANDSET] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_HAC_HANDSET] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER] = strdup("SLIMBUS_0_RX");
+ hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_STEREO] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_2] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("SLIMBUS_0_RX");
@@ -2108,7 +2156,9 @@
strdup("SLIMBUS_0_RX-and-SLIMBUS_7_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_HANDSET_TMUS] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = strdup("SLIMBUS_6_RX");
+ hw_interface_table[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADSET] = strdup("SLIMBUS_6_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = strdup("SLIMBUS_6_RX");
+ hw_interface_table[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADSET] = strdup("SLIMBUS_6_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_TTY_FULL_USB] = strdup("USB_AUDIO_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_TTY_VCO_USB] = strdup("USB_AUDIO_RX");
@@ -2151,6 +2201,10 @@
Need snd_device to route voice call and use specific acdb tuning.
Also, BT_RX is a virtual port to indicate bluetooth hearing aid. */
hw_interface_table[SND_DEVICE_OUT_VOICE_HEARING_AID] = strdup("BT_RX"),
+ hw_interface_table[SND_DEVICE_OUT_BUS_MEDIA] = strdup("TERT_TDM_RX_0");
+ hw_interface_table[SND_DEVICE_OUT_BUS_SYS] = strdup("TERT_TDM_RX_0");
+ hw_interface_table[SND_DEVICE_OUT_BUS_NAV] = strdup("TERT_TDM_RX_1");
+ hw_interface_table[SND_DEVICE_OUT_BUS_PHN] = strdup("TERT_TDM_RX_2");
hw_interface_table[SND_DEVICE_IN_HANDSET_MIC] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_HANDSET_MIC_SB] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_HANDSET_MIC_EXTERNAL] = strdup("SLIMBUS_0_TX");
@@ -2277,6 +2331,7 @@
hw_interface_table[SND_DEVICE_IN_CAMCORDER_SELFIE_INVERT_LANDSCAPE] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_CAMCORDER_SELFIE_PORTRAIT] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_VOICE_HEARING_AID] = strdup("SLIMBUS_0_TX");
+ hw_interface_table[SND_DEVICE_IN_BUS] = strdup("TERT_TDM_TX_0");
my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
/*remove ALAC & APE from DSP decoder list based on software decoder availability*/
@@ -3051,11 +3106,11 @@
}
}
/* Check for Ambisonic Capture Enablement */
- if (property_get_bool("persist.vendor.audio.ambisonic.capture",false))
+ if (property_get_bool("vendor.audio.ambisonic.capture",false))
my_data->ambisonic_capture = true;
/* Check for Ambisonic Profile Assignment*/
- if (property_get_bool("persist.vendor.audio.ambisonic.auto.profile",false))
+ if (property_get_bool("vendor.audio.ambisonic.auto.profile",false))
my_data->ambisonic_profile = true;
if (check_and_get_wsa_info((char *)snd_card_name, &wsaCount, &is_wsa_combo_supported)
@@ -3097,6 +3152,7 @@
list_init(&my_data->acdb_meta_key_list);
list_init(&my_data->custom_mtmx_params_list);
+ list_init(&my_data->custom_mtmx_in_params_list);
ret = audio_extn_is_hifi_audio_supported();
if (ret || !my_data->is_internal_codec)
@@ -3691,6 +3747,66 @@
}
}
+struct audio_custom_mtmx_in_params *platform_get_custom_mtmx_in_params(void *platform,
+ struct audio_custom_mtmx_in_params_info *info)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ struct listnode *node = NULL;
+ struct audio_custom_mtmx_in_params *params = NULL;
+
+ list_for_each(node, &my_data->custom_mtmx_in_params_list) {
+ params = node_to_item(node, struct audio_custom_mtmx_in_params, list);
+ if (params &&
+ params->in_info.op_channels == info->op_channels &&
+ params->in_info.usecase_id == info->usecase_id) {
+ ALOGV("%s: found params with op_ch %d uc_id %d",
+ __func__, info->op_channels, info->usecase_id);
+ return params;
+ }
+ }
+
+ ALOGI("%s: no matching param with op_ch %d uc_id %d",
+ __func__, info->op_channels, info->usecase_id);
+ return NULL;
+}
+
+int platform_add_custom_mtmx_in_params(void *platform,
+ struct audio_custom_mtmx_in_params_info *info)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ struct audio_custom_mtmx_in_params *params = NULL;
+ uint32_t size = sizeof(*params);
+
+ if (info->op_channels > AUDIO_CHANNEL_COUNT_MAX) {
+ ALOGE("%s: unusupported channels in %d", __func__, info->op_channels);
+ return -EINVAL;
+ }
+
+ params = (struct audio_custom_mtmx_in_params *)calloc(1, size);
+ if (!params) {
+ ALOGE("%s: failed to add custom mtmx in params", __func__);
+ return -ENOMEM;
+ }
+
+ ALOGI("%s: adding mtmx in params with op_ch %d uc_id %d",
+ __func__, info->op_channels, info->usecase_id);
+
+ params->in_info = *info;
+ list_add_tail(&my_data->custom_mtmx_in_params_list, ¶ms->list);
+ return 0;
+}
+
+static void platform_release_custom_mtmx_in_params(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ struct listnode *node = NULL, *tempnode = NULL;
+
+ list_for_each_safe(node, tempnode, &my_data->custom_mtmx_in_params_list) {
+ list_remove(node);
+ free(node_to_item(node, struct audio_custom_mtmx_in_params, list));
+ }
+}
+
void platform_release_acdb_metainfo_key(void *platform)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -3840,6 +3956,7 @@
/* free acdb_meta_key_list */
platform_release_acdb_metainfo_key(platform);
platform_release_custom_mtmx_params(platform);
+ platform_release_custom_mtmx_in_params(platform);
if (my_data->acdb_deallocate)
my_data->acdb_deallocate();
@@ -4697,6 +4814,9 @@
else if (strncmp(backend_tag_table[snd_device], "headphones",
sizeof("headphones")) == 0)
port = HEADPHONE_BACKEND;
+ else if (strncmp(backend_tag_table[snd_device], "headset",
+ sizeof("headset")) == 0)
+ port = HEADPHONE_BACKEND;
else if (strcmp(backend_tag_table[snd_device], "hdmi") == 0)
port = HDMI_RX_BACKEND;
else if (strcmp(backend_tag_table[snd_device], "display-port") == 0)
@@ -4845,6 +4965,7 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_STEREO ||
out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
@@ -5377,8 +5498,29 @@
new_snd_devices[0] = SND_DEVICE_IN_INCALL_REC_RX;
new_snd_devices[1] = SND_DEVICE_IN_INCALL_REC_TX;
ret = 0;
+ } else if (SND_DEVICE_IN_HANDSET_DMIC_AND_EC_REF_LOOPBACK == snd_device) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_IN_HANDSET_DMIC;
+ new_snd_devices[1] = SND_DEVICE_IN_EC_REF_LOOPBACK;
+ ret = 0;
+ } else if (SND_DEVICE_IN_HANDSET_QMIC_AND_EC_REF_LOOPBACK == snd_device) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_IN_UNPROCESSED_QUAD_MIC;
+ new_snd_devices[1] = SND_DEVICE_IN_EC_REF_LOOPBACK;
+ ret = 0;
+ } else if (SND_DEVICE_IN_HANDSET_6MIC_AND_EC_REF_LOOPBACK == snd_device) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_IN_HANDSET_6MIC;
+ new_snd_devices[1] = SND_DEVICE_IN_EC_REF_LOOPBACK;
+ ret = 0;
+ } else if (SND_DEVICE_IN_HANDSET_8MIC_AND_EC_REF_LOOPBACK == snd_device) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_IN_HANDSET_8MIC;
+ new_snd_devices[1] = SND_DEVICE_IN_EC_REF_LOOPBACK;
+ ret = 0;
}
+
ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
snd_device, *num_devices, *new_snd_devices);
@@ -5430,9 +5572,10 @@
audio_devices_t devices = out->devices;
unsigned int sample_rate = out->sample_rate;
int na_mode = platform_get_native_support();
+ struct stream_in *in = adev_get_active_input(adev);
- audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
- AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
+ audio_channel_mask_t channel_mask = (in == NULL) ?
+ AUDIO_CHANNEL_IN_MONO : in->channel_mask;
int channel_count = popcount(channel_mask);
ALOGV("%s: enter: output devices(%#x)", __func__, devices);
@@ -5451,7 +5594,7 @@
* enforced audible (e.g. Camera shutter sound).
*/
if ((mode == AUDIO_MODE_IN_CALL) ||
- voice_is_in_call(adev) ||
+ voice_check_voicecall_usecases_active(adev) ||
voice_extn_compress_voip_is_active(adev))
is_active_voice_call = true;
@@ -5575,7 +5718,7 @@
}
if ((mode == AUDIO_MODE_IN_CALL) ||
- voice_is_in_call(adev) ||
+ voice_check_voicecall_usecases_active(adev) ||
voice_extn_compress_voip_is_active(adev) ||
adev->enable_voicerx ||
audio_extn_hfp_is_active(adev)) {
@@ -5586,10 +5729,22 @@
!voice_extn_compress_voip_is_active(adev)) {
switch (adev->voice.tty_mode) {
case TTY_MODE_FULL:
- snd_device = SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES;
+ if (audio_extn_is_concurrent_capture_enabled() &&
+ (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET)) {
+ //Separate backend is added for headset-mic as part of concurrent capture
+ snd_device = SND_DEVICE_OUT_VOICE_TTY_FULL_HEADSET;
+ } else {
+ snd_device = SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES;
+ }
break;
case TTY_MODE_VCO:
- snd_device = SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES;
+ if (audio_extn_is_concurrent_capture_enabled() &&
+ (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET)) {
+ //Separate backend is added for headset-mic as part of concurrent capture
+ snd_device = SND_DEVICE_OUT_VOICE_TTY_VCO_HEADSET;
+ } else {
+ snd_device = SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES;
+ }
break;
case TTY_MODE_HCO:
snd_device = SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET;
@@ -5818,6 +5973,8 @@
ALOGD("%s: setting sink capability(%d) for Proxy", __func__, channel_count);
snd_device = SND_DEVICE_OUT_AFE_PROXY;
audio_extn_set_afe_proxy_channel_mixer(adev, channel_count, snd_device);
+ } else if (devices & AUDIO_DEVICE_OUT_BUS) {
+ snd_device = audio_extn_auto_hal_get_snd_device_for_car_audio_stream(out);
} else {
ALOGE("%s: Unknown device(s) %#x", __func__, devices);
}
@@ -5827,7 +5984,8 @@
}
static snd_device_t get_snd_device_for_voice_comm_ecns_enabled(struct platform_data *my_data,
- audio_devices_t out_device,
+ struct stream_in *in,
+ audio_devices_t out_device __unused,
audio_devices_t in_device)
{
struct audio_device *adev = my_data->adev;
@@ -5873,20 +6031,20 @@
} else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_HEADSET_MIC;
}
- platform_set_echo_reference(adev, true, out_device);
+ in->enable_ec_port = true;
return snd_device;
}
static snd_device_t get_snd_device_for_voice_comm_ecns_disabled(struct platform_data *my_data,
+ struct stream_in *in,
audio_devices_t out_device,
audio_devices_t in_device)
{
struct audio_device *adev = my_data->adev;
snd_device_t snd_device = SND_DEVICE_NONE;
- if (adev->active_input->enable_aec &&
- adev->active_input->enable_ns) {
+ if (in != NULL && in->enable_aec && in->enable_ns) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
if (my_data->fluence_in_spkr_mode) {
if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
@@ -5931,8 +6089,7 @@
} else if (audio_extn_usb_connected(NULL) && audio_is_usb_in_device(in_device | AUDIO_DEVICE_BIT_IN)) {
snd_device = SND_DEVICE_IN_USB_HEADSET_MIC_AEC;
}
- platform_set_echo_reference(adev, true, out_device);
- } else if (adev->active_input->enable_aec) {
+ } else if (in != NULL && in->enable_aec) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
if (my_data->fluence_in_spkr_mode) {
if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
@@ -5975,8 +6132,7 @@
} else if (audio_extn_usb_connected(NULL) && audio_is_usb_in_device(in_device | AUDIO_DEVICE_BIT_IN)) {
snd_device = SND_DEVICE_IN_USB_HEADSET_MIC_AEC;
}
- platform_set_echo_reference(adev, true, out_device);
- } else if (adev->active_input->enable_ns) {
+ } else if (in != NULL && in->enable_ns) {
if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
if (my_data->fluence_in_spkr_mode) {
if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
@@ -6025,42 +6181,44 @@
}
static snd_device_t get_snd_device_for_voice_comm(struct platform_data *my_data,
+ struct stream_in *in,
audio_devices_t out_device,
audio_devices_t in_device)
{
if(voice_extn_is_dynamic_ecns_enabled())
- return get_snd_device_for_voice_comm_ecns_enabled(my_data, out_device, in_device);
+ return get_snd_device_for_voice_comm_ecns_enabled(my_data, in, out_device, in_device);
else
- return get_snd_device_for_voice_comm_ecns_disabled(my_data, out_device, in_device);
+ return get_snd_device_for_voice_comm_ecns_disabled(my_data, in, out_device, in_device);
}
-snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device)
+snd_device_t platform_get_input_snd_device(void *platform,
+ struct stream_in *in,
+ audio_devices_t out_device)
{
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
- /*
- * TODO: active_input always points to last opened input. Source returned will
- * be wrong if more than one active inputs are present.
- */
- audio_source_t source = (adev->active_input == NULL) ?
- AUDIO_SOURCE_DEFAULT : adev->active_input->source;
-
- audio_mode_t mode = adev->mode;
- audio_devices_t in_device = ((adev->active_input == NULL) ?
- AUDIO_DEVICE_NONE : adev->active_input->device)
- & ~AUDIO_DEVICE_BIT_IN;
- audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
- AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
+ audio_mode_t mode = adev->mode;
snd_device_t snd_device = SND_DEVICE_NONE;
- int channel_count = popcount(channel_mask);
- int str_bitwidth = (adev->active_input == NULL) ?
- CODEC_BACKEND_DEFAULT_BIT_WIDTH : adev->active_input->bit_width;
+
+ if (in == NULL)
+ in = adev_get_active_input(adev);
+
+ audio_source_t source = (in == NULL) ? AUDIO_SOURCE_DEFAULT : in->source;
+ audio_devices_t in_device =
+ ((in == NULL) ? AUDIO_DEVICE_NONE : in->device) & ~AUDIO_DEVICE_BIT_IN;
+ audio_channel_mask_t channel_mask = (in == NULL) ? AUDIO_CHANNEL_IN_MONO : in->channel_mask;
+ int channel_count = audio_channel_count_from_in_mask(channel_mask);
+ int str_bitwidth = (in == NULL) ? CODEC_BACKEND_DEFAULT_BIT_WIDTH : in->bit_width;
+ int sample_rate = (in == NULL) ? 8000 : in->sample_rate;
+ struct audio_usecase *usecase = NULL;
ALOGV("%s: enter: out_device(%#x) in_device(%#x) channel_count (%d) channel_mask (0x%x)",
__func__, out_device, in_device, channel_count, channel_mask);
if (my_data->external_mic) {
- if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) || voice_is_in_call(adev) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_check_voicecall_usecases_active(adev) ||
+ voice_extn_compress_voip_is_active(adev) ||
+ audio_extn_hfp_is_active(adev))) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
out_device & AUDIO_DEVICE_OUT_EARPIECE ||
out_device & AUDIO_DEVICE_OUT_SPEAKER )
@@ -6074,8 +6232,10 @@
if (snd_device != AUDIO_DEVICE_NONE)
goto exit;
- if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) || voice_is_in_call(adev) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_check_voicecall_usecases_active(adev) ||
+ voice_extn_compress_voip_is_active(adev) ||
+ audio_extn_hfp_is_active(adev))) {
if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
!voice_extn_compress_voip_is_active(adev)) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
@@ -6265,23 +6425,25 @@
(in_device & AUDIO_DEVICE_IN_BACK_MIC)) && // OR Back-mic
(source == AUDIO_SOURCE_MIC || // AND source is MIC for 16bit
source == AUDIO_SOURCE_UNPROCESSED || // OR unprocessed for 24bit
- source == AUDIO_SOURCE_CAMCORDER)) { // OR camera usecase
+ source == AUDIO_SOURCE_CAMCORDER) && // OR camera usecase
+ ((int)channel_mask == (int)AUDIO_CHANNEL_INDEX_MASK_4) && // AND 4mic channel mask
+ (sample_rate == 48000)) { // AND sample rate is 48Khz
snd_device = SND_DEVICE_IN_HANDSET_GENERIC_QMIC;
/* Below check is true only in LA build to set
ambisonic profile. In LE hal client will set profile
*/
- if (my_data->ambisonic_profile == true) {
- strlcpy(adev->active_input->profile, "record_ambisonic",
- sizeof(adev->active_input->profile));
- }
+ if (my_data->ambisonic_profile == true &&
+ in != NULL)
+ strlcpy(in->profile, "record_ambisonic",
+ sizeof(in->profile));
- if (!strncmp(adev->active_input->profile, "record_ambisonic",
- strlen("record_ambisonic"))) {
+ if (in != NULL && !strncmp(in->profile, "record_ambisonic",
+ strlen("record_ambisonic"))) {
/* Validate input stream configuration for
Ambisonic capture.
*/
if (((int)channel_mask != (int)AUDIO_CHANNEL_INDEX_MASK_4) ||
- (adev->active_input->sample_rate != 48000)) {
+ (sample_rate != 48000)) {
snd_device = SND_DEVICE_NONE;
ALOGW("Unsupported Input configuration for ambisonic capture");
goto exit;
@@ -6347,24 +6509,24 @@
if (my_data->fluence_in_voice_rec && channel_count == 1) {
if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
(my_data->source_mic_type & SOURCE_QUAD_MIC)) {
- if (adev->active_input->enable_aec)
+ if (in != NULL && in->enable_aec)
snd_device = SND_DEVICE_IN_HANDSET_QMIC_AEC;
else
snd_device = SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE;
} else if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
(my_data->source_mic_type & SOURCE_THREE_MIC)) {
- if (adev->active_input->enable_aec)
+ if (in != NULL && in->enable_aec)
snd_device = SND_DEVICE_IN_HANDSET_TMIC_AEC;
else
snd_device = SND_DEVICE_IN_VOICE_REC_TMIC;
} else if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
(my_data->source_mic_type & SOURCE_DUAL_MIC)) {
- if (adev->active_input->enable_aec)
+ if (in != NULL && in->enable_aec)
snd_device = SND_DEVICE_IN_HANDSET_DMIC_AEC;
else
snd_device = SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE;
}
- platform_set_echo_reference(adev, true, out_device);
+ in->enable_ec_port = true;
} else if (((channel_mask == AUDIO_CHANNEL_IN_FRONT_BACK) ||
(channel_mask == AUDIO_CHANNEL_IN_STEREO)) &&
(my_data->source_mic_type & SOURCE_DUAL_MIC)) {
@@ -6377,14 +6539,13 @@
snd_device = SND_DEVICE_IN_QUAD_MIC;
}
if (snd_device == SND_DEVICE_NONE) {
- if (adev->active_input->enable_aec) {
- if (adev->active_input->enable_ns) {
+ if (in != NULL && in->enable_aec) {
+ if (in->enable_ns) {
snd_device = SND_DEVICE_IN_VOICE_REC_MIC_AEC_NS;
} else {
snd_device = SND_DEVICE_IN_VOICE_REC_MIC_AEC;
}
- platform_set_echo_reference(adev, true, out_device);
- } else if (adev->active_input->enable_ns)
+ } else if (in != NULL && in->enable_ns)
snd_device = SND_DEVICE_IN_VOICE_REC_MIC_NS;
else
snd_device = SND_DEVICE_IN_VOICE_REC_MIC;
@@ -6435,9 +6596,8 @@
in_device = ((out_device == AUDIO_DEVICE_NONE) ?
AUDIO_DEVICE_IN_BUILTIN_MIC : in_device) & ~AUDIO_DEVICE_BIT_IN;
- if (adev->active_input) {
- snd_device = get_snd_device_for_voice_comm(my_data, out_device, in_device);
- }
+ if (in)
+ snd_device = get_snd_device_for_voice_comm(my_data, in, out_device, in_device);
} else if (source == AUDIO_SOURCE_MIC) {
if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC &&
channel_count == 1 ) {
@@ -6458,6 +6618,20 @@
platform_set_echo_reference(adev, true, out_device);
}
}
+ } else if (in_device & AUDIO_DEVICE_IN_LOOPBACK) {
+ if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ usecase = get_usecase_from_list(adev, USECASE_AUDIO_RECORD);
+ if (usecase == NULL) {
+ ALOGE("%s: Could not find the record usecase", __func__);
+ snd_device = SND_DEVICE_NONE;
+ goto exit;
+ }
+
+ int ch_count = audio_channel_count_from_in_mask(channel_mask);
+ snd_device = audio_extn_get_loopback_snd_device(adev, usecase,
+ ch_count);
+ ALOGD("%s: snd device %d", __func__, snd_device);
+ }
}
} else if (source == AUDIO_SOURCE_FM_TUNER) {
snd_device = SND_DEVICE_IN_CAPTURE_FM;
@@ -6465,7 +6639,7 @@
goto exit;
}
- if (adev->active_input && (audio_extn_ssr_get_stream() == adev->active_input))
+ if (in && (audio_extn_ssr_get_stream() == in))
snd_device = SND_DEVICE_IN_THREE_MIC;
if (snd_device != SND_DEVICE_NONE) {
@@ -6476,7 +6650,7 @@
!(in_device & AUDIO_DEVICE_IN_VOICE_CALL) &&
!(in_device & AUDIO_DEVICE_IN_COMMUNICATION)) {
if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- if ((adev->active_input && (audio_extn_ssr_get_stream() == adev->active_input)) ||
+ if ((in && (audio_extn_ssr_get_stream() == in)) ||
((my_data->source_mic_type & SOURCE_QUAD_MIC) &&
channel_mask == AUDIO_CHANNEL_INDEX_MASK_4))
snd_device = SND_DEVICE_IN_QUAD_MIC;
@@ -6879,9 +7053,11 @@
goto done_key_audcal;
}
- if(cal.dev_id) {
- if(audio_is_input_device(cal.dev_id)) {
- cal.snd_dev_id = platform_get_input_snd_device(platform, cal.dev_id);
+ if (cal.dev_id) {
+ if (audio_is_input_device(cal.dev_id)) {
+ // FIXME: why pass an input device whereas
+ // platform_get_input_snd_device() expects as an output device?
+ cal.snd_dev_id = platform_get_input_snd_device(platform, NULL, cal.dev_id);
} else {
out.devices = cal.dev_id;
out.sample_rate = cal.sampling_rate;
@@ -7436,8 +7612,8 @@
goto done;
}
- if(cal.dev_id & AUDIO_DEVICE_BIT_IN) {
- cal.snd_dev_id = platform_get_input_snd_device(platform, cal.dev_id);
+ if (cal.dev_id & AUDIO_DEVICE_BIT_IN) {
+ cal.snd_dev_id = platform_get_input_snd_device(platform, NULL, cal.dev_id);
} else if(cal.dev_id) {
out.devices = cal.dev_id;
out.sample_rate = cal.sampling_rate;
@@ -7654,9 +7830,13 @@
{
switch (usecase) {
case USECASE_AUDIO_PLAYBACK_DEEP_BUFFER:
+ case USECASE_AUDIO_PLAYBACK_MEDIA:
+ case USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE:
return DEEP_BUFFER_PLATFORM_DELAY;
case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
case USECASE_AUDIO_PLAYBACK_WITH_HAPTICS:
+ case USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION:
+ case USECASE_AUDIO_PLAYBACK_PHONE:
return LOW_LATENCY_PLATFORM_DELAY;
case USECASE_AUDIO_PLAYBACK_OFFLOAD:
case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
@@ -7723,15 +7903,20 @@
case USECASE_AUDIO_PLAYBACK_MULTI_CH:
case USECASE_AUDIO_PLAYBACK_OFFLOAD:
case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
- needs_event = true;
- break;
- /* concurrent playback in low latency allowed */
- case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
- break;
- /* concurrent playback FM needs event */
case USECASE_AUDIO_PLAYBACK_FM:
needs_event = true;
break;
+ case USECASE_AUDIO_PLAYBACK_ULL:
+ case USECASE_AUDIO_PLAYBACK_MMAP:
+ if (property_get_bool("persist.vendor.audio.ull_playback_bargein",
+ false))
+ needs_event = true;
+ break;
+ case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
+ if (property_get_bool("persist.vendor.audio.ll_playback_bargein",
+ false))
+ needs_event = true;
+ break;
/* concurrent capture usecases which needs event */
case USECASE_AUDIO_RECORD:
@@ -7775,6 +7960,9 @@
{
char value[PROPERTY_VALUE_MAX] = {0};
uint32_t fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ uint32_t new_fragment_size = 0;
+ int32_t duration_ms = 0;
+ int channel_count = 0;
if((property_get("vendor.audio.offload.buffer.size.kb", value, "")) &&
atoi(value)) {
fragment_size = atoi(value) * 1024;
@@ -7788,6 +7976,17 @@
fragment_size = info->offload_buffer_size;
}
+ /* Use client specified buffer size if mentioned */
+ if ((info != NULL) && (info->duration_us > 0)) {
+ duration_ms = info->duration_us / 1000;
+ channel_count = audio_channel_count_from_in_mask(info->channel_mask);
+
+ new_fragment_size = (duration_ms * info->sample_rate * channel_count * audio_bytes_per_sample(info->format)) / 1000;
+ ALOGI("%s:: Overwriting offload buffer size with client requested size old:%d new:%d", __func__, fragment_size, new_fragment_size);
+
+ fragment_size = new_fragment_size;
+ }
+
if (info != NULL) {
if (info->is_streaming && info->has_video) {
fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
@@ -8615,6 +8814,7 @@
unsigned int channels;
unsigned int format;
struct platform_data *my_data = (struct platform_data *)adev->platform;
+ struct stream_in *in = adev_get_active_input(adev);
bit_width = backend_cfg->bit_width;
sample_rate = backend_cfg->sample_rate;
@@ -8638,7 +8838,7 @@
} else if (my_data->is_internal_codec && !audio_is_usb_in_device(snd_device)) {
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
- if (adev->active_input && adev->active_input->bit_width == 24)
+ if (in && in->bit_width == 24)
bit_width = platform_get_snd_device_bit_width(snd_device);
} else {
struct listnode *node;
@@ -8722,6 +8922,14 @@
backend_cfg.bit_width= usecase->stream.in->bit_width;
backend_cfg.format= usecase->stream.in->format;
backend_cfg.channels = audio_channel_count_from_in_mask(usecase->stream.in->channel_mask);
+ if (is_loopback_input_device(usecase->stream.in->device)) {
+ int bw = platform_get_snd_device_bit_width(snd_device);
+ if ((-ENOSYS != bw) && (backend_cfg.bit_width > (uint32_t)bw)) {
+ backend_cfg.bit_width = bw;
+ ALOGD("%s:txbecf: set bitwidth to %d from platform info",
+ __func__, bw);
+ }
+ }
} else {
backend_cfg.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
backend_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
@@ -10533,7 +10741,7 @@
size_t actual_mic_count = 0;
snd_device_t active_input_snd_device =
- platform_get_input_snd_device(platform, usecase->stream.in->device);
+ platform_get_input_snd_device(platform, usecase->stream.in, AUDIO_DEVICE_NONE);
if (active_input_snd_device == SND_DEVICE_NONE) {
ALOGI("%s: No active microphones found", __func__);
goto end;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 5160959..1d56a7e 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -129,7 +129,9 @@
SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB,
SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB,
SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
+ SND_DEVICE_OUT_VOICE_TTY_FULL_HEADSET,
SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
+ SND_DEVICE_OUT_VOICE_TTY_VCO_HEADSET,
SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
SND_DEVICE_OUT_VOICE_TTY_FULL_USB,
SND_DEVICE_OUT_VOICE_TTY_VCO_USB,
@@ -175,6 +177,10 @@
SND_DEVICE_OUT_VOICE_SPEAKER_STEREO_AND_VOICE_ANC_HEADSET,
SND_DEVICE_OUT_VOICE_SPEAKER_STEREO_AND_VOICE_ANC_FB_HEADSET,
SND_DEVICE_OUT_VOICE_HEARING_AID,
+ SND_DEVICE_OUT_BUS_MEDIA,
+ SND_DEVICE_OUT_BUS_SYS,
+ SND_DEVICE_OUT_BUS_NAV,
+ SND_DEVICE_OUT_BUS_PHN,
SND_DEVICE_OUT_END,
/*
@@ -311,6 +317,12 @@
SND_DEVICE_IN_CAMCORDER_SELFIE_INVERT_LANDSCAPE,
SND_DEVICE_IN_CAMCORDER_SELFIE_PORTRAIT,
SND_DEVICE_IN_VOICE_HEARING_AID,
+ SND_DEVICE_IN_BUS,
+ SND_DEVICE_IN_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_DMIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_QMIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_6MIC_AND_EC_REF_LOOPBACK,
+ SND_DEVICE_IN_HANDSET_8MIC_AND_EC_REF_LOOPBACK,
SND_DEVICE_IN_END,
SND_DEVICE_MAX = SND_DEVICE_IN_END,
@@ -625,6 +637,11 @@
#define PLAYBACK_INTERACTIVE_STRM_DEVICE7 48
#define PLAYBACK_INTERACTIVE_STRM_DEVICE8 49
+#define MEDIA_PCM_DEVICE DEEP_BUFFER_PCM_DEVICE
+#define SYS_NOTIFICATION_PCM_DEVICE 9
+#define NAV_GUIDANCE_PCM_DEVICE MULTIMEDIA2_PCM_DEVICE
+#define PHONE_PCM_DEVICE 12
+
#ifdef PLATFORM_APQ8084
#define FM_RX_VOLUME "Quat MI2S FM RX Volume"
#elif PLATFORM_MSM8994
diff --git a/hal/platform_api.h b/hal/platform_api.h
index fd85aab..394310a 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -34,6 +34,7 @@
#define LICENSE_STR_MAX_LEN (64)
#define PRODUCT_FFV "ffv"
#define PRODUCT_ALLPLAY "allplay"
+#define MAX_IN_CHANNELS 32
typedef enum {
PLATFORM,
@@ -104,6 +105,7 @@
uint32_t op_channels;
uint32_t usecase_id;
uint32_t snd_device;
+ char fe_name[128];
};
struct audio_custom_mtmx_params {
@@ -112,6 +114,26 @@
uint32_t coeffs[0];
};
+struct audio_custom_mtmx_in_params_info {
+ uint32_t op_channels;
+ uint32_t usecase_id;
+};
+
+struct audio_custom_mtmx_params_in_ch_info {
+ uint32_t ch_count;
+ char device[128];
+ char hw_interface[128];
+};
+
+struct audio_custom_mtmx_in_params {
+ struct listnode list;
+ struct audio_custom_mtmx_in_params_info in_info;
+ uint32_t ip_channels;
+ uint32_t mic_ch;
+ uint32_t ec_ref_ch;
+ struct audio_custom_mtmx_params_in_ch_info in_ch_info[MAX_IN_CHANNELS];
+};
+
enum card_status_t;
void *platform_init(struct audio_device *adev);
@@ -179,7 +201,9 @@
int platform_get_sample_rate(void *platform, uint32_t *rate);
int platform_set_device_mute(void *platform, bool state, char *dir);
snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out);
-snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device);
+snd_device_t platform_get_input_snd_device(void *platform,
+ struct stream_in *in,
+ audio_devices_t out_device);
int platform_set_hdmi_channels(void *platform, int channel_count);
int platform_edid_get_max_channels(void *platform);
void platform_add_operator_specific_device(snd_device_t snd_device,
@@ -356,4 +380,11 @@
struct audio_custom_mtmx_params_info *info);
int platform_add_custom_mtmx_params(void *platform,
struct audio_custom_mtmx_params_info *info);
+/* callback functions from platform to common audio HAL */
+struct stream_in *adev_get_active_input(const struct audio_device *adev);
+
+struct audio_custom_mtmx_in_params * platform_get_custom_mtmx_in_params(void *platform,
+ struct audio_custom_mtmx_in_params_info *info);
+int platform_add_custom_mtmx_in_params(void *platform,
+ struct audio_custom_mtmx_in_params_info *info);
#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/platform_info.c b/hal/platform_info.c
index 05ee9cd..8ee8b07 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -73,6 +73,8 @@
CUSTOM_MTMX_PARAMS,
CUSTOM_MTMX_PARAM_COEFFS,
EXTERNAL_DEVICE_SPECIFIC,
+ CUSTOM_MTMX_IN_PARAMS,
+ CUSTOM_MTMX_PARAM_IN_CH_INFO,
} section_t;
typedef void (* section_process_fn)(const XML_Char **attr);
@@ -97,6 +99,8 @@
static void process_custom_mtmx_params(const XML_Char **attr);
static void process_custom_mtmx_param_coeffs(const XML_Char **attr);
static void process_external_dev(const XML_Char **attr);
+static void process_custom_mtmx_in_params(const XML_Char **attr);
+static void process_custom_mtmx_param_in_ch_info(const XML_Char **attr);
static section_process_fn section_table[] = {
[ROOT] = process_root,
@@ -118,6 +122,8 @@
[CUSTOM_MTMX_PARAMS] = process_custom_mtmx_params,
[CUSTOM_MTMX_PARAM_COEFFS] = process_custom_mtmx_param_coeffs,
[EXTERNAL_DEVICE_SPECIFIC] = process_external_dev,
+ [CUSTOM_MTMX_IN_PARAMS] = process_custom_mtmx_in_params,
+ [CUSTOM_MTMX_PARAM_IN_CH_INFO] = process_custom_mtmx_param_in_ch_info,
};
static section_t section;
@@ -224,6 +230,7 @@
}
static struct audio_custom_mtmx_params_info mtmx_params_info;
+static struct audio_custom_mtmx_in_params_info mtmx_in_params_info;
/*
* <audio_platform_info>
@@ -1003,6 +1010,82 @@
return;
}
+static void process_custom_mtmx_param_in_ch_info(const XML_Char **attr)
+{
+ uint32_t attr_idx = 0;
+ int32_t in_ch_idx = -1;
+ struct audio_custom_mtmx_in_params *mtmx_in_params = NULL;
+
+ mtmx_in_params = platform_get_custom_mtmx_in_params((void *)my_data.platform,
+ &mtmx_in_params_info);
+ if (mtmx_in_params == NULL) {
+ ALOGE("%s: mtmx in params with given param info, not found", __func__);
+ return;
+ }
+
+ if (strcmp(attr[attr_idx++], "in_channel_index") != 0) {
+ ALOGE("%s: 'in_channel_index' not found", __func__);
+ return;
+ }
+
+ in_ch_idx = atoi((char *)attr[attr_idx++]);
+ if (in_ch_idx < 0 || in_ch_idx >= MAX_IN_CHANNELS) {
+ ALOGE("%s: invalid input channel index(%d)", __func__, in_ch_idx);
+ return;
+ }
+
+ if (strcmp(attr[attr_idx++], "channel_count") != 0) {
+ ALOGE("%s: 'channel_count' not found", __func__);
+ return;
+ }
+ mtmx_in_params->in_ch_info[in_ch_idx].ch_count = atoi((char *)attr[attr_idx++]);
+
+ if (strcmp(attr[attr_idx++], "device") != 0) {
+ ALOGE("%s: 'device' not found", __func__);
+ return;
+ }
+ strlcpy(mtmx_in_params->in_ch_info[in_ch_idx].device, attr[attr_idx++],
+ sizeof(mtmx_in_params->in_ch_info[in_ch_idx].device));
+
+ if (strcmp(attr[attr_idx++], "interface") != 0) {
+ ALOGE("%s: 'interface' not found", __func__);
+ return;
+ }
+ strlcpy(mtmx_in_params->in_ch_info[in_ch_idx].hw_interface, attr[attr_idx++],
+ sizeof(mtmx_in_params->in_ch_info[in_ch_idx].hw_interface));
+
+ if (!strncmp(mtmx_in_params->in_ch_info[in_ch_idx].device,
+ ENUM_TO_STRING(AUDIO_DEVICE_IN_BUILTIN_MIC),
+ sizeof(mtmx_in_params->in_ch_info[in_ch_idx].device)))
+ mtmx_in_params->mic_ch = mtmx_in_params->in_ch_info[in_ch_idx].ch_count;
+ else if (!strncmp(mtmx_in_params->in_ch_info[in_ch_idx].device,
+ ENUM_TO_STRING(AUDIO_DEVICE_IN_LOOPBACK),
+ sizeof(mtmx_in_params->in_ch_info[in_ch_idx].device)))
+ mtmx_in_params->ec_ref_ch = mtmx_in_params->in_ch_info[in_ch_idx].ch_count;
+
+ mtmx_in_params->ip_channels += mtmx_in_params->in_ch_info[in_ch_idx].ch_count;
+}
+
+static void process_custom_mtmx_in_params(const XML_Char **attr)
+{
+ int attr_idx = 0;
+
+ if (strcmp(attr[attr_idx++], "usecase") != 0) {
+ ALOGE("%s: 'usecase' not found", __func__);
+ return;
+ }
+ mtmx_in_params_info.usecase_id = platform_get_usecase_index((char *)attr[attr_idx++]);
+
+ if (strcmp(attr[attr_idx++], "out_channel_count") != 0) {
+ ALOGE("%s: 'out_channel_count' not found", __func__);
+ return;
+ }
+ mtmx_in_params_info.op_channels = atoi((char *)attr[attr_idx++]);
+
+ platform_add_custom_mtmx_in_params((void *)my_data.platform, &mtmx_in_params_info);
+
+}
+
static void process_custom_mtmx_param_coeffs(const XML_Char **attr)
{
uint32_t attr_idx = 0, out_ch_idx = -1, ch_coeff_count = 0;
@@ -1034,7 +1117,7 @@
ch_coeff_value = strtok_r((char *)attr[attr_idx++], " ", &context);
ip_channels = mtmx_params->info.ip_channels;
op_channels = mtmx_params->info.op_channels;
- while(ch_coeff_value && ch_coeff_count < op_channels) {
+ while(ch_coeff_value && ch_coeff_count < ip_channels) {
mtmx_params->coeffs[ip_channels * out_ch_idx + ch_coeff_count++]
= atoi(ch_coeff_value);
ch_coeff_value = strtok_r(NULL, " ", &context);
@@ -1077,6 +1160,15 @@
return;
}
mtmx_params_info.snd_device = platform_get_snd_device_index((char *)attr[attr_idx++]);
+
+ if ((attr[attr_idx] != NULL) && (strcmp(attr[attr_idx++], "fe_name") == 0)) {
+ strlcpy(mtmx_params_info.fe_name, (char *)attr[attr_idx++],
+ sizeof(mtmx_params_info.fe_name));
+ } else {
+ ALOGD("%s: 'fe_name' not found", __func__);
+ mtmx_params_info.fe_name[0] = '\0';
+ }
+
platform_add_custom_mtmx_params((void *)my_data.platform, &mtmx_params_info);
}
@@ -1244,6 +1336,22 @@
} else if (strcmp(tag_name, "ext_device") == 0) {
section_process_fn fn = section_table[section];
fn(attr);
+ } else if (strcmp(tag_name, "custom_mtmx_in_params") == 0) {
+ if (section != ROOT) {
+ ALOGE("custom_mtmx_in_params tag supported only in ROOT section");
+ return;
+ }
+ section = CUSTOM_MTMX_IN_PARAMS;
+ section_process_fn fn = section_table[section];
+ fn(attr);
+ } else if (strcmp(tag_name, "custom_mtmx_param_in_chs") == 0) {
+ if (section != CUSTOM_MTMX_IN_PARAMS) {
+ ALOGE("custom_mtmx_param_in_chs tag supported only with CUSTOM_MTMX_IN_PARAMS section");
+ return;
+ }
+ section = CUSTOM_MTMX_PARAM_IN_CH_INFO;
+ section_process_fn fn = section_table[section];
+ fn(attr);
}
} else {
if(strcmp(tag_name, "config_params") == 0) {
@@ -1306,6 +1414,10 @@
section = ROOT;
} else if (strcmp(tag_name, "custom_mtmx_param_coeffs") == 0) {
section = CUSTOM_MTMX_PARAMS;
+ } else if (strcmp(tag_name, "custom_mtmx_in_params") == 0) {
+ section = ROOT;
+ } else if (strcmp(tag_name, "custom_mtmx_param_in_chs") == 0) {
+ section = CUSTOM_MTMX_IN_PARAMS;
}
}
diff --git a/hal/voice.c b/hal/voice.c
index 729ab27..006dd08 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -377,6 +377,22 @@
return session_id;
}
+bool voice_check_voicecall_usecases_active(struct audio_device *adev)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase = NULL;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == VOICE_CALL) {
+ ALOGV("%s: voice usecase:%s is active", __func__,
+ use_case_table[usecase->id]);
+ return true;
+ }
+ }
+ return false;
+}
+
int voice_check_and_set_incall_rec_usecase(struct audio_device *adev,
struct stream_in *in)
{
diff --git a/hal/voice.h b/hal/voice.h
index 9612edd..188345d 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -101,4 +101,5 @@
bool voice_is_call_state_active(struct audio_device *adev);
void voice_set_device_mute_flag (struct audio_device *adev, bool state);
snd_device_t voice_get_incall_rec_backend_device(struct stream_in *in);
+bool voice_check_voicecall_usecases_active(struct audio_device *adev);
#endif //VOICE_H
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index e4bc210..fbd6d6f 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -501,7 +501,6 @@
if (!voip_data.in_stream_count)
ret = compress_voip_open_input_stream(in);
- adev->active_input = in;
ret = voip_start_call(adev, &in->config);
in->pcm = voip_data.pcm_tx;
@@ -539,7 +538,6 @@
if(voip_data.in_stream_count > 0) {
voip_data.in_stream_count--;
status = voip_stop_call(adev);
- adev->active_input = get_next_active_input(adev);
in->pcm = NULL;
}
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 6b1afd3..fb42514 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -30,8 +30,7 @@
virtualizer.c \
reverb.c \
effect_api.c \
- effect_util.c \
- asphere.c
+ effect_util.c
# HW_ACCELERATED has been disabled by default since msm8996. File doesn't
# compile cleanly on tip so don't want to include it, but keeping this
diff --git a/post_proc/Makefile.am b/post_proc/Makefile.am
index bd29473..8bd41ae 100644
--- a/post_proc/Makefile.am
+++ b/post_proc/Makefile.am
@@ -19,10 +19,6 @@
c_sources += hw_accelerator.c
endif
-if AUDIOSPHERE
-c_sources += asphere.c
-endif
-
library_include_HEADERS = $(h_sources)
library_includedir = $(includedir)
diff --git a/post_proc/asphere.c b/post_proc/asphere.c
deleted file mode 100644
index efe07c6..0000000
--- a/post_proc/asphere.c
+++ /dev/null
@@ -1,323 +0,0 @@
-/* Copyright (c) 2015, 2017, 2019 The Linux Foundation. All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are
- * met:
- * * Redistributions of source code must retain the above copyright
- * notice, this list of conditions and the following disclaimer.
- * * Redistributions in binary form must reproduce the above
- * copyright notice, this list of conditions and the following
- * disclaimer in the documentation and/or other materials provided
- * with the distribution.
- * * Neither the name of The Linux Foundation nor the names of its
- * contributors may be used to endorse or promote products derived
- * from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
- * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
- * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
- * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
- * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
- * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
- * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
- * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- *
- */
-#define LOG_TAG "audio_pp_asphere"
-/*#define LOG_NDEBUG 0*/
-
-#include <errno.h>
-#include <fcntl.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <stdbool.h>
-#include <sys/stat.h>
-#include <log/log.h>
-#include <cutils/list.h>
-#include <cutils/str_parms.h>
-#include <cutils/properties.h>
-#include <hardware/audio_effect.h>
-#include <pthread.h>
-#include "bundle.h"
-#include "equalizer.h"
-#include "bass_boost.h"
-#include "virtualizer.h"
-#include "reverb.h"
-#include "asphere.h"
-
-#define ASPHERE_MIXER_NAME "MSM ASphere Set Param"
-
-#define AUDIO_PARAMETER_KEY_ASPHERE_STATUS "asphere_status"
-#define AUDIO_PARAMETER_KEY_ASPHERE_ENABLE "asphere_enable"
-#define AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH "asphere_strength"
-
-#define AUDIO_ASPHERE_EVENT_NODE "/data/misc/audio_pp/event_node"
-
-enum {
- ASPHERE_ACTIVE = 0,
- ASPHERE_SUSPENDED,
- ASPHERE_ERROR
-};
-
-#ifdef AUDIO_FEATURE_ENABLED_GCOV
-extern void __gcov_flush();
-static void enable_gcov()
-{
- __gcov_flush();
-}
-#else
-static void enable_gcov()
-{
-}
-#endif
-
-struct asphere_module {
- bool enabled;
- int status;
- int strength;
- pthread_mutex_t lock;
- int init_status;
-};
-
-static struct asphere_module asphere;
-pthread_once_t asphere_once = PTHREAD_ONCE_INIT;
-
-static int asphere_create_app_notification_node(void)
-{
- int fd;
- if ((fd = open(AUDIO_ASPHERE_EVENT_NODE, O_CREAT|O_TRUNC|O_WRONLY,
- S_IRUSR|S_IWUSR|S_IRGRP|S_IROTH)) < 0) {
- ALOGE("creating notification node failed %d", errno);
- return -EINVAL;
- }
- chmod(AUDIO_ASPHERE_EVENT_NODE, S_IRWXU|S_IRGRP|S_IXGRP|S_IROTH);
- close(fd);
- ALOGD("%s: successfully created notification node %s",
- __func__, AUDIO_ASPHERE_EVENT_NODE);
- return 0;
-}
-
-static int asphere_notify_app(void)
-{
- int fd;
- if ((fd = open(AUDIO_ASPHERE_EVENT_NODE, O_TRUNC|O_WRONLY)) < 0) {
- ALOGE("opening notification node failed %d", errno);
- return -EINVAL;
- }
- close(fd);
- ALOGD("%s: successfully opened notification node", __func__);
- return 0;
-}
-
-static int asphere_get_values_from_mixer(void)
-{
- int ret = 0;
- long val[2] = {-1, -1};
- struct mixer_ctl *ctl = NULL;
- struct mixer *mixer = mixer_open(MIXER_CARD);
- if (mixer)
- ctl = mixer_get_ctl_by_name(mixer, ASPHERE_MIXER_NAME);
- if (!ctl) {
- ALOGE("%s: could not get ctl for mixer cmd - %s",
- __func__, ASPHERE_MIXER_NAME);
- return -EINVAL;
- }
- ret = mixer_ctl_get_array(ctl, val, sizeof(val)/sizeof(val[0]));
- if (!ret) {
- asphere.enabled = (val[0] == 0) ? false : true;
- asphere.strength = val[1];
- }
- ALOGD("%s: returned %d, enabled:%ld, strength:%ld",
- __func__, ret, val[0], val[1]);
-
- return ret;
-}
-
-static int asphere_set_values_to_mixer(void)
-{
- int ret = 0;
- long val[2] = {-1, -1};
- struct mixer_ctl *ctl = NULL;
- struct mixer *mixer = mixer_open(MIXER_CARD);
- if (mixer)
- ctl = mixer_get_ctl_by_name(mixer, ASPHERE_MIXER_NAME);
- if (!ctl) {
- ALOGE("%s: could not get ctl for mixer cmd - %s",
- __func__, ASPHERE_MIXER_NAME);
- return -EINVAL;
- }
- val[0] = ((asphere.status == ASPHERE_ACTIVE) && asphere.enabled) ? 1 : 0;
- val[1] = asphere.strength;
-
- ret = mixer_ctl_set_array(ctl, val, sizeof(val)/sizeof(val[0]));
- ALOGD("%s: returned %d, enabled:%ld, strength:%ld",
- __func__, ret, val[0], val[1]);
-
- return ret;
-}
-
-static void asphere_init_once() {
- ALOGD("%s", __func__);
- pthread_mutex_init(&asphere.lock, NULL);
-
- if (property_get_bool("vendor.audio.feature.audio_sphere.enable", false)) {
- asphere.init_status = 1;
- asphere_get_values_from_mixer();
- asphere_create_app_notification_node();
- return;
- } else {
- ALOGW("%s: asphere feature not enabled", __func__);
- }
-
- asphere.init_status = 0;
-}
-
-static int asphere_init() {
- pthread_once(&asphere_once, asphere_init_once);
- enable_gcov();
- return asphere.init_status;
-}
-
-static bool asphere_parms_allowed(struct str_parms *parms)
-{
- if (str_parms_has_key(parms, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE))
- return true;
- if (str_parms_has_key(parms, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH))
- return true;
- if (str_parms_has_key(parms, AUDIO_PARAMETER_KEY_ASPHERE_STATUS))
- return true;
-
- return false;
-}
-
-void asphere_set_parameters(struct str_parms *parms)
-{
- int ret = 0;
- bool enable = false;
- int strength = -1;
- char value[32] = {0};
- bool set_enable = false, set_strength = false;
-
- if (!asphere_parms_allowed(parms)) {
- return;
- }
-
- if (asphere_init() != 1) {
- ALOGW("%s: init check failed!!!", __func__);
- return;
- }
-
- ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
- value, sizeof(value));
- if (ret > 0) {
- enable = (atoi(value) == 1) ? true : false;
- set_enable = true;
- }
-
- ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
- value, sizeof(value));
- if (ret > 0) {
- strength = atoi(value);
- if (strength >= 0 && strength <= 1000)
- set_strength = true;
- }
-
- if (set_enable || set_strength) {
- pthread_mutex_lock(&asphere.lock);
- asphere.enabled = set_enable ? enable : asphere.enabled;
- asphere.strength = set_strength ? strength : asphere.strength;
- ret = asphere_set_values_to_mixer();
- pthread_mutex_unlock(&asphere.lock);
- ALOGV("%s: exit ret %d", __func__, ret);
- }
-}
-
-void asphere_get_parameters(struct str_parms *query,
- struct str_parms *reply)
-{
- char value[32] = {0};
- int ret;
-
- if (asphere_init() != 1) {
- ALOGW("%s: init check failed!!!", __func__);
- return;
- }
-
- ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_STATUS,
- value, sizeof(value));
- if (ret >= 0) {
- str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_STATUS,
- asphere.status);
- }
-
- ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
- value, sizeof(value));
- if (ret >= 0) {
- str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
- asphere.enabled ? 1 : 0);
- }
-
- ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
- value, sizeof(value));
- if (ret >= 0) {
- str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
- asphere.strength);
- }
-}
-
-static bool effect_needs_asphere_concurrency_handling(effect_context_t *context)
-{
- if (memcmp(&context->desc->type,
- &equalizer_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
- memcmp(&context->desc->type,
- &bassboost_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
- memcmp(&context->desc->type,
- &virtualizer_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
- memcmp(&context->desc->type,
- &ins_preset_reverb_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
- memcmp(&context->desc->type,
- &ins_env_reverb_descriptor.type, sizeof(effect_uuid_t)) == 0)
- return true;
-
- return false;
-}
-
-void handle_asphere_on_effect_enabled(bool enable,
- effect_context_t *context,
- struct listnode *created_effects)
-{
- struct listnode *node;
-
- ALOGV("%s: effect %0x", __func__, context->desc->type.timeLow);
- if (asphere_init() != 1) {
- ALOGW("%s: init check failed!!!", __func__);
- return;
- }
-
- if (!effect_needs_asphere_concurrency_handling(context)) {
- ALOGV("%s: effect %0x, do not need concurrency handling",
- __func__, context->desc->type.timeLow);
- return;
- }
-
- list_for_each(node, created_effects) {
- effect_context_t *fx_ctxt = node_to_item(node,
- effect_context_t,
- effects_list_node);
- if (fx_ctxt != NULL &&
- effect_needs_asphere_concurrency_handling(fx_ctxt) == true &&
- fx_ctxt != context && effect_is_active(fx_ctxt) == true) {
- ALOGV("%s: found another effect %0x, skip processing %0x", __func__,
- fx_ctxt->desc->type.timeLow, context->desc->type.timeLow);
- return;
- }
- }
- pthread_mutex_lock(&asphere.lock);
- asphere.status = enable ? ASPHERE_SUSPENDED : ASPHERE_ACTIVE;
- asphere_set_values_to_mixer();
- asphere_notify_app();
- pthread_mutex_unlock(&asphere.lock);
-}
diff --git a/post_proc/asphere.h b/post_proc/asphere.h
deleted file mode 100644
index 3babd1d..0000000
--- a/post_proc/asphere.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/* Copyright (c) 2015, The Linux Foundation. All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are
- * met:
- * * Redistributions of source code must retain the above copyright
- * notice, this list of conditions and the following disclaimer.
- * * Redistributions in binary form must reproduce the above
- * copyright notice, this list of conditions and the following
- * disclaimer in the documentation and/or other materials provided
- * with the distribution.
- * * Neither the name of The Linux Foundation nor the names of its
- * contributors may be used to endorse or promote products derived
- * from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
- * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
- * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
- * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
- * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
- * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
- * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
- * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- *
- */
-
-#ifndef OFFLOAD_ASPHERE_H_
-#define OFFLOAD_ASPHERE_H_
-
-#include <cutils/str_parms.h>
-#include <cutils/list.h>
-#include "bundle.h"
-
-void asphere_get_parameters(struct str_parms *query,
- struct str_parms *reply);
-void asphere_set_parameters(struct str_parms *reply);
-void handle_asphere_on_effect_enabled(bool enable,
- effect_context_t *context,
- struct listnode *created_effects);
-
-#endif /* OFFLOAD_ASPHERE_H_ */
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index 1e6b91d..0dbf27b 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -40,6 +40,7 @@
#include <stdlib.h>
#include <cutils/list.h>
+#include <cutils/str_parms.h>
#include <log/log.h>
#include <system/thread_defs.h>
#include <tinyalsa/asoundlib.h>
@@ -53,7 +54,6 @@
#include "bass_boost.h"
#include "virtualizer.h"
#include "reverb.h"
-#include "asphere.h"
#ifdef DTS_EAGLE
#include "effect_util.h"
@@ -455,20 +455,16 @@
/*
* Effect Bundle Set and get param operations.
- * currently only handles audio sphere scenario,
- * but the interface itself can be utilized for any effect.
*/
__attribute__ ((visibility ("default")))
-void offload_effects_bundle_get_parameters(struct str_parms *query,
- struct str_parms *reply)
+void offload_effects_bundle_get_parameters(struct str_parms *query __unused,
+ struct str_parms *reply __unused)
{
- asphere_get_parameters(query, reply);
}
__attribute__ ((visibility ("default")))
-void offload_effects_bundle_set_parameters(struct str_parms *parms)
+void offload_effects_bundle_set_parameters(struct str_parms *parms __unused)
{
- asphere_set_parameters(parms);
}
/*
@@ -826,7 +822,6 @@
status = -ENOSYS;
goto exit;
}
- handle_asphere_on_effect_enabled(true, context, &created_effects_list);
context->state = EFFECT_STATE_ACTIVE;
if (context->ops.enable)
context->ops.enable(context);
@@ -841,7 +836,6 @@
status = -ENOSYS;
goto exit;
}
- handle_asphere_on_effect_enabled(false, context, &created_effects_list);
context->state = EFFECT_STATE_INITIALIZED;
if (context->ops.disable)
context->ops.disable(context);
diff --git a/qahw/inc/qahw.h b/qahw/inc/qahw.h
index dd5b403..5020c8f 100644
--- a/qahw/inc/qahw.h
+++ b/qahw/inc/qahw.h
@@ -358,6 +358,10 @@
ssize_t qahw_in_read_l(qahw_stream_handle_t *in_handle,
qahw_in_buffer_t *in_buf);
/*
+ * Stop input stream. Returns zero on success.
+ */
+int qahw_in_stop_l(qahw_stream_handle_t *in_handle);
+/*
* Return the amount of input frames lost in the audio driver since the
* last call of this function.
* Audio driver is expected to reset the value to 0 and restart counting
diff --git a/qahw/src/qahw.c b/qahw/src/qahw.c
index 3390c26..545152c 100644
--- a/qahw/src/qahw.c
+++ b/qahw/src/qahw.c
@@ -61,6 +61,8 @@
typedef uint64_t (*qahwi_in_read_v2_t)(audio_stream_in_t *in, void* buffer,
size_t bytes, int64_t *timestamp);
+typedef int (*qahwi_in_stop_t)(audio_stream_in_t *in);
+
typedef int (*qahwi_out_set_param_data_t)(struct audio_stream_out *out,
qahw_param_id param_id,
qahw_param_payload *payload);
@@ -109,6 +111,7 @@
struct listnode list;
pthread_mutex_t lock;
qahwi_in_read_v2_t qahwi_in_read_v2;
+ qahwi_in_stop_t qahwi_in_stop;
} qahw_stream_in_t;
typedef enum {
@@ -1035,6 +1038,31 @@
}
/*
+ * Stop input stream. Returns zero on success.
+ */
+int qahw_in_stop_l(qahw_stream_handle_t *in_handle)
+{
+ int rc = -EINVAL;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+ ALOGD("%s", __func__);
+
+ in = qahw_stream_in->stream;
+
+ if (qahw_stream_in->qahwi_in_stop)
+ rc = qahw_stream_in->qahwi_in_stop(in);
+ ALOGD("%s: exit", __func__);
+
+exit:
+ return rc;
+}
+
+/*
* Return the amount of input frames lost in the audio driver since the
* last call of this function.
* Audio driver is expected to reset the value to 0 and restart counting
@@ -1718,6 +1746,7 @@
qahw_module_t *qahw_module_temp = NULL;
audio_hw_device_t *audio_device = NULL;
qahw_stream_in_t *qahw_stream_in = NULL;
+ const char *error;
pthread_mutex_lock(&qahw_module_init_lock);
qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
@@ -1747,6 +1776,7 @@
if (rc) {
ALOGE("%s::open input stream failed %d",__func__, rc);
free(qahw_stream_in);
+ goto exit;
} else {
qahw_stream_in->module = hw_module;
*in_handle = (void *)qahw_stream_in;
@@ -1757,7 +1787,6 @@
/* dlsym qahwi_in_read_v2 if timestamp flag is used */
if (!rc && ((flags & QAHW_INPUT_FLAG_TIMESTAMP) ||
(flags & QAHW_INPUT_FLAG_PASSTHROUGH))) {
- const char *error;
/* clear any existing errors */
dlerror();
@@ -1769,7 +1798,16 @@
}
}
-exit:
+ /* clear any existing errors */
+ dlerror();
+ qahw_stream_in->qahwi_in_stop = (qahwi_in_stop_t)
+ dlsym(qahw_module->module->dso, "qahwi_in_stop");
+ if ((error = dlerror()) != NULL) {
+ ALOGI("%s: dlsym error %s for qahwi_in_stop", __func__, error);
+ qahw_stream_in->qahwi_in_stop = NULL;
+ }
+
+ exit:
pthread_mutex_unlock(&qahw_module->lock);
return rc;
}
diff --git a/qahw_api/inc/qahw_api.h b/qahw_api/inc/qahw_api.h
index 823c6bb..b37757d 100644
--- a/qahw_api/inc/qahw_api.h
+++ b/qahw_api/inc/qahw_api.h
@@ -354,6 +354,10 @@
ssize_t qahw_in_read(qahw_stream_handle_t *in_handle,
qahw_in_buffer_t *in_buf);
/*
+ * Stop input stream. Returns zero on success.
+ */
+int qahw_in_stop(qahw_stream_handle_t *in_handle);
+/*
* Return the amount of input frames lost in the audio driver since the
* last call of this function.
* Audio driver is expected to reset the value to 0 and restart counting
diff --git a/qahw_api/src/qahw_api.cpp b/qahw_api/src/qahw_api.cpp
index f1c75f4..0810ede 100644
--- a/qahw_api/src/qahw_api.cpp
+++ b/qahw_api/src/qahw_api.cpp
@@ -678,6 +678,22 @@
}
}
+int qahw_in_stop(qahw_stream_handle_t *in_handle)
+{
+ if (g_binder_enabled) {
+ if (!g_qas_died) {
+ sp<Iqti_audio_server> qas = get_qti_audio_server();
+ if (qas_status(qas) == -1)
+ return -ENODEV;
+ return qas->qahw_in_stop(in_handle);
+ } else {
+ return -ENODEV;
+ }
+ } else {
+ return qahw_in_stop_l(in_handle);
+ }
+}
+
uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle)
{
ALOGV("%d:%s",__LINE__, __func__);
@@ -1544,6 +1560,11 @@
return qahw_in_read_l(in_handle, in_buf);
}
+int qahw_in_stop(qahw_stream_handle_t *in_handle)
+{
+ return qahw_in_stop_l(in_handle);
+}
+
uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle)
{
ALOGV("%d:%s",__LINE__, __func__);
diff --git a/qahw_api/test/qahw_multi_record_test.c b/qahw_api/test/qahw_multi_record_test.c
index eccfe76..e033921 100644
--- a/qahw_api/test/qahw_multi_record_test.c
+++ b/qahw_api/test/qahw_multi_record_test.c
@@ -283,6 +283,15 @@
case 8:
params->config.channel_mask = AUDIO_CHANNEL_INDEX_MASK_8;
break;
+ case 10:
+ params->config.channel_mask = AUDIO_CHANNEL_INDEX_MASK_10;
+ break;
+ case 12:
+ params->config.channel_mask = AUDIO_CHANNEL_INDEX_MASK_12;
+ break;
+ case 14:
+ params->config.channel_mask = AUDIO_CHANNEL_INDEX_MASK_14;
+ break;
default:
fprintf(log_file, "ERROR :::: channle count %d not supported, handle(%d)", params->channels, params->handle);
if (log_file != stdout)
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
index 9f1489c..12be83d 100644
--- a/qahw_api/test/qahw_playback_test.c
+++ b/qahw_api/test/qahw_playback_test.c
@@ -353,9 +353,11 @@
switch (event) {
case QAHW_STREAM_CBK_EVENT_WRITE_READY:
fprintf(log_file, "stream %d: received event - QAHW_STREAM_CBK_EVENT_WRITE_READY\n", params->stream_index);
+
pthread_mutex_lock(¶ms->write_lock);
pthread_cond_signal(¶ms->write_cond);
pthread_mutex_unlock(¶ms->write_lock);
+
break;
case QAHW_STREAM_CBK_EVENT_DRAIN_READY:
fprintf(log_file, "stream %d: received event - QAHW_STREAM_CBK_EVENT_DRAIN_READY\n", params->stream_index);
@@ -534,7 +536,7 @@
stream_config *stream_params = (stream_config*) params_ptr;
ssize_t ret;
- pthread_mutex_lock(&stream_params->write_lock);
+
qahw_out_buffer_t out_buf;
memset(&out_buf,0, sizeof(qahw_out_buffer_t));
@@ -545,13 +547,14 @@
if (ret < 0) {
fprintf(log_file, "stream %d: writing data to hal failed (ret = %zd)\n", stream_params->stream_index, ret);
} else if ((ret != bytes) && (!stop_playback)) {
+ pthread_mutex_lock(&stream_params->write_lock);
fprintf(log_file, "stream %d: provided bytes %zd, written bytes %d\n",stream_params->stream_index, bytes, ret);
fprintf(log_file, "stream %d: waiting for event write ready\n", stream_params->stream_index);
pthread_cond_wait(&stream_params->write_cond, &stream_params->write_lock);
fprintf(log_file, "stream %d: out of wait for event write ready\n", stream_params->stream_index);
+ pthread_mutex_unlock(&stream_params->write_lock);
}
- pthread_mutex_unlock(&stream_params->write_lock);
return ret;
}
@@ -2111,6 +2114,7 @@
{"intr-strm", required_argument, 0, 'i'},
{"device-config", required_argument, 0, 'C'},
{"play-list", required_argument, 0, 'g'},
+ {"ec-ref", no_argument, 0, 'L'},
{"help", no_argument, 0, 'h'},
{"bt-wbs", no_argument, 0, 'z'},
{0, 0, 0, 0}
@@ -2135,7 +2139,7 @@
while ((opt = getopt_long(argc,
argv,
- "-f:r:c:b:d:s:v:V:l:t:a:w:k:PD:KF:Ee:A:u:m:S:C:p::x:y:qQzh:i:h:g:O:",
+ "-f:r:c:b:d:s:v:V:l:t:a:w:k:PD:KF:Ee:A:u:m:S:C:p::x:y:qQzLh:i:h:g:O:",
long_options,
&option_index)) != -1) {
@@ -2335,6 +2339,9 @@
case 'x':
render_format = atoi(optarg);
break;
+ case 'L':
+ ec_ref = true;
+ break;
case 'y':
stream_param[i].timestamp_filename = optarg;
break;
diff --git a/qahw_api/test/qahw_playback_test.h b/qahw_api/test/qahw_playback_test.h
index 6f33338..4c78813 100644
--- a/qahw_api/test/qahw_playback_test.h
+++ b/qahw_api/test/qahw_playback_test.h
@@ -41,6 +41,7 @@
bool enable_dump;
float vol_level;
uint8_t render_format;
+bool ec_ref;
enum {
diff --git a/qahw_api/test/qap_wrapper_extn.c b/qahw_api/test/qap_wrapper_extn.c
index de954bf..b3da21a 100644
--- a/qahw_api/test/qap_wrapper_extn.c
+++ b/qahw_api/test/qap_wrapper_extn.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2016-2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2016-2017,2019 The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2015 The Android Open Source Project *
@@ -88,6 +88,7 @@
FILE *fp_output_writer_hp = NULL;
FILE *fp_output_writer_hdmi = NULL;
FILE *fp_output_timestamp_file = NULL;
+FILE *fp_ecref = NULL;
unsigned char data_buf[MAX_BUFFER_SIZE];
uint32_t output_device_id = 0;
uint16_t input_streams_count = 0;
@@ -207,6 +208,7 @@
bool enable_hdmi = false;
bool combo_enabled = false;
char dev_kv_pair[16] = {0};
+ bool enable_ecref = false;
ALOGV("%s:%d output device id %d render format = %d", __func__, __LINE__, output_device_id, hdmi_render_format);
@@ -217,6 +219,8 @@
enable_hp = true;
if (output_device_id & AUDIO_DEVICE_OUT_SPEAKER)
enable_spk = true;
+ if (ec_ref)
+ enable_ecref = true;
if (enable_hdmi) {
session_output_config.output_config[session_output_config.num_output].id = AUDIO_DEVICE_OUT_HDMI;
@@ -270,6 +274,20 @@
session_output_config.num_output++;
}
+ if (enable_ecref) {
+ session_output_config.output_config[session_output_config.num_output].channels = popcount(AUDIO_CHANNEL_OUT_STEREO);
+ session_output_config.output_config[session_output_config.num_output].id = AUDIO_DEVICE_OUT_PROXY;
+ session_output_config.output_config[session_output_config.num_output].sample_rate = smpl_rate;
+ if (bitwidth == PCM_24_BITWIDTH) {
+ session_output_config.output_config[session_output_config.num_output].format = QAP_AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ session_output_config.output_config[session_output_config.num_output].bit_width = PCM_24_BITWIDTH;
+ } else {
+ session_output_config.output_config[session_output_config.num_output].format = QAP_AUDIO_FORMAT_PCM_16_BIT;
+ session_output_config.output_config[session_output_config.num_output].bit_width = PCM_16_BITWIDTH;
+ }
+ session_output_config.num_output++;
+ }
+
ALOGV("%s:%d num_output = %d", __func__, __LINE__, session_output_config.num_output);
return;
}
@@ -1235,6 +1253,20 @@
ALOGD("%s::%d Measuring Kpi cold stop %lf", __func__, __LINE__, cold_stop);
}
}
+ if (buffer->buffer_parms.output_buf_params.output_id == AUDIO_DEVICE_OUT_PROXY) {
+
+ if (fp_ecref == NULL) {
+ fp_ecref = fopen("/data/vendor/misc/audio/ecref", "w+");
+ }
+
+ if (fp_ecref) {
+ ALOGD("%s: write %d bytes to ecref dump",__func__,buffer->common_params.size);
+ fwrite((unsigned char *)buffer->common_params.data, 1, buffer->common_params.size, fp_ecref);
+ } else {
+ ALOGE("%s: failed to open ecref dump file",__func__);
+ }
+
+ }
}
}
break;