Merge "configs: Add audio configs file for msm8998"
diff --git a/configs/kona/audio_configs.xml b/configs/kona/audio_configs.xml
index 8c24bb2..bcc617d 100644
--- a/configs/kona/audio_configs.xml
+++ b/configs/kona/audio_configs.xml
@@ -70,6 +70,7 @@
<flag name="a2dp_offload_enabled" value="true" />
<flag name="anc_headset_enabled" value="true" />
<flag name="audiosphere_enabled" value="true" />
+ <flag name="audio_zoom_enabled" value="false" />
<flag name="battery_listener_enabled" value="true" />
<flag name="compress_capture_enabled" value="false" />
<flag name="compress_in_enabled" value="true" />
diff --git a/configs/kona/audio_configs_stock.xml b/configs/kona/audio_configs_stock.xml
index b9ecf45..e26ef14 100644
--- a/configs/kona/audio_configs_stock.xml
+++ b/configs/kona/audio_configs_stock.xml
@@ -32,6 +32,7 @@
<flag name="a2dp_offload_enabled" value="true" />
<flag name="afe_proxy_enabled" value="false" />
<flag name="anc_headset_enabled" value="false" />
+ <flag name="audio_zoom_enabled" value="false" />
<flag name="audiosphere_enabled" value="false" />
<flag name="battery_listener_enabled" value="false" />
<flag name="compress_capture_enabled" value="false" />
diff --git a/configs/kona/audio_policy_configuration.xml b/configs/kona/audio_policy_configuration.xml
index 50920b3..829f181 100644
--- a/configs/kona/audio_policy_configuration.xml
+++ b/configs/kona/audio_policy_configuration.xml
@@ -263,17 +263,17 @@
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_TWSP">
<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_TWSP">
<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_TWSP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
diff --git a/configs/lito/sound_trigger_mixer_paths.xml b/configs/lito/sound_trigger_mixer_paths.xml
index 3ab9c0f..4db6412 100644
--- a/configs/lito/sound_trigger_mixer_paths.xml
+++ b/configs/lito/sound_trigger_mixer_paths.xml
@@ -213,7 +213,7 @@
<path name="listen-ape-handset-mic">
<ctl name="VA_AIF1_CAP Mixer DEC0" value="1" />
<ctl name="VA DEC0 MUX" value="MSM_DMIC" />
- <ctl name="VA DMIC MUX0" value="DMIC0" />
+ <ctl name="VA DMIC MUX0" value="DMIC4" />
</path>
<path name="listen-ape-handset-mic-preproc">
@@ -226,8 +226,8 @@
<ctl name="VA_AIF1_CAP Mixer DEC1" value="1" />
<ctl name="VA DEC0 MUX" value="MSM_DMIC" />
<ctl name="VA DEC1 MUX" value="MSM_DMIC" />
- <ctl name="VA DMIC MUX0" value="DMIC0" />
- <ctl name="VA DMIC MUX1" value="DMIC1" />
+ <ctl name="VA DMIC MUX0" value="DMIC4" />
+ <ctl name="VA DMIC MUX1" value="DMIC0" />
</path>
<path name="listen-ape-handset-tmic">
@@ -238,9 +238,9 @@
<ctl name="VA DEC0 MUX" value="MSM_DMIC" />
<ctl name="VA DEC1 MUX" value="MSM_DMIC" />
<ctl name="VA DEC2 MUX" value="MSM_DMIC" />
- <ctl name="VA DMIC MUX0" value="DMIC0" />
- <ctl name="VA DMIC MUX1" value="DMIC1" />
- <ctl name="VA DMIC MUX2" value="DMIC4" />
+ <ctl name="VA DMIC MUX0" value="DMIC4" />
+ <ctl name="VA DMIC MUX1" value="DMIC0" />
+ <ctl name="VA DMIC MUX2" value="DMIC1" />
</path>
<path name="listen-ape-handset-qmic">
@@ -253,10 +253,10 @@
<ctl name="VA DEC1 MUX" value="MSM_DMIC" />
<ctl name="VA DEC2 MUX" value="MSM_DMIC" />
<ctl name="VA DEC3 MUX" value="MSM_DMIC" />
- <ctl name="VA DMIC MUX0" value="DMIC0" />
- <ctl name="VA DMIC MUX1" value="DMIC1" />
- <ctl name="VA DMIC MUX2" value="DMIC2" />
- <ctl name="VA DMIC MUX3" value="DMIC4" />
+ <ctl name="VA DMIC MUX0" value="DMIC4" />
+ <ctl name="VA DMIC MUX1" value="DMIC0" />
+ <ctl name="VA DMIC MUX2" value="DMIC1" />
+ <ctl name="VA DMIC MUX3" value="DMIC2" />
</path>
<path name="listen-ape-headset-mic">
diff --git a/configs/msmnile/audio_configs.xml b/configs/msmnile/audio_configs.xml
index 8c24bb2..40d04d7 100644
--- a/configs/msmnile/audio_configs.xml
+++ b/configs/msmnile/audio_configs.xml
@@ -69,6 +69,7 @@
<!-- AHAL Configs -->
<flag name="a2dp_offload_enabled" value="true" />
<flag name="anc_headset_enabled" value="true" />
+ <flag name="audio_zoom_enabled" value="false" />
<flag name="audiosphere_enabled" value="true" />
<flag name="battery_listener_enabled" value="true" />
<flag name="compress_capture_enabled" value="false" />
diff --git a/configs/msmnile/audio_configs_stock.xml b/configs/msmnile/audio_configs_stock.xml
index 6414675..bbc44c3 100644
--- a/configs/msmnile/audio_configs_stock.xml
+++ b/configs/msmnile/audio_configs_stock.xml
@@ -32,6 +32,7 @@
<flag name="a2dp_offload_enabled" value="true" />
<flag name="afe_proxy_enabled" value="false" />
<flag name="anc_headset_enabled" value="false" />
+ <flag name="audio_zoom_enabled" value="true" />
<flag name="audiosphere_enabled" value="false" />
<flag name="battery_listener_enabled" value="false" />
<flag name="compress_capture_enabled" value="false" />
diff --git a/configs/msmnile/audio_policy_configuration.xml b/configs/msmnile/audio_policy_configuration.xml
index 50920b3..829f181 100644
--- a/configs/msmnile/audio_policy_configuration.xml
+++ b/configs/msmnile/audio_policy_configuration.xml
@@ -263,17 +263,17 @@
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_TWSP">
<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_TWSP">
<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_TWSP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
diff --git a/configs/msmsteppe/audio_policy_configuration.xml b/configs/msmsteppe/audio_policy_configuration.xml
index 5435fad..61ea4e1 100644
--- a/configs/msmsteppe/audio_policy_configuration.xml
+++ b/configs/msmsteppe/audio_policy_configuration.xml
@@ -257,17 +257,17 @@
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_APTX_TWSP">
<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_APTX_TWSP">
<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_APTX_TWSP">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</devicePort>
diff --git a/hal/Android.mk b/hal/Android.mk
index 383a272..a6bd9bc 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -340,12 +340,6 @@
LOCAL_STATIC_LIBRARIES += libprofile_rt
endif
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUDIO_ZOOM)), true)
- LOCAL_CFLAGS += -DAUDIOZOOM_QDSP_ENABLED
- LOCAL_SRC_FILES += audio_extn/audiozoom.c
-endif
-
-
#ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUTO_HAL)),true)
# LOCAL_CFLAGS += -DAUDIO_EXTN_AUTO_HAL_ENABLED
# LOCAL_SRC_FILES += audio_extn/auto_hal.c
diff --git a/hal/acdb.c b/hal/acdb.c
index 7394906..d2b2b94 100644
--- a/hal/acdb.c
+++ b/hal/acdb.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2018, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2019, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -28,6 +28,8 @@
#include <time.h>
#include "acdb.h"
#include "platform_api.h"
+#include "audio_extn.h"
+#include <platform.h>
#ifdef INSTANCE_ID_ENABLED
int check_and_set_instance_id_support(struct mixer* mixer, bool acdb_support)
@@ -57,6 +59,22 @@
#define check_and_set_instance_id_support(x, y) -ENOSYS
#endif
+void get_platform_file_for_device(struct mixer *mixer, char* platform_info_file)
+{
+ const char *snd_card_name = NULL;
+
+ if (mixer != NULL) {
+ /* Get Sound card name */
+ snd_card_name = mixer_get_name(mixer);
+ if (!snd_card_name) {
+ ALOGE("failed to allocate memory for snd_card_name");
+ return;
+ }
+ /* Get platform info file for target */
+ audio_extn_utils_get_platform_info(snd_card_name, platform_info_file);
+ }
+}
+
int acdb_init(int snd_card_num)
{
@@ -84,7 +102,7 @@
int result = -1;
char *cvd_version = NULL;
-
+ char platform_info_file[MIXER_PATH_MAX_LENGTH] = PLATFORM_INFO_XML_PATH;
const char *snd_card_name = NULL;
struct acdb_platform_data *my_data = NULL;
@@ -100,9 +118,9 @@
}
list_init(&my_data->acdb_meta_key_list);
-
+ get_platform_file_for_device(mixer, platform_info_file);
/* Extract META KEY LIST INFO */
- platform_info_init(PLATFORM_INFO_XML_PATH, my_data, ACDB_EXTN);
+ platform_info_init(platform_info_file, my_data, ACDB_EXTN);
my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
if (my_data->acdb_handle == NULL) {
@@ -164,13 +182,13 @@
/* Get Sound card name */
snd_card_name = mixer_get_name(mixer);
+ snd_card_name = platform_get_snd_card_name_for_acdb_loader(snd_card_name);
if (!snd_card_name) {
- ALOGE("failed to allocate memory for snd_card_name");
+ ALOGE("failed to get snd_card_name");
result = -1;
goto cleanup;
}
- snd_card_name = platform_get_snd_card_name_for_acdb_loader(snd_card_name);
int key = 0;
struct listnode *node = NULL;
struct meta_key_list *key_info = NULL;
diff --git a/hal/ahal_config_helper.cpp b/hal/ahal_config_helper.cpp
index e46b8f3..9313e5f 100644
--- a/hal/ahal_config_helper.cpp
+++ b/hal/ahal_config_helper.cpp
@@ -109,7 +109,8 @@
true, /* COMPRESS_METADATA_NEEDED */
false, /* INCALL_MUSIC */
false, /* COMPRESS_VOIP */
- true /* DYNAMIC_ECNS */
+ true, /* DYNAMIC_ECNS */
+ false, /* AUDIO_ZOOM */
};
#else
if (isVendorEnhancedFwk) {
@@ -155,6 +156,7 @@
true, /* INCALL_MUSIC */
false, /* COMPRESS_VOIP */
true, /* DYNAMIC_ECNS */
+ false, /* AUDIO_ZOOM */
};
} else {
defaultConfigs = {
@@ -198,7 +200,8 @@
false, /* COMPRESS_METADATA_NEEDED */
true, /* INCALL_MUSIC */
false, /* COMPRESS_VOIP */
- false /* DYNAMIC_ECNS */
+ false, /* DYNAMIC_ECNS */
+ true, /* AUDIO_ZOOM */
};
}
#endif
diff --git a/hal/ahal_config_helper.h b/hal/ahal_config_helper.h
index 39ed68e..d658f7d 100644
--- a/hal/ahal_config_helper.h
+++ b/hal/ahal_config_helper.h
@@ -76,6 +76,7 @@
bool incall_music_enabled;
bool compress_voip_enabled;
bool dynamic_ecns_enabled;
+ bool audio_zoom_enabled;
} AHalValues;
#ifdef __cplusplus
diff --git a/hal/audio_extn/Android.mk b/hal/audio_extn/Android.mk
index feae999..255bc4e 100644
--- a/hal/audio_extn/Android.mk
+++ b/hal/audio_extn/Android.mk
@@ -63,7 +63,7 @@
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 qcs605 msmnile kona lito atoll sdm660 msm8937 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 qcs605 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -121,7 +121,7 @@
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -182,7 +182,7 @@
PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 $(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
endif
@@ -234,7 +234,7 @@
#--------------------------------------------
include $(CLEAR_VARS)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 $(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
endif
@@ -289,7 +289,7 @@
include $(CLEAR_VARS)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 $(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
endif
@@ -350,7 +350,7 @@
PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
-ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito atoll sdm660 msm8937 msm8998 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -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 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE),$(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 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE),$(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 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE),$(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 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE),$(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 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter sdm845 sdm710 msmnile kona lito sdm660 msm8937 msm8998 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
# B-family platform uses msm8974 code base
AUDIO_PLATFORM := msm8974
MULTIPLE_HW_VARIANTS_ENABLED := true
@@ -765,3 +765,62 @@
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_HEADER_LIBRARIES += libsystem_headers
include $(BUILD_SHARED_LIBRARY)
+#-------------------------------------------
+# Build AUDIOZOOM
+#-------------------------------------------
+include $(CLEAR_VARS)
+
+LOCAL_MODULE:= libaudiozoom
+LOCAL_VENDOR_MODULE := true
+
+PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
+AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
+
+ifneq ($(filter sdm845 sdm710 msmnile kona sdm660 msm8937 $(MSMSTEPPE),$(TARGET_BOARD_PLATFORM)),)
+ # B-family platform uses msm8974 code base
+ AUDIO_PLATFORM = msm8974
+ MULTIPLE_HW_VARIANTS_ENABLED := true
+endif
+
+LOCAL_SRC_FILES:= \
+ audiozoom.c
+
+LOCAL_CFLAGS += \
+ -Wall \
+ -Werror \
+ -Wno-unused-function \
+ -Wno-unused-variable
+
+LOCAL_SHARED_LIBRARIES := \
+ libaudioutils \
+ libcutils \
+ liblog \
+ libtinyalsa \
+ libtinycompress \
+ libaudioroute \
+ libdl \
+ libexpat
+
+LOCAL_C_INCLUDES := \
+ $(PRIMARY_HAL_PATH) \
+ $(PRIMARY_HAL_PATH)/$(AUDIO_PLATFORM) \
+ external/tinyalsa/include \
+ external/tinycompress/include \
+ external/expat/lib \
+ system/media/audio_utils/include \
+ $(call include-path-for, audio-route) \
+
+LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
+LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/techpack/audio/include
+LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_DLKM)),true)
+ LOCAL_HEADER_LIBRARIES += audio_kernel_headers
+ LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/vendor/qcom/opensource/audio-kernel/include
+endif
+
+LOCAL_HEADER_LIBRARIES += libhardware_headers
+LOCAL_HEADER_LIBRARIES += libsystem_headers
+include $(BUILD_SHARED_LIBRARY)
+
+
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
index c9ab383..0d58df6 100644
--- a/hal/audio_extn/a2dp.c
+++ b/hal/audio_extn/a2dp.c
@@ -2678,7 +2678,7 @@
goto param_handled;
list_for_each(node, &a2dp.adev->usecase_list) {
uc_info = node_to_item(node, struct audio_usecase, list);
- if (uc_info->type == PCM_PLAYBACK &&
+ if (uc_info->stream.out && uc_info->type == PCM_PLAYBACK &&
(uc_info->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
pthread_mutex_unlock(&a2dp.adev->lock);
fp_check_a2dp_restore(a2dp.adev, uc_info->stream.out, false);
@@ -2719,7 +2719,7 @@
}
list_for_each(node, &a2dp.adev->usecase_list) {
uc_info = node_to_item(node, struct audio_usecase, list);
- if (uc_info->type == PCM_PLAYBACK &&
+ if (uc_info->stream.out && uc_info->type == PCM_PLAYBACK &&
(uc_info->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
pthread_mutex_unlock(&a2dp.adev->lock);
fp_check_a2dp_restore(a2dp.adev, uc_info->stream.out, true);
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 1688b9d..3d6cd4a 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -46,6 +46,7 @@
#include <cutils/properties.h>
#include <log/log.h>
#include <unistd.h>
+#include <sched.h>
#include "audio_hw.h"
#include "audio_extn.h"
@@ -190,6 +191,7 @@
static bool audio_extn_compress_in_enabled = false;
static bool audio_extn_battery_listener_enabled = false;
static bool audio_extn_maxx_audio_enabled = false;
+static bool audio_extn_audiozoom_enabled = false;
#define AUDIO_PARAMETER_KEY_AANC_NOISE_LEVEL "aanc_noise_level"
#define AUDIO_PARAMETER_KEY_ANC "anc_enabled"
@@ -821,7 +823,7 @@
// Refresh device selection for anc playback
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->type != PCM_CAPTURE) {
+ if (usecase->stream.out && usecase->type != PCM_CAPTURE) {
if (usecase->stream.out->devices == \
AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
usecase->stream.out->devices == \
@@ -4761,6 +4763,105 @@
}
// END: BATTERY_LISTENER ================================================================
+// START: AUDIOZOOM_FEATURE =====================================================================
+#ifdef __LP64__
+#define AUDIOZOOM_LIB_PATH "/vendor/lib64/libaudiozoom.so"
+#else
+#define AUDIOZOOM_LIB_PATH "/vendor/lib/libaudiozoom.so"
+#endif
+
+static void *audiozoom_lib_handle = NULL;
+
+typedef int (*audiozoom_init_t)(audiozoom_init_config_t);
+static audiozoom_init_t audiozoom_init;
+
+typedef int (*audiozoom_set_microphone_direction_t)(struct stream_in *,
+ audio_microphone_direction_t);
+static audiozoom_set_microphone_direction_t audiozoom_set_microphone_direction;
+
+typedef int (*audiozoom_set_microphone_field_dimension_t)(struct stream_in *, float);
+static audiozoom_set_microphone_field_dimension_t audiozoom_set_microphone_field_dimension;
+
+int audiozoom_feature_init(bool is_feature_enabled)
+{
+ audio_extn_audiozoom_enabled = is_feature_enabled;
+ ALOGD("%s: Called with feature %s", __func__,
+ is_feature_enabled ? "Enabled" : "NOT Enabled");
+ if (is_feature_enabled) {
+ // dlopen lib
+ audiozoom_lib_handle = dlopen(AUDIOZOOM_LIB_PATH, RTLD_NOW);
+
+ if (!audiozoom_lib_handle) {
+ ALOGE("%s: dlopen failed", __func__);
+ goto feature_disabled;
+ }
+
+ if (!(audiozoom_init =
+ (audiozoom_init_t)dlsym(audiozoom_lib_handle, "audiozoom_init")) ||
+ !(audiozoom_set_microphone_direction =
+ (audiozoom_set_microphone_direction_t)dlsym(audiozoom_lib_handle,
+ "audiozoom_set_microphone_direction")) ||
+ !(audiozoom_set_microphone_field_dimension =
+ (audiozoom_set_microphone_field_dimension_t)dlsym(audiozoom_lib_handle,
+ "audiozoom_set_microphone_field_dimension"))) {
+ ALOGE("%s: dlsym failed", __func__);
+ goto feature_disabled;
+ }
+
+ ALOGD("%s:: ---- Feature AUDIOZOOM is Enabled ----", __func__);
+ return 0;
+ }
+feature_disabled:
+ if (audiozoom_lib_handle) {
+ dlclose(audiozoom_lib_handle);
+ audiozoom_lib_handle = NULL;
+ }
+
+ audiozoom_init = NULL;
+ audiozoom_set_microphone_direction = NULL;
+ audiozoom_set_microphone_field_dimension = NULL;
+ ALOGW(":: %s: ---- Feature AUDIOZOOM is disabled ----", __func__);
+ return -ENOSYS;
+}
+
+bool audio_extn_is_audiozoom_enabled()
+{
+ return audio_extn_audiozoom_enabled;
+}
+
+int audio_extn_audiozoom_init()
+{
+ int ret_val = 0;
+ if (audiozoom_init) {
+ audiozoom_init_config_t init_config;
+ init_config.fp_platform_set_parameters = platform_set_parameters;
+ ret_val = audiozoom_init(init_config);
+ }
+
+ return ret_val;
+}
+
+int audio_extn_audiozoom_set_microphone_direction(struct stream_in *stream,
+ audio_microphone_direction_t dir)
+{
+ int ret_val = -ENOSYS;
+ if (audiozoom_set_microphone_direction)
+ ret_val = audiozoom_set_microphone_direction(stream, dir);
+
+ return ret_val;
+}
+
+int audio_extn_audiozoom_set_microphone_field_dimension(struct stream_in *stream,
+ float zoom)
+{
+ int ret_val = -ENOSYS;
+ if (audiozoom_set_microphone_field_dimension)
+ ret_val = audiozoom_set_microphone_field_dimension(stream, zoom);
+
+ return ret_val;
+}
+// END: AUDIOZOOM_FEATURE =====================================================================
+
// START: MAXX_AUDIO =====================================================================
#ifdef __LP64__
#define MAXX_AUDIO_LIB_PATH "/vendor/lib64/libmaxxaudio.so"
@@ -4999,6 +5100,8 @@
case MAXX_AUDIO:
maxx_audio_feature_init(enable);
break;
+ case AUDIO_ZOOM:
+ audiozoom_feature_init(enable);
default:
break;
}
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 60b2610..043ce48 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -327,6 +327,22 @@
};
typedef struct a2dp_offload_init_config a2dp_offload_init_config_t;
// END: A2DP_OFFLOAD FEATURE ====================================================
+
+typedef int (*fp_platform_set_parameters_t)(void*, struct str_parms*);
+
+// START: AUDIOZOOM FEATURE ==================================================
+int audio_extn_audiozoom_init();
+int audio_extn_audiozoom_set_microphone_direction(struct stream_in *stream,
+ audio_microphone_direction_t dir);
+int audio_extn_audiozoom_set_microphone_field_dimension(struct stream_in *stream, float zoom);
+bool audio_extn_is_audiozoom_enabled();
+
+struct audiozoom_init_config {
+ fp_platform_set_parameters_t fp_platform_set_parameters;
+};
+typedef struct audiozoom_init_config audiozoom_init_config_t;
+// END: AUDIOZOOM FEATURE ==================================================
+
// START: MAXX_AUDIO FEATURE ==================================================
void audio_extn_ma_init(void *platform);
void audio_extn_ma_deinit();
@@ -337,7 +353,6 @@
struct str_parms *parms);
bool audio_extn_ma_supported_usb();
bool audio_extn_is_maxx_audio_enabled();
-typedef int (*fp_platform_set_parameters_t)(void*, struct str_parms*);
// --- Function pointers from audio_extn needed by MAXX_AUDIO
struct maxx_audio_init_config {
fp_platform_set_parameters_t fp_platform_set_parameters;
diff --git a/hal/audio_extn/audio_feature_manager.c b/hal/audio_extn/audio_feature_manager.c
index e121426..2de5af3 100644
--- a/hal/audio_extn/audio_feature_manager.c
+++ b/hal/audio_extn/audio_feature_manager.c
@@ -178,6 +178,8 @@
return confValues->compress_voip_enabled;
case DYNAMIC_ECNS:
return confValues->dynamic_ecns_enabled;
+ case AUDIO_ZOOM:
+ return confValues->audio_zoom_enabled;
default:
return false;
}
diff --git a/hal/audio_extn/audio_feature_manager.h b/hal/audio_extn/audio_feature_manager.h
index 8df076c..908b0cb 100644
--- a/hal/audio_extn/audio_feature_manager.h
+++ b/hal/audio_extn/audio_feature_manager.h
@@ -33,6 +33,7 @@
#include <ahal_config_helper.h>
enum audio_ext_feature_t {
+ // Start Audio feature flags
SND_MONITOR = 0,
COMPRESS_CAPTURE,
SOURCE_TRACK,
@@ -71,10 +72,14 @@
BATTERY_LISTENER,
COMPRESS_METADATA_NEEDED,
MAXX_AUDIO,
+ AUDIO_ZOOM,
+ // End Audio feature flags
+ // Start Voice feature flags
COMPRESS_VOIP,
VOICE_START = COMPRESS_VOIP,
DYNAMIC_ECNS,
INCALL_MUSIC,
+ // End Voice feature flags
MAX_SUPPORTED_FEATURE
};
diff --git a/hal/audio_extn/audiozoom.c b/hal/audio_extn/audiozoom.c
index 73e4862..9958cc4 100644
--- a/hal/audio_extn/audiozoom.c
+++ b/hal/audio_extn/audiozoom.c
@@ -23,7 +23,6 @@
#include <expat.h>
#include <audio_hw.h>
#include <system/audio.h>
-#include <platform_api.h>
#include "audio_extn.h"
#include "audiozoom.h"
@@ -32,6 +31,9 @@
#define AUDIOZOOM_PRESET_FILE "/vendor/etc/audiozoom.xml"
+// --- external function dependency ---
+fp_platform_set_parameters_t fp_platform_set_parameters;
+
typedef struct qdsp_audiozoom_cfg {
uint32_t topo_id;
uint32_t module_id;
@@ -116,7 +118,7 @@
}
}
-static int audio_extn_audiozoom_parse_info(const char *filename)
+static int audiozoom_parse_info(const char *filename)
{
XML_Parser parser;
FILE *file;
@@ -176,7 +178,7 @@
return ret;
}
-int audio_extn_audiozoom_set_microphone_direction(
+int audiozoom_set_microphone_direction(
struct stream_in *in, audio_microphone_direction_t dir)
{
(void)in;
@@ -184,7 +186,7 @@
return 0;
}
-static int audio_extn_audiozoom_set_microphone_field_dimension_zoom(
+static int audiozoom_set_microphone_field_dimension_zoom(
struct stream_in *in, float zoom)
{
struct audio_device *adev = in->dev;
@@ -212,7 +214,7 @@
if (ret > 0) {
str_parms_add_str(parms, "cal_data", data);
- platform_set_parameters(adev->platform, parms);
+ fp_platform_set_parameters(adev->platform, parms);
} else {
ALOGE("%s: failed to convert data to string, ret %d", __func__, ret);
}
@@ -222,7 +224,7 @@
return 0;
}
-static int audio_extn_audiozoom_set_microphone_field_dimension_wide_angle(
+static int audiozoom_set_microphone_field_dimension_wide_angle(
struct stream_in *in, float zoom)
{
(void)in;
@@ -230,24 +232,25 @@
return 0;
}
-int audio_extn_audiozoom_set_microphone_field_dimension(
+int audiozoom_set_microphone_field_dimension(
struct stream_in *in, float zoom)
{
if (zoom > 1.0 || zoom < -1.0)
return -EINVAL;
if (zoom >= 0 && zoom <= 1.0)
- return audio_extn_audiozoom_set_microphone_field_dimension_zoom(in, zoom);
+ return audiozoom_set_microphone_field_dimension_zoom(in, zoom);
if (zoom >= -1.0 && zoom <= 0)
- return audio_extn_audiozoom_set_microphone_field_dimension_wide_angle(in, zoom);
+ return audiozoom_set_microphone_field_dimension_wide_angle(in, zoom);
return 0;
}
-int audio_extn_audiozoom_init()
+int audiozoom_init(audiozoom_init_config_t init_config)
{
- audio_extn_audiozoom_parse_info(AUDIOZOOM_PRESET_FILE);
+ fp_platform_set_parameters = init_config.fp_platform_set_parameters;
+ audiozoom_parse_info(AUDIOZOOM_PRESET_FILE);
ALOGV("%s: topo_id=%d, module_id=%d, instance_id=%d, zoom__id=%d, dir_id=%d, app_type=%d",
__func__, qdsp_audiozoom.topo_id, qdsp_audiozoom.module_id, qdsp_audiozoom.instance_id,
diff --git a/hal/audio_extn/audiozoom.h b/hal/audio_extn/audiozoom.h
index 2c0ad71..cb00828 100644
--- a/hal/audio_extn/audiozoom.h
+++ b/hal/audio_extn/audiozoom.h
@@ -17,15 +17,9 @@
#ifndef AUDIOZOOM_H_
#define AUDIOZOOM_H_
-#ifndef AUDIOZOOM_QDSP_ENABLED
-#define audio_extn_audiozoom_init() (0)
-#define audio_extn_audiozoom_set_microphone_direction(stream, dir) (-ENOSYS)
-#define audio_extn_audiozoom_set_microphone_field_dimension(stream, zoom) (-ENOSYS)
-#else
-int audio_extn_audiozoom_init();
-int audio_extn_audiozoom_set_microphone_direction(struct stream_in *stream,
+int audiozoom_init(audiozoom_init_config_t init_config);
+int audiozoom_set_microphone_direction(struct stream_in *stream,
audio_microphone_direction_t dir);
-int audio_extn_audiozoom_set_microphone_field_dimension(struct stream_in *stream, float zoom);
-#endif
+int audiozoom_set_microphone_field_dimension(struct stream_in *stream, float zoom);
#endif /* AUDIOZOOM_H_ */
diff --git a/hal/audio_extn/dolby.c b/hal/audio_extn/dolby.c
index 906c234..335cfbd 100644
--- a/hal/audio_extn/dolby.c
+++ b/hal/audio_extn/dolby.c
@@ -245,7 +245,7 @@
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if ((usecase->type == PCM_PLAYBACK) &&
+ if (usecase->stream.out && (usecase->type == PCM_PLAYBACK) &&
(usecase->devices & ddp_dev) &&
(usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
((usecase->stream.out->format == AUDIO_FORMAT_AC3) ||
@@ -263,7 +263,7 @@
struct audio_usecase *usecase;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if ((usecase->type == PCM_PLAYBACK) &&
+ if (usecase->stream.out && (usecase->type == PCM_PLAYBACK) &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL) &&
(usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
((usecase->stream.out->format == AUDIO_FORMAT_AC3) ||
diff --git a/hal/audio_extn/dts_eagle.c b/hal/audio_extn/dts_eagle.c
index 3771a9d..be7bab1 100644
--- a/hal/audio_extn/dts_eagle.c
+++ b/hal/audio_extn/dts_eagle.c
@@ -112,7 +112,7 @@
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
/* set/get eagle params for offload usecases only */
- if ((usecase->type == PCM_PLAYBACK) && is_offload_usecase(usecase->id)) {
+ if (usecase->stream.out && (usecase->type == PCM_PLAYBACK) && is_offload_usecase(usecase->id)) {
tret = do_DTS_Eagle_params_stream(usecase->stream.out, t, get);
if (tret < 0)
ret = tret;
diff --git a/hal/audio_extn/maxxaudio.c b/hal/audio_extn/maxxaudio.c
index b6249ef..31feb02 100644
--- a/hal/audio_extn/maxxaudio.c
+++ b/hal/audio_extn/maxxaudio.c
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2018-2019 The Android Open Source Project
+ * Copyright (C) 2018 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -284,7 +284,7 @@
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (valid_usecase(usecase)) {
+ if (usecase->stream.out && valid_usecase(usecase)) {
ma_cal.common.app_type = usecase->stream.out->app_type_cfg.app_type;
ma_cal.common.device = usecase->stream.out->devices;
ALOGV("%s: send usecase(%d) app_type(%d) device(%d)",
diff --git a/hal/audio_extn/maxxaudio.h b/hal/audio_extn/maxxaudio.h
index 1ab7f80..4f45f79 100644
--- a/hal/audio_extn/maxxaudio.h
+++ b/hal/audio_extn/maxxaudio.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2018-2019 The Android Open Source Project
+ * Copyright (C) 2018 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -25,4 +25,4 @@
void ma_set_parameters(struct audio_device *adev,
struct str_parms *parms);
bool ma_supported_usb();
-#endif /* MAXXAUDIO_H_ */
\ No newline at end of file
+#endif /* MAXXAUDIO_H_ */
diff --git a/hal/audio_extn/passthru.c b/hal/audio_extn/passthru.c
index a59717c..e900932 100644
--- a/hal/audio_extn/passthru.c
+++ b/hal/audio_extn/passthru.c
@@ -284,7 +284,7 @@
/* find max period time among active playback use cases */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->type == PCM_PLAYBACK &&
+ if (usecase->stream.out && usecase->type == PCM_PLAYBACK &&
usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
o = usecase->stream.out;
temp = o->config.period_size * 1000000LL / o->sample_rate;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 1afeda1..abd1aba 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -75,7 +75,6 @@
#include "voice_extn.h"
#include "ip_hdlr_intf.h"
#include "audio_feature_manager.h"
-#include "audio_extn/audiozoom.h"
#include "sound/compress_params.h"
#include "sound/asound.h"
@@ -1069,6 +1068,8 @@
audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_BUSY);
audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_BUSY);
audio_extn_utils_send_app_type_cfg(adev, usecase);
+ if (audio_extn_is_maxx_audio_enabled())
+ audio_extn_ma_set_device(usecase);
audio_extn_utils_send_audio_calibration(adev, usecase);
if ((usecase->type == PCM_PLAYBACK) && is_offload_usecase(usecase->id)) {
out = usecase->stream.out;
@@ -1615,7 +1616,7 @@
usecase->out_snd_device,
platform_get_input_snd_device(adev->platform, uc_info->devices));
enable_audio_route(adev, usecase);
- if (usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
+ if (usecase->stream.out && usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
out_set_voip_volume(&usecase->stream.out->stream,
usecase->stream.out->volume_l,
usecase->stream.out->volume_r);
@@ -5098,6 +5099,7 @@
pthread_mutex_lock(&adev->lock);
select_devices(adev, out->usecase);
if (!audio_extn_passthru_is_supported_backend_edid_cfg(adev, out)) {
+ pthread_mutex_unlock(&adev->lock);
ret = -EINVAL;
goto exit;
}
@@ -7811,7 +7813,7 @@
struct listnode *node;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if ((usecase->type == PCM_PLAYBACK) &&
+ if (usecase->stream.out && (usecase->type == PCM_PLAYBACK) &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP)){
ALOGD("reconfigure a2dp... forcing device switch");
@@ -8834,6 +8836,8 @@
pthread_mutex_lock(&adev_init_lock);
if ((--audio_device_ref_count) == 0) {
+ if (audio_extn_spkr_prot_is_enabled())
+ audio_extn_spkr_prot_deinit();
audio_extn_snd_mon_unregister_listener(adev);
audio_extn_sound_trigger_deinit(adev);
audio_extn_listen_deinit(adev);
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 1cb3c1d..ace49a0 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -1993,6 +1993,10 @@
snd_card_name = mixer_get_name(my_data->adev->mixer);
snd_card_name = platform_get_snd_card_name_for_acdb_loader(snd_card_name);
+ if (!snd_card_name) {
+ ALOGE("Failed to get snd_card_name");
+ goto cleanup;
+ }
my_data->acdb_init_data.cvd_version = cvd_version;
my_data->acdb_init_data.snd_card_name = strdup(snd_card_name);
@@ -2014,6 +2018,7 @@
strlcpy(my_data->snd_card_name, snd_card_name,
MAX_SND_CARD_STRING_SIZE);
+cleanup:
if (cvd_version)
free(cvd_version);
if (!result) {
@@ -2934,8 +2939,6 @@
/* deinit usb */
audio_extn_usb_deinit();
audio_extn_dap_hal_deinit();
- if (audio_extn_spkr_prot_is_enabled())
- audio_extn_spkr_prot_deinit();
#ifdef DYNAMIC_LOG_ENABLED
log_utils_deinit();
#endif
@@ -3503,7 +3506,7 @@
list_for_each(node, &(platform->adev)->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (is_offload_usecase(usecase->id) &&
+ if (usecase->stream.out && is_offload_usecase(usecase->id) &&
(usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) &&
OUTPUT_SAMPLING_RATE_44100 == usecase->stream.out->sample_rate) {
@@ -7807,7 +7810,8 @@
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase != NULL && usecase->type == PCM_PLAYBACK) {
+ if (usecase != NULL && usecase->stream.out &&
+ usecase->type == PCM_PLAYBACK) {
int new_snd_device[2] = {0};
int i, num_devices = 1;
@@ -8306,7 +8310,7 @@
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->type == PCM_PLAYBACK &&
+ if (usecase->stream.out && usecase->type == PCM_PLAYBACK &&
usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
/*
* If acdb tuning is different for SPEAKER_REVERSE, it is must
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 67845de..e02b9f0 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -1684,7 +1684,8 @@
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase != NULL && usecase->type == PCM_PLAYBACK) {
+ if (usecase != NULL && usecase->stream.out &&
+ usecase->type == PCM_PLAYBACK) {
int new_snd_device[2] = {0};
int i, num_devices = 1;
@@ -2514,11 +2515,11 @@
}
snd_card_name = mixer_get_name(my_data->adev->mixer);
+ snd_card_name = platform_get_snd_card_name_for_acdb_loader(snd_card_name);
if (!snd_card_name) {
ALOGE("Failed to get snd_card_name");
goto cleanup;
}
- snd_card_name = platform_get_snd_card_name_for_acdb_loader(snd_card_name);
my_data->acdb_init_data.cvd_version = cvd_version;
my_data->acdb_init_data.snd_card_name = strdup(snd_card_name);
@@ -3832,8 +3833,6 @@
/* deinit usb */
audio_extn_usb_deinit();
audio_extn_dap_hal_deinit();
- if (audio_extn_spkr_prot_is_enabled())
- audio_extn_spkr_prot_deinit();
#ifdef DYNAMIC_LOG_ENABLED
log_utils_deinit();
#endif
@@ -4123,7 +4122,7 @@
goto done;
}
- if(effect_config == NULL) {
+ if (effect_config == NULL) {
ALOGE("%s: Invalid effect_config", __func__);
ret = -EINVAL;
goto done;
@@ -4151,6 +4150,10 @@
}
device = (struct external_specific_device *)calloc(1, sizeof(struct external_specific_device));
+ if (device == NULL) {
+ ALOGE("%s: memory allocation failed", __func__);
+ return;
+ }
device->usbid = strdup(usbid);
device->acdb_id = acdb_id;
@@ -4426,7 +4429,7 @@
list_for_each(node, &(platform->adev)->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (is_offload_usecase(usecase->id) &&
+ if (usecase->stream.out && is_offload_usecase(usecase->id) &&
(usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
usecase->stream.out->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) &&
OUTPUT_SAMPLING_RATE_44100 == usecase->stream.out->sample_rate) {
@@ -10046,7 +10049,7 @@
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- if (usecase->type == PCM_PLAYBACK &&
+ if (usecase->stream.out && usecase->type == PCM_PLAYBACK &&
usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
/*
* If acdb tuning is different for SPEAKER_REVERSE, it is must
diff --git a/visualizer/offload_visualizer.c b/visualizer/offload_visualizer.c
index ccbd8bc..a478ff6 100644
--- a/visualizer/offload_visualizer.c
+++ b/visualizer/offload_visualizer.c
@@ -505,6 +505,7 @@
pcm_close(pcm);
pcm = NULL;
configure_proxy_capture(mixer, 0);
+ pthread_cond_wait(&cond, &lock);
} else {
capture_enabled = true;
ALOGD("%s: capture ENABLED", __func__);