Merge "configs: Declare low-latency and pro audio capability"
diff --git a/configs/atoll/atoll.mk b/configs/atoll/atoll.mk
index 62cb134..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
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_policy_configuration.xml b/configs/kona/audio_policy_configuration.xml
index 1594f3d..8bb3328 100644
--- a/configs/kona/audio_policy_configuration.xml
+++ b/configs/kona/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/kona/kona.mk b/configs/kona/kona.mk
index a9be2e6..252c462 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
@@ -300,7 +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 += \
diff --git a/configs/kona/mixer_paths_qrd.xml b/configs/kona/mixer_paths_qrd.xml
index b8671d6..08176d5 100644
--- a/configs/kona/mixer_paths_qrd.xml
+++ b/configs/kona/mixer_paths_qrd.xml
@@ -2523,6 +2523,10 @@
<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/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_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 cefd152..dd88b9c 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
@@ -283,7 +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 += \
@@ -394,7 +396,7 @@
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 188d5ee..0d8585b 100644
--- a/configs/lito/mixer_paths_qrd.xml
+++ b/configs/lito/mixer_paths_qrd.xml
@@ -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>
diff --git a/configs/msm8998/msm8998.mk b/configs/msm8998/msm8998.mk
index bee32c8..56cc95d 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 += \
diff --git a/configs/msmnile/msmnile.mk b/configs/msmnile/msmnile.mk
index 5357e61..29f277b 100644
--- a/configs/msmnile/msmnile.mk
+++ b/configs/msmnile/msmnile.mk
@@ -281,7 +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 += \
diff --git a/configs/msmnile_au/audio_policy_configuration.xml b/configs/msmnile_au/audio_policy_configuration.xml
index fcba319..b00e62f 100644
--- a/configs/msmnile_au/audio_policy_configuration.xml
+++ b/configs/msmnile_au/audio_policy_configuration.xml
@@ -168,6 +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">
diff --git a/configs/msmnile_au/mixer_paths_adp.xml b/configs/msmnile_au/mixer_paths_adp.xml
index 7978e97..63012be 100644
--- a/configs/msmnile_au/mixer_paths_adp.xml
+++ b/configs/msmnile_au/mixer_paths_adp.xml
@@ -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">
diff --git a/configs/msmnile_au/msmnile_au.mk b/configs/msmnile_au/msmnile_au.mk
index 7dd0a3e..394dfea 100644
--- a/configs/msmnile_au/msmnile_au.mk
+++ b/configs/msmnile_au/msmnile_au.mk
@@ -195,6 +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
@@ -286,3 +294,16 @@
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
index 01e279d..0274f9e 100644
--- 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
@@ -23,7 +23,7 @@
<resources>
<!-- Car uses hardware amplifier for volume. -->
- <bool name="config_useFixedVolume">true</bool>
+ <bool name="config_useFixedVolume">false</bool>
<!--
Handle volume keys directly in CarAudioService without passing them to the foreground app
-->
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_policy_configuration.xml b/configs/msmsteppe/audio_policy_configuration.xml
index 71cac8b..b092687 100644
--- a/configs/msmsteppe/audio_policy_configuration.xml
+++ b/configs/msmsteppe/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/msmsteppe/mixer_paths_idp.xml b/configs/msmsteppe/mixer_paths_idp.xml
index 85de3eb..2ce8d12 100644
--- a/configs/msmsteppe/mixer_paths_idp.xml
+++ b/configs/msmsteppe/mixer_paths_idp.xml
@@ -2150,6 +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 19afb26..db33f96 100644
--- a/configs/msmsteppe/mixer_paths_qrd.xml
+++ b/configs/msmsteppe/mixer_paths_qrd.xml
@@ -2175,6 +2175,10 @@
<path name="handset" />
</path>
+ <path name="voice-handset-tmus">
+ <path name="handset" />
+ </path>
+
<path name="voice-speaker">
<path name="speaker-mono" />
</path>
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 59a3abd..5c648b5 100644
--- a/configs/msmsteppe/msmsteppe.mk
+++ b/configs/msmsteppe/msmsteppe.mk
@@ -289,7 +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 += \
@@ -402,6 +402,10 @@
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 \
@@ -415,6 +419,15 @@
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_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 858077c..51829bd 100644
--- a/configs/msmsteppe_au/msmsteppe_au.mk
+++ b/configs/msmsteppe_au/msmsteppe_au.mk
@@ -186,6 +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
index 01e279d..0274f9e 100644
--- 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
@@ -23,7 +23,7 @@
<resources>
<!-- Car uses hardware amplifier for volume. -->
- <bool name="config_useFixedVolume">true</bool>
+ <bool name="config_useFixedVolume">false</bool>
<!--
Handle volume keys directly in CarAudioService without passing them to the foreground app
-->
diff --git a/configs/sdm660/sdm660.mk b/configs/sdm660/sdm660.mk
index 84f0f1e..0f66da9 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 += \
diff --git a/configs/sdm710/sdm710.mk b/configs/sdm710/sdm710.mk
index c47a146..37fe4a4 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 += \
@@ -418,6 +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/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index 0ede765..659262b 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -233,7 +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 += \
@@ -367,6 +367,15 @@
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/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 a5b7eac..5dc2ec4 100644
--- a/configs/trinket/trinket.mk
+++ b/configs/trinket/trinket.mk
@@ -187,7 +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 += \
@@ -235,6 +235,10 @@
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 \
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 f47cf70..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)
{
@@ -5063,9 +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)
{
@@ -5075,15 +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 ====================================================
@@ -5529,6 +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 b7942ae..61434b5 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 +1150,7 @@
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);
#ifdef AUDIO_HW_LOOPBACK_ENABLED
/* API to create audio patch */
int audio_extn_hw_loopback_create_audio_patch(struct audio_hw_device *dev,
diff --git a/hal/audio_extn/compress_in.c b/hal/audio_extn/compress_in.c
index 6cf6b81..bc630a3 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);
@@ -332,6 +343,6 @@
return ret;
err_config:
- cin_close_input_stream(in);
+ cin_free_input_stream_resources(in);
return ret;
}
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/utils.c b/hal/audio_extn/utils.c
index d66b368..8d9cd33 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>
@@ -1249,6 +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)
{
@@ -1442,11 +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
@@ -1481,7 +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) {
@@ -2860,3 +2948,42 @@
return is_running_with_enhanced_fwk;
}
+
+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 a8f3ad8..1fb7c21 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -2910,7 +2910,7 @@
ALOGE("%s: failed to start ext hw plugin", __func__);
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
@@ -4109,8 +4109,6 @@
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;
@@ -6169,8 +6167,6 @@
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))
@@ -6232,7 +6228,7 @@
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) {
@@ -7282,8 +7278,17 @@
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 ||
@@ -7458,6 +7463,10 @@
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.
@@ -8074,6 +8083,7 @@
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);
@@ -8176,6 +8186,21 @@
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);
@@ -8824,7 +8849,7 @@
(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 {
@@ -8875,7 +8900,7 @@
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;
}
@@ -9002,7 +9027,7 @@
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__);
@@ -9210,9 +9235,9 @@
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);
@@ -9228,6 +9253,8 @@
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);
@@ -9338,7 +9365,7 @@
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);
}
@@ -9508,6 +9535,20 @@
}
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 df29413..d84a8f7 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -399,6 +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;
diff --git a/hal/audio_hw_extn_api.c b/hal/audio_hw_extn_api.c
index af73375..fa3a818 100644
--- a/hal/audio_hw_extn_api.c
+++ b/hal/audio_hw_extn_api.c
@@ -190,6 +190,27 @@
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_format_supported(in->format)) {
+ audio_extn_cin_stop_input_stream(in);
+ }
+ }
+
+ 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/msm8974/platform.c b/hal/msm8974/platform.c
index 459c37c..3d89032 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -4814,6 +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)
@@ -7896,15 +7899,20 @@
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:
@@ -7948,6 +7956,9 @@
{
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;
@@ -7961,6 +7972,17 @@
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;
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_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;