audio: Add get param support and resolve compile errors
- Add get param support in AHAL for Direction of Arrival and Channel
index from STHAL.
- Correct feature flag enable syntax and resolve compile errors.
- Remove codec loopback and add afe loopback mixer controls in mixer xml.
- Update sound_trigger_platform xml file.
CRs-Fixed: 2225936
Change-Id: I2e477013977a03a599d6bb4a52c66b897e967219
diff --git a/configs/msm8953/audio_platform_info_extcodec.xml b/configs/msm8953/audio_platform_info_extcodec.xml
index 81d34c1..a268b65 100644
--- a/configs/msm8953/audio_platform_info_extcodec.xml
+++ b/configs/msm8953/audio_platform_info_extcodec.xml
@@ -40,11 +40,6 @@
<device name="AUDIO_DEVICE_IN_BUILTIN_MIC" interface="TERT_MI2S" codec_type="internal"/>
<device name="AUDIO_DEVICE_IN_BACK_MIC" interface="TERT_MI2S" codec_type="internal"/>
</interface_names>
- <config_params>
- <param key="input_mic_max_count" value="6"/>
- <param key="ffv_split_ec_ref_data" value="false"/>
- <param key="ffv_ec_ref_channel_count" value="1"/>
- </config_params>
<pcm_ids>
<usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD2" type="out" id="24"/>
<usecase name="USECASE_AUDIO_PLAYBACK_OFFLOAD3" type="out" id="29"/>
@@ -61,14 +56,16 @@
<usecase name="USECASE_AUDIO_SPKR_CALIB_TX" type="in" id="37"/>
<usecase name="USECASE_QCHAT_CALL" type="in" id="42"/>
<usecase name="USECASE_QCHAT_CALL" type="out" id="42"/>
- <usecase name="USECASE_AUDIO_EC_REF_LOOPBACK" type="in" id="12"/>
- <usecase name="USECASE_AUDIO_PLAYBACK_SILENCE" type="out" id="12"/>
+ <usecase name="USECASE_AUDIO_EC_REF_LOOPBACK" type="in" id="14"/>
+ <usecase name="USECASE_AUDIO_PLAYBACK_SILENCE" type="out" id="14"/>
</pcm_ids>
<config_params>
<param key="spkr_1_tz_name" value="wsatz.11"/>
<param key="spkr_2_tz_name" value="wsatz.12"/>
<param key="native_audio_mode" value="src"/>
- <param key="input_mic_max_count" value="4"/>
+ <param key="input_mic_max_count" value="6"/>
+ <param key="ffv_split_ec_ref_data" value="false"/>
+ <param key="ffv_ec_ref_channel_count" value="1"/>
</config_params>
<backend_names>
<device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="SLIMBUS_6_RX"/>
diff --git a/configs/msm8953/mixer_paths_wcd9335.xml b/configs/msm8953/mixer_paths_wcd9335.xml
index 5b53ff3..db68041 100644
--- a/configs/msm8953/mixer_paths_wcd9335.xml
+++ b/configs/msm8953/mixer_paths_wcd9335.xml
@@ -563,14 +563,17 @@
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
<ctl name="SLIMBUS6_DL_HL Switch" value="0" />
<!-- ADSP testfwk end-->
- <ctl name="MultiMedia5 Mixer SLIM_2_TX" value="0" />
- <ctl name="PCM_Dev 12 Topology Capture" value="DTS" />
<!-- These are audio route (FE to BE) specific mixer settings -->
<path name="deep-buffer-playback">
<ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia1" value="1" />
</path>
+ <path name="silence-playback">
+ <ctl name="SLIMBUS_0_RX Audio Mixer MultiMedia9" value="1" />
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
+ </path>
+
<path name="deep-buffer-playback speaker-protected">
<path name="deep-buffer-playback" />
</path>
@@ -2485,10 +2488,9 @@
<path name="speaker-and-headphones" />
</path>
-
<path name="ec-ref-audio-capture">
- <ctl name="PCM_Dev 12 Topology Capture" value="LEGACY" />
- <ctl name="MultiMedia5 Mixer SLIM_2_TX" value="1" />
+ <ctl name="MultiMedia9 Mixer AFE_LOOPBACK_TX" value="1" />
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
</path>
<path name="handset-6mic">
@@ -2567,37 +2569,15 @@
</path>
<path name="ec-ref-loopback-mono">
- <ctl name="SLIM_2_TX Channels" value="One" />
- <ctl name="AIF4_CAP Mixer SLIM TX7" value ="1"/>
- <ctl name="SLIM TX7 MUX" value="RX_MIX_TX7"/>
- <ctl name="RX MIX TX7 MUX" value="RX_MIX7"/>
</path>
<path name="ec-ref-loopback-stereo">
- <ctl name="SLIM_2_TX Channels" value="Two" />
- <ctl name="AIF4_CAP Mixer SLIM TX7" value ="1"/>
- <ctl name="SLIM TX7 MUX" value="RX_MIX_TX7"/>
- <ctl name="RX MIX TX7 MUX" value="RX_MIX7"/>
- <ctl name="AIF4_CAP Mixer SLIM TX8" value ="1"/>
- <ctl name="SLIM TX8 MUX" value="RX_MIX_TX8"/>
- <ctl name="RX MIX TX8 MUX" value="RX_MIX8"/>
</path>
<path name="ec-ref-loopback-mono lineout">
- <ctl name="SLIM_2_TX Channels" value="One" />
- <ctl name="AIF4_CAP Mixer SLIM TX7" value ="1"/>
- <ctl name="SLIM TX7 MUX" value="RX_MIX_TX7"/>
- <ctl name="RX MIX TX7 MUX" value="RX_MIX5"/>
</path>
<path name="ec-ref-loopback-stereo lineout">
- <ctl name="SLIM_2_TX Channels" value="Two" />
- <ctl name="AIF4_CAP Mixer SLIM TX7" value ="1"/>
- <ctl name="SLIM TX7 MUX" value="RX_MIX_TX7"/>
- <ctl name="RX MIX TX7 MUX" value="RX_MIX5"/>
- <ctl name="AIF4_CAP Mixer SLIM TX8" value ="1"/>
- <ctl name="SLIM TX8 MUX" value="RX_MIX_TX8"/>
- <ctl name="RX MIX TX8 MUX" value="RX_MIX6"/>
</path>
</mixer>
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index 1e51212..c9a9746 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -34,10 +34,10 @@
USE_XML_AUDIO_POLICY_CONF := 1
BOARD_SUPPORTS_SOUND_TRIGGER := true
-BOARD_SUPPORTS_SOUND_TRIGGER_ARM: = true
-AUDIO_FEATURE_ENABLED_FFV: = true
-AUDIO_FEATURE_ENABLED_KEEP_ALIVE_ARM_FFV: = true
-AUDIO_FEATURE_ENABLED_KEEP_ALIVE: = true
+BOARD_SUPPORTS_SOUND_TRIGGER_ARM := true
+AUDIO_FEATURE_ENABLED_FFV := true
+AUDIO_FEATURE_ENABLED_KEEP_ALIVE_ARM_FFV := true
+AUDIO_FEATURE_ENABLED_KEEP_ALIVE := true
AUDIO_USE_LL_AS_PRIMARY_OUTPUT := true
AUDIO_FEATURE_ENABLED_HIFI_AUDIO := true
AUDIO_FEATURE_ENABLED_VBAT_MONITOR := true
diff --git a/configs/msm8953/sound_trigger_mixer_paths_wcd9335.xml b/configs/msm8953/sound_trigger_mixer_paths_wcd9335.xml
index 1756772..8121c0d 100644
--- a/configs/msm8953/sound_trigger_mixer_paths_wcd9335.xml
+++ b/configs/msm8953/sound_trigger_mixer_paths_wcd9335.xml
@@ -96,9 +96,11 @@
<ctl name="AIF1_CAP Mixer SLIM TX8" value="0" />
<ctl name="AIF4_CAP Mixer SLIM TX8" value="0" />
<ctl name="AIF4_CAP Mixer SLIM TX7" value="0" />
+ <ctl name="MultiMedia9 Mixer AFE_LOOPBACK_TX" value="0" />
<ctl name="MultiMedia2 Mixer SLIM_0_TX" value="0" />
<ctl name="MultiMedia9 Mixer SLIM_2_TX" value="0" />
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="None"/>
<path name="listen-voice-wakeup-1">
<ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
@@ -195,7 +197,8 @@
</path>
<path name="ec-ref-audio-capture">
- <ctl name="MultiMedia9 Mixer SLIM_2_TX" value="1" />
+ <ctl name="MultiMedia9 Mixer AFE_LOOPBACK_TX" value="1" />
+ <ctl name="AUDIO_REF_EC_UL1 MUX" value="SLIM_RX" />
</path>
<path name="listen-cpe-handset-mic">
@@ -361,37 +364,15 @@
</path>
<path name="ec-ref-loopback-mono">
- <ctl name="SLIM_2_TX Channels" value="One" />
- <ctl name="AIF4_CAP Mixer SLIM TX7" value ="1"/>
- <ctl name="SLIM TX7 MUX" value="RX_MIX_TX7"/>
- <ctl name="RX MIX TX7 MUX" value="RX_MIX7"/>
</path>
<path name="ec-ref-loopback-stereo">
- <ctl name="SLIM_2_TX Channels" value="Two" />
- <ctl name="AIF4_CAP Mixer SLIM TX7" value ="1"/>
- <ctl name="SLIM TX7 MUX" value="RX_MIX_TX7"/>
- <ctl name="RX MIX TX7 MUX" value="RX_MIX7"/>
- <ctl name="AIF4_CAP Mixer SLIM TX8" value ="1"/>
- <ctl name="SLIM TX8 MUX" value="RX_MIX_TX8"/>
- <ctl name="RX MIX TX8 MUX" value="RX_MIX8"/>
</path>
<path name="ec-ref-loopback-mono lineout">
- <ctl name="SLIM_2_TX Channels" value="One" />
- <ctl name="AIF4_CAP Mixer SLIM TX7" value ="1"/>
- <ctl name="SLIM TX7 MUX" value="RX_MIX_TX7"/>
- <ctl name="RX MIX TX7 MUX" value="RX_MIX5"/>
</path>
<path name="ec-ref-loopback-stereo lineout">
- <ctl name="SLIM_2_TX Channels" value="Two" />
- <ctl name="AIF4_CAP Mixer SLIM TX7" value ="1"/>
- <ctl name="SLIM TX7 MUX" value="RX_MIX_TX7"/>
- <ctl name="RX MIX TX7 MUX" value="RX_MIX5"/>
- <ctl name="AIF4_CAP Mixer SLIM TX8" value ="1"/>
- <ctl name="SLIM TX8 MUX" value="RX_MIX_TX8"/>
- <ctl name="RX MIX TX8 MUX" value="RX_MIX6"/>
</path>
<path name="echo-reference">
diff --git a/configs/msm8953/sound_trigger_platform_info.xml b/configs/msm8953/sound_trigger_platform_info.xml
index f5280e4..a9e1387 100644
--- a/configs/msm8953/sound_trigger_platform_info.xml
+++ b/configs/msm8953/sound_trigger_platform_info.xml
@@ -41,6 +41,14 @@
<param backend_dai_name="TERT_MI2S_TX" /-->
<param backend_port_name="SLIM_0_TX" />
<param backend_dai_name="SLIMBUS_0_TX" />
+ <param sw_mad="false"/>
+ <!-- Enable concurrent VA & audio capture excluding voip/voice call -->
+ <!-- using concurrent_capture param. -->
+ <!-- Enable VA & voip/voice call concurrency using concurrent_capture -->
+ <!-- param along with concurrent_voip_call/concurrent_voice_call params -->
+ <param concurrent_capture="false" />
+ <param concurrent_voip_call="false" />
+ <param concurrent_voice_call="false" />
</common_config>
<acdb_ids>
@@ -69,15 +77,7 @@
<param max_cpe_users="3" />
<param max_ape_phrases="10" />
<param max_ape_users="10" />
- <param sample_rate="16000" />
- <param bit_width="16" />
- <param channel_count="1"/>
- <!-- 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="FLUENCE"-->
- <param fluence_type="NONE" />
+ <param event_timestamp_mode="false" />
<!-- Module and param ids with which the algorithm is integrated in firmware -->
<lsm_usecase>
@@ -93,6 +93,17 @@
<!-- kw_duration is in milli seconds. It is valid only for FTRT transfer mode -->
<param capture_keyword="PCM_packet, RT, 2000" />
<param client_capture_read_delay="2000" />
+
+ <!-- Profile specific data which the algorithm can support -->
+ <param sample_rate="16000" />
+ <param bit_width="16" />
+ <param channel_count="1"/>
+ <!-- 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_QMIC" -->
+ <!-- param value is valid when profile type is fluence -->
+ <param fluence_type="FLUENCE_QMIC" />
</sound_model_config>
<!-- ARM based SVA sound_model_config -->
diff --git a/hal/Android.mk b/hal/Android.mk
index 921252c..36bcca9 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -395,12 +395,13 @@
LOCAL_CFLAGS += -DDYNAMIC_ECNS_ENABLED
endif
-ifeq ($(strip $($AUDIO_FEATURE_ENABLED_KEEP_ALIVE_ARM_FFV)), true)
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_KEEP_ALIVE_ARM_FFV)), true)
LOCAL_CFLAGS += -DRUN_KEEP_ALIVE_IN_ARM_FFV
endif
-ifeq ($(strip $($AUDIO_FEATURE_ENABLED_FFV_FFV)), true)
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FFV)), true)
LOCAL_CFLAGS += -DFFV_ENABLED
+ LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio-noship/include/ffv
LOCAL_SRC_FILES += audio_extn/ffv.c
endif
diff --git a/hal/audio_extn/ffv.c b/hal/audio_extn/ffv.c
index 54ce30c..82140b1 100755
--- a/hal/audio_extn/ffv.c
+++ b/hal/audio_extn/ffv.c
@@ -81,6 +81,7 @@
#define FFV_CHANNEL_MODE_MONO 1
#define FFV_CHANNEL_MODE_STEREO 2
+#define FFV_CHANNEL_MODE_QUAD 6
#define FFV_CHANNEL_MODE_HEX 6
#define FFV_CHANNEL_MODE_OCT 8
@@ -407,7 +408,7 @@
config->period_size = ffvmod.capture_config.period_size;
}
-int32_t audio_extn_ffv_init(struct audio_device *adev)
+int32_t audio_extn_ffv_init(struct audio_device *adev __unused)
{
int ret = 0;
@@ -487,7 +488,7 @@
__func__, num_ec_ref_ch, num_tx_in_ch, num_out_ch, frame_len, sample_rate);
ALOGD("%s: config file path %s", __func__, config_file_path);
status_type = ffv_init_fn(&ffvmod.handle, num_tx_in_ch, num_out_ch, num_ec_ref_ch,
- frame_len, sample_rate, config_file_path, sm_buffer, 0,
+ frame_len, sample_rate, config_file_path, (char *)sm_buffer, 0,
&total_mem_size, key, lic);
if (status_type) {
ALOGE("%s: ERROR. ffv_init returned %d", __func__, status_type);
@@ -580,6 +581,8 @@
return SND_DEVICE_IN_HANDSET_8MIC;
} else if (ffvmod.capture_config.channels == FFV_CHANNEL_MODE_HEX) {
return SND_DEVICE_IN_HANDSET_6MIC;
+ } else if (ffvmod.capture_config.channels == FFV_CHANNEL_MODE_QUAD) {
+ return SND_DEVICE_IN_HANDSET_QMIC;
} else {
ALOGE("%s: Invalid channels configured for capture", __func__);
return SND_DEVICE_NONE;
@@ -587,7 +590,7 @@
}
int audio_extn_ffv_init_ec_ref_loopback(struct audio_device *adev,
- snd_device_t snd_device)
+ snd_device_t snd_device __unused)
{
struct audio_usecase *uc_info_tx = NULL;
snd_device_t in_snd_device;
@@ -679,7 +682,7 @@
}
int audio_extn_ffv_deinit_ec_ref_loopback(struct audio_device *adev,
- snd_device_t snd_device)
+ snd_device_t snd_device __unused)
{
struct audio_usecase *uc_info_tx = NULL;
snd_device_t in_snd_device;
@@ -709,7 +712,7 @@
return ret;
}
-int32_t audio_extn_ffv_read(struct audio_stream_in *stream,
+int32_t audio_extn_ffv_read(struct audio_stream_in *stream __unused,
void *buffer, size_t bytes)
{
int status = 0;
@@ -717,7 +720,7 @@
int16_t *process_ec_ref_ptr = NULL;
size_t in_buf_size, out_buf_size, bytes_to_copy;
int retry_num = 0;
- int i, j, ch;
+ int i, ch;
int total_in_ch, in_ch, ec_ref_ch;
if (!ffvmod.ffv_lib_handle) {
@@ -805,7 +808,7 @@
total_in_ch = ffvmod.capture_config.channels;
ec_ref_ch = ffvmod.ec_ref_config.channels;
in_ch = total_in_ch - ec_ref_ch;
- for (i = 0; i < ffvmod.capture_config.period_size; i++) {
+ for (i = 0; i < (int)ffvmod.capture_config.period_size; i++) {
for (ch = 0; ch < in_ch; ch++) {
process_in_ptr[i*in_ch+ch] =
in_ptr[i*total_in_ch+ch];
@@ -846,7 +849,6 @@
void audio_extn_ffv_set_parameters(struct audio_device *adev __unused,
struct str_parms *parms)
{
- int err;
int val;
int ret = 0;
char value[128];
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index 041eb53..061616c 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -79,6 +79,7 @@
AUDIO_EVENT_SVA_EXEC_MODE_STATUS,
AUDIO_EVENT_CAPTURE_STREAM_INACTIVE,
AUDIO_EVENT_CAPTURE_STREAM_ACTIVE,
+ AUDIO_EVENT_GET_PARAM
} audio_event_type_t;
typedef enum {
@@ -121,6 +122,12 @@
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;
@@ -129,6 +136,7 @@
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;
};
@@ -165,6 +173,8 @@
#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"
/*
* Current proprietary API version used by AHAL. Queried by STHAL
* for compatibility check with AHAL
@@ -597,7 +607,7 @@
struct str_parms *query, struct str_parms *reply)
{
audio_event_info_t event;
- int ret;
+ int ret, val;
char value[32];
ret = str_parms_get_str(query, "SVA_EXEC_MODE_STATUS", value,
@@ -606,6 +616,22 @@
st_dev->st_callback(AUDIO_EVENT_SVA_EXEC_MODE_STATUS, &event);
str_parms_add_int(reply, "SVA_EXEC_MODE_STATUS", event.u.value);
}
+
+ ret = str_parms_get_int(query, SVA_PARAM_DIRECTION_OF_ARRIVAL, &val);
+ if (ret >= 0) {
+ event.u.st_get_param_data.sm_handle = val;
+ 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);
+ }
+
+ ret = str_parms_get_int(query, SVA_PARAM_CHANNEL_INDEX, &val);
+ if (ret >= 0) {
+ event.u.st_get_param_data.sm_handle = val;
+ 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)