Merge "hal: Add support to use compress path for ec ref loopback"
diff --git a/configs/atoll/atoll.mk b/configs/atoll/atoll.mk
old mode 100644
new mode 100755
index 65d42cc..3517085
--- a/configs/atoll/atoll.mk
+++ b/configs/atoll/atoll.mk
@@ -71,87 +71,6 @@
BOARD_SUPPORTS_OPENSOURCE_STHAL := true
-AUDIO_HARDWARE := audio.a2dp.default
-AUDIO_HARDWARE += audio.usb.default
-AUDIO_HARDWARE += audio.r_submix.default
-AUDIO_HARDWARE += audio.primary.atoll
-
-#HAL Wrapper
-AUDIO_WRAPPER := libqahw
-AUDIO_WRAPPER += libqahwwrapper
-
-#HAL Test app
-AUDIO_HAL_TEST_APPS := hal_play_test
-AUDIO_HAL_TEST_APPS += hal_rec_test
-
-PRODUCT_PACKAGES += $(AUDIO_HARDWARE)
-PRODUCT_PACKAGES += $(AUDIO_WRAPPER)
-PRODUCT_PACKAGES += $(AUDIO_HAL_TEST_APPS)
-
-ifeq ($(AUDIO_FEATURE_ENABLED_DLKM),true)
-BOARD_VENDOR_KERNEL_MODULES := \
- $(KERNEL_MODULES_OUT)/audio_apr.ko \
- $(KERNEL_MODULES_OUT)/audio_q6_pdr.ko \
- $(KERNEL_MODULES_OUT)/audio_q6_notifier.ko \
- $(KERNEL_MODULES_OUT)/audio_adsp_loader.ko \
- $(KERNEL_MODULES_OUT)/audio_q6.ko \
- $(KERNEL_MODULES_OUT)/audio_usf.ko \
- $(KERNEL_MODULES_OUT)/audio_pinctrl_lpi.ko \
- $(KERNEL_MODULES_OUT)/audio_swr.ko \
- $(KERNEL_MODULES_OUT)/audio_wcd_core.ko \
- $(KERNEL_MODULES_OUT)/audio_swr_ctrl.ko \
- $(KERNEL_MODULES_OUT)/audio_wsa881x.ko \
- $(KERNEL_MODULES_OUT)/audio_platform.ko \
- $(KERNEL_MODULES_OUT)/audio_hdmi.ko \
- $(KERNEL_MODULES_OUT)/audio_stub.ko \
- $(KERNEL_MODULES_OUT)/audio_wcd9xxx.ko \
- $(KERNEL_MODULES_OUT)/audio_mbhc.ko \
- $(KERNEL_MODULES_OUT)/audio_wcd938x.ko \
- $(KERNEL_MODULES_OUT)/audio_wcd938x_slave.ko \
- $(KERNEL_MODULES_OUT)/audio_wcd937x.ko \
- $(KERNEL_MODULES_OUT)/audio_wcd937x_slave.ko \
- $(KERNEL_MODULES_OUT)/audio_bolero_cdc.ko \
- $(KERNEL_MODULES_OUT)/audio_wsa_macro.ko \
- $(KERNEL_MODULES_OUT)/audio_va_macro.ko \
- $(KERNEL_MODULES_OUT)/audio_rx_macro.ko \
- $(KERNEL_MODULES_OUT)/audio_tx_macro.ko \
- $(KERNEL_MODULES_OUT)/audio_native.ko \
- $(KERNEL_MODULES_OUT)/audio_machine_atoll.ko \
- $(KERNEL_MODULES_OUT)/audio_snd_event.ko
-endif
-
-#Audio DLKM
-AUDIO_DLKM := audio_apr.ko
-AUDIO_DLKM += audio_q6_pdr.ko
-AUDIO_DLKM += audio_q6_notifier.ko
-AUDIO_DLKM += audio_adsp_loader.ko
-AUDIO_DLKM += audio_q6.ko
-AUDIO_DLKM += audio_usf.ko
-AUDIO_DLKM += audio_pinctrl_wcd.ko
-AUDIO_DLKM += audio_swr.ko
-AUDIO_DLKM += audio_wcd_core.ko
-AUDIO_DLKM += audio_swr_ctrl.ko
-AUDIO_DLKM += audio_wsa881x.ko
-AUDIO_DLKM += audio_platform.ko
-AUDIO_DLKM += audio_hdmi.ko
-AUDIO_DLKM += audio_stub.ko
-AUDIO_DLKM += audio_wcd9xxx.ko
-AUDIO_DLKM += audio_mbhc.ko
-AUDIO_DLKM += audio_native.ko
-AUDIO_DLKM += audio_wcd938x.ko
-AUDIO_DLKM += audio_wcd938x_slave.ko
-AUDIO_DLKM += audio_wcd937x.ko
-AUDIO_DLKM += audio_wcd937x_slave.ko
-AUDIO_DLKM += audio_bolero_cdc.ko
-AUDIO_DLKM += audio_wsa_macro.ko
-AUDIO_DLKM += audio_va_macro.ko
-AUDIO_DLKM += audio_rx_macro.ko
-AUDIO_DLKM += audio_tx_macro.ko
-AUDIO_DLKM += audio_machine_atoll.ko
-AUDIO_DLKM += audio_snd_event.ko
-
-PRODUCT_PACKAGES += $(AUDIO_DLKM)
-
#Audio Specific device overlays
DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
@@ -169,6 +88,8 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/atoll/mixer_paths_wcd9375.xml:$(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)
@@ -199,19 +120,6 @@
persist.vendor.audio.fluence.voicerec=false\
persist.vendor.audio.fluence.speaker=true
-#
-#snapdragon value add features
-#
-PRODUCT_PROPERTY_OVERRIDES += \
-ro.qc.sdk.audio.ssr=false
-
-##fluencetype can be "fluence" or "fluencepro" or "none"
-PRODUCT_PROPERTY_OVERRIDES += \
-ro.qc.sdk.audio.fluencetype=none\
-persist.audio.fluence.voicecall=true\
-persist.audio.fluence.voicerec=false\
-persist.audio.fluence.speaker=true
-
##speaker protection v3 switch and ADSP AFE API version
PRODUCT_PROPERTY_OVERRIDES += \
persist.vendor.audio.spv3.enable=true\
diff --git a/configs/atoll/audio_policy_configuration.xml b/configs/atoll/audio_policy_configuration.xml
index e96660a..bcf3e4b 100644
--- a/configs/atoll/audio_policy_configuration.xml
+++ b/configs/atoll/audio_policy_configuration.xml
@@ -176,6 +176,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -266,17 +271,20 @@
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>
@@ -358,6 +366,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="usb_surround_sound"
diff --git a/configs/kona/audio_policy_configuration.xml b/configs/kona/audio_policy_configuration.xml
index 8bb3328..1e4e338 100644
--- a/configs/kona/audio_policy_configuration.xml
+++ b/configs/kona/audio_policy_configuration.xml
@@ -173,6 +173,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="usb_surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
@@ -350,6 +355,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="usb_surround_sound"
sources="USB Device In,USB Headset In"/>
<route type="mix" sink="record_24"
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/kona/mixer_paths.xml b/configs/kona/mixer_paths.xml
index 3949c7c..36595c0 100644
--- a/configs/kona/mixer_paths.xml
+++ b/configs/kona/mixer_paths.xml
@@ -236,6 +236,7 @@
<ctl name="WSA_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
<ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
<ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="0" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="0" />
<ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
<ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
<ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
@@ -1740,6 +1741,10 @@
<ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
</path>
+ <path name="compress-voip-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="1" />
+ </path>
<path name="compress-voip-call bt-sco">
<ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
@@ -1887,6 +1892,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
</path>
+ <path name="audio-playback-voip headset">
+ <path name="audio-playback-voip headphones" />
+ </path>
+
<path name="audio-playback-voip bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
</path>
@@ -2703,6 +2712,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
</path>
+ <path name="mmap-playback headset">
+ <path name="mmap-playback headphones" />
+ </path>
+
<path name="mmap-playback speaker-and-headphones">
<path name="mmap-playback" />
<path name="mmap-playback headphones" />
diff --git a/configs/kona/mixer_paths_cdp.xml b/configs/kona/mixer_paths_cdp.xml
index 67041e6..ed9bfc0 100644
--- a/configs/kona/mixer_paths_cdp.xml
+++ b/configs/kona/mixer_paths_cdp.xml
@@ -237,6 +237,7 @@
<ctl name="WSA_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
<ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
<ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="0" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="0" />
<ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
<ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
<ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
@@ -1746,6 +1747,10 @@
<ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
</path>
+ <path name="compress-voip-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="1" />
+ </path>
<path name="compress-voip-call bt-sco">
<ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
@@ -1893,6 +1898,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
</path>
+ <path name="audio-playback-voip headset">
+ <path name="audio-playback-voip headphones" />
+ </path>
+
<path name="audio-playback-voip bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
</path>
@@ -2725,6 +2734,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
</path>
+ <path name="mmap-playback headset">
+ <path name="mmap-playback headphones" />
+ </path>
+
<path name="mmap-playback speaker-and-headphones">
<path name="mmap-playback" />
<path name="mmap-playback headphones" />
diff --git a/configs/kona/mixer_paths_qrd.xml b/configs/kona/mixer_paths_qrd.xml
index 08176d5..65f15db 100644
--- a/configs/kona/mixer_paths_qrd.xml
+++ b/configs/kona/mixer_paths_qrd.xml
@@ -236,6 +236,7 @@
<ctl name="WSA_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
<ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="0" />
<ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="0" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="0" />
<ctl name="SLIM_7_RX_Voice Mixer Voip" value="0" />
<ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="0" />
<ctl name="AFE_PCM_RX_Voice Mixer Voip" value="0" />
@@ -1757,6 +1758,11 @@
<ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_3_Voip" value="1" />
</path>
+ <path name="compress-voip-call headset">
+ <ctl name="RX_CDC_DMA_RX_0_Voice Mixer Voip" value="1" />
+ <ctl name="Voip_Tx Mixer TX_CDC_DMA_TX_4_Voip" value="1" />
+ </path>
+
<path name="compress-voip-call bt-sco">
<ctl name="SLIM_7_RX_Voice Mixer Voip" value="1" />
<ctl name="Voip_Tx Mixer SLIM_7_TX_Voip" value="1" />
@@ -1917,6 +1923,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia10" value="1" />
</path>
+ <path name="audio-playback-voip headset">
+ <path name="audio-playback-voip headphones" />
+ </path>
+
<path name="audio-playback-voip bt-sco">
<ctl name="SLIMBUS_7_RX Audio Mixer MultiMedia10" value="1" />
</path>
@@ -2110,6 +2120,10 @@
<ctl name="RX_CDC_DMA_RX_0 Audio Mixer MultiMedia16" value="1" />
</path>
+ <path name="mmap-playback headset">
+ <path name="mmap-playback headphones" />
+ </path>
+
<path name="mmap-playback speaker-and-headphones">
<path name="mmap-playback" />
<path name="mmap-playback headphones" />
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..b719ff0 100644
--- a/configs/lito/audio_policy_configuration.xml
+++ b/configs/lito/audio_policy_configuration.xml
@@ -173,6 +173,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -263,17 +268,20 @@
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>
@@ -355,6 +363,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="usb_surround_sound"
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/audio_policy_configuration.xml b/configs/msm8998/audio_policy_configuration.xml
index 505a205..5f861d9 100644
--- a/configs/msm8998/audio_policy_configuration.xml
+++ b/configs/msm8998/audio_policy_configuration.xml
@@ -167,6 +167,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -346,6 +351,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="record_24"
diff --git a/configs/msm8998/msm8998.mk b/configs/msm8998/msm8998.mk
index bee32c8..7004379 100644
--- a/configs/msm8998/msm8998.mk
+++ b/configs/msm8998/msm8998.mk
@@ -99,7 +99,9 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/sound_trigger_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_platform_info.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/graphite_ipc_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/graphite_ipc_platform_info.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/audio_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_platform_info.xml \
- vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml
+ vendor/qcom/opensource/audio-hal/primary-hal/configs/msm8998/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.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
ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
@@ -191,7 +193,7 @@
#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 +257,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/configs/msmnile/audio_policy_configuration.xml b/configs/msmnile/audio_policy_configuration.xml
index 5c05206..92430bd 100644
--- a/configs/msmnile/audio_policy_configuration.xml
+++ b/configs/msmnile/audio_policy_configuration.xml
@@ -173,6 +173,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="usb_surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,88200,96000,176400,192000"
@@ -350,6 +355,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="usb_surround_sound"
sources="USB Device In,USB Headset In"/>
<route type="mix" sink="record_24"
diff --git a/configs/msmnile/mixer_paths_tavil.xml b/configs/msmnile/mixer_paths_tavil.xml
index fb315bf..cb9d2af 100644
--- a/configs/msmnile/mixer_paths_tavil.xml
+++ b/configs/msmnile/mixer_paths_tavil.xml
@@ -3025,7 +3025,12 @@
</path>
<path name="voice-tty-hco-headset-mic">
- <path name="voice-tty-full-headset-mic" />
+ <ctl name="AIF1_CAP Mixer SLIM TX0" value="1"/>
+ <ctl name="SLIM_0_TX Channels" value="One" />
+ <ctl name="CDC_IF TX0 MUX" value="DEC0" />
+ <ctl name="ADC MUX0" value="AMIC" />
+ <ctl name="AMIC MUX0" value="ADC2" />
+ <ctl name="IIR0 INP0 MUX" value="DEC0" />
</path>
<path name="voice-tty-vco-handset-mic">
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/msmnile_au/msmnile_au.mk b/configs/msmnile_au/msmnile_au.mk
index 394dfea..7d17be7 100644
--- a/configs/msmnile_au/msmnile_au.mk
+++ b/configs/msmnile_au/msmnile_au.mk
@@ -80,7 +80,9 @@
endif
#Automotive audio specific device overlays
+ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/msmnile_au/overlay
+endif
PRODUCT_COPY_FILES += \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmnile_au/audio_output_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_output_policy.conf \
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 b092687..99f03bf 100644
--- a/configs/msmsteppe/audio_policy_configuration.xml
+++ b/configs/msmsteppe/audio_policy_configuration.xml
@@ -167,6 +167,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="incall_music_uplink" role="source"
flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
@@ -350,6 +355,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="usb_surround_sound"
sources="USB Device In,USB Headset In"/>
<route type="mix" sink="record_24"
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/msmsteppe_au/audio_platform_info.xml b/configs/msmsteppe_au/audio_platform_info.xml
index a33ae3f..e90675c 100644
--- a/configs/msmsteppe_au/audio_platform_info.xml
+++ b/configs/msmsteppe_au/audio_platform_info.xml
@@ -30,6 +30,10 @@
<device name="SND_DEVICE_OUT_HANDSET" acdb_id="78"/>
<device name="SND_DEVICE_OUT_SPEAKER" acdb_id="78"/>
<device name="SND_DEVICE_OUT_HEADPHONES" acdb_id="78"/>
+ <device name="SND_DEVICE_OUT_BUS_MEDIA" acdb_id="78"/>
+ <device name="SND_DEVICE_OUT_BUS_SYS" acdb_id="78"/>
+ <device name="SND_DEVICE_OUT_BUS_NAV" acdb_id="14"/>
+ <device name="SND_DEVICE_OUT_BUS_PHN" acdb_id="94"/>
<device name="SND_DEVICE_OUT_BT_SCO" acdb_id="94"/>
<device name="SND_DEVICE_OUT_BT_SCO_WB" acdb_id="94"/>
<device name="SND_DEVICE_OUT_BT_A2DP" acdb_id="78"/>
@@ -37,6 +41,7 @@
<device name="SND_DEVICE_OUT_VOICE_SPEAKER" acdb_id="94"/>
<device name="SND_DEVICE_IN_HANDSET_MIC" acdb_id="11"/>
<device name="SND_DEVICE_IN_SPEAKER_MIC" acdb_id="11"/>
+ <device name="SND_DEVICE_IN_BUS" acdb_id="11"/>
<device name="SND_DEVICE_IN_HEADSET_MIC" acdb_id="11"/>
<device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC" acdb_id="95"/>
<device name="SND_DEVICE_IN_VOICE_HEADSET_MIC" acdb_id="95"/>
diff --git a/configs/msmsteppe_au/audio_policy_configuration.xml b/configs/msmsteppe_au/audio_policy_configuration.xml
index 4d9340d..6ab75d8 100644
--- a/configs/msmsteppe_au/audio_policy_configuration.xml
+++ b/configs/msmsteppe_au/audio_policy_configuration.xml
@@ -33,6 +33,11 @@
“devicePorts”: a list of device descriptors for all input and output devices accessible via this
module.
This contains both permanently attached devices and removable devices.
+ "gain": constraints applied to the millibel values:
+ - maxValueMB >= minValueMB
+ - defaultValueMB >= minValueMB && defaultValueMB <= maxValueMB
+ - (maxValueMB - minValueMB) % stepValueMB == 0
+ - (defaultValueMB - minValueMB) % stepValueMB == 0
“mixPorts”: listing all output and input streams exposed by the audio HAL
“routes”: list of possible connections between input and output devices or between stream and
devices.
@@ -47,18 +52,37 @@
-->
<modules>
<!-- Primary Audio HAL -->
- <module name="primary" halVersion="2.0">
+ <module name="primary" halVersion="3.0">
<attachedDevices>
- <item>Earpiece</item>
- <item>Speaker</item>
+ <item>Media Bus</item>
+ <item>Sys Notification Bus</item>
+ <item>Nav Guidance Bus</item>
+ <item>Phone Bus</item>
<item>Telephony Tx</item>
<item>Built-In Mic</item>
<item>Built-In Back Mic</item>
<item>FM Tuner</item>
<item>Telephony Rx</item>
</attachedDevices>
- <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <defaultOutputDevice>Media Bus</defaultOutputDevice>
<mixPorts>
+ <mixPort name="media" role="source"
+ flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="sys_notification" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="nav_guidance" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="phone" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
<mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
@@ -203,6 +227,42 @@
<devicePorts>
<!-- Output devices declaration, i.e. Sink DEVICE PORT -->
+ <devicePort tagName="Media Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="BUS00_MEDIA">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Sys Notification Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="BUS01_SYS_NOTIFICATION">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Nav Guidance Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="BUS02_NAV_GUIDANCE">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Phone Bus" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="BUS03_PHONE">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-6000" maxValueMB="600" defaultValueMB="0" stepValueMB="100"/>
+ </gains>
+ </devicePort>
<devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
@@ -301,6 +361,14 @@
</devicePorts>
<!-- route declaration, i.e. list all available sources for a given sink -->
<routes>
+ <route type="mix" sink="Media Bus"
+ sources="media,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
+ <route type="mix" sink="Sys Notification Bus"
+ sources="sys_notification"/>
+ <route type="mix" sink="Nav Guidance Bus"
+ sources="nav_guidance"/>
+ <route type="mix" sink="Phone Bus"
+ sources="phone"/>
<route type="mix" sink="Earpiece"
sources="primary output,raw,deep_buffer,direct_pcm,compressed_offload,voip_rx,mmap_no_irq_out"/>
<route type="mix" sink="Speaker"
diff --git a/configs/msmsteppe_au/mixer_paths_adp.xml b/configs/msmsteppe_au/mixer_paths_adp.xml
index e2de539..75ce9c5 100644
--- a/configs/msmsteppe_au/mixer_paths_adp.xml
+++ b/configs/msmsteppe_au/mixer_paths_adp.xml
@@ -755,6 +755,11 @@
<path name="compress-offload-playback4" />
</path>
+ <path name="voice-call">
+ <!-- Added AFE loopback ctrl path for CS-Voice call-->
+ <ctl name="TERT_TDM_RX_0 Port Mixer AUX_PCM_UL_TX" value="1" />
+ <ctl name="AUX_PCM_RX Port Mixer TERT_TDM_TX_0" value="1" />
+ </path>
<path name="compress-offload-playback4 afe-proxy">
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia11" value="1" />
@@ -1135,6 +1140,27 @@
<path name="compress-offload-playback9" />
</path>
+ <!-- The following use cases are used for car streams -->
+ <path name="media-playback">
+ <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="sys-notification-playback">
+ <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="nav-guidance-playback">
+ <ctl name="TERT_TDM_RX_1 Channels" value="One" />
+ <ctl name="TERT_TDM_RX_1 Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="phone-playback">
+ <ctl name="TERT_TDM_RX_2 Channels" value="One" />
+ <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
<path name="audio-record">
<ctl name="TERT_TDM_TX_0 Channels" value="One" />
<ctl name="MultiMedia1 Mixer TERT_TDM_TX_0" value="1" />
@@ -1622,9 +1648,15 @@
</path>
<path name="speaker-adp">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
</path>
<path name="speaker-custom">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
+ </path>
+
+ <path name="bus-speaker">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
</path>
<path name="sidetone-iir">
diff --git a/configs/msmsteppe_au/mixer_paths_custom.xml b/configs/msmsteppe_au/mixer_paths_custom.xml
index 3de6f0f..5175d9f 100644
--- a/configs/msmsteppe_au/mixer_paths_custom.xml
+++ b/configs/msmsteppe_au/mixer_paths_custom.xml
@@ -1131,6 +1131,27 @@
<path name="compress-offload-playback9" />
</path>
+ <!-- The following use cases are used for car streams -->
+ <path name="media-playback">
+ <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia1" value="1" />
+ </path>
+
+ <path name="sys-notification-playback">
+ <ctl name="TERT_TDM_RX_0 Channels" value="Six" />
+ <ctl name="TERT_TDM_RX_0 Audio Mixer MultiMedia5" value="1" />
+ </path>
+
+ <path name="nav-guidance-playback">
+ <ctl name="TERT_TDM_RX_1 Channels" value="One" />
+ <ctl name="TERT_TDM_RX_1 Audio Mixer MultiMedia2" value="1" />
+ </path>
+
+ <path name="phone-playback">
+ <ctl name="TERT_TDM_RX_2 Channels" value="One" />
+ <ctl name="TERT_TDM_RX_2 Audio Mixer MultiMedia10" value="1" />
+ </path>
+
<path name="audio-record">
<ctl name="TERT_TDM_TX_0 Channels" value="One" />
<ctl name="MultiMedia1 Mixer TERT_TDM_TX_0" value="1" />
@@ -1618,9 +1639,15 @@
</path>
<path name="speaker-adp">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
</path>
<path name="speaker-custom">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
+ </path>
+
+ <path name="bus-speaker">
+ <!--ctl name="TERT_TDM_RX_0 Channels" value="Six" /-->
</path>
<path name="sidetone-iir">
diff --git a/configs/msmsteppe_au/msmsteppe_au.mk b/configs/msmsteppe_au/msmsteppe_au.mk
index 51829bd..a0b9cf1 100644
--- a/configs/msmsteppe_au/msmsteppe_au.mk
+++ b/configs/msmsteppe_au/msmsteppe_au.mk
@@ -69,13 +69,20 @@
AUDIO_FEATURE_ENABLED_BATTERY_LISTENER := false
##AUDIO_FEATURE_FLAGS
+AUDIO_FEATURE_ENABLED_AUTO_HAL := true
+AUDIO_FEATURE_ENABLED_EXT_HW_PLUGIN := true
+AUDIO_FEATURE_ENABLED_AUDIO_CONTROL_HAL := true
+##AUTOMOTIVE_AUDIO_FEATURE_FLAGS
+
ifneq ($(strip $(TARGET_USES_RRO)), true)
#Audio Specific device overlays
DEVICE_PACKAGE_OVERLAYS += vendor/qcom/opensource/audio-hal/primary-hal/configs/common/overlay
endif
#Automotive audio specific device overlays
+ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
DEVICE_PACKAGE_OVERLAYS += hardware/qcom/audio/configs/msmsteppe_au/overlay
+endif
PRODUCT_COPY_FILES += \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/audio_output_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_output_policy.conf \
@@ -109,6 +116,10 @@
PRODUCT_COPY_FILES += \
vendor/qcom/opensource/audio-hal/primary-hal/configs/msmsteppe_au/listen_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/listen_platform_info.xml
+#Audio HAL version
+PRODUCT_PROPERTY_OVERRIDES += \
+vendor.audio.hal.maj.version=3
+
# Reduce client buffer size for fast audio output tracks
PRODUCT_PROPERTY_OVERRIDES += \
af.fast_track_multiplier=1
@@ -261,13 +272,20 @@
android.hardware.audio.effect@4.0 \
android.hardware.audio.effect@4.0-impl
+# for HIDL related audiocontrol packages
+PRODUCT_PACKAGES += \
+ vendor.qti.hardware.automotive.audiocontrol@1.0-service \
+ android.hardware.automotive.audiocontrol@1.0
+
+ifeq ($(ENABLE_HYP),true)
PRODUCT_PROPERTY_OVERRIDES += \
-persist.audio.calfile0=/vendor/etc/acdbdata/adsp_avs_config.acdb\
-persist.audio.calfile1=/vendor/etc/acdbdata/ADP/Bluetooth_cal.acdb\
-persist.audio.calfile2=/vendor/etc/acdbdata/ADP/Codec_cal.acdb\
-persist.audio.calfile3=/vendor/etc/acdbdata/ADP/General_cal.acdb\
-persist.audio.calfile4=/vendor/etc/acdbdata/ADP/Global_cal.acdb\
-persist.audio.calfile5=/vendor/etc/acdbdata/ADP/Handset_cal.acdb\
-persist.audio.calfile6=/vendor/etc/acdbdata/ADP/Hdmi_cal.acdb\
-persist.audio.calfile7=/vendor/etc/acdbdata/ADP/Headset_cal.acdb\
-persist.audio.calfile8=/vendor/etc/acdbdata/ADP/Speaker_cal.acdb
+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/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/audio_policy_configuration.xml b/configs/sdm660/audio_policy_configuration.xml
index 662764f..e1a0181 100644
--- a/configs/sdm660/audio_policy_configuration.xml
+++ b/configs/sdm660/audio_policy_configuration.xml
@@ -163,6 +163,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -337,6 +342,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="record_24"
diff --git a/configs/sdm660/sdm660.mk b/configs/sdm660/sdm660.mk
index 84f0f1e..03e59e2 100644
--- a/configs/sdm660/sdm660.mk
+++ b/configs/sdm660/sdm660.mk
@@ -106,7 +106,9 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/sound_trigger_mixer_paths_wcd9340.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths_wcd9340.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/sound_trigger_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_platform_info.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/graphite_ipc_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/graphite_ipc_platform_info.xml \
- vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml
+ vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm660/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.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
ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
@@ -202,7 +204,7 @@
#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 +268,6 @@
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/audio_policy_configuration.xml b/configs/sdm710/audio_policy_configuration.xml
index a7f4869..145a811 100644
--- a/configs/sdm710/audio_policy_configuration.xml
+++ b/configs/sdm710/audio_policy_configuration.xml
@@ -167,6 +167,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="surround_sound" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
@@ -346,6 +351,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="record_24"
diff --git a/configs/sdm710/sdm710.mk b/configs/sdm710/sdm710.mk
index 177562a..ea71582 100644
--- a/configs/sdm710/sdm710.mk
+++ b/configs/sdm710/sdm710.mk
@@ -170,7 +170,9 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/sound_trigger_mixer_paths_wcd9340.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths_wcd9340.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/sound_trigger_mixer_paths_wcd9340.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_mixer_paths_wcd9340.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/graphite_ipc_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/graphite_ipc_platform_info.xml \
- vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml
+ vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm710/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.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
ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
@@ -296,7 +298,7 @@
#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 +367,6 @@
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/audio_policy_configuration.xml b/configs/sdm845/audio_policy_configuration.xml
index 76f5236..fcd0119 100644
--- a/configs/sdm845/audio_policy_configuration.xml
+++ b/configs/sdm845/audio_policy_configuration.xml
@@ -161,12 +161,16 @@
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
</mixPort>
-
<mixPort name="primary input" role="sink" maxOpenCount="2" maxActiveCount="2">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="record_24" role="sink">
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000,192000"
@@ -341,6 +345,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="record_24"
sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic"/>
<route type="mix" sink="mmap_no_irq_in"
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index 80ff74b..6e56597 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -124,7 +124,9 @@
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/sound_trigger_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/sound_trigger_platform_info.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/graphite_ipc_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/graphite_ipc_platform_info.xml \
vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/audio_platform_info.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_platform_info.xml \
- vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.xml
+ vendor/qcom/opensource/audio-hal/primary-hal/configs/sdm845/audio_configs.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_configs.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
ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
@@ -233,7 +235,7 @@
#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 +316,6 @@
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..043df15 100644
--- a/configs/trinket/audio_policy_configuration.xml
+++ b/configs/trinket/audio_policy_configuration.xml
@@ -167,6 +167,11 @@
samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
</mixPort>
+ <mixPort name="voip_tx" role="sink"
+ flags="AUDIO_INPUT_FLAG_VOIP_TX">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,16000,32000,48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
<mixPort name="incall_music_uplink" role="source"
flags="AUDIO_OUTPUT_FLAG_INCALL_MUSIC">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
@@ -263,17 +268,20 @@
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>
@@ -355,6 +363,8 @@
sources="Telephony Rx"/>
<route type="mix" sink="primary input"
sources="Wired Headset Mic,BT SCO Headset Mic,FM Tuner,USB Device In,USB Headset In,Telephony Rx"/>
+ <route type="mix" sink="voip_tx"
+ sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic,USB Device In,USB Headset In"/>
<route type="mix" sink="surround_sound"
sources="Built-In Mic,Built-In Back Mic"/>
<route type="mix" sink="usb_surround_sound"
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 ($(TARGET_USES_AOSP_FOR_AUDIO), true)
@@ -185,7 +187,7 @@
#enable pbe effects
PRODUCT_PROPERTY_OVERRIDES += \
-vendor.audio.safx.pbe.enabled=true
+vendor.audio.safx.pbe.enabled=false
#parser input buffer size(256kb) in byte stream mode
PRODUCT_PROPERTY_OVERRIDES += \
@@ -242,7 +244,6 @@
vendor.audio.feature.a2dp_offload.enable=true \
vendor.audio.feature.afe_proxy.enable=true \
vendor.audio.feature.anc_headset.enable=true \
-vendor.audio.feature.audio_sphere.enable=true \
vendor.audio.feature.battery_listener.enable=false \
vendor.audio.feature.compr_cap.enable=false \
vendor.audio.feature.compress_in.enable=false \
diff --git a/hal/Android.mk b/hal/Android.mk
index a671373..1a0c2e2 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -88,6 +88,10 @@
LOCAL_CFLAGS := -DPLATFORM_LITO
LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
endif
+ifneq ($(filter atoll,$(TARGET_BOARD_PLATFORM)),)
+ LOCAL_CFLAGS := -DPLATFORM_ATOLL
+ LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
+endif
ifneq ($(filter sdm660,$(TARGET_BOARD_PLATFORM)),)
LOCAL_CFLAGS := -DPLATFORM_MSMFALCON
LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="8"
diff --git a/hal/audio_extn/Android.mk b/hal/audio_extn/Android.mk
index e944260..fee6977 100644
--- a/hal/audio_extn/Android.mk
+++ b/hal/audio_extn/Android.mk
@@ -410,7 +410,7 @@
PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -469,7 +469,7 @@
PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -528,7 +528,7 @@
PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -590,7 +590,7 @@
PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -660,7 +660,7 @@
PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE) $(TRINKET),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 03c07e3..8e65471 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)
{
@@ -5064,9 +5065,9 @@
{
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)
{
@@ -5076,15 +5077,19 @@
{
(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 05c50bf..d407f80 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);
@@ -998,7 +1001,8 @@
#define audio_extn_gef_init(adev) (0)
#define audio_extn_gef_deinit(adev) (0)
-#define audio_extn_gef_notify_device_config(devices, cmask, sample_rate, acdb_id) (0)
+#define audio_extn_gef_notify_device_config(devices, cmask, sample_rate, \
+ acdb_id, app_type) (0)
#ifndef INSTANCE_ID_ENABLED
#define audio_extn_gef_send_audio_cal(dev, acdb_dev_id, acdb_device_type,\
@@ -1030,7 +1034,7 @@
void audio_extn_gef_deinit(struct audio_device *adev);
void audio_extn_gef_notify_device_config(audio_devices_t audio_device,
- audio_channel_mask_t channel_mask, int sample_rate, int acdb_id);
+ audio_channel_mask_t channel_mask, int sample_rate, int acdb_id, int app_type);
#ifndef INSTANCE_ID_ENABLED
int audio_extn_gef_send_audio_cal(void* adev, int acdb_dev_id, int acdb_device_type,
int app_type, int topology_id, int sample_rate, uint32_t module_id,
@@ -1066,12 +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 ==============================================
@@ -1146,6 +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,
@@ -1268,8 +1275,6 @@
#ifndef AUDIO_EXTN_AUTO_HAL_ENABLED
#define audio_extn_auto_hal_init(adev) (0)
#define audio_extn_auto_hal_deinit() (0)
-#define audio_extn_auto_hal_enable_hostless() (0)
-#define audio_extn_auto_hal_disable_hostless() (0)
#define audio_extn_auto_hal_create_audio_patch(dev, num_sources,\
sources, num_sinks, sinks, handle) (0)
#define audio_extn_auto_hal_release_audio_patch(dev, handle) (0)
@@ -1283,8 +1288,6 @@
#else
int32_t audio_extn_auto_hal_init(struct audio_device *adev);
void audio_extn_auto_hal_deinit(void);
-int32_t audio_extn_auto_hal_enable_hostless(void);
-void audio_extn_auto_hal_disable_hostless(void);
int audio_extn_auto_hal_create_audio_patch(struct audio_hw_device *dev,
unsigned int num_sources,
const struct audio_port_config *sources,
diff --git a/hal/audio_extn/auto_hal.c b/hal/audio_extn/auto_hal.c
index f008a47..7f2163d 100644
--- a/hal/audio_extn/auto_hal.c
+++ b/hal/audio_extn/auto_hal.c
@@ -47,15 +47,9 @@
#ifdef AUDIO_EXTN_AUTO_HAL_ENABLED
-struct hostless_config {
- struct pcm *pcm_tx;
- struct pcm *pcm_rx;
-};
-
typedef struct auto_hal_module {
struct audio_device *adev;
card_status_t card_status;
- struct hostless_config hostless;
} auto_hal_module_t;
/* Auto hal module struct */
@@ -71,104 +65,6 @@
USECASE_AUDIO_PLAYBACK_PHONE,
};
-/* Note: Due to ADP H/W design, SoC TERT/SEC TDM CLK and FSYNC lines are
- * both connected with CODEC and a single master is needed to provide
- * consistent CLK and FSYNC to slaves, hence configuring SoC TERT TDM as
- * single master and bring up a dummy hostless from TERT to SEC to ensure
- * both slave SoC SEC TDM and CODEC are driven upon system boot. */
-int32_t audio_extn_auto_hal_enable_hostless(void)
-{
- int32_t ret = 0;
- char mixer_path[MIXER_PATH_MAX_LENGTH];
-
- ALOGD("%s: Enable TERT -> SEC Hostless", __func__);
-
- if (auto_hal == NULL) {
- ALOGE("%s: Invalid device", __func__);
- return -EINVAL;
- }
-
- strlcpy(mixer_path, "dummy-hostless", MIXER_PATH_MAX_LENGTH);
- ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path);
- if (audio_route_apply_and_update_path(auto_hal->adev->audio_route,
- mixer_path)) {
- ALOGD("%s: %s not supported, continue", __func__, mixer_path);
- return ret;
- }
-
- /* TERT TDM TX 7 HOSTLESS to SEC TDM RX 7 HOSTLESS */
- int pcm_dev_rx = 48, pcm_dev_tx = 49;
- struct pcm_config pcm_config_lb = {
- .channels = 1,
- .rate = 48000,
- .period_size = 240,
- .period_count = 2,
- .format = PCM_FORMAT_S16_LE,
- .start_threshold = 0,
- .stop_threshold = INT_MAX,
- .avail_min = 0,
- };
-
- auto_hal->hostless.pcm_tx = pcm_open(auto_hal->adev->snd_card,
- pcm_dev_tx,
- PCM_IN, &pcm_config_lb);
- if (auto_hal->hostless.pcm_tx &&
- !pcm_is_ready(auto_hal->hostless.pcm_tx)) {
- ALOGE("%s: %s", __func__,
- pcm_get_error(auto_hal->hostless.pcm_tx));
- ret = -EIO;
- goto error;
- }
- auto_hal->hostless.pcm_rx = pcm_open(auto_hal->adev->snd_card,
- pcm_dev_rx,
- PCM_OUT, &pcm_config_lb);
- if (auto_hal->hostless.pcm_rx &&
- !pcm_is_ready(auto_hal->hostless.pcm_rx)) {
- ALOGE("%s: %s", __func__,
- pcm_get_error(auto_hal->hostless.pcm_rx));
- ret = -EIO;
- goto error;
- }
-
- if (pcm_start(auto_hal->hostless.pcm_tx) < 0) {
- ALOGE("%s: pcm start for pcm tx failed", __func__);
- ret = -EIO;
- goto error;
- }
- if (pcm_start(auto_hal->hostless.pcm_rx) < 0) {
- ALOGE("%s: pcm start for pcm rx failed", __func__);
- ret = -EIO;
- goto error;
- }
- return ret;
-
-error:
- if (auto_hal->hostless.pcm_rx)
- pcm_close(auto_hal->hostless.pcm_rx);
- if (auto_hal->hostless.pcm_tx)
- pcm_close(auto_hal->hostless.pcm_tx);
- return ret;
-}
-
-void audio_extn_auto_hal_disable_hostless(void)
-{
- ALOGD("%s: Disable TERT -> SEC Hostless", __func__);
-
- if (auto_hal == NULL) {
- ALOGE("%s: Invalid device", __func__);
- return;
- }
-
- if (auto_hal->hostless.pcm_tx) {
- pcm_close(auto_hal->hostless.pcm_tx);
- auto_hal->hostless.pcm_tx = NULL;
- }
- if (auto_hal->hostless.pcm_rx) {
- pcm_close(auto_hal->hostless.pcm_rx);
- auto_hal->hostless.pcm_rx = NULL;
- }
-}
-
#define MAX_SOURCE_PORTS_PER_PATCH 1
#define MAX_SINK_PORTS_PER_PATCH 1
@@ -445,7 +341,7 @@
*/
#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 +394,10 @@
/* 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 */
@@ -575,11 +474,9 @@
ALOGV("%s: snd card status %s", __func__, snd_card_status);
if (strstr(snd_card_status, "OFFLINE")) {
auto_hal->card_status = CARD_STATUS_OFFLINE;
- audio_extn_auto_hal_disable_hostless();
}
else if (strstr(snd_card_status, "ONLINE")) {
auto_hal->card_status = CARD_STATUS_ONLINE;
- audio_extn_auto_hal_enable_hostless();
}
}
diff --git a/hal/audio_extn/compress_in.c b/hal/audio_extn/compress_in.c
index 6cf6b81..6b525b0 100644
--- a/hal/audio_extn/compress_in.c
+++ b/hal/audio_extn/compress_in.c
@@ -100,7 +100,7 @@
* only after validating that input against cin_attached_usecase
* except below calls
* 1. cin_applicable_stream(in)
- * 2. cin_configure_input_stream(in)
+ * 2. cin_configure_input_stream(in, in_config)
*/
bool cin_attached_usecase(audio_usecase_t uc_id)
@@ -179,7 +179,7 @@
return sz;
}
-int cin_start_input_stream(struct stream_in *in)
+int cin_open_input_stream(struct stream_in *in)
{
int ret = -EINVAL;
struct audio_device *adev = in->dev;
@@ -208,12 +208,23 @@
ALOGV("%s: in %p, cin_data %p", __func__, in, cin_data);
if (cin_data->compr) {
+ compress_stop(cin_data->compr);
+ }
+}
+
+
+void cin_close_input_stream(struct stream_in *in)
+{
+ cin_private_data_t *cin_data = (cin_private_data_t *) in->cin_extn;
+
+ ALOGV("%s: in %p, cin_data %p", __func__, in, cin_data);
+ if (cin_data->compr) {
compress_close(cin_data->compr);
cin_data->compr = NULL;
}
}
-void cin_close_input_stream(struct stream_in *in)
+void cin_free_input_stream_resources(struct stream_in *in)
{
cin_private_data_t *cin_data = (cin_private_data_t *) in->cin_extn;
@@ -265,9 +276,8 @@
return ret;
}
-int cin_configure_input_stream(struct stream_in *in)
+int cin_configure_input_stream(struct stream_in *in, struct audio_config *in_config)
{
- struct audio_device *adev = in->dev;
struct audio_config config = {.format = 0};
int ret = 0, buffer_size = 0, meta_size = sizeof(struct snd_codec_metadata);
cin_private_data_t *cin_data = NULL;
@@ -304,7 +314,8 @@
config.channel_mask = in->channel_mask;
config.format = in->format;
in->config.channels = audio_channel_count_from_in_mask(in->channel_mask);
- buffer_size = adev->device.get_input_buffer_size(&adev->device, &config);
+ buffer_size = audio_extn_utils_get_input_buffer_size(config.sample_rate, config.format,
+ in->config.channels, in_config->offload_info.duration_us / 1000, false);
cin_data->compr_config.fragment_size = buffer_size;
cin_data->compr_config.codec->id = get_snd_codec_id(in->format);
@@ -321,6 +332,11 @@
else
cin_data->compr_config.codec->compr_passthr = PASSTHROUGH_GEN;
+ if (in->flags & AUDIO_INPUT_FLAG_FAST) {
+ ALOGD("%s: Setting latency mode to true", __func__);
+ cin_data->compr_config.codec->flags |= audio_extn_utils_get_perf_mode_flag();
+ }
+
if ((in->flags & AUDIO_INPUT_FLAG_TIMESTAMP) ||
(in->flags & AUDIO_INPUT_FLAG_PASSTHROUGH)) {
compress_config_set_timstamp_flag(&cin_data->compr_config);
@@ -332,6 +348,6 @@
return ret;
err_config:
- cin_close_input_stream(in);
+ cin_free_input_stream_resources(in);
return ret;
}
diff --git a/hal/audio_extn/ext_hw_plugin.c b/hal/audio_extn/ext_hw_plugin.c
index 6b4a718..619ecfc 100644
--- a/hal/audio_extn/ext_hw_plugin.c
+++ b/hal/audio_extn/ext_hw_plugin.c
@@ -77,85 +77,6 @@
/* This can be defined in platform specific file or use compile flag */
#define LIB_PLUGIN_DRIVER "libaudiohalplugin.so"
-/* Note: Due to ADP H/W design, SoC TERT/SEC TDM CLK and FSYNC lines are
- * both connected with CODEC and a single master is needed to provide
- * consistent CLK and FSYNC to slaves, hence configuring SoC TERT TDM as
- * single master and bring up a dummy hostless from TERT to SEC to ensure
- * both slave SoC SEC TDM and CODEC are driven upon system boot. */
-static void ext_hw_plugin_enable_adev_hostless(void *plugin)
-{
- struct ext_hw_plugin_data *my_plugin =
- (struct ext_hw_plugin_data *)plugin;
- char mixer_path[MIXER_PATH_MAX_LENGTH];
-
- ALOGI("%s: Enable TERT -> SEC Hostless", __func__);
-
- strlcpy(mixer_path, "dummy-hostless", MIXER_PATH_MAX_LENGTH);
- ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path);
- if (audio_route_apply_and_update_path(my_plugin->adev->audio_route,
- mixer_path)) {
- ALOGE("%s: %s not supported, continue", __func__, mixer_path);
- return;
- }
-
- /* TERT TDM TX 7 HOSTLESS to SEC TDM RX 7 HOSTLESS */
- int pcm_dev_rx = 48, pcm_dev_tx = 49;
- struct pcm_config pcm_config_lb = {
- .channels = 1,
- .rate = 48000,
- .period_size = 240,
- .period_count = 2,
- .format = PCM_FORMAT_S16_LE,
- .start_threshold = 0,
- .stop_threshold = INT_MAX,
- .avail_min = 0,
- };
-
- my_plugin->adev_hostless.pcm_tx = pcm_open(my_plugin->adev->snd_card,
- pcm_dev_tx,
- PCM_IN, &pcm_config_lb);
- if (my_plugin->adev_hostless.pcm_tx &&
- !pcm_is_ready(my_plugin->adev_hostless.pcm_tx)) {
- ALOGE("%s: %s", __func__,
- pcm_get_error(my_plugin->adev_hostless.pcm_tx));
- return;
- }
- my_plugin->adev_hostless.pcm_rx = pcm_open(my_plugin->adev->snd_card,
- pcm_dev_rx,
- PCM_OUT, &pcm_config_lb);
- if (my_plugin->adev_hostless.pcm_rx &&
- !pcm_is_ready(my_plugin->adev_hostless.pcm_rx)) {
- ALOGE("%s: %s", __func__,
- pcm_get_error(my_plugin->adev_hostless.pcm_rx));
- return;
- }
-
- if (pcm_start(my_plugin->adev_hostless.pcm_tx) < 0) {
- ALOGE("%s: pcm start for pcm tx failed", __func__);
- return;
- }
- if (pcm_start(my_plugin->adev_hostless.pcm_rx) < 0) {
- ALOGE("%s: pcm start for pcm rx failed", __func__);
- return;
- }
-}
-
-static void ext_hw_plugin_disable_adev_hostless(void *plugin)
-{
- struct ext_hw_plugin_data *my_plugin = (struct ext_hw_plugin_data *)plugin;
-
- ALOGI("%s: Disable TERT -> SEC Hostless", __func__);
-
- if (my_plugin->adev_hostless.pcm_tx) {
- pcm_close(my_plugin->adev_hostless.pcm_tx);
- my_plugin->adev_hostless.pcm_tx = NULL;
- }
- if (my_plugin->adev_hostless.pcm_rx) {
- pcm_close(my_plugin->adev_hostless.pcm_rx);
- my_plugin->adev_hostless.pcm_rx = NULL;
- }
-}
-
void* ext_hw_plugin_init(struct audio_device *adev, ext_hw_plugin_init_config_t init_config)
{
int32_t ret = 0;
@@ -170,7 +91,6 @@
my_plugin->adev = adev;
fp_audio_route_apply_and_update_path = init_config.fp_audio_route_apply_and_update_path;
- (void)audio_extn_auto_hal_enable_hostless();
my_plugin->plugin_handle = dlopen(LIB_PLUGIN_DRIVER, RTLD_NOW);
if (my_plugin->plugin_handle == NULL) {
@@ -209,7 +129,6 @@
goto plugin_init_fail;
}
}
- ext_hw_plugin_enable_adev_hostless(my_plugin);
my_plugin->mic_mute = false;
return my_plugin;
@@ -229,7 +148,6 @@
ALOGE("[%s] NULL plugin pointer",__func__);
return -EINVAL;
}
- ext_hw_plugin_disable_adev_hostless(my_plugin);
if (my_plugin->audio_hal_plugin_deinit) {
ret = my_plugin->audio_hal_plugin_deinit();
if (ret) {
@@ -240,8 +158,6 @@
if(my_plugin->plugin_handle != NULL)
dlclose(my_plugin->plugin_handle);
- audio_extn_auto_hal_disable_hostless();
-
free(my_plugin);
return ret;
}
diff --git a/hal/audio_extn/gef.c b/hal/audio_extn/gef.c
index ca1a16b..83e9d45 100644
--- a/hal/audio_extn/gef.c
+++ b/hal/audio_extn/gef.c
@@ -64,7 +64,7 @@
typedef void* (*gef_init_t)(void*);
typedef void (*gef_deinit_t)(void*);
typedef void (*gef_device_config_cb_t)(void*, audio_devices_t,
- audio_channel_mask_t, int, int);
+ audio_channel_mask_t, int, int, int);
typedef struct {
void* handle;
@@ -428,14 +428,14 @@
//this will be called from HAL to notify GEF of new device configuration
void audio_extn_gef_notify_device_config(audio_devices_t audio_device,
- audio_channel_mask_t channel_mask, int sample_rate, int acdb_id)
+ audio_channel_mask_t channel_mask, int sample_rate, int acdb_id, int app_type)
{
ALOGV("%s: Enter", __func__);
//call into GEF to share channel mask and device info
if (gef_hal_handle.handle && gef_hal_handle.device_config_cb) {
gef_hal_handle.device_config_cb(gef_hal_handle.gef_ptr, audio_device, channel_mask,
- sample_rate, acdb_id);
+ sample_rate, acdb_id, app_type);
}
ALOGV("%s: Exit", __func__);
diff --git a/hal/audio_extn/ip_hdlr_intf.c b/hal/audio_extn/ip_hdlr_intf.c
index 0afc705..3214c03 100644
--- a/hal/audio_extn/ip_hdlr_intf.c
+++ b/hal/audio_extn/ip_hdlr_intf.c
@@ -62,8 +62,8 @@
#define ADSP_DEC_SERVICE_ID 1
#define ADSP_EVENT_ID_RTIC 0x00013239
#define ADSP_EVENT_ID_RTIC_FAIL 0x0001323A
-#define TRUMPET_TOPOLOGY 0x11000099
-#define TRUMPET_MODULE 0x0001099A
+#define TRUMPET_TOPOLOGY 0x11000099
+#define TRUMPET_MODULE 0x0001099A
struct lib_fd_info {
int32_t fd;
@@ -212,10 +212,12 @@
return ret;
}
+
bool audio_extn_ip_hdlr_intf_supported_for_copp(void *platform)
{
return adm_event_enable;
}
+
bool audio_extn_ip_hdlr_intf_supported(audio_format_t format,
bool is_direct_passthrough,
bool is_transcode_loopback)
@@ -224,28 +226,30 @@
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DOLBY_TRUEHD) {
asm_event_enable = true;
return true;
+ } else if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_MAT) {
+ asm_event_enable = true;
+ return true;
} else if (!is_direct_passthrough && !audio_extn_qaf_is_enabled() &&
(((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_E_AC3) ||
((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AC3))) {
asm_event_enable = true;
return true;
- } else if (is_transcode_loopback &&
+ } else if (is_transcode_loopback &&
(((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_E_AC3) ||
((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AC3))) {
- asm_event_enable = true;
- return true;
- } else {
- asm_event_enable = false;
- return false;
- }
+ asm_event_enable = true;
+ return true;
+ } else {
+ asm_event_enable = false;
+ return false;
}
int audio_extn_ip_hdlr_intf_event_adm(void *stream_handle __unused,
void *payload, void *ip_hdlr_handle )
{
- ALOGVV("%s:[%d] handle = %p\n",__func__, ip_hdlr->ref_cnt, ip_hdlr_handle);
+ ALOGVV("%s:[%d] handle = %p\n",__func__, ip_hdlr->ref_cnt, ip_hdlr_handle);
- return ip_hdlr->event_adm(ip_hdlr_handle, payload);
+ return ip_hdlr->event_adm(ip_hdlr_handle, payload);
}
int audio_extn_ip_hdlr_intf_event(void *stream_handle __unused, void *payload, void *ip_hdlr_handle )
@@ -887,9 +891,6 @@
return -EINVAL;
}
ALOGD("%s:[%d] handle = %p",__func__, ip_hdlr->ref_cnt, handle);
- ret = ip_hdlr->deinit(handle);
- if (ret < 0)
- ALOGE("%s:[%d] deinit failed ret = %d", __func__, ip_hdlr->ref_cnt, ret);
if (--ip_hdlr->ref_cnt == 0) {
ip_hdlr->get_lib_fd(handle, &lib_fd.fd);
@@ -917,8 +918,11 @@
goto dlclose;
}
- ret = ip_hdlr->deinit_lib(ip_hdlr->ip_lib_handle);
+ ret = ip_hdlr->deinit_lib(handle);
ip_hdlr->lib_fd_created = false;
+ ret = ip_hdlr->deinit(handle);
+ if (ret < 0)
+ ALOGE("%s:[%d] deinit failed ret = %d", __func__, ip_hdlr->ref_cnt, ret);
if (ip_hdlr->lib_hdl)
dlclose(ip_hdlr->lib_hdl);
dlclose:
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index aa13c2b..1e28b86 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -274,10 +274,42 @@
static void stdev_snd_mon_cb(void * stream __unused, struct str_parms * parms)
{
+ audio_event_info_t event;
+ char value[32];
+ int ret = 0;
+
if (!parms)
return;
- audio_extn_sound_trigger_set_parameters(NULL, parms);
+ ret = str_parms_get_str(parms, "SND_CARD_STATUS", value,
+ sizeof(value));
+ if (ret > 0) {
+ if (strstr(value, "OFFLINE")) {
+ event.u.status = SND_CARD_STATUS_OFFLINE;
+ st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+ }
+ else if (strstr(value, "ONLINE")) {
+ event.u.status = SND_CARD_STATUS_ONLINE;
+ st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+ }
+ else
+ ALOGE("%s: unknown snd_card_status", __func__);
+ }
+
+ ret = str_parms_get_str(parms, "CPE_STATUS", value, sizeof(value));
+ if (ret > 0) {
+ if (strstr(value, "OFFLINE")) {
+ event.u.status = CPE_STATUS_OFFLINE;
+ st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+ }
+ else if (strstr(value, "ONLINE")) {
+ event.u.status = CPE_STATUS_ONLINE;
+ st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+ }
+ else
+ ALOGE("%s: unknown CPE status", __func__);
+ }
+
return;
}
@@ -630,34 +662,7 @@
return;
}
- ret = str_parms_get_str(params, "SND_CARD_STATUS", value,
- sizeof(value));
- if (ret > 0) {
- if (strstr(value, "OFFLINE")) {
- event.u.status = SND_CARD_STATUS_OFFLINE;
- st_dev->st_callback(AUDIO_EVENT_SSR, &event);
- }
- else if (strstr(value, "ONLINE")) {
- event.u.status = SND_CARD_STATUS_ONLINE;
- st_dev->st_callback(AUDIO_EVENT_SSR, &event);
- }
- else
- ALOGE("%s: unknown snd_card_status", __func__);
- }
-
- ret = str_parms_get_str(params, "CPE_STATUS", value, sizeof(value));
- if (ret > 0) {
- if (strstr(value, "OFFLINE")) {
- event.u.status = CPE_STATUS_OFFLINE;
- st_dev->st_callback(AUDIO_EVENT_SSR, &event);
- }
- else if (strstr(value, "ONLINE")) {
- event.u.status = CPE_STATUS_ONLINE;
- st_dev->st_callback(AUDIO_EVENT_SSR, &event);
- }
- else
- ALOGE("%s: unknown CPE status", __func__);
- }
+ stdev_snd_mon_cb(NULL, params);
ret = str_parms_get_int(params, "SVA_NUM_SESSIONS", &val);
if (ret >= 0) {
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 1f7fb5f..b759f4d 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -505,6 +505,10 @@
static int out_set_voip_volume(struct audio_stream_out *stream, float left, float right);
static int out_set_pcm_volume(struct audio_stream_out *stream, float left, float right);
+static void adev_snd_mon_cb(void *cookie, struct str_parms *parms);
+static void in_snd_mon_cb(void * stream, struct str_parms * parms);
+static void out_snd_mon_cb(void * stream, struct str_parms * parms);
+
#ifdef AUDIO_FEATURE_ENABLED_GCOV
extern void __gcov_flush();
static void enable_gcov()
@@ -2698,15 +2702,18 @@
(usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
}
+ }
+ enable_audio_route(adev, usecase);
- /* Notify device change info to effect clients registered */
+ /* Notify device change info to effect clients registered */
+ if (usecase->type == PCM_PLAYBACK) {
audio_extn_gef_notify_device_config(
usecase->stream.out->devices,
usecase->stream.out->channel_mask,
usecase->stream.out->app_type_cfg.sample_rate,
- platform_get_snd_device_acdb_id(usecase->out_snd_device));
+ platform_get_snd_device_acdb_id(usecase->out_snd_device),
+ usecase->stream.out->app_type_cfg.app_type);
}
- enable_audio_route(adev, usecase);
audio_extn_qdsp_set_device(usecase);
@@ -2909,8 +2916,10 @@
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
@@ -5117,6 +5126,24 @@
volume[1] = (long)(AmpToDb(right));
mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
return 0;
+ } else if ((out->devices & AUDIO_DEVICE_OUT_BUS) &&
+ (audio_extn_auto_hal_get_snd_device_for_car_audio_stream(out) ==
+ SND_DEVICE_OUT_BUS_MEDIA)) {
+ ALOGD("%s: Overriding offload set volume for media bus stream", __func__);
+ struct listnode *node = NULL;
+ list_for_each(node, &adev->active_outputs_list) {
+ streams_output_ctxt_t *out_ctxt = node_to_item(node,
+ streams_output_ctxt_t,
+ list);
+ if (out_ctxt->output->usecase == USECASE_AUDIO_PLAYBACK_MEDIA) {
+ out->volume_l = out_ctxt->output->volume_l;
+ out->volume_r = out_ctxt->output->volume_r;
+ }
+ }
+ if (!out->a2dp_compress_mute) {
+ ret = out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
+ }
+ return ret;
} else {
pthread_mutex_lock(&out->compr_mute_lock);
ALOGV("%s: compress mute %d", __func__, out->a2dp_compress_mute);
@@ -6228,7 +6255,7 @@
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) {
@@ -6532,6 +6559,13 @@
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;
@@ -7380,6 +7414,11 @@
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 &&
@@ -7463,6 +7502,10 @@
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.
@@ -8031,6 +8074,23 @@
if (!parms)
goto error;
+ /* notify adev and input/output streams on the snd card status */
+ adev_snd_mon_cb((void *)adev, parms);
+
+ list_for_each(node, &adev->active_outputs_list) {
+ streams_output_ctxt_t *out_ctxt = node_to_item(node,
+ streams_output_ctxt_t,
+ list);
+ out_snd_mon_cb((void *)out_ctxt->output, parms);
+ }
+
+ list_for_each(node, &adev->active_inputs_list) {
+ streams_input_ctxt_t *in_ctxt = node_to_item(node,
+ streams_input_ctxt_t,
+ list);
+ in_snd_mon_cb((void *)in_ctxt->input, parms);
+ }
+
pthread_mutex_lock(&adev->lock);
ret = str_parms_get_str(parms, "BT_SCO", value, sizeof(value));
if (ret >= 0) {
@@ -8211,13 +8271,17 @@
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;
@@ -8725,6 +8789,8 @@
}
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
@@ -8845,7 +8911,7 @@
(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 {
@@ -8896,7 +8962,7 @@
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;
}
@@ -9023,7 +9089,7 @@
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 98f587a..5fffd87 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -4229,7 +4229,7 @@
* 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;
@@ -4312,7 +4312,7 @@
}
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 ||
@@ -4688,8 +4688,10 @@
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 )
@@ -4703,8 +4705,10 @@
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 8ce1d58..0321714 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -89,7 +89,9 @@
defined (PLATFORM_QCS605) || defined (PLATFORM_MSMNILE) || \
defined (PLATFORM_KONA) || defined (PLATFORM_MSMSTEPPE) || \
defined (PLATFORM_QCS405) || defined (PLATFORM_TRINKET) || \
- defined (PLATFORM_LITO) || defined (PLATFORM_MSMFALCON)
+ defined (PLATFORM_LITO) || defined (PLATFORM_MSMFALCON) || \
+ defined (PLATFORM_ATOLL)
+
#include <sound/devdep_params.h>
#endif
@@ -2017,7 +2019,9 @@
!strncmp(snd_card_name, "sdx-tavil-i2s-snd-card",
sizeof("sdx-tavil-i2s-snd-card")) ||
!strncmp(snd_card_name, "sda845-tavil-i2s-snd-card",
- sizeof("sda845-tavil-i2s-snd-card"))) {
+ sizeof("sda845-tavil-i2s-snd-card")) ||
+ !strncmp(snd_card_name, "sa6155-adp-star-snd-card",
+ sizeof("sa6155-adp-star-snd-card"))) {
plat_data->is_i2s_ext_modem = true;
}
ALOGV("%s, is_i2s_ext_modem:%d soundcard name is %s",__func__,
@@ -2026,6 +2030,20 @@
return plat_data->is_i2s_ext_modem;
}
+static bool is_auto_snd_card(const char *snd_card_name)
+{
+ bool is_auto_snd_card = false;
+
+ if (!strncmp(snd_card_name, "sa6155-adp-star-snd-card",
+ sizeof("sa6155-adp-star-snd-card"))) {
+ is_auto_snd_card = true;
+ ALOGV("%s : Auto snd card detected: soundcard name is %s",__func__,
+ snd_card_name);
+ }
+
+ return is_auto_snd_card;
+}
+
static void set_platform_defaults(struct platform_data * my_data)
{
int32_t dev;
@@ -2923,7 +2941,8 @@
return NULL;
}
- if (platform_is_i2s_ext_modem(snd_card_name, my_data)) {
+ if (platform_is_i2s_ext_modem(snd_card_name, my_data) &&
+ !is_auto_snd_card(snd_card_name)) {
ALOGD("%s: Call MIXER_XML_PATH_I2S", __func__);
adev->audio_route = audio_route_init(adev->snd_card,
@@ -3161,7 +3180,7 @@
/* Initialize ACDB ID's */
- if (my_data->is_i2s_ext_modem)
+ if (my_data->is_i2s_ext_modem && !is_auto_snd_card(snd_card_name))
platform_info_init(PLATFORM_INFO_XML_PATH_I2S, my_data, PLATFORM);
else if (!strncmp(snd_card_name, "sdm660-snd-card-skush",
sizeof("sdm660-snd-card-skush")))
@@ -3360,6 +3379,7 @@
property_get("ro.baseband", baseband, "");
if ((!strncmp("apq8084", platform, sizeof("apq8084")) ||
!strncmp("msm8996", platform, sizeof("msm8996")) ||
+ !strncmp("sm6150", platform, sizeof("sm6150")) ||
!strncmp("sdx", platform, sizeof("sdx")) ||
!strncmp("sdm845", platform, sizeof("sdm845"))) &&
( !strncmp("mdm", baseband, (sizeof("mdm")-1)) ||
@@ -4859,6 +4879,9 @@
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)
@@ -5636,7 +5659,7 @@
* 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;
@@ -5760,7 +5783,7 @@
}
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)) {
@@ -6258,8 +6281,10 @@
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 )
@@ -6273,8 +6298,10 @@
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 ||
@@ -7999,6 +8026,9 @@
{
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;
@@ -8012,6 +8042,17 @@
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;
@@ -10642,7 +10683,8 @@
defined (PLATFORM_QCS605) || defined (PLATFORM_MSMNILE) || \
defined (PLATFORM_KONA) || defined (PLATFORM_MSMSTEPPE) || \
defined (PLATFORM_QCS405) || defined (PLATFORM_TRINKET) || \
- defined (PLATFORM_LITO) || defined (PLATFORM_MSMFALCON)
+ defined (PLATFORM_LITO) || defined (PLATFORM_MSMFALCON) || \
+ defined (PLATFORM_ATOLL)
int platform_get_mmap_data_fd(void *platform, int fe_dev, int dir, int *fd,
uint32_t *size)
{
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 1d56a7e..7b4647a 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -480,7 +480,8 @@
defined (PLATFORM_QCS605) ||defined (PLATFORM_SDX24) || \
defined (PLATFORM_MSMNILE) || defined (PLATFORM_KONA) || \
defined (PLATFORM_MSMSTEPPE) || defined (PLATFORM_QCS405) || \
- defined (PLATFORM_TRINKET) || defined (PLATFORM_LITO)
+ defined (PLATFORM_TRINKET) || defined (PLATFORM_LITO) || \
+ defined (PLATFORM_ATOLL)
#define PLAYBACK_OFFLOAD_DEVICE2 17
#elif defined (PLATFORM_MSMFALCON) || defined (PLATFORM_MSM8937)
#define PLAYBACK_OFFLOAD_DEVICE2 24
@@ -493,7 +494,7 @@
defined (PLATFORM_KONA) || defined (PLATFORM_MSMSTEPPE) || \
defined (PLATFORM_QCS405) || defined (PLATFORM_TRINKET) || \
defined (PLATFORM_LITO) || defined (PLATFORM_MSMFALCON) || \
- defined (PLATFORM_MSM8937)
+ defined (PLATFORM_MSM8937) || defined (PLATFORM_ATOLL)
#define PLAYBACK_OFFLOAD_DEVICE3 18
#define PLAYBACK_OFFLOAD_DEVICE4 34
#define PLAYBACK_OFFLOAD_DEVICE5 35
@@ -585,6 +586,14 @@
#define VOLTE_CALL_PCM_DEVICE 15
#define QCHAT_CALL_PCM_DEVICE 37
#define VOWLAN_CALL_PCM_DEVICE 16
+#elif PLATFORM_AUTO
+#define HOST_LESS_RX_ID 41
+#define HOST_LESS_TX_ID 42
+#define VOICE_CALL_PCM_DEVICE 8
+#define VOICE2_CALL_PCM_DEVICE -1
+#define VOLTE_CALL_PCM_DEVICE -1
+#define QCHAT_CALL_PCM_DEVICE -1
+#define VOWLAN_CALL_PCM_DEVICE -1
#else
#define VOICE_CALL_PCM_DEVICE 2
#define VOICE2_CALL_PCM_DEVICE 22
diff --git a/hal/platform_info.c b/hal/platform_info.c
index 0f5f0f9..d73792c 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -653,6 +653,8 @@
}
str_parms_add_str(my_data.kvpairs, (char*)attr[1], (char*)attr[3]);
+ if (my_data.caller == PLATFORM)
+ platform_set_parameters(my_data.platform, my_data.kvpairs);
done:
return;
}
@@ -1405,9 +1407,6 @@
section = ROOT;
} else if (strcmp(tag_name, "config_params") == 0) {
section = ROOT;
- if (my_data.caller == PLATFORM) {
- platform_set_parameters(my_data.platform, my_data.kvpairs);
- }
} else if (strcmp(tag_name, "operator_specific") == 0) {
section = ROOT;
} else if (strcmp(tag_name, "interface_names") == 0) {
diff --git a/hal/voice.c b/hal/voice.c
index 729ab27..c455537 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -48,6 +48,10 @@
.format = PCM_FORMAT_S16_LE,
};
+#ifdef PLATFORM_AUTO
+struct pcm *voice_loopback_tx = NULL;
+struct pcm *voice_loopback_rx = NULL;
+#endif
static struct voice_session *voice_get_session_from_use_case(struct audio_device *adev,
audio_usecase_t usecase_id)
{
@@ -182,6 +186,16 @@
session->pcm_tx = NULL;
}
+#ifdef PLATFORM_AUTO
+ if(voice_loopback_rx) {
+ pcm_close(voice_loopback_rx);
+ voice_loopback_rx = NULL;
+ }
+ if(voice_loopback_tx) {
+ pcm_close(voice_loopback_tx);
+ voice_loopback_tx = NULL;
+ }
+#endif
/* 2. Get and set stream specific mixer controls */
disable_audio_route(adev, uc_info);
@@ -201,6 +215,9 @@
int ret = 0;
struct audio_usecase *uc_info;
int pcm_dev_rx_id, pcm_dev_tx_id;
+#ifdef PLATFORM_AUTO
+ int pcm_dev_loopback_rx_id, pcm_dev_loopback_tx_id;
+#endif
uint32_t sample_rate = 8000;
struct voice_session *session = NULL;
struct pcm_config voice_config = pcm_config_voice_call;
@@ -246,6 +263,10 @@
select_devices(adev, usecase_id);
+#ifdef PLATFORM_AUTO
+ pcm_dev_loopback_rx_id = HOST_LESS_RX_ID;
+ pcm_dev_loopback_tx_id = HOST_LESS_TX_ID;
+#endif
pcm_dev_rx_id = platform_get_pcm_device_id(uc_info->id, PCM_PLAYBACK);
pcm_dev_tx_id = platform_get_pcm_device_id(uc_info->id, PCM_CAPTURE);
@@ -287,6 +308,28 @@
goto error_start_voice;
}
+#ifdef PLATFORM_AUTO
+ voice_loopback_rx = pcm_open(adev->snd_card,
+ pcm_dev_loopback_rx_id,
+ PCM_OUT, &voice_config);
+ if (voice_loopback_rx < 0 || !pcm_is_ready(voice_loopback_rx)) {
+ ALOGE("%s: Either could not open pcm_dev_loopback_rx_id %d or %s",
+ __func__, pcm_dev_loopback_rx_id, pcm_get_error(voice_loopback_rx));
+ ret = -EIO;
+ goto error_start_voice;
+ }
+
+ voice_loopback_tx = pcm_open(adev->snd_card,
+ pcm_dev_loopback_tx_id,
+ PCM_IN, &voice_config);
+ if (voice_loopback_tx < 0 || !pcm_is_ready(voice_loopback_tx)) {
+ ALOGE("%s: Either could not open pcm_dev_loopback_tx_id %d or %s",
+ __func__, pcm_dev_loopback_tx_id, pcm_get_error(voice_loopback_tx));
+ ret = -EIO;
+ goto error_start_voice;
+ }
+#endif
+
if(adev->mic_break_enabled)
platform_set_mic_break_det(adev->platform, true);
@@ -302,6 +345,20 @@
goto error_start_voice;
}
+#ifdef PLATFORM_AUTO
+ ret = pcm_start(voice_loopback_tx);
+ if (ret != 0) {
+ ALOGE("%s: %s", __func__, pcm_get_error(voice_loopback_tx));
+ goto error_start_voice;
+ }
+
+ ret = pcm_start(voice_loopback_rx);
+ if (ret != 0) {
+ ALOGE("%s: %s", __func__, pcm_get_error(voice_loopback_rx));
+ goto error_start_voice;
+ }
+#endif
+
/* Enable aanc only when no calls are active */
if (!voice_is_call_state_active(adev))
voice_check_and_update_aanc_path(adev, uc_info->out_snd_device, true);
@@ -377,6 +434,22 @@
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__);