Merge "hal: add apptype as a part of event_value"
diff --git a/configs/atoll/atoll.mk b/configs/atoll/atoll.mk
index 65d42cc..5acb773 100644
--- a/configs/atoll/atoll.mk
+++ b/configs/atoll/atoll.mk
@@ -169,6 +169,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/mixer_paths_wcd9375.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_wcd9375.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_qrd.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/mixer_paths_wcd9375qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_wcd9375qrd.xml \
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
diff --git a/configs/atoll/audio_policy_configuration.xml b/configs/atoll/audio_policy_configuration.xml
index e96660a..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>
diff --git a/configs/kona/kona.mk b/configs/kona/kona.mk
index a8991fc..33a678f 100644
--- a/configs/kona/kona.mk
+++ b/configs/kona/kona.mk
@@ -187,7 +187,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/audio_configs_stock.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs_stock.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/kona/audio_tuning_mixer.txt:$(TARGET_COPY_OUT_VENDOR)/etc/audio_tuning_mixer.txt \
- frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -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 += \
@@ -400,7 +401,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=true \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=true \
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 e8990fb..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>
diff --git a/configs/lito/lito.mk b/configs/lito/lito.mk
index 56fd30f..00876db 100644
--- a/configs/lito/lito.mk
+++ b/configs/lito/lito.mk
@@ -179,7 +179,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/lito/mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_qrd.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/lito/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/lito/audio_configs_stock.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs_stock.xml \
- frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -284,7 +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 += \
@@ -384,7 +385,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=true \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=true \
diff --git a/configs/lito/mixer_paths_qrd.xml b/configs/lito/mixer_paths_qrd.xml
index 0d8585b..b246c5a 100644
--- a/configs/lito/mixer_paths_qrd.xml
+++ b/configs/lito/mixer_paths_qrd.xml
@@ -2284,7 +2284,7 @@
</path>
<path name="speaker-protected">
- <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="1" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
<ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
<path name="speaker" />
<ctl name="VI_FEED_TX Channels" value="One" />
@@ -2292,7 +2292,7 @@
</path>
<path name="voice-speaker-protected">
- <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_1" value="1" />
+ <ctl name="WSA_AIF_VI Mixer WSA_SPKR_VI_2" value="1" />
<ctl name="WSA_CDC_DMA_0 TX Format" value="PACKED_16B" />
<path name="speaker-mono" />
<ctl name="VI_FEED_TX Channels" value="One" />
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index 72fa6f3..a41740f 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -199,10 +199,6 @@
vendor.audio.use.sw.alac.decoder=true\
vendor.audio.use.sw.ape.decoder=true
-#property for AudioSphere Post processing
-PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.pp.asphere.enabled=false
-
#Audio voice concurrency related flags
PRODUCT_PROPERTY_OVERRIDES += \
vendor.voice.playback.conc.disabled=true\
@@ -245,7 +241,6 @@
vendor.audio.feature.a2dp_offload.enable=false \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index 25d42cf..0b0e6be 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -212,10 +212,6 @@
vendor.audio.use.sw.alac.decoder=true\
vendor.audio.use.sw.ape.decoder=true
-#property for AudioSphere Post processing
-PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.pp.asphere.enabled=false
-
#Audio voice concurrency related flags
PRODUCT_PROPERTY_OVERRIDES += \
vendor.voice.playback.conc.disabled=true\
diff --git a/configs/msm8998/msm8998.mk b/configs/msm8998/msm8998.mk
index bee32c8..6b77f69 100644
--- a/configs/msm8998/msm8998.mk
+++ b/configs/msm8998/msm8998.mk
@@ -191,7 +191,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -255,7 +255,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/msmnile/msmnile.mk b/configs/msmnile/msmnile.mk
index ae79cd6..3315b11 100644
--- a/configs/msmnile/msmnile.mk
+++ b/configs/msmnile/msmnile.mk
@@ -176,7 +176,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile/sound_trigger_mixer_paths.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile/audio_configs_stock.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs_stock.xml \
- frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -280,7 +281,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -379,7 +380,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=false \
vendor.audio.feature.anc_headset.enable=false \
-vendor.audio.feature.audio_sphere.enable=false \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
@@ -424,7 +424,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=true \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=true \
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/mixer_paths_qrd.xml b/configs/msmsteppe/mixer_paths_qrd.xml
index db33f96..5665322 100644
--- a/configs/msmsteppe/mixer_paths_qrd.xml
+++ b/configs/msmsteppe/mixer_paths_qrd.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="ISO-8859-1"?>
-<!-- Copyright (c) 2015-2018, The Linux Foundation. All rights reserved. -->
+<!-- Copyright (c) 2015-2019, The Linux Foundation. All rights reserved. -->
<!-- -->
<!-- Redistribution and use in source and binary forms, with or without -->
<!-- modification, are permitted provided that the following conditions are -->
@@ -2196,7 +2196,15 @@
</path>
<path name="voice-headphones">
- <path name="headphones" />
+ <ctl name="RX_MACRO RX0 MUX" value="AIF1_PB" />
+ <ctl name="RX_MACRO RX1 MUX" value="AIF1_PB" />
+ <ctl name="RX_CDC_DMA_RX_0 Channels" value="Two" />
+ <ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
+ <ctl name="RX INT1_1 MIX1 INP0" value="RX1" />
+ <ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="HPHL_RDAC Switch" value="1" />
+ <ctl name="HPHR_RDAC Switch" value="1" />
</path>
<path name="voice-line">
diff --git a/configs/msmsteppe/msmsteppe.mk b/configs/msmsteppe/msmsteppe.mk
index 528bd7c..ec546ac 100644
--- a/configs/msmsteppe/msmsteppe.mk
+++ b/configs/msmsteppe/msmsteppe.mk
@@ -176,6 +176,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe/mixer_paths_qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_qrd.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe/mixer_paths_wcd9375qrd.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_wcd9375qrd.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe/mixer_paths_tavil.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tavil.xml \
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -287,7 +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 += \
@@ -360,7 +362,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/msmsteppe/sound_trigger_mixer_paths.xml b/configs/msmsteppe/sound_trigger_mixer_paths.xml
index a489e7f..90de0d3 100644
--- a/configs/msmsteppe/sound_trigger_mixer_paths.xml
+++ b/configs/msmsteppe/sound_trigger_mixer_paths.xml
@@ -206,11 +206,11 @@
<ctl name="TX_DEC3 Volume" value="102" />
<ctl name="TX DMIC MUX0" value="DMIC2" />
<ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
- <ctl name="TX DMIC MUX1" value="DMIC1" />
+ <ctl name="TX DMIC MUX1" value="DMIC0" />
<ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
<ctl name="TX DMIC MUX2" value="DMIC3" />
<ctl name="TX_AIF1_CAP Mixer DEC3" value="1" />
- <ctl name="TX DMIC MUX3" value="DMIC0" />
+ <ctl name="TX DMIC MUX3" value="DMIC1" />
</path>
<path name="echo-reference">
diff --git a/configs/msmsteppe/sound_trigger_mixer_paths_wcd9340.xml b/configs/msmsteppe/sound_trigger_mixer_paths_wcd9340.xml
index 55dd42f..f74c4fe 100644
--- a/configs/msmsteppe/sound_trigger_mixer_paths_wcd9340.xml
+++ b/configs/msmsteppe/sound_trigger_mixer_paths_wcd9340.xml
@@ -199,7 +199,7 @@
<ctl name= "DMIC MUX0" value="DMIC2" />
<ctl name= "DEC0 Volume" value="84" />
<ctl name= "ADC MUX1" value="DMIC" />
- <ctl name= "DMIC MUX1" value="DMIC0" />
+ <ctl name= "DMIC MUX1" value="DMIC5" />
<ctl name= "DEC1 Volume" value="84" />
<ctl name= "WDMA3 PORT0 MUX" value="DEC0" />
<ctl name= "WDMA3 PORT1 MUX" value="DEC1" />
@@ -217,7 +217,7 @@
<ctl name= "DMIC MUX1" value="DMIC0" />
<ctl name= "DEC1 Volume" value="84" />
<ctl name= "ADC MUX2" value="DMIC" />
- <ctl name= "DMIC MUX2" value="DMIC1" />
+ <ctl name= "DMIC MUX2" value="DMIC5" />
<ctl name= "DEC2 Volume" value="84" />
<ctl name= "WDMA3 PORT0 MUX" value="DEC0" />
<ctl name= "WDMA3 PORT1 MUX" value="DEC1" />
@@ -237,10 +237,10 @@
<ctl name= "DMIC MUX1" value="DMIC0" />
<ctl name= "DEC1 Volume" value="84" />
<ctl name= "ADC MUX2" value="DMIC" />
- <ctl name= "DMIC MUX2" value="DMIC1" />
+ <ctl name= "DMIC MUX2" value="DMIC5" />
<ctl name= "DEC2 Volume" value="84" />
<ctl name= "ADC MUX3" value="DMIC" />
- <ctl name= "DMIC MUX3" value="DMIC3" />
+ <ctl name= "DMIC MUX3" value="DMIC1" />
<ctl name= "DEC3 Volume" value="84" />
<ctl name= "WDMA3 PORT0 MUX" value="DEC0" />
<ctl name= "WDMA3 PORT1 MUX" value="DEC1" />
@@ -298,7 +298,7 @@
<ctl name="AIF1_CAP Mixer SLIM TX8" value="1" />
<ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC1" />
+ <ctl name="DMIC MUX7" value="DMIC2" />
<ctl name="CDC_IF TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
<ctl name="DMIC MUX8" value="DMIC5" />
@@ -312,13 +312,13 @@
<ctl name="SLIM_0_TX Channels" value="Three" />
<ctl name="CDC_IF TX5 MUX" value="DEC5" />
<ctl name="ADC MUX5" value="DMIC" />
- <ctl name="DMIC MUX5" value="DMIC1" />
+ <ctl name="DMIC MUX5" value="DMIC2" />
<ctl name="CDC_IF TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
- <ctl name="DMIC MUX6" value="DMIC5" />
+ <ctl name="DMIC MUX6" value="DMIC0" />
<ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="DMIC MUX7" value="DMIC5" />
</path>
<path name="listen-ape-handset-qmic">
@@ -329,16 +329,16 @@
<ctl name="SLIM_0_TX Channels" value="Four" />
<ctl name="CDC_IF TX5 MUX" value="DEC5" />
<ctl name="ADC MUX5" value="DMIC" />
- <ctl name="DMIC MUX5" value="DMIC1" />
+ <ctl name="DMIC MUX5" value="DMIC2" />
<ctl name="CDC_IF TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
- <ctl name="DMIC MUX6" value="DMIC5" />
+ <ctl name="DMIC MUX6" value="DMIC0" />
<ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
- <ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="DMIC MUX7" value="DMIC5" />
<ctl name="CDC_IF TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC0" />
+ <ctl name="DMIC MUX8" value="DMIC1" />
</path>
<path name="echo-reference">
diff --git a/configs/msmsteppe/sound_trigger_platform_info.xml b/configs/msmsteppe/sound_trigger_platform_info.xml
index 413f4c6..a85a180 100644
--- a/configs/msmsteppe/sound_trigger_platform_info.xml
+++ b/configs/msmsteppe/sound_trigger_platform_info.xml
@@ -54,6 +54,8 @@
</common_config>
<acdb_ids>
+ <!--For internal codec please enable below device-->
+ <!--param DEVICE_HANDSET_MIC_APE="130" /-->
<param DEVICE_HANDSET_MIC_APE="100" />
<param DEVICE_HANDSET_MIC_CPE="128" />
<param DEVICE_HANDSET_MIC_ECPP_CPE="128" />
@@ -127,6 +129,28 @@
<param read_rsp_ids="0x00020013, 0x3, 0x00020016" />
<param custom_config_ids="0x00012C0D, 0x3, 0x00012C20" />
</gcs_usecase>
+ <gcs_usecase>
+ <param uid="0x7" />
+ <param acdb_devices="DEVICE_HANDSET_DMIC_CPE" />
+ <param load_sound_model_ids="0x00012C0D, 0x7, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x7, 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x7, 0x00012B05" />
+ <param read_cmd_ids="0x00020013, 0x7, 0x00020015" />
+ <param read_rsp_ids="0x00020013, 0x7, 0x00020016" />
+ <param custom_config_ids="0x00012C0D, 0x7, 0x00012C20" />
+ <param det_event_type_ids="0x00012C0D, 0x7, 0x00012C2A" />
+ </gcs_usecase>
+ <gcs_usecase>
+ <param uid="0x8" />
+ <param acdb_devices="DEVICE_HANDSET_DMIC_CPE" />
+ <param load_sound_model_ids="0x00012C0D, 0x8, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x8, 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x8, 0x00012B05" />
+ <param read_cmd_ids="0x00020013, 0x8, 0x00020015" />
+ <param read_rsp_ids="0x00020013, 0x8, 0x00020016" />
+ <param custom_config_ids="0x00012C0D, 0x8, 0x00012C20" />
+ <param det_event_type_ids="0x00012C0D, 0x8, 0x00012C2A" />
+ </gcs_usecase>
<!-- Module and param ids with which the algorithm is integrated
in non-graphite firmware (note these must come after gcs params)
Extends flexibility to have different ids based on execution type.
diff --git a/configs/qssi/qssi.mk b/configs/qssi/qssi.mk
index 39569fb..653c177 100644
--- a/configs/qssi/qssi.mk
+++ b/configs/qssi/qssi.mk
@@ -106,6 +106,10 @@
PRODUCT_PRODUCT_PROPERTIES += \
audio.sys.noisy.broadcast.delay=600
+#offload minimum duration set to 30sec
+PRODUCT_PRODUCT_PROPERTIES += \
+audio.offload.min.duration.secs=30
+
#offload pausetime out duration to 3 secs to inline with other outputs
PRODUCT_PRODUCT_PROPERTIES += \
audio.sys.offload.pstimeout.secs=3
diff --git a/configs/sdm660/sdm660.mk b/configs/sdm660/sdm660.mk
index 84f0f1e..5695851 100644
--- a/configs/sdm660/sdm660.mk
+++ b/configs/sdm660/sdm660.mk
@@ -202,7 +202,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -266,7 +266,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/sdm710/sdm710.mk b/configs/sdm710/sdm710.mk
index 177562a..fb01728 100644
--- a/configs/sdm710/sdm710.mk
+++ b/configs/sdm710/sdm710.mk
@@ -296,7 +296,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -365,7 +365,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index 80ff74b..c3c3578 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 += \
@@ -314,7 +314,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
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 5d74497..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>
diff --git a/configs/trinket/trinket.mk b/configs/trinket/trinket.mk
index e5f2f56..5176889 100644
--- a/configs/trinket/trinket.mk
+++ b/configs/trinket/trinket.mk
@@ -92,6 +92,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/trinket/mixer_paths_tavil.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tavil.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/trinket/mixer_paths_tasha.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tasha.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/trinket/mixer_paths_tashalite.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths_tashalite.xml \
+ frameworks/native/data/etc/android.hardware.audio.pro.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.pro.xml \
+ frameworks/native/data/etc/android.hardware.audio.low_latency.xml:$(TARGET_COPY_OUT_VENDOR)/etc/permissions/android.hardware.audio.low_latency.xml
#XML Audio configuration files
ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -185,7 +187,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -242,7 +244,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index a2a52a8..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 ====================================================
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index d2b8309..18bfc2a 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);
@@ -1067,12 +1070,13 @@
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 ==============================================
@@ -1147,6 +1151,8 @@
uint64_t *frames, struct timespec *timestamp, int32_t clock_id);
int audio_extn_utils_pcm_get_dsp_presentation_pos(struct stream_out *out,
uint64_t *frames, struct timespec *timestamp, int32_t clock_id);
+size_t audio_extn_utils_get_input_buffer_size(uint32_t, audio_format_t, int, int64_t, bool);
+int audio_extn_utils_get_perf_mode_flag(void);
#ifdef AUDIO_HW_LOOPBACK_ENABLED
/* API to create audio patch */
int audio_extn_hw_loopback_create_audio_patch(struct audio_hw_device *dev,
diff --git a/hal/audio_extn/auto_hal.c b/hal/audio_extn/auto_hal.c
index f008a47..ad5e331 100644
--- a/hal/audio_extn/auto_hal.c
+++ b/hal/audio_extn/auto_hal.c
@@ -445,7 +445,7 @@
*/
#define MIN_VOLUME_VALUE_MB -6000
#define MAX_VOLUME_VALUE_MB 600
-
+#define STEP_VALUE_MB 100
int audio_extn_auto_hal_set_audio_port_config(struct audio_hw_device *dev,
const struct audio_port_config *config)
{
@@ -498,7 +498,10 @@
/* millibel = 1/100 dB = 1/1000 bel
* q13 = (10^(mdb/100/20))*(2^13)
*/
- volume = powf(10.0, ((float)config->gain.values[0] / 2000));
+ if(config->gain.values[0] <= (MIN_VOLUME_VALUE_MB + STEP_VALUE_MB))
+ volume = 0.0 ;
+ else
+ volume = powf(10.0, ((float)config->gain.values[0] / 2000));
ALOGV("%s: set volume to stream: %p", __func__,
&out_ctxt->output->stream);
/* set gain if output stream is active */
diff --git a/hal/audio_extn/compress_in.c b/hal/audio_extn/compress_in.c
index 6cf6b81..6b525b0 100644
--- a/hal/audio_extn/compress_in.c
+++ b/hal/audio_extn/compress_in.c
@@ -100,7 +100,7 @@
* only after validating that input against cin_attached_usecase
* except below calls
* 1. cin_applicable_stream(in)
- * 2. cin_configure_input_stream(in)
+ * 2. cin_configure_input_stream(in, in_config)
*/
bool cin_attached_usecase(audio_usecase_t uc_id)
@@ -179,7 +179,7 @@
return sz;
}
-int cin_start_input_stream(struct stream_in *in)
+int cin_open_input_stream(struct stream_in *in)
{
int ret = -EINVAL;
struct audio_device *adev = in->dev;
@@ -208,12 +208,23 @@
ALOGV("%s: in %p, cin_data %p", __func__, in, cin_data);
if (cin_data->compr) {
+ compress_stop(cin_data->compr);
+ }
+}
+
+
+void cin_close_input_stream(struct stream_in *in)
+{
+ cin_private_data_t *cin_data = (cin_private_data_t *) in->cin_extn;
+
+ ALOGV("%s: in %p, cin_data %p", __func__, in, cin_data);
+ if (cin_data->compr) {
compress_close(cin_data->compr);
cin_data->compr = NULL;
}
}
-void cin_close_input_stream(struct stream_in *in)
+void cin_free_input_stream_resources(struct stream_in *in)
{
cin_private_data_t *cin_data = (cin_private_data_t *) in->cin_extn;
@@ -265,9 +276,8 @@
return ret;
}
-int cin_configure_input_stream(struct stream_in *in)
+int cin_configure_input_stream(struct stream_in *in, struct audio_config *in_config)
{
- struct audio_device *adev = in->dev;
struct audio_config config = {.format = 0};
int ret = 0, buffer_size = 0, meta_size = sizeof(struct snd_codec_metadata);
cin_private_data_t *cin_data = NULL;
@@ -304,7 +314,8 @@
config.channel_mask = in->channel_mask;
config.format = in->format;
in->config.channels = audio_channel_count_from_in_mask(in->channel_mask);
- buffer_size = adev->device.get_input_buffer_size(&adev->device, &config);
+ buffer_size = audio_extn_utils_get_input_buffer_size(config.sample_rate, config.format,
+ in->config.channels, in_config->offload_info.duration_us / 1000, false);
cin_data->compr_config.fragment_size = buffer_size;
cin_data->compr_config.codec->id = get_snd_codec_id(in->format);
@@ -321,6 +332,11 @@
else
cin_data->compr_config.codec->compr_passthr = PASSTHROUGH_GEN;
+ if (in->flags & AUDIO_INPUT_FLAG_FAST) {
+ ALOGD("%s: Setting latency mode to true", __func__);
+ cin_data->compr_config.codec->flags |= audio_extn_utils_get_perf_mode_flag();
+ }
+
if ((in->flags & AUDIO_INPUT_FLAG_TIMESTAMP) ||
(in->flags & AUDIO_INPUT_FLAG_PASSTHROUGH)) {
compress_config_set_timstamp_flag(&cin_data->compr_config);
@@ -332,6 +348,6 @@
return ret;
err_config:
- cin_close_input_stream(in);
+ cin_free_input_stream_resources(in);
return ret;
}
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index d66b368..30bc10d 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -37,6 +37,7 @@
#include "platform.h"
#include "platform_api.h"
#include "audio_extn.h"
+#include "voice_extn.h"
#include "voice.h"
#include <sound/compress_params.h>
#include <sound/compress_offload.h>
@@ -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 +2948,51 @@
return is_running_with_enhanced_fwk;
}
+
+int audio_extn_utils_get_perf_mode_flag(void)
+{
+#ifdef COMPRESSED_PERF_MODE_FLAG
+ return COMPRESSED_PERF_MODE_FLAG;
+#else
+ return 0;
+#endif
+}
+
+size_t audio_extn_utils_get_input_buffer_size(uint32_t sample_rate,
+ audio_format_t format,
+ int channel_count,
+ int64_t duration_ms,
+ bool is_low_latency)
+{
+ size_t size = 0;
+ size_t capture_duration = AUDIO_CAPTURE_PERIOD_DURATION_MSEC;
+ uint32_t bytes_per_period_sample = 0;
+
+
+ if (audio_extn_utils_check_input_parameters(sample_rate, format, channel_count) != 0)
+ return 0;
+
+ if (duration_ms >= MIN_OFFLOAD_BUFFER_DURATION_MS && duration_ms <= MAX_OFFLOAD_BUFFER_DURATION_MS)
+ capture_duration = duration_ms;
+
+ size = (sample_rate * capture_duration) / 1000;
+ if (is_low_latency)
+ size = LOW_LATENCY_CAPTURE_PERIOD_SIZE;
+
+
+ bytes_per_period_sample = audio_bytes_per_sample(format) * channel_count;
+ size *= bytes_per_period_sample;
+
+ /* make sure the size is multiple of 32 bytes and additionally multiple of
+ * the frame_size (required for 24bit samples and non-power-of-2 channel counts)
+ * At 48 kHz mono 16-bit PCM:
+ * 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
+ * 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
+ *
+ * The loop reaches result within 32 iterations, as initial size is
+ * already a multiple of frame_size
+ */
+ size = audio_extn_utils_nearest_multiple(size, audio_extn_utils_lcm(32, bytes_per_period_sample));
+
+ return size;
+}
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 5baa9ad..7d8103b 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -2912,8 +2912,10 @@
if (audio_extn_ext_hw_plugin_usecase_start(adev->ext_hw_plugin, uc_info))
ALOGE("%s: failed to start ext hw plugin", __func__);
+ android_atomic_acquire_cas(true, false, &(in->capture_stopped));
+
if (audio_extn_cin_attached_usecase(in->usecase)) {
- ret = audio_extn_cin_start_input_stream(in);
+ ret = audio_extn_cin_open_input_stream(in);
if (ret)
goto error_open;
else
@@ -6231,7 +6233,7 @@
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) {
@@ -6535,6 +6537,13 @@
in->standby = 0;
}
+ /* Avoid read if capture_stopped is set */
+ if (android_atomic_acquire_load(&(in->capture_stopped)) > 0) {
+ ALOGD("%s: force stopped catpure session, ignoring read request", __func__);
+ ret = -EINVAL;
+ goto exit;
+ }
+
// what's the duration requested by the client?
long ns = 0;
@@ -7383,6 +7392,11 @@
ALOGV("non-offload DIRECT_usecase ... usecase selected %d ", out->usecase);
}
+ if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
+ ALOGD("%s: Setting latency mode to true", __func__);
+ out->compr_config.codec->flags |= audio_extn_utils_get_perf_mode_flag();
+ }
+
if (out->usecase == USECASE_INVALID) {
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
config->format == 0 && config->sample_rate == 0 &&
@@ -7466,6 +7480,10 @@
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.
@@ -8214,13 +8232,17 @@
if (usecase->stream.out && (usecase->type == PCM_PLAYBACK) &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP)){
ALOGD("reconfigure a2dp... forcing device switch");
-
pthread_mutex_unlock(&adev->lock);
lock_output_stream(usecase->stream.out);
pthread_mutex_lock(&adev->lock);
audio_extn_a2dp_set_handoff_mode(true);
+ ALOGD("Switching to speaker and muting the stream before select_devices");
+ check_a2dp_restore_l(adev, usecase->stream.out, false);
//force device switch to re configure encoder
select_devices(adev, usecase->id);
+ ALOGD("Unmuting the stream after select_devices");
+ usecase->stream.out->a2dp_compress_mute = false;
+ out_set_compr_volume(&usecase->stream.out->stream, usecase->stream.out->volume_l, usecase->stream.out->volume_r);
audio_extn_a2dp_set_handoff_mode(false);
pthread_mutex_unlock(&usecase->stream.out->lock);
break;
@@ -8728,6 +8750,8 @@
}
if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
+ (flags & AUDIO_INPUT_FLAG_TIMESTAMP) == 0 &&
+ (flags & AUDIO_INPUT_FLAG_COMPRESS) == 0 &&
(flags & AUDIO_INPUT_FLAG_FAST) != 0) {
is_low_latency = true;
#if LOW_LATENCY_CAPTURE_USE_CASE
@@ -8848,7 +8872,7 @@
(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 {
@@ -8899,7 +8923,7 @@
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;
}
@@ -9026,7 +9050,7 @@
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__);
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index d84a8f7..4810896 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -473,6 +473,8 @@
float zoom;
audio_microphone_direction_t direction;
+ volatile int32_t capture_stopped;
+
/* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
audio_format_t supported_formats[MAX_SUPPORTED_FORMATS + 1];
diff --git a/hal/audio_hw_extn_api.c b/hal/audio_hw_extn_api.c
index af73375..22c8685 100644
--- a/hal/audio_hw_extn_api.c
+++ b/hal/audio_hw_extn_api.c
@@ -34,6 +34,7 @@
#include <inttypes.h>
#include <errno.h>
#include <log/log.h>
+#include <cutils/atomic.h>
#include <hardware/audio.h>
#include "sound/compress_params.h"
@@ -190,6 +191,31 @@
return ret;
}
+int qahwi_in_stop(struct audio_stream_in* stream) {
+ struct stream_in *in = (struct stream_in *)stream;
+ struct audio_device *adev = in->dev;
+
+ ALOGD("%s processing, in %p", __func__, in);
+
+ pthread_mutex_lock(&adev->lock);
+
+ if (!in->standby) {
+ if (in->pcm != NULL ) {
+ pcm_stop(in->pcm);
+ } else if (audio_extn_cin_attached_usecase(in->usecase)) {
+ audio_extn_cin_stop_input_stream(in);
+ }
+
+ /* Set the atomic variable when the session is stopped */
+ if (android_atomic_acquire_cas(false, true, &(in->capture_stopped)) == 0)
+ ALOGI("%s: capture_stopped bit set", __func__);
+ }
+
+ pthread_mutex_unlock(&adev->lock);
+
+ return 0;
+}
+
ssize_t qahwi_in_read_v2(struct audio_stream_in *stream, void* buffer,
size_t bytes, uint64_t *timestamp)
{
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index e6989f6..8b9b53d 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -4206,7 +4206,7 @@
* enforced audible (e.g. Camera shutter sound).
*/
if ((mode == AUDIO_MODE_IN_CALL) ||
- voice_is_in_call(adev) ||
+ voice_check_voicecall_usecases_active(adev) ||
voice_extn_compress_voip_is_active(adev))
is_active_voice_call = true;
@@ -4289,7 +4289,7 @@
}
if ((mode == AUDIO_MODE_IN_CALL) ||
- voice_is_in_call(adev) ||
+ voice_check_voicecall_usecases_active(adev) ||
voice_extn_compress_voip_is_active(adev)) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
@@ -4665,8 +4665,10 @@
ALOGV("%s: enter: out_device(%#x) in_device(%#x) channel_count (%d) channel_mask (0x%x)",
__func__, out_device, in_device, channel_count, channel_mask);
if (my_data->external_mic) {
- if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) || voice_is_in_call(adev) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_check_voicecall_usecases_active(adev) ||
+ voice_extn_compress_voip_is_active(adev) ||
+ audio_extn_hfp_is_active(adev))) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
out_device & AUDIO_DEVICE_OUT_EARPIECE ||
out_device & AUDIO_DEVICE_OUT_SPEAKER )
@@ -4680,8 +4682,10 @@
if (snd_device != AUDIO_DEVICE_NONE)
goto exit;
- if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) || voice_is_in_call(adev) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_check_voicecall_usecases_active(adev) ||
+ voice_extn_compress_voip_is_active(adev) ||
+ audio_extn_hfp_is_active(adev))) {
if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
!voice_extn_compress_voip_is_active(adev)) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 676a30d..1b14d63 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)
@@ -5591,7 +5594,7 @@
* enforced audible (e.g. Camera shutter sound).
*/
if ((mode == AUDIO_MODE_IN_CALL) ||
- voice_is_in_call(adev) ||
+ voice_check_voicecall_usecases_active(adev) ||
voice_extn_compress_voip_is_active(adev))
is_active_voice_call = true;
@@ -5715,7 +5718,7 @@
}
if ((mode == AUDIO_MODE_IN_CALL) ||
- voice_is_in_call(adev) ||
+ voice_check_voicecall_usecases_active(adev) ||
voice_extn_compress_voip_is_active(adev) ||
adev->enable_voicerx ||
audio_extn_hfp_is_active(adev)) {
@@ -6212,8 +6215,10 @@
ALOGV("%s: enter: out_device(%#x) in_device(%#x) channel_count (%d) channel_mask (0x%x)",
__func__, out_device, in_device, channel_count, channel_mask);
if (my_data->external_mic) {
- if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) || voice_is_in_call(adev) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_check_voicecall_usecases_active(adev) ||
+ voice_extn_compress_voip_is_active(adev) ||
+ audio_extn_hfp_is_active(adev))) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
out_device & AUDIO_DEVICE_OUT_EARPIECE ||
out_device & AUDIO_DEVICE_OUT_SPEAKER )
@@ -6227,8 +6232,10 @@
if (snd_device != AUDIO_DEVICE_NONE)
goto exit;
- if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) || voice_is_in_call(adev) ||
- voice_extn_compress_voip_is_active(adev) || audio_extn_hfp_is_active(adev))) {
+ if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+ voice_check_voicecall_usecases_active(adev) ||
+ voice_extn_compress_voip_is_active(adev) ||
+ audio_extn_hfp_is_active(adev))) {
if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
!voice_extn_compress_voip_is_active(adev)) {
if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
@@ -7953,6 +7960,9 @@
{
char value[PROPERTY_VALUE_MAX] = {0};
uint32_t fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ uint32_t new_fragment_size = 0;
+ int32_t duration_ms = 0;
+ int channel_count = 0;
if((property_get("vendor.audio.offload.buffer.size.kb", value, "")) &&
atoi(value)) {
fragment_size = atoi(value) * 1024;
@@ -7966,6 +7976,17 @@
fragment_size = info->offload_buffer_size;
}
+ /* Use client specified buffer size if mentioned */
+ if ((info != NULL) && (info->duration_us > 0)) {
+ duration_ms = info->duration_us / 1000;
+ channel_count = audio_channel_count_from_in_mask(info->channel_mask);
+
+ new_fragment_size = (duration_ms * info->sample_rate * channel_count * audio_bytes_per_sample(info->format)) / 1000;
+ ALOGI("%s:: Overwriting offload buffer size with client requested size old:%d new:%d", __func__, fragment_size, new_fragment_size);
+
+ fragment_size = new_fragment_size;
+ }
+
if (info != NULL) {
if (info->is_streaming && info->has_video) {
fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
diff --git a/hal/voice.c b/hal/voice.c
index 729ab27..006dd08 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -377,6 +377,22 @@
return session_id;
}
+bool voice_check_voicecall_usecases_active(struct audio_device *adev)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase = NULL;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == VOICE_CALL) {
+ ALOGV("%s: voice usecase:%s is active", __func__,
+ use_case_table[usecase->id]);
+ return true;
+ }
+ }
+ return false;
+}
+
int voice_check_and_set_incall_rec_usecase(struct audio_device *adev,
struct stream_in *in)
{
diff --git a/hal/voice.h b/hal/voice.h
index 9612edd..188345d 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -101,4 +101,5 @@
bool voice_is_call_state_active(struct audio_device *adev);
void voice_set_device_mute_flag (struct audio_device *adev, bool state);
snd_device_t voice_get_incall_rec_backend_device(struct stream_in *in);
+bool voice_check_voicecall_usecases_active(struct audio_device *adev);
#endif //VOICE_H
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 6b1afd3..fb42514 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -30,8 +30,7 @@
virtualizer.c \
reverb.c \
effect_api.c \
- effect_util.c \
- asphere.c
+ effect_util.c
# HW_ACCELERATED has been disabled by default since msm8996. File doesn't
# compile cleanly on tip so don't want to include it, but keeping this
diff --git a/post_proc/Makefile.am b/post_proc/Makefile.am
index bd29473..8bd41ae 100644
--- a/post_proc/Makefile.am
+++ b/post_proc/Makefile.am
@@ -19,10 +19,6 @@
c_sources += hw_accelerator.c
endif
-if AUDIOSPHERE
-c_sources += asphere.c
-endif
-
library_include_HEADERS = $(h_sources)
library_includedir = $(includedir)
diff --git a/post_proc/asphere.c b/post_proc/asphere.c
deleted file mode 100644
index efe07c6..0000000
--- a/post_proc/asphere.c
+++ /dev/null
@@ -1,323 +0,0 @@
-/* Copyright (c) 2015, 2017, 2019 The Linux Foundation. All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are
- * met:
- * * Redistributions of source code must retain the above copyright
- * notice, this list of conditions and the following disclaimer.
- * * Redistributions in binary form must reproduce the above
- * copyright notice, this list of conditions and the following
- * disclaimer in the documentation and/or other materials provided
- * with the distribution.
- * * Neither the name of The Linux Foundation nor the names of its
- * contributors may be used to endorse or promote products derived
- * from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
- * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
- * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
- * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
- * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
- * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
- * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
- * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- *
- */
-#define LOG_TAG "audio_pp_asphere"
-/*#define LOG_NDEBUG 0*/
-
-#include <errno.h>
-#include <fcntl.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <stdbool.h>
-#include <sys/stat.h>
-#include <log/log.h>
-#include <cutils/list.h>
-#include <cutils/str_parms.h>
-#include <cutils/properties.h>
-#include <hardware/audio_effect.h>
-#include <pthread.h>
-#include "bundle.h"
-#include "equalizer.h"
-#include "bass_boost.h"
-#include "virtualizer.h"
-#include "reverb.h"
-#include "asphere.h"
-
-#define ASPHERE_MIXER_NAME "MSM ASphere Set Param"
-
-#define AUDIO_PARAMETER_KEY_ASPHERE_STATUS "asphere_status"
-#define AUDIO_PARAMETER_KEY_ASPHERE_ENABLE "asphere_enable"
-#define AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH "asphere_strength"
-
-#define AUDIO_ASPHERE_EVENT_NODE "/data/misc/audio_pp/event_node"
-
-enum {
- ASPHERE_ACTIVE = 0,
- ASPHERE_SUSPENDED,
- ASPHERE_ERROR
-};
-
-#ifdef AUDIO_FEATURE_ENABLED_GCOV
-extern void __gcov_flush();
-static void enable_gcov()
-{
- __gcov_flush();
-}
-#else
-static void enable_gcov()
-{
-}
-#endif
-
-struct asphere_module {
- bool enabled;
- int status;
- int strength;
- pthread_mutex_t lock;
- int init_status;
-};
-
-static struct asphere_module asphere;
-pthread_once_t asphere_once = PTHREAD_ONCE_INIT;
-
-static int asphere_create_app_notification_node(void)
-{
- int fd;
- if ((fd = open(AUDIO_ASPHERE_EVENT_NODE, O_CREAT|O_TRUNC|O_WRONLY,
- S_IRUSR|S_IWUSR|S_IRGRP|S_IROTH)) < 0) {
- ALOGE("creating notification node failed %d", errno);
- return -EINVAL;
- }
- chmod(AUDIO_ASPHERE_EVENT_NODE, S_IRWXU|S_IRGRP|S_IXGRP|S_IROTH);
- close(fd);
- ALOGD("%s: successfully created notification node %s",
- __func__, AUDIO_ASPHERE_EVENT_NODE);
- return 0;
-}
-
-static int asphere_notify_app(void)
-{
- int fd;
- if ((fd = open(AUDIO_ASPHERE_EVENT_NODE, O_TRUNC|O_WRONLY)) < 0) {
- ALOGE("opening notification node failed %d", errno);
- return -EINVAL;
- }
- close(fd);
- ALOGD("%s: successfully opened notification node", __func__);
- return 0;
-}
-
-static int asphere_get_values_from_mixer(void)
-{
- int ret = 0;
- long val[2] = {-1, -1};
- struct mixer_ctl *ctl = NULL;
- struct mixer *mixer = mixer_open(MIXER_CARD);
- if (mixer)
- ctl = mixer_get_ctl_by_name(mixer, ASPHERE_MIXER_NAME);
- if (!ctl) {
- ALOGE("%s: could not get ctl for mixer cmd - %s",
- __func__, ASPHERE_MIXER_NAME);
- return -EINVAL;
- }
- ret = mixer_ctl_get_array(ctl, val, sizeof(val)/sizeof(val[0]));
- if (!ret) {
- asphere.enabled = (val[0] == 0) ? false : true;
- asphere.strength = val[1];
- }
- ALOGD("%s: returned %d, enabled:%ld, strength:%ld",
- __func__, ret, val[0], val[1]);
-
- return ret;
-}
-
-static int asphere_set_values_to_mixer(void)
-{
- int ret = 0;
- long val[2] = {-1, -1};
- struct mixer_ctl *ctl = NULL;
- struct mixer *mixer = mixer_open(MIXER_CARD);
- if (mixer)
- ctl = mixer_get_ctl_by_name(mixer, ASPHERE_MIXER_NAME);
- if (!ctl) {
- ALOGE("%s: could not get ctl for mixer cmd - %s",
- __func__, ASPHERE_MIXER_NAME);
- return -EINVAL;
- }
- val[0] = ((asphere.status == ASPHERE_ACTIVE) && asphere.enabled) ? 1 : 0;
- val[1] = asphere.strength;
-
- ret = mixer_ctl_set_array(ctl, val, sizeof(val)/sizeof(val[0]));
- ALOGD("%s: returned %d, enabled:%ld, strength:%ld",
- __func__, ret, val[0], val[1]);
-
- return ret;
-}
-
-static void asphere_init_once() {
- ALOGD("%s", __func__);
- pthread_mutex_init(&asphere.lock, NULL);
-
- if (property_get_bool("vendor.audio.feature.audio_sphere.enable", false)) {
- asphere.init_status = 1;
- asphere_get_values_from_mixer();
- asphere_create_app_notification_node();
- return;
- } else {
- ALOGW("%s: asphere feature not enabled", __func__);
- }
-
- asphere.init_status = 0;
-}
-
-static int asphere_init() {
- pthread_once(&asphere_once, asphere_init_once);
- enable_gcov();
- return asphere.init_status;
-}
-
-static bool asphere_parms_allowed(struct str_parms *parms)
-{
- if (str_parms_has_key(parms, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE))
- return true;
- if (str_parms_has_key(parms, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH))
- return true;
- if (str_parms_has_key(parms, AUDIO_PARAMETER_KEY_ASPHERE_STATUS))
- return true;
-
- return false;
-}
-
-void asphere_set_parameters(struct str_parms *parms)
-{
- int ret = 0;
- bool enable = false;
- int strength = -1;
- char value[32] = {0};
- bool set_enable = false, set_strength = false;
-
- if (!asphere_parms_allowed(parms)) {
- return;
- }
-
- if (asphere_init() != 1) {
- ALOGW("%s: init check failed!!!", __func__);
- return;
- }
-
- ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
- value, sizeof(value));
- if (ret > 0) {
- enable = (atoi(value) == 1) ? true : false;
- set_enable = true;
- }
-
- ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
- value, sizeof(value));
- if (ret > 0) {
- strength = atoi(value);
- if (strength >= 0 && strength <= 1000)
- set_strength = true;
- }
-
- if (set_enable || set_strength) {
- pthread_mutex_lock(&asphere.lock);
- asphere.enabled = set_enable ? enable : asphere.enabled;
- asphere.strength = set_strength ? strength : asphere.strength;
- ret = asphere_set_values_to_mixer();
- pthread_mutex_unlock(&asphere.lock);
- ALOGV("%s: exit ret %d", __func__, ret);
- }
-}
-
-void asphere_get_parameters(struct str_parms *query,
- struct str_parms *reply)
-{
- char value[32] = {0};
- int ret;
-
- if (asphere_init() != 1) {
- ALOGW("%s: init check failed!!!", __func__);
- return;
- }
-
- ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_STATUS,
- value, sizeof(value));
- if (ret >= 0) {
- str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_STATUS,
- asphere.status);
- }
-
- ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
- value, sizeof(value));
- if (ret >= 0) {
- str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_ENABLE,
- asphere.enabled ? 1 : 0);
- }
-
- ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
- value, sizeof(value));
- if (ret >= 0) {
- str_parms_add_int(reply, AUDIO_PARAMETER_KEY_ASPHERE_STRENGTH,
- asphere.strength);
- }
-}
-
-static bool effect_needs_asphere_concurrency_handling(effect_context_t *context)
-{
- if (memcmp(&context->desc->type,
- &equalizer_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
- memcmp(&context->desc->type,
- &bassboost_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
- memcmp(&context->desc->type,
- &virtualizer_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
- memcmp(&context->desc->type,
- &ins_preset_reverb_descriptor.type, sizeof(effect_uuid_t)) == 0 ||
- memcmp(&context->desc->type,
- &ins_env_reverb_descriptor.type, sizeof(effect_uuid_t)) == 0)
- return true;
-
- return false;
-}
-
-void handle_asphere_on_effect_enabled(bool enable,
- effect_context_t *context,
- struct listnode *created_effects)
-{
- struct listnode *node;
-
- ALOGV("%s: effect %0x", __func__, context->desc->type.timeLow);
- if (asphere_init() != 1) {
- ALOGW("%s: init check failed!!!", __func__);
- return;
- }
-
- if (!effect_needs_asphere_concurrency_handling(context)) {
- ALOGV("%s: effect %0x, do not need concurrency handling",
- __func__, context->desc->type.timeLow);
- return;
- }
-
- list_for_each(node, created_effects) {
- effect_context_t *fx_ctxt = node_to_item(node,
- effect_context_t,
- effects_list_node);
- if (fx_ctxt != NULL &&
- effect_needs_asphere_concurrency_handling(fx_ctxt) == true &&
- fx_ctxt != context && effect_is_active(fx_ctxt) == true) {
- ALOGV("%s: found another effect %0x, skip processing %0x", __func__,
- fx_ctxt->desc->type.timeLow, context->desc->type.timeLow);
- return;
- }
- }
- pthread_mutex_lock(&asphere.lock);
- asphere.status = enable ? ASPHERE_SUSPENDED : ASPHERE_ACTIVE;
- asphere_set_values_to_mixer();
- asphere_notify_app();
- pthread_mutex_unlock(&asphere.lock);
-}
diff --git a/post_proc/asphere.h b/post_proc/asphere.h
deleted file mode 100644
index 3babd1d..0000000
--- a/post_proc/asphere.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/* Copyright (c) 2015, The Linux Foundation. All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are
- * met:
- * * Redistributions of source code must retain the above copyright
- * notice, this list of conditions and the following disclaimer.
- * * Redistributions in binary form must reproduce the above
- * copyright notice, this list of conditions and the following
- * disclaimer in the documentation and/or other materials provided
- * with the distribution.
- * * Neither the name of The Linux Foundation nor the names of its
- * contributors may be used to endorse or promote products derived
- * from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
- * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
- * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
- * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
- * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
- * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
- * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
- * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- *
- */
-
-#ifndef OFFLOAD_ASPHERE_H_
-#define OFFLOAD_ASPHERE_H_
-
-#include <cutils/str_parms.h>
-#include <cutils/list.h>
-#include "bundle.h"
-
-void asphere_get_parameters(struct str_parms *query,
- struct str_parms *reply);
-void asphere_set_parameters(struct str_parms *reply);
-void handle_asphere_on_effect_enabled(bool enable,
- effect_context_t *context,
- struct listnode *created_effects);
-
-#endif /* OFFLOAD_ASPHERE_H_ */
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index 1e6b91d..0dbf27b 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -40,6 +40,7 @@
#include <stdlib.h>
#include <cutils/list.h>
+#include <cutils/str_parms.h>
#include <log/log.h>
#include <system/thread_defs.h>
#include <tinyalsa/asoundlib.h>
@@ -53,7 +54,6 @@
#include "bass_boost.h"
#include "virtualizer.h"
#include "reverb.h"
-#include "asphere.h"
#ifdef DTS_EAGLE
#include "effect_util.h"
@@ -455,20 +455,16 @@
/*
* Effect Bundle Set and get param operations.
- * currently only handles audio sphere scenario,
- * but the interface itself can be utilized for any effect.
*/
__attribute__ ((visibility ("default")))
-void offload_effects_bundle_get_parameters(struct str_parms *query,
- struct str_parms *reply)
+void offload_effects_bundle_get_parameters(struct str_parms *query __unused,
+ struct str_parms *reply __unused)
{
- asphere_get_parameters(query, reply);
}
__attribute__ ((visibility ("default")))
-void offload_effects_bundle_set_parameters(struct str_parms *parms)
+void offload_effects_bundle_set_parameters(struct str_parms *parms __unused)
{
- asphere_set_parameters(parms);
}
/*
@@ -826,7 +822,6 @@
status = -ENOSYS;
goto exit;
}
- handle_asphere_on_effect_enabled(true, context, &created_effects_list);
context->state = EFFECT_STATE_ACTIVE;
if (context->ops.enable)
context->ops.enable(context);
@@ -841,7 +836,6 @@
status = -ENOSYS;
goto exit;
}
- handle_asphere_on_effect_enabled(false, context, &created_effects_list);
context->state = EFFECT_STATE_INITIALIZED;
if (context->ops.disable)
context->ops.disable(context);
diff --git a/qahw/inc/qahw.h b/qahw/inc/qahw.h
index dd5b403..5020c8f 100644
--- a/qahw/inc/qahw.h
+++ b/qahw/inc/qahw.h
@@ -358,6 +358,10 @@
ssize_t qahw_in_read_l(qahw_stream_handle_t *in_handle,
qahw_in_buffer_t *in_buf);
/*
+ * Stop input stream. Returns zero on success.
+ */
+int qahw_in_stop_l(qahw_stream_handle_t *in_handle);
+/*
* Return the amount of input frames lost in the audio driver since the
* last call of this function.
* Audio driver is expected to reset the value to 0 and restart counting
diff --git a/qahw/src/qahw.c b/qahw/src/qahw.c
index 3390c26..545152c 100644
--- a/qahw/src/qahw.c
+++ b/qahw/src/qahw.c
@@ -61,6 +61,8 @@
typedef uint64_t (*qahwi_in_read_v2_t)(audio_stream_in_t *in, void* buffer,
size_t bytes, int64_t *timestamp);
+typedef int (*qahwi_in_stop_t)(audio_stream_in_t *in);
+
typedef int (*qahwi_out_set_param_data_t)(struct audio_stream_out *out,
qahw_param_id param_id,
qahw_param_payload *payload);
@@ -109,6 +111,7 @@
struct listnode list;
pthread_mutex_t lock;
qahwi_in_read_v2_t qahwi_in_read_v2;
+ qahwi_in_stop_t qahwi_in_stop;
} qahw_stream_in_t;
typedef enum {
@@ -1035,6 +1038,31 @@
}
/*
+ * Stop input stream. Returns zero on success.
+ */
+int qahw_in_stop_l(qahw_stream_handle_t *in_handle)
+{
+ int rc = -EINVAL;
+ qahw_stream_in_t *qahw_stream_in = (qahw_stream_in_t *)in_handle;
+ audio_stream_in_t *in = NULL;
+
+ if (!is_valid_qahw_stream_l((void *)qahw_stream_in, STREAM_DIR_IN)) {
+ ALOGV("%s::Invalid in handle %p", __func__, in_handle);
+ goto exit;
+ }
+ ALOGD("%s", __func__);
+
+ in = qahw_stream_in->stream;
+
+ if (qahw_stream_in->qahwi_in_stop)
+ rc = qahw_stream_in->qahwi_in_stop(in);
+ ALOGD("%s: exit", __func__);
+
+exit:
+ return rc;
+}
+
+/*
* Return the amount of input frames lost in the audio driver since the
* last call of this function.
* Audio driver is expected to reset the value to 0 and restart counting
@@ -1718,6 +1746,7 @@
qahw_module_t *qahw_module_temp = NULL;
audio_hw_device_t *audio_device = NULL;
qahw_stream_in_t *qahw_stream_in = NULL;
+ const char *error;
pthread_mutex_lock(&qahw_module_init_lock);
qahw_module_temp = get_qahw_module_by_ptr_l(qahw_module);
@@ -1747,6 +1776,7 @@
if (rc) {
ALOGE("%s::open input stream failed %d",__func__, rc);
free(qahw_stream_in);
+ goto exit;
} else {
qahw_stream_in->module = hw_module;
*in_handle = (void *)qahw_stream_in;
@@ -1757,7 +1787,6 @@
/* dlsym qahwi_in_read_v2 if timestamp flag is used */
if (!rc && ((flags & QAHW_INPUT_FLAG_TIMESTAMP) ||
(flags & QAHW_INPUT_FLAG_PASSTHROUGH))) {
- const char *error;
/* clear any existing errors */
dlerror();
@@ -1769,7 +1798,16 @@
}
}
-exit:
+ /* clear any existing errors */
+ dlerror();
+ qahw_stream_in->qahwi_in_stop = (qahwi_in_stop_t)
+ dlsym(qahw_module->module->dso, "qahwi_in_stop");
+ if ((error = dlerror()) != NULL) {
+ ALOGI("%s: dlsym error %s for qahwi_in_stop", __func__, error);
+ qahw_stream_in->qahwi_in_stop = NULL;
+ }
+
+ exit:
pthread_mutex_unlock(&qahw_module->lock);
return rc;
}
diff --git a/qahw_api/inc/qahw_api.h b/qahw_api/inc/qahw_api.h
index 823c6bb..b37757d 100644
--- a/qahw_api/inc/qahw_api.h
+++ b/qahw_api/inc/qahw_api.h
@@ -354,6 +354,10 @@
ssize_t qahw_in_read(qahw_stream_handle_t *in_handle,
qahw_in_buffer_t *in_buf);
/*
+ * Stop input stream. Returns zero on success.
+ */
+int qahw_in_stop(qahw_stream_handle_t *in_handle);
+/*
* Return the amount of input frames lost in the audio driver since the
* last call of this function.
* Audio driver is expected to reset the value to 0 and restart counting
diff --git a/qahw_api/src/qahw_api.cpp b/qahw_api/src/qahw_api.cpp
index f1c75f4..0810ede 100644
--- a/qahw_api/src/qahw_api.cpp
+++ b/qahw_api/src/qahw_api.cpp
@@ -678,6 +678,22 @@
}
}
+int qahw_in_stop(qahw_stream_handle_t *in_handle)
+{
+ if (g_binder_enabled) {
+ if (!g_qas_died) {
+ sp<Iqti_audio_server> qas = get_qti_audio_server();
+ if (qas_status(qas) == -1)
+ return -ENODEV;
+ return qas->qahw_in_stop(in_handle);
+ } else {
+ return -ENODEV;
+ }
+ } else {
+ return qahw_in_stop_l(in_handle);
+ }
+}
+
uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle)
{
ALOGV("%d:%s",__LINE__, __func__);
@@ -1544,6 +1560,11 @@
return qahw_in_read_l(in_handle, in_buf);
}
+int qahw_in_stop(qahw_stream_handle_t *in_handle)
+{
+ return qahw_in_stop_l(in_handle);
+}
+
uint32_t qahw_in_get_input_frames_lost(qahw_stream_handle_t *in_handle)
{
ALOGV("%d:%s",__LINE__, __func__);