Merge "audio: hal: Add support for Ambisonic capture"
diff --git a/configs/kona/mixer_paths_mtp.xml b/configs/kona/mixer_paths_mtp.xml
index f7f4110..fbde6bf 100644
--- a/configs/kona/mixer_paths_mtp.xml
+++ b/configs/kona/mixer_paths_mtp.xml
@@ -2261,7 +2261,6 @@
<path name="ADSP testfwk">
<ctl name="WSA_CDC_DMA_RX_0_DL_HL Switch" value="1" />
</path>
- </path>
<path name="bt-a2dp">
<ctl name="SLIM7_RX ADM Channels" value="Two" />
diff --git a/configs/msmnile/sound_trigger_mixer_paths_wcd9340.xml b/configs/msmnile/sound_trigger_mixer_paths_wcd9340.xml
index f5de4b5..276652c 100644
--- a/configs/msmnile/sound_trigger_mixer_paths_wcd9340.xml
+++ b/configs/msmnile/sound_trigger_mixer_paths_wcd9340.xml
@@ -371,6 +371,13 @@
<ctl name="EC Reference SampleRate" value="48000"/>
</path>
+ <path name="echo-reference headset">
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_6_RX"/>
+ <ctl name="EC Reference Channels" value="One"/>
+ <ctl name="EC Reference Bit Format" value="S16_LE"/>
+ <ctl name="EC Reference SampleRate" value="48000"/>
+ </path>
+
<path name="echo-reference a2dp">
<ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_7_RX"/>
<ctl name="EC Reference Channels" value="Two"/>
diff --git a/configs/msmnile/sound_trigger_platform_info.xml b/configs/msmnile/sound_trigger_platform_info.xml
index 9d51d59..e0f48c3 100644
--- a/configs/msmnile/sound_trigger_platform_info.xml
+++ b/configs/msmnile/sound_trigger_platform_info.xml
@@ -26,13 +26,15 @@
<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -->
<sound_trigger_platform_info>
- <param version="0x0103" /> <!-- this must be the first param -->
+ <param version="0x0105" /> <!-- this must be the first param -->
<!--- Version History: -->
<!--- 0x0101: Legacy version. -->
<!--- 0x0102: Includes acdb_ids param with the gcs_usecase tag. This matches -->
<!--- the gcs_usecase with the acdb device that uses it. -->
<!--- 0x0103: app_type and in_channels added to <lsm usecase> and out_channels -->
<!--- added to <adm_config> -->
+<!--- 0x0104: instance id support for both WDSP<CPE> and ADSP lsm usecases -->
+<!--- 0x0105: Select <lsm_usecase> based on capture device -->
<common_config>
<param implementer_version="0x0100" />
@@ -83,14 +85,6 @@
<param sample_rate="16000" />
<param bit_width="16" />
<param out_channels="1"/> <!-- Module output channels -->
-
- <!-- adm_cfg_profile should match with the one defined under adm_config -->
- <!-- Set it to NONE if LSM directly connects to AFE -->
- <param adm_cfg_profile="FFECNS" />
- <!-- fluence_type: "FLUENCE", "FLUENCE_DMIC", "FLUENCE_TMIC" -->
- <!-- "FLUENCE_QMIC". Param value is valid when adm_cfg_profile -->
- <!-- is one of FLUENCE, FLUENCE_STEREO, FFECNS values -->
- <param fluence_type="FLUENCE_TMIC" />
<!-- wdsp_fluence_type: fluence disabled: "NONE" -->
<!-- fluence enabled: "FLUENCE_DMIC", "FLUENCE_TMIC", "FLUENCE_QMIC" -->
<param wdsp_fluence_type="NONE" />
@@ -161,16 +155,39 @@
Extends flexibility to have different ids based on execution type.
valid execution_type values: "WDSP" "ADSP" -->
<lsm_usecase>
+ <param capture_device="HANDSET" />
+ <!-- adm_cfg_profile should match with the one defined under adm_config -->
+ <!-- Set it to NONE if LSM directly connects to AFE -->
+ <param adm_cfg_profile="FFECNS" />
+ <!-- fluence_type: "FLUENCE_MIC", "FLUENCE_DMIC", "FLUENCE_TMIC" -->
+ <!-- "FLUENCE_QMIC". Param value is valid when adm_cfg_profile -->
+ <!-- is FFECNS -->
+ <param fluence_type="FLUENCE_TMIC" />
<param execution_mode="ADSP" />
<param app_type="2" /> <!-- app type used in ACDB -->
<param in_channels="5"/> <!-- Module input channels -->
- <param load_sound_model_ids="0x00012C1C, 0x00012C14" />
- <param unload_sound_model_ids="0x00012C1C, 0x00012C15" />
- <param confidence_levels_ids="0x00012C1C, 0x00012C07" />
- <param operation_mode_ids="0x00012C1C, 0x00012C02" />
- <param polling_enable_ids="0x00012C1C, 0x00012C1B" />
- <param custom_config_ids="0x00012C1C, 0x00012C20" />
- <param det_event_type_ids="0x00012C1C, 0x00012C2C" />
+ <param load_sound_model_ids="0x00012C1C, 0x0, 0x00012C14" />
+ <param unload_sound_model_ids="0x00012C1C, 0x0, 0x00012C15" />
+ <param confidence_levels_ids="0x00012C1C, 0x0, 0x00012C07" />
+ <param operation_mode_ids="0x00012C1C, 0x0, 0x00012C02" />
+ <param polling_enable_ids="0x00012C1C, 0x0, 0x00012C1B" />
+ <param custom_config_ids="0x00012C1C, 0x0, 0x00012C20" />
+ <param det_event_type_ids="0x00012C1C, 0x0, 0x00012C2C" />
+ </lsm_usecase>
+ <lsm_usecase>
+ <param capture_device="HEADSET" />
+ <param adm_cfg_profile="FFECNS" />
+ <param fluence_type="FLUENCE_MIC" />
+ <param execution_mode="ADSP" />
+ <param app_type="2" /> <!-- app type used in ACDB -->
+ <param in_channels="1"/> <!-- Module input channels -->
+ <param load_sound_model_ids="0x00012C1C, 0x0, 0x00012C14" />
+ <param unload_sound_model_ids="0x00012C1C, 0x0, 0x00012C15" />
+ <param confidence_levels_ids="0x00012C1C, 0x0, 0x00012C07" />
+ <param operation_mode_ids="0x00012C1C, 0x0, 0x00012C02" />
+ <param polling_enable_ids="0x00012C1C, 0x0, 0x00012C1B" />
+ <param custom_config_ids="0x00012C1C, 0x0, 0x00012C20" />
+ <param det_event_type_ids="0x00012C1C, 0x0, 0x00012C2C" />
</lsm_usecase>
<!-- format: "ADPCM_packet" or "PCM_packet" !-->
@@ -194,13 +211,6 @@
<param sample_rate="16000" />
<param bit_width="16" />
<param out_channels="1"/> <!-- Module output channels -->
- <!-- adm_cfg_profile should match with the one defined under adm_config -->
- <!-- Set it to NONE if LSM directly connects to AFE -->
- <param adm_cfg_profile="NONE" />
- <!-- fluence_type: "FLUENCE", "FLUENCE_DMIC", "FLUENCE_TMIC" -->
- <!-- "FLUENCE_QMIC". Param value is valid when adm_cfg_profile -->
- <!-- is one of FLUENCE, FLUENCE_STEREO, FFECNS values -->
- <param fluence_type="FLUENCE_DMIC" />
<!-- wdsp_fluence_type: fluence disabled: "NONE" -->
<!-- fluence enabled: "FLUENCE_DMIC", "FLUENCE_TMIC", "FLUENCE_QMIC" -->
<param wdsp_fluence_type="NONE" />
@@ -221,14 +231,35 @@
Extends flexibility to have different ids based on execution type.
valid execution_type values: only "ADSP" -->
<lsm_usecase>
+ <param capture_device="HANDSET" />
+ <!-- adm_cfg_profile should match with the one defined under adm_config -->
+ <!-- Set it to NONE if LSM directly connects to AFE -->
+ <param adm_cfg_profile="NONE" />
+ <!-- fluence_type: "FLUENCE_MIC", "FLUENCE_DMIC", "FLUENCE_TMIC" -->
+ <!-- "FLUENCE_QMIC". Param value is valid when adm_cfg_profile -->
+ <!-- is FFECNS -->
+ <param fluence_type="NONE" />
<param execution_mode="ADSP" />
<param app_type="4" /> <!-- app type for MD used in ACDB -->
<param in_channels="1"/> <!-- Module input channels -->
- <param load_sound_model_ids="0x00012C22, 0x00012C14" />
- <param unload_sound_model_ids="0x00012C22, 0x00012C15" />
- <param confidence_levels_ids="0x00012C22, 0x00012C07" />
- <param det_event_type_ids="0x00012C22, 0x00012C2C" />
- <param custom_config_ids="0x00012C22, 0x00012C30" />
+ <param load_sound_model_ids="0x00012C22, 0x0, 0x00012C14" />
+ <param unload_sound_model_ids="0x00012C22, 0x0, 0x00012C15" />
+ <param confidence_levels_ids="0x00012C22, 0x0, 0x00012C07" />
+ <param det_event_type_ids="0x00012C22, 0x0, 0x00012C2C" />
+ <param custom_config_ids="0x00012C22, 0x0, 0x00012C30" />
+ </lsm_usecase>
+ <lsm_usecase>
+ <param capture_device="HEADSET" />
+ <param adm_cfg_profile="NONE" />
+ <param fluence_type="NONE" />
+ <param execution_mode="ADSP" />
+ <param app_type="4" /> <!-- app type for MD used in ACDB -->
+ <param in_channels="1"/> <!-- Module input channels -->
+ <param load_sound_model_ids="0x00012C22, 0x0, 0x00012C14" />
+ <param unload_sound_model_ids="0x00012C22, 0x0, 0x00012C15" />
+ <param confidence_levels_ids="0x00012C22, 0x0, 0x00012C07" />
+ <param det_event_type_ids="0x00012C22, 0x0, 0x00012C2C" />
+ <param custom_config_ids="0x00012C22, 0x0, 0x00012C30" />
</lsm_usecase>
<!-- format: "ADPCM_packet" or "PCM_packet" !-->
@@ -252,13 +283,6 @@
<param sample_rate="16000" />
<param bit_width="16" />
<param out_channels="1"/> <!-- Module output channels -->
- <!-- adm_cfg_profile should match with the one defined under adm_config -->
- <!-- Set it to NONE if LSM directly connects to AFE -->
- <param adm_cfg_profile="NONE" />
- <!-- fluence_type: "FLUENCE", "FLUENCE_DMIC", "FLUENCE_TMIC" -->
- <!-- "FLUENCE_QMIC". Param value is valid when adm_cfg_profile -->
- <!-- is one of FLUENCE, FLUENCE_STEREO, FFECNS values -->
- <param fluence_type="FLUENCE_DMIC" />
<!-- wdsp_fluence_type: fluence disabled: "NONE" -->
<!-- fluence enabled: "FLUENCE_DMIC", "FLUENCE_QMIC" -->
<param wdsp_fluence_type="NONE" />
@@ -286,15 +310,37 @@
</gcs_usecase>
<lsm_usecase>
+ <param capture_device="HANDSET" />
+ <!-- adm_cfg_profile should match with the one defined under adm_config -->
+ <!-- Set it to NONE if LSM directly connects to AFE -->
+ <param adm_cfg_profile="NONE" />
+ <!-- fluence_type: "FLUENCE_MIC", "FLUENCE_DMIC", "FLUENCE_TMIC" -->
+ <!-- "FLUENCE_QMIC". Param value is valid when adm_cfg_profile -->
+ <!-- is FFECNS -->
+ <param fluence_type="NONE" />
<param execution_mode="ADSP" />
<param app_type="3" /> <!-- app type used in ACDB -->
<param in_channels="1"/> <!-- Module input channels -->
- <param load_sound_model_ids="0x18000001, 0x00012C14" />
- <param unload_sound_model_ids="0x18000001, 0x00012C15" />
- <param confidence_levels_ids="0x18000001, 0x00012C07" />
- <param operation_mode_ids="0x18000001, 0x00012C02" />
- <param polling_enable_ids="0x18000001, 0x00012C1B" />
- <param custom_config_ids="0x18000001, 0x00012C20" />
+ <param load_sound_model_ids="0x18000001, 0x0, 0x00012C14" />
+ <param unload_sound_model_ids="0x18000001, 0x0, 0x00012C15" />
+ <param confidence_levels_ids="0x18000001, 0x0, 0x00012C07" />
+ <param operation_mode_ids="0x18000001, 0x0, 0x00012C02" />
+ <param polling_enable_ids="0x18000001, 0x0, 0x00012C1B" />
+ <param custom_config_ids="0x18000001, 0x0, 0x00012C20" />
+ </lsm_usecase>
+ <lsm_usecase>
+ <param capture_device="HEADSET" />
+ <param adm_cfg_profile="NONE" />
+ <param fluence_type="NONE" />
+ <param execution_mode="ADSP" />
+ <param app_type="3" /> <!-- app type used in ACDB -->
+ <param in_channels="1"/> <!-- Module input channels -->
+ <param load_sound_model_ids="0x18000001, 0x0, 0x00012C14" />
+ <param unload_sound_model_ids="0x18000001, 0x0, 0x00012C15" />
+ <param confidence_levels_ids="0x18000001, 0x0, 0x00012C07" />
+ <param operation_mode_ids="0x18000001, 0x0, 0x00012C02" />
+ <param polling_enable_ids="0x18000001, 0x0, 0x00012C1B" />
+ <param custom_config_ids="0x18000001, 0x0, 0x00012C20" />
</lsm_usecase>
<!-- format: "ADPCM_packet" or "PCM_packet" !-->
@@ -332,27 +378,6 @@
<!-- Multiple adm_config tags can be listed, each with unique profile name. -->
<!-- app_type to match corresponding value from ACDB -->
<adm_config>
- <param adm_cfg_profile="DEFAULT" />
- <param app_type="69938" />
- <param sample_rate="16000" />
- <param bit_width="16" />
- </adm_config>
-
- <adm_config>
- <param adm_cfg_profile="FLUENCE" />
- <param app_type="69944" />
- <param sample_rate="16000" />
- <param bit_width="16" />
- </adm_config>
-
- <adm_config>
- <param adm_cfg_profile="FLUENCE_STEREO" />
- <param app_type="69948" />
- <param sample_rate="16000" />
- <param bit_width="16" />
- </adm_config>
-
- <adm_config>
<param adm_cfg_profile="FFECNS" />
<param app_type="69947" />
<param sample_rate="16000" />
diff --git a/configs/msmsteppe/mixer_paths_qrd.xml b/configs/msmsteppe/mixer_paths_qrd.xml
index 1186f61..fc0f760 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 -->
@@ -297,6 +297,7 @@
<ctl name="HPHL_RDAC Switch" value="0" />
<ctl name="HPHR_RDAC Switch" value="0" />
<ctl name="RDAC3_MUX" value="ZERO" />
+ <ctl name="RX_EAR Mode" value="OFF" />
<!-- TX Controls -->
<ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
@@ -2125,6 +2126,7 @@
<ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="EAR_RDAC Switch" value="1" />
<ctl name="RDAC3_MUX" value="RX1" />
+ <ctl name="RX_EAR Mode" value="ON" />
</path>
<path name="handset-mic">
@@ -2139,6 +2141,8 @@
<ctl name="RX INT1_1 MIX1 INP0" value="RX1" />
<ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
+ <ctl name="RX_COMP1 Switch" value="1" />
+ <ctl name="RX_COMP2 Switch" value="1" />
<ctl name="HPHL_COMP Switch" value="1" />
<ctl name="HPHR_COMP Switch" value="1" />
<ctl name="HPHL_RDAC Switch" value="1" />
diff --git a/configs/msmsteppe/mixer_paths_wcd9375qrd.xml b/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
index 1659977..8f5e15d 100644
--- a/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
+++ b/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
@@ -303,6 +303,7 @@
<ctl name="HPHL_RDAC Switch" value="0" />
<ctl name="HPHR_RDAC Switch" value="0" />
<ctl name="RDAC3_MUX" value="ZERO" />
+ <ctl name="RX_EAR Mode" value="OFF" />
<!-- TX Controls -->
<ctl name="TX_CDC_DMA_TX_3 Channels" value="One" />
@@ -2177,13 +2178,13 @@
</path>
<path name="handset">
- <ctl name="RX HPH Mode" value="CLS_AB" />
<ctl name="RX_MACRO RX0 MUX" value="AIF1_PB" />
<ctl name="RX_CDC_DMA_RX_0 Channels" value="One" />
<ctl name="RX INT0_1 MIX1 INP0" value="RX0" />
<ctl name="RX INT0 DEM MUX" value="CLSH_DSM_OUT" />
<ctl name="EAR_RDAC Switch" value="1" />
<ctl name="RDAC3_MUX" value="RX1" />
+ <ctl name="RX_EAR Mode" value="ON" />
</path>
<path name="handset-mic">
diff --git a/configs/sdm710/sdm710.mk b/configs/sdm710/sdm710.mk
index 6710eef..cda0360 100644
--- a/configs/sdm710/sdm710.mk
+++ b/configs/sdm710/sdm710.mk
@@ -263,6 +263,10 @@
PRODUCT_PROPERTY_OVERRIDES += \
ro.af.client_heap_size_kbyte=7168
+#keep alive is needed by default for ffv
+PRODUCT_PROPERTY_OVERRIDES += \
+vendor.audio.keep_alive.disabled=false
+
# for HIDL related packages
PRODUCT_PACKAGES += \
android.hardware.audio@2.0-service \
diff --git a/configs/sdm710/sound_trigger_mixer_paths_wcd9340.xml b/configs/sdm710/sound_trigger_mixer_paths_wcd9340.xml
index 4d1bb31..df61226 100644
--- a/configs/sdm710/sound_trigger_mixer_paths_wcd9340.xml
+++ b/configs/sdm710/sound_trigger_mixer_paths_wcd9340.xml
@@ -342,15 +342,19 @@
<ctl name="CDC_IF TX5 MUX" value="DEC5" />
<ctl name="ADC MUX5" value="DMIC" />
<ctl name="DMIC MUX5" value="DMIC1" />
+ <ctl name="DEC5 Volume" value="96" />
<ctl name="CDC_IF TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
<ctl name="DMIC MUX6" value="DMIC5" />
+ <ctl name="DEC6 Volume" value="96" />
<ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
<ctl name="DMIC MUX7" value="DMIC2" />
+ <ctl name="DEC7 Volume" value="96" />
<ctl name="CDC_IF TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
<ctl name="DMIC MUX8" value="DMIC0" />
+ <ctl name="DEC8 Volume" value="96" />
</path>
<path name="audio-capture">
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index 50bebf7..a233bad 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -81,6 +81,7 @@
AUDIO_EVENT_CAPTURE_STREAM_INACTIVE,
AUDIO_EVENT_CAPTURE_STREAM_ACTIVE,
AUDIO_EVENT_BATTERY_STATUS_CHANGED,
+ AUDIO_EVENT_GET_PARAM
} audio_event_type_t;
typedef enum {
@@ -123,6 +124,13 @@
int device;
};
+struct sound_trigger_get_param_data {
+ char *param;
+ int sm_handle;
+ struct str_parms *reply;
+};
+
+
struct audio_event_info {
union {
ssr_event_status_t status;
@@ -131,6 +139,7 @@
struct audio_read_samples_info aud_info;
char str_value[ST_EVENT_CONFIG_MAX_STR_VALUE];
struct audio_hal_usecase usecase;
+ struct sound_trigger_get_param_data st_get_param_data;
} u;
struct sound_trigger_device_info device_info;
};
@@ -166,6 +175,11 @@
#define SOUND_TRIGGER_LIBRARY_PATH "/vendor/lib/hw/sound_trigger.primary.%s.so"
#endif
+#define SVA_PARAM_DIRECTION_OF_ARRIVAL "st_direction_of_arrival"
+#define SVA_PARAM_CHANNEL_INDEX "st_channel_index"
+
+#define MAX_STR_LENGTH_FFV_PARAMS 30
+#define MAX_FFV_SESSION_ID 100
/*
* Current proprietary API version used by AHAL. Queried by STHAL
* for compatibility check with AHAL
@@ -585,12 +599,46 @@
}
}
+static int extract_sm_handle(const char *keys, char *paramstr) {
+ char *tmpstr, *token;
+ char *inputstr = NULL;
+ int value = -EINVAL;
+
+ if (keys == NULL || paramstr == NULL)
+ goto exit;
+
+ inputstr = strdup(keys);
+ token =strtok_r(inputstr,":",&tmpstr);
+
+ if (token == NULL)
+ goto exit;
+
+ ALOGD("%s input string <%s> param string <%s>", __func__, keys,token);
+ strlcpy(paramstr, token, MAX_STR_LENGTH_FFV_PARAMS);
+ token =strtok_r(NULL,":=",&tmpstr);
+
+ if (token == NULL)
+ goto exit;
+
+ value = atoi(token);
+ if (value > 0 && value < MAX_FFV_SESSION_ID)
+ ALOGD(" %s SVA sm handle<=%d>",__func__, value);
+
+exit:
+ if (inputstr != NULL)
+ free(inputstr);
+
+ return value;
+}
void audio_extn_sound_trigger_get_parameters(const struct audio_device *adev __unused,
- struct str_parms *query, struct str_parms *reply)
+ struct str_parms *query,
+ struct str_parms *reply)
{
audio_event_info_t event;
int ret;
- char value[32];
+ char value[32], paramstr[MAX_STR_LENGTH_FFV_PARAMS];
+
+ ALOGD("%s input string<%s>", __func__, str_parms_to_str(query));
ret = str_parms_get_str(query, "SVA_EXEC_MODE_STATUS", value,
sizeof(value));
@@ -598,6 +646,23 @@
st_dev->st_callback(AUDIO_EVENT_SVA_EXEC_MODE_STATUS, &event);
str_parms_add_int(reply, "SVA_EXEC_MODE_STATUS", event.u.value);
}
+
+ ret = extract_sm_handle(str_parms_to_str(query), paramstr);
+
+ if ((ret >= 0) && !strncmp(paramstr, SVA_PARAM_DIRECTION_OF_ARRIVAL,
+ MAX_STR_LENGTH_FFV_PARAMS)) {
+ event.u.st_get_param_data.sm_handle = ret;
+ event.u.st_get_param_data.param = SVA_PARAM_DIRECTION_OF_ARRIVAL;
+ event.u.st_get_param_data.reply = reply;
+ st_dev->st_callback(AUDIO_EVENT_GET_PARAM, &event);
+ } else if ((ret >=0) && !strncmp(paramstr, SVA_PARAM_CHANNEL_INDEX,
+ MAX_STR_LENGTH_FFV_PARAMS)) {
+ event.u.st_get_param_data.sm_handle = ret;
+ event.u.st_get_param_data.param = SVA_PARAM_CHANNEL_INDEX;
+ event.u.st_get_param_data.reply = reply;
+ st_dev->st_callback(AUDIO_EVENT_GET_PARAM, &event);
+ }
+
}
int audio_extn_sound_trigger_init(struct audio_device *adev)
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 99c291c..7d6db15 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -6928,9 +6928,12 @@
static char* adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
+ ALOGD("%s:%s", __func__, keys);
+
struct audio_device *adev = (struct audio_device *)dev;
struct str_parms *reply = str_parms_create();
struct str_parms *query = str_parms_create_str(keys);
+
char *str;
char value[256] = {0};
int ret = 0;
@@ -6982,7 +6985,7 @@
str_parms_destroy(query);
str_parms_destroy(reply);
- ALOGV("%s: exit: returns - %s", __func__, str);
+ ALOGD("%s: exit: returns - %s", __func__, str);
return str;
}