Merge "configs: rename vendor properties"
diff --git a/configs/kona/audio_configs.xml b/configs/kona/audio_configs.xml
index fa07ca2..8c24bb2 100644
--- a/configs/kona/audio_configs.xml
+++ b/configs/kona/audio_configs.xml
@@ -92,6 +92,7 @@
<flag name="incall_music_enabled" value="true" />
<flag name="keep_alive_enabled" value="true" />
<flag name="kpi_optimize_enabled" value="true" />
+ <flag name="maxx_audio_enabled" value="false" />
<flag name="receiver_aided_stereo" value="true" />
<flag name="snd_monitor_enabled" value="true" />
<flag name="source_track_enabled" value="true" />
diff --git a/configs/msm8937/audio_configs.xml b/configs/msm8937/audio_configs.xml
index e221be7..ed3ed8d 100644
--- a/configs/msm8937/audio_configs.xml
+++ b/configs/msm8937/audio_configs.xml
@@ -92,6 +92,7 @@
<flag name="incall_music_enabled" value="false" />
<flag name="keep_alive_enabled" value="false" />
<flag name="kpi_optimize_enabled" value="true" />
+ <flag name="maxx_audio_enabled" value="false" />
<flag name="receiver_aided_stereo" value="false" />
<flag name="snd_monitor_enabled" value="true" />
<flag name="source_track_enabled" value="true" />
diff --git a/configs/msmnile/audio_configs.xml b/configs/msmnile/audio_configs.xml
index fa07ca2..8c24bb2 100644
--- a/configs/msmnile/audio_configs.xml
+++ b/configs/msmnile/audio_configs.xml
@@ -92,6 +92,7 @@
<flag name="incall_music_enabled" value="true" />
<flag name="keep_alive_enabled" value="true" />
<flag name="kpi_optimize_enabled" value="true" />
+ <flag name="maxx_audio_enabled" value="false" />
<flag name="receiver_aided_stereo" value="true" />
<flag name="snd_monitor_enabled" value="true" />
<flag name="source_track_enabled" value="true" />
diff --git a/configs/msmnile/audio_configs_stock.xml b/configs/msmnile/audio_configs_stock.xml
index b9ecf45..6414675 100644
--- a/configs/msmnile/audio_configs_stock.xml
+++ b/configs/msmnile/audio_configs_stock.xml
@@ -57,6 +57,7 @@
<flag name="incall_music_enabled" value="true" />
<flag name="keep_alive_enabled" value="false" />
<flag name="kpi_optimize_enabled" value="false" />
+ <flag name="maxx_audio_enabled" value="true" />
<flag name="receiver_aided_stereo" value="false" />
<flag name="record_play_concurrency" value="false" />
<flag name="snd_monitor_enabled" value="true" />
diff --git a/configs/sdm660/audio_configs.xml b/configs/sdm660/audio_configs.xml
index 853f7e7..3688697 100644
--- a/configs/sdm660/audio_configs.xml
+++ b/configs/sdm660/audio_configs.xml
@@ -92,6 +92,7 @@
<flag name="incall_music_enabled" value="false" />
<flag name="keep_alive_enabled" value="false" />
<flag name="kpi_optimize_enabled" value="true" />
+ <flag name="maxx_audio_enabled" value="false" />
<flag name="receiver_aided_stereo" value="true" />
<flag name="snd_monitor_enabled" value="true" />
<flag name="source_track_enabled" value="true" />
diff --git a/configs/sdm710/audio_configs.xml b/configs/sdm710/audio_configs.xml
index 0410f4b..6a6fb7d 100644
--- a/configs/sdm710/audio_configs.xml
+++ b/configs/sdm710/audio_configs.xml
@@ -92,6 +92,7 @@
<flag name="incall_music_enabled" value="false" />
<flag name="keep_alive_enabled" value="false" />
<flag name="kpi_optimize_enabled" value="true" />
+ <flag name="maxx_audio_enabled" value="false" />
<flag name="receiver_aided_stereo" value="true" />
<flag name="snd_monitor_enabled" value="true" />
<flag name="source_track_enabled" value="true" />
diff --git a/configs/sdm845/audio_configs.xml b/configs/sdm845/audio_configs.xml
index 95b7d97..4b0159d 100644
--- a/configs/sdm845/audio_configs.xml
+++ b/configs/sdm845/audio_configs.xml
@@ -92,6 +92,7 @@
<flag name="incall_music_enabled" value="false" />
<flag name="keep_alive_enabled" value="false" />
<flag name="kpi_optimize_enabled" value="true" />
+ <flag name="maxx_audio_enabled" value="false" />
<flag name="receiver_aided_stereo" value="true" />
<flag name="snd_monitor_enabled" value="true" />
<flag name="source_track_enabled" value="true" />
diff --git a/hal/ahal_config_helper.cpp b/hal/ahal_config_helper.cpp
index 753d8a8..e46b8f3 100644
--- a/hal/ahal_config_helper.cpp
+++ b/hal/ahal_config_helper.cpp
@@ -105,6 +105,7 @@
false, /* CONCURRENT_CAPTURE */
false, /* COMPRESS_IN */
false, /* BATTERY_LISTENER */
+ false, /* MAXX_AUDIO */
true, /* COMPRESS_METADATA_NEEDED */
false, /* INCALL_MUSIC */
false, /* COMPRESS_VOIP */
@@ -149,6 +150,7 @@
true, /* CONCURRENT_CAPTURE */
true, /* COMPRESS_IN */
true, /* BATTERY_LISTENER */
+ false, /* MAXX_AUDIO */
true, /* COMPRESS_METADATA_NEEDED */
true, /* INCALL_MUSIC */
false, /* COMPRESS_VOIP */
@@ -192,6 +194,7 @@
true, /* CONCURRENT_CAPTURE */
false, /* COMPRESS_IN */
false, /* BATTERY_LISTENER */
+ true, /* MAXX_AUDIO */
false, /* COMPRESS_METADATA_NEEDED */
true, /* INCALL_MUSIC */
false, /* COMPRESS_VOIP */
diff --git a/hal/ahal_config_helper.h b/hal/ahal_config_helper.h
index 3251961..39ed68e 100644
--- a/hal/ahal_config_helper.h
+++ b/hal/ahal_config_helper.h
@@ -71,6 +71,7 @@
bool concurrent_capture_enabled;
bool compress_in_enabled;
bool battery_listener_enabled;
+ bool maxx_audio_enabled;
bool compress_metadata_needed;
bool incall_music_enabled;
bool compress_voip_enabled;
diff --git a/hal/audio_extn/Android.mk b/hal/audio_extn/Android.mk
old mode 100755
new mode 100644
index 59b18a2..117ee27
--- a/hal/audio_extn/Android.mk
+++ b/hal/audio_extn/Android.mk
@@ -398,11 +398,13 @@
include $(BUILD_SHARED_LIBRARY)
#-------------------------------------------
+
# Build EXT_HW_PLUGIN LIB
#-------------------------------------------
include $(CLEAR_VARS)
LOCAL_MODULE := libexthwplugin
+
LOCAL_VENDOR_MODULE := true
PRIMARY_HAL_PATH := vendor/qcom/opensource/audio-hal/primary-hal/hal
@@ -705,3 +707,61 @@
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_HEADER_LIBRARIES += libsystem_headers
#include $(BUILD_SHARED_LIBRARY)
+
+#-------------------------------------------
+# Build MAXX_AUDIO
+#-------------------------------------------
+include $(CLEAR_VARS)
+
+LOCAL_MODULE:= libmaxxaudio
+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:= \
+ maxxaudio.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 400d7d0..19c839c 100644
--- a/hal/audio_extn/a2dp.c
+++ b/hal/audio_extn/a2dp.c
@@ -62,6 +62,7 @@
#define MEDIA_FMT_LDAC 0x00013224
#define MEDIA_FMT_MP3 0x00010BE9
#define MEDIA_FMT_APTX_ADAPTIVE 0x00013204
+#define MEDIA_FMT_APTX_AD_SPEECH 0x00013208
#define MEDIA_FMT_AAC_AOT_LC 2
#define MEDIA_FMT_AAC_AOT_SBR 5
#define MEDIA_FMT_AAC_AOT_PS 29
@@ -85,6 +86,7 @@
#define MIXER_SAMPLE_RATE_DEFAULT "BT SampleRate"
#define MIXER_AFE_IN_CHANNELS "AFE Input Channels"
#define MIXER_ABR_TX_FEEDBACK_PATH "A2DP_SLIM7_UL_HL Switch"
+#define MIXER_ABR_RX_FEEDBACK_PATH "SCO_SLIM7_DL_HL Switch"
#define MIXER_SET_FEEDBACK_CHANNEL "BT set feedback channel"
#define MIXER_SINK_SAMPLE_RATE "BT_TX SampleRate"
#define MIXER_AFE_SINK_CHANNELS "AFE Output Channels"
@@ -125,6 +127,9 @@
// Slimbus Tx sample rate for ABR feedback channel
#define ABR_TX_SAMPLE_RATE "KHZ_8"
+// Slimbus Tx sample rate for APTX AD SPEECH
+#define SPEECH_TX_SAMPLE_RATE "KHZ_96"
+
// Purpose ID for Inter Module Communication (IMC) in AFE
#define IMC_PURPOSE_ID_BT_INFO 0x000132E2
@@ -134,8 +139,13 @@
// Instance identifier for A2DP
#define MAX_INSTANCE_ID (UINT32_MAX / 2)
+// Instance identifier for SWB
+#define APTX_AD_SPEECH_INSTANCE_ID 37
+
+#define SAMPLING_RATE_96K 96000
#define SAMPLING_RATE_48K 48000
#define SAMPLING_RATE_441K 44100
+#define SAMPLING_RATE_32K 32000
#define CH_STEREO 2
#define CH_MONO 1
#define SOURCE 0
@@ -172,6 +182,7 @@
CODEC_TYPE_LDAC = AUDIO_FORMAT_LDAC, // 0x23000000UL
CODEC_TYPE_CELT = 603979776u, // 0x24000000UL
CODEC_TYPE_APTX_AD = 620756992u, // 0x25000000UL
+ CODEC_TYPE_APTX_AD_SPEECH = 637534208u, //0x26000000UL
CODEC_TYPE_PCM = AUDIO_FORMAT_PCM_16_BIT, // 0x1u
}codec_t;
@@ -237,6 +248,11 @@
} imc_status_t;
typedef enum {
+ SWAP_DISABLE,
+ SWAP_ENABLE,
+} swap_status_t;
+
+typedef enum {
MTU_SIZE,
PEAK_BIT_RATE,
} frame_control_type_t;
@@ -265,6 +281,8 @@
bool abr_started;
/* ABR Tx path pcm handle */
struct pcm *abr_tx_handle;
+ /* ABR Rx path pcm handle */
+ struct pcm *abr_rx_handle;
/* ABR Inter Module Communication (IMC) instance ID */
uint32_t imc_instance;
};
@@ -319,6 +337,7 @@
uint32_t dec_channels;
bool a2dp_sink_started;
int a2dp_sink_total_active_session_requests;
+ bool swb_configured;
};
struct a2dp_data a2dp;
@@ -400,6 +419,18 @@
struct imc_dec_enc_info imc_info;
} __attribute__ ((packed));
+struct aptx_ad_speech_mode_cfg_t
+{
+ uint32_t mode;
+ uint32_t swapping;
+} __attribute__ ((packed));
+
+/* Structure for SWB voice dec config */
+struct aptx_ad_speech_dec_cfg_t {
+ struct abr_dec_cfg_t abr_cfg;
+ struct aptx_ad_speech_mode_cfg_t speech_mode;
+} __attribute__ ((packed));
+
/* START of DSP configurable structures
* These values should match with DSP interface defintion
*/
@@ -546,6 +577,15 @@
struct abr_enc_cfg_t abr_cfg;
} __attribute__ ((packed));
+/* APTX AD SPEECH structure */
+struct aptx_ad_speech_enc_cfg_t
+{
+ struct custom_enc_cfg_t custom_cfg;
+ /* Information to set up IMC between decoder and encoder */
+ struct imc_dec_enc_info imc_info;
+ struct aptx_ad_speech_mode_cfg_t speech_mode;
+} __attribute__ ((packed));
+
struct ldac_specific_enc_cfg_t
{
uint32_t bit_rate;
@@ -715,7 +755,9 @@
static int stop_abr()
{
struct mixer_ctl *ctl_abr_tx_path = NULL;
+ struct mixer_ctl *ctl_abr_rx_path = NULL;
struct mixer_ctl *ctl_set_bt_feedback_channel = NULL;
+ int ret = 0;
/* This function can be used if !abr_started for clean up */
ALOGV("%s: enter", __func__);
@@ -725,6 +767,10 @@
pcm_close(a2dp.abr_config.abr_tx_handle);
a2dp.abr_config.abr_tx_handle = NULL;
}
+ if (a2dp.abr_config.abr_rx_handle != NULL) {
+ pcm_close(a2dp.abr_config.abr_rx_handle);
+ a2dp.abr_config.abr_rx_handle = NULL;
+ }
a2dp.abr_config.abr_started = false;
a2dp.abr_config.imc_instance = 0;
@@ -733,11 +779,10 @@
MIXER_SET_FEEDBACK_CHANNEL);
if (!ctl_set_bt_feedback_channel) {
ALOGE("%s: ERROR Set usecase mixer control not identifed", __func__);
- return -ENOSYS;
- }
- if (mixer_ctl_set_value(ctl_set_bt_feedback_channel, 0, 0) != 0) {
+ ret = -ENOSYS;
+ } else if (mixer_ctl_set_value(ctl_set_bt_feedback_channel, 0, 0) != 0) {
ALOGE("%s: Failed to set BT usecase", __func__);
- return -ENOSYS;
+ ret = -ENOSYS;
}
// Reset ABR Tx feedback path
@@ -746,19 +791,31 @@
MIXER_ABR_TX_FEEDBACK_PATH);
if (!ctl_abr_tx_path) {
ALOGE("%s: ERROR ABR Tx feedback path mixer control not identifed", __func__);
- return -ENOSYS;
- }
- if (mixer_ctl_set_value(ctl_abr_tx_path, 0, 0) != 0) {
+ ret = -ENOSYS;
+ } else if (mixer_ctl_set_value(ctl_abr_tx_path, 0, 0) != 0) {
ALOGE("%s: Failed to set ABR Tx feedback path", __func__);
- return -ENOSYS;
+ ret = -ENOSYS;
}
- return 0;
+ // Reset ABR Rx feedback path
+ ALOGV("%s: Disable ABR Rx feedback path", __func__);
+ ctl_abr_rx_path = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ABR_RX_FEEDBACK_PATH);
+ if (!ctl_abr_rx_path) {
+ ALOGE("%s: ERROR ABR Rx feedback path mixer control not identifed", __func__);
+ ret = -ENOSYS;
+ } else if (mixer_ctl_set_value(ctl_abr_rx_path, 0, 0) != 0) {
+ ALOGE("%s: Failed to set ABR Rx feedback path", __func__);
+ ret = -ENOSYS;
+ }
+
+ return ret;
}
static int start_abr()
{
struct mixer_ctl *ctl_abr_tx_path = NULL;
+ struct mixer_ctl *ctl_abr_rx_path = NULL;
struct mixer_ctl *ctl_set_bt_feedback_channel = NULL;
int abr_device_id;
int ret = 0;
@@ -792,11 +849,11 @@
MIXER_SET_FEEDBACK_CHANNEL);
if (!ctl_set_bt_feedback_channel) {
ALOGE("%s: ERROR Set usecase mixer control not identifed", __func__);
- return -ENOSYS;
+ goto fail;
}
if (mixer_ctl_set_value(ctl_set_bt_feedback_channel, 0, 1) != 0) {
ALOGE("%s: Failed to set BT usecase", __func__);
- return -ENOSYS;
+ goto fail;
}
// Open hostless front end and prepare ABR Tx path
@@ -806,19 +863,60 @@
a2dp.abr_config.abr_tx_handle = pcm_open(a2dp.adev->snd_card,
abr_device_id, PCM_IN,
&pcm_config_abr);
- if (a2dp.abr_config.abr_tx_handle == NULL ||
- !pcm_is_ready(a2dp.abr_config.abr_tx_handle))
+ if (a2dp.abr_config.abr_tx_handle == NULL) {
+ ALOGE("%s: Can't open abr tx device", __func__);
goto fail;
+ }
+ if (!(pcm_is_ready(a2dp.abr_config.abr_tx_handle) &&
+ !pcm_start(a2dp.abr_config.abr_tx_handle))) {
+ ALOGE("%s: tx: %s", __func__, pcm_get_error(a2dp.abr_config.abr_tx_handle));
+ goto fail;
+ }
}
- ret = pcm_start(a2dp.abr_config.abr_tx_handle);
- if (ret < 0)
- goto fail;
+
+ // Enable Slimbus 7 Rx feedback path for HD Voice use case
+ if (a2dp.bt_encoder_format == CODEC_TYPE_APTX_AD_SPEECH) {
+ ALOGV("%s: Enable ABR Rx feedback path", __func__);
+ ctl_abr_rx_path = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_ABR_RX_FEEDBACK_PATH);
+ if (!ctl_abr_rx_path) {
+ ALOGE("%s: ERROR ABR Rx feedback path mixer control not identifed", __func__);
+ goto fail;
+ }
+ if (mixer_ctl_set_value(ctl_abr_rx_path, 0, 1) != 0) {
+ ALOGE("%s: Failed to set ABR Rx feedback path", __func__);
+ goto fail;
+ }
+
+ if (mixer_ctl_set_value(ctl_set_bt_feedback_channel, 0, 1) != 0) {
+ ALOGE("%s: Failed to set BT usecase", __func__);
+ goto fail;
+ }
+
+ // Open hostless front end and prepare ABR Rx path
+ abr_device_id = fp_platform_get_pcm_device_id(USECASE_AUDIO_A2DP_ABR_FEEDBACK,
+ PCM_PLAYBACK);
+ if (!a2dp.abr_config.abr_rx_handle) {
+ a2dp.abr_config.abr_rx_handle = pcm_open(a2dp.adev->snd_card,
+ abr_device_id, PCM_OUT,
+ &pcm_config_abr);
+ if (a2dp.abr_config.abr_rx_handle == NULL) {
+ ALOGE("%s: Can't open abr rx device", __func__);
+ goto fail;
+ }
+ if (!(pcm_is_ready(a2dp.abr_config.abr_rx_handle) &&
+ !pcm_start(a2dp.abr_config.abr_rx_handle))) {
+ ALOGE("%s: rx: %s", __func__, pcm_get_error(a2dp.abr_config.abr_rx_handle));
+ goto fail;
+ }
+ }
+ }
+
a2dp.abr_config.abr_started = true;
return ret;
fail:
- ALOGE("%s: %s", __func__, pcm_get_error(a2dp.abr_config.abr_tx_handle));
stop_abr();
return -ENOSYS;
}
@@ -965,6 +1063,7 @@
a2dp.abr_config.abr_started = false;
a2dp.abr_config.imc_instance = 0;
a2dp.abr_config.abr_tx_handle = NULL;
+ a2dp.abr_config.abr_rx_handle = NULL;
a2dp.bt_state_source = A2DP_STATE_DISCONNECTED;
return 0;
@@ -1079,9 +1178,12 @@
if (direction == SOURCE) {
/* Set Tx backend sample rate */
- if (a2dp.abr_config.is_abr_enabled)
- rate_str = ABR_TX_SAMPLE_RATE;
-
+ if (a2dp.abr_config.is_abr_enabled) {
+ if (a2dp.bt_encoder_format == CODEC_TYPE_APTX_AD_SPEECH)
+ rate_str = SPEECH_TX_SAMPLE_RATE;
+ else
+ rate_str = ABR_TX_SAMPLE_RATE;
+ }
ALOGD("%s: set backend tx sample rate = %s", __func__, rate_str);
ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
MIXER_SOURCE_SAMPLE_RATE_TX);
@@ -1282,17 +1384,18 @@
ALOGE("%s: Failed to reset backend sample rate = %s", __func__, rate_str);
return -ENOSYS;
}
-
- ctl_sample_rate_tx = mixer_get_ctl_by_name(a2dp.adev->mixer,
- MIXER_SOURCE_SAMPLE_RATE_TX);
- if (!ctl_sample_rate_tx) {
+ if (a2dp.abr_config.is_abr_enabled) {
+ ctl_sample_rate_tx = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_SOURCE_SAMPLE_RATE_TX);
+ if (!ctl_sample_rate_tx) {
ALOGE("%s: ERROR Tx backend sample rate mixer control not identifed", __func__);
return -ENOSYS;
- }
+ }
- if (mixer_ctl_set_enum_by_string(ctl_sample_rate_tx, rate_str) != 0) {
- ALOGE("%s: Failed to reset Tx backend sample rate = %s", __func__, rate_str);
- return -ENOSYS;
+ if (mixer_ctl_set_enum_by_string(ctl_sample_rate_tx, rate_str) != 0) {
+ ALOGE("%s: Failed to reset Tx backend sample rate = %s", __func__, rate_str);
+ return -ENOSYS;
+ }
}
} else {
@@ -1707,7 +1810,7 @@
bool configure_aptx_enc_format(audio_aptx_encoder_config *aptx_bt_cfg)
{
struct mixer_ctl *ctl_enc_data = NULL;
- int mixer_size;
+ int mixer_size = 0;
bool is_configured = false;
int ret = 0;
int sample_rate_backup;
@@ -2209,7 +2312,7 @@
return -ENOSYS;
}
- if (a2dp.a2dp_source_suspended == true) {
+ if (a2dp.a2dp_source_suspended || a2dp.swb_configured) {
//session will be restarted after suspend completion
ALOGD("a2dp start requested during suspend state");
return -ENOSYS;
@@ -2434,6 +2537,16 @@
}
}
+static void reset_codec_config()
+{
+ reset_a2dp_enc_config_params();
+ reset_a2dp_source_dec_config_params();
+ a2dp_reset_backend_cfg(SOURCE);
+ if (a2dp.abr_config.is_abr_enabled && a2dp.abr_config.abr_started)
+ stop_abr();
+ a2dp.abr_config.is_abr_enabled = false;
+}
+
int a2dp_stop_playback()
{
int ret =0;
@@ -2456,14 +2569,9 @@
ALOGE("stop stream to BT IPC lib failed");
else
ALOGV("stop steam to BT IPC lib successful");
- reset_a2dp_enc_config_params();
- reset_a2dp_source_dec_config_params();
- a2dp_reset_backend_cfg(SOURCE);
- if (a2dp.abr_config.is_abr_enabled && a2dp.abr_config.abr_started)
- stop_abr();
- a2dp.abr_config.is_abr_enabled = false;
+ if (!a2dp.a2dp_source_suspended && !a2dp.swb_configured)
+ reset_codec_config();
a2dp.a2dp_source_started = false;
- a2dp_reset_backend_cfg(SOURCE);
}
if (!a2dp.a2dp_source_total_active_session_requests)
a2dp.a2dp_source_started = false;
@@ -2578,8 +2686,8 @@
pthread_mutex_lock(&a2dp.adev->lock);
}
}
- reset_a2dp_enc_config_params();
- reset_a2dp_source_dec_config_params();
+ if (!a2dp.swb_configured)
+ reset_codec_config();
if (a2dp.audio_source_suspend)
a2dp.audio_source_suspend();
} else if (a2dp.a2dp_source_suspended == true) {
@@ -2699,8 +2807,11 @@
a2dp.abr_config.abr_started = false;
a2dp.abr_config.imc_instance = 0;
a2dp.abr_config.abr_tx_handle = NULL;
+ a2dp.abr_config.abr_rx_handle = NULL;
a2dp.is_tws_mono_mode_on = false;
a2dp_source_init();
+ a2dp.swb_configured = false;
+
// init function pointers
fp_platform_get_pcm_device_id =
init_config.fp_platform_get_pcm_device_id;
@@ -2802,3 +2913,112 @@
return 0;
}
+
+
+bool configure_aptx_ad_speech_enc_fmt() {
+ struct mixer_ctl *ctl_enc_data = NULL;
+ int mixer_size = 0;
+ int ret = 0;
+ struct aptx_ad_speech_enc_cfg_t aptx_dsp_cfg;
+
+ ctl_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
+ if (!ctl_enc_data) {
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identifed");
+ return false;
+ }
+
+ /* Initialize dsp configuration params */
+ memset(&aptx_dsp_cfg, 0x0, sizeof(struct aptx_ad_speech_enc_cfg_t));
+ aptx_dsp_cfg.custom_cfg.enc_format = MEDIA_FMT_APTX_AD_SPEECH;
+ aptx_dsp_cfg.custom_cfg.sample_rate = SAMPLING_RATE_32K;
+ aptx_dsp_cfg.custom_cfg.num_channels = CH_MONO;
+ aptx_dsp_cfg.custom_cfg.channel_mapping[0] = PCM_CHANNEL_L;
+ aptx_dsp_cfg.imc_info.direction = IMC_RECEIVE;
+ aptx_dsp_cfg.imc_info.enable = IMC_ENABLE;
+ aptx_dsp_cfg.imc_info.purpose = IMC_PURPOSE_ID_BT_INFO;
+ aptx_dsp_cfg.imc_info.comm_instance = APTX_AD_SPEECH_INSTANCE_ID;
+ aptx_dsp_cfg.speech_mode.mode = a2dp.adev->swb_speech_mode;
+ aptx_dsp_cfg.speech_mode.swapping = SWAP_ENABLE;
+
+ /* Configure AFE DSP configuration */
+ mixer_size = sizeof(struct aptx_ad_speech_enc_cfg_t);
+ ret = mixer_ctl_set_array(ctl_enc_data, (void *)&aptx_dsp_cfg,
+ mixer_size);
+ if (ret != 0) {
+ ALOGE("%s: Failed to set SWB encoder config", __func__);
+ return false;
+ }
+
+ /* Configure AFE Input Bit Format as PCM_16 */
+ ret = a2dp_set_bit_format(DEFAULT_ENCODER_BIT_FORMAT);
+ if (ret != 0) {
+ ALOGE("%s: Failed to set SWB bit format", __func__);
+ return false;
+ }
+
+ return true;
+}
+
+bool configure_aptx_ad_speech_dec_fmt()
+{
+ struct mixer_ctl *ctl_dec_data = NULL;
+ struct aptx_ad_speech_dec_cfg_t dec_cfg;
+ int ret = 0;
+
+ ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_SOURCE_DEC_CONFIG_BLOCK);
+ if (!ctl_dec_data) {
+ ALOGE("%s: ERROR codec config data mixer control not identifed", __func__);
+ return false;
+ }
+ memset(&dec_cfg, 0x0, sizeof(dec_cfg));
+ dec_cfg.abr_cfg.dec_format = MEDIA_FMT_APTX_AD_SPEECH;
+ dec_cfg.abr_cfg.imc_info.direction = IMC_TRANSMIT;
+ dec_cfg.abr_cfg.imc_info.enable = IMC_ENABLE;
+ dec_cfg.abr_cfg.imc_info.purpose = IMC_PURPOSE_ID_BT_INFO;
+ dec_cfg.abr_cfg.imc_info.comm_instance = APTX_AD_SPEECH_INSTANCE_ID;
+ dec_cfg.speech_mode.mode = a2dp.adev->swb_speech_mode;
+ dec_cfg.speech_mode.swapping = SWAP_ENABLE;
+
+ ret = mixer_ctl_set_array(ctl_dec_data, &dec_cfg,
+ sizeof(dec_cfg));
+ if (ret != 0) {
+ ALOGE("%s: Failed to set decoder config", __func__);
+ return false;
+ }
+ return true;
+}
+
+int sco_start_configuration()
+{
+ ALOGD("sco_start_configuration start");
+
+ if (!a2dp.swb_configured) {
+ a2dp.bt_encoder_format = CODEC_TYPE_APTX_AD_SPEECH;
+ /* Configure AFE codec*/
+ if (configure_aptx_ad_speech_enc_fmt() &&
+ configure_aptx_ad_speech_dec_fmt()) {
+ ALOGD("%s: SCO enc/dec configured successfully", __func__);
+ } else {
+ ALOGE("%s: failed to send SCO configuration", __func__);
+ return -ETIMEDOUT;
+ }
+ /* Configure backend*/
+ a2dp.enc_sampling_rate = SAMPLING_RATE_96K;
+ a2dp.enc_channels = CH_MONO;
+ a2dp.abr_config.is_abr_enabled = true;
+ a2dp_set_backend_cfg(SOURCE);
+ /* Start abr*/
+ start_abr();
+ a2dp.swb_configured = true;
+ }
+ return 0;
+}
+
+void sco_reset_configuration()
+{
+ ALOGD("sco_reset_configuration start");
+
+ reset_codec_config();
+ a2dp.bt_encoder_format = CODEC_TYPE_INVALID;
+ a2dp.swb_configured = false;
+}
diff --git a/hal/audio_extn/adsp_hdlr.c b/hal/audio_extn/adsp_hdlr.c
index 31a6b16..f8c8133 100644
--- a/hal/audio_extn/adsp_hdlr.c
+++ b/hal/audio_extn/adsp_hdlr.c
@@ -1,5 +1,5 @@
/*
-* Copyright (c) 2017, The Linux Foundation. All rights reserved.
+* Copyright (c) 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
@@ -457,7 +457,7 @@
int audio_extn_adsp_hdlr_stream_register_event(void *handle, void *data,
adsp_event_callback_t cb,
- void *cookie)
+ void *cookie, bool is_adm_event)
{
int ret = 0;
char mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = {0};
@@ -480,14 +480,22 @@
ALOGE("%s: Invalid payload_length %d",__func__, param->payload_length);
return -EINVAL;
}
- ret = snprintf(cb_mixer_ctl_name, sizeof(cb_mixer_ctl_name),
+
+ if (is_adm_event)
+ ret = snprintf(cb_mixer_ctl_name, sizeof(cb_mixer_ctl_name),
+ "ADSP COPP Callback Event");
+ else
+ ret = snprintf(cb_mixer_ctl_name, sizeof(cb_mixer_ctl_name),
"ADSP Stream Callback Event %d", config->pcm_device_id);
+
if (ret < 0) {
ALOGE("%s: snprintf failed",__func__);
ret = -EINVAL;
goto done;
}
+
ctl = mixer_get_ctl_by_name(adsp_hdlr_inst->mixer, cb_mixer_ctl_name);
+
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s", __func__,
cb_mixer_ctl_name);
@@ -495,8 +503,13 @@
goto done;
}
- ret = snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ if (is_adm_event)
+ ret = snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "COPP Event Cmd");
+ else
+ ret = snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
"ADSP Stream Cmd %d", config->pcm_device_id);
+
if (ret < 0) {
ALOGE("%s: snprintf failed",__func__);
ret = -EINVAL;
@@ -504,12 +517,14 @@
}
ctl = mixer_get_ctl_by_name(adsp_hdlr_inst->mixer, mixer_ctl_name);
+
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s", __func__,
mixer_ctl_name);
ret = -EINVAL;
goto done;
}
+
ALOGD("%s: event = %d, payload_length %d", __func__, param->event_type, param->payload_length);
/* copy event_type, payload size and payload */
@@ -521,6 +536,7 @@
param->payload, param->payload_length);
ret = mixer_ctl_set_array(ctl, payload, (sizeof(param->event_type) +
sizeof(param->payload_length) + param->payload_length));
+
if (ret < 0) {
ALOGE("%s: Could not set ctl for mixer cmd - %s, ret %d", __func__,
mixer_ctl_name, ret);
@@ -537,23 +553,27 @@
/* create event threads during first event registration */
pthread_mutex_lock(&adsp_hdlr_inst->event_wait_lock);
+
if (!adsp_hdlr_inst->event_wait_thread_active)
create_event_wait_thread(adsp_hdlr_inst);
- pthread_mutex_unlock(&adsp_hdlr_inst->event_wait_lock);
+ pthread_mutex_unlock(&adsp_hdlr_inst->event_wait_lock);
pthread_mutex_lock(&adsp_hdlr_inst->event_callback_lock);
+
if (!adsp_hdlr_inst->event_callback_thread_active)
create_event_callback_thread(adsp_hdlr_inst);
- pthread_mutex_unlock(&adsp_hdlr_inst->event_callback_lock);
+ pthread_mutex_unlock(&adsp_hdlr_inst->event_callback_lock);
send_cmd_event_wait_thread(adsp_hdlr_inst, EVENT_CMD_WAIT);
}
+
event_info = (struct adsp_hdlr_event_info *) calloc(1,
sizeof(struct adsp_hdlr_event_info));
if (event_info == NULL) {
ret = -ENOMEM;
goto done;
}
+
event_info->event_type = param->event_type;
event_info->cb = cb;
event_info->cookie = cookie;
@@ -582,7 +602,7 @@
switch (cmd) {
case ADSP_HDLR_STREAM_CMD_REGISTER_EVENT :
- ret = audio_extn_adsp_hdlr_stream_register_event(handle, param, NULL, NULL);
+ ret = audio_extn_adsp_hdlr_stream_register_event(handle, param, NULL, NULL, false);
if (ret)
ALOGE("%s:adsp_hdlr_stream_register_event failed error %d",
__func__, ret);
diff --git a/hal/audio_extn/adsp_hdlr.h b/hal/audio_extn/adsp_hdlr.h
index 1c257fc..4ba9cb3 100644
--- a/hal/audio_extn/adsp_hdlr.h
+++ b/hal/audio_extn/adsp_hdlr.h
@@ -57,7 +57,7 @@
adsp_hdlr_cmd_t cmd,
void *param);
int audio_extn_adsp_hdlr_stream_register_event(void *handle,
- void *param, adsp_event_callback_t cb, void *cookie);
+ void *param, adsp_event_callback_t cb, void *cookie, bool is_adm_event);
int audio_extn_adsp_hdlr_stream_deregister_event(void *handle, void *param);
#else
#define audio_extn_adsp_hdlr_init(mixer) (0)
diff --git a/hal/audio_extn/audio_defs.h b/hal/audio_extn/audio_defs.h
index a0b1949..3fd4b85 100644
--- a/hal/audio_extn/audio_defs.h
+++ b/hal/audio_extn/audio_defs.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014-2015, 2017-2018, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014-2015, 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
@@ -217,6 +217,7 @@
typedef enum {
AUDIO_STREAM_PP_EVENT = 0,
AUDIO_STREAM_ENCDEC_EVENT = 1,
+ AUDIO_COPP_EVENT = 3,
} audio_event_id;
/* payload format for HAL parameter
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 431c248..24ed4c5 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -189,6 +189,7 @@
static bool audio_extn_concurrent_capture_enabled = false;
static bool audio_extn_compress_in_enabled = false;
static bool audio_extn_battery_listener_enabled = false;
+static bool audio_extn_maxx_audio_enabled = false;
#define AUDIO_PARAMETER_KEY_AANC_NOISE_LEVEL "aanc_noise_level"
#define AUDIO_PARAMETER_KEY_ANC "anc_enabled"
@@ -3791,6 +3792,13 @@
typedef int (*a2dp_stop_capture_t)();
static a2dp_stop_capture_t a2dp_stop_capture;
+typedef int (*sco_start_configuration_t)();
+static sco_start_configuration_t sco_start_configuration;
+
+typedef void (*sco_reset_configuration_t)();
+static sco_reset_configuration_t sco_reset_configuration;
+
+
int a2dp_offload_feature_init(bool is_feature_enabled)
{
ALOGD("%s: Called with feature %s", __func__,
@@ -3842,6 +3850,15 @@
ALOGE("%s: dlsym failed", __func__);
goto feature_disabled;
}
+ // initialize APIs for SWB extension
+ if (!(sco_start_configuration =
+ (sco_start_configuration_t)dlsym(a2dp_lib_handle, "sco_start_configuration")) ||
+ !(sco_reset_configuration =
+ (sco_reset_configuration_t)dlsym(a2dp_lib_handle, "sco_reset_configuration"))) {
+ ALOGE("%s: dlsym failed for swb APIs", __func__);
+ sco_start_configuration = NULL;
+ sco_reset_configuration = NULL;
+ }
ALOGD("%s:: ---- Feature A2DP_OFFLOAD is Enabled ----", __func__);
return 0;
}
@@ -3965,6 +3982,16 @@
return (a2dp_stop_capture ? a2dp_stop_capture() : 0);
}
+int audio_extn_sco_start_configuration()
+{
+ return (sco_start_configuration? sco_start_configuration() : 0);
+}
+
+void audio_extn_sco_reset_configuration()
+{
+ return (sco_reset_configuration? sco_reset_configuration() : 0);
+}
+
// END: A2DP_OFFLOAD =====================================================================
// START: HFP ======================================================================
@@ -4722,6 +4749,134 @@
{
return (batt_prop_is_charging)? batt_prop_is_charging(): false;
}
+// END: BATTERY_LISTENER ================================================================
+
+// START: MAXX_AUDIO =====================================================================
+#ifdef __LP64__
+#define MAXX_AUDIO_LIB_PATH "/vendor/lib64/libmaxxaudio.so"
+#else
+#define MAXX_AUDIO_LIB_PATH "/vendor/lib/libmaxxaudio.so"
+#endif
+
+static void *maxxaudio_lib_handle = NULL;
+
+typedef void (*maxxaudio_init_t)(void *, maxx_audio_init_config_t);
+static maxxaudio_init_t maxxaudio_init;
+
+typedef void (*maxxaudio_deinit_t)();
+static maxxaudio_deinit_t maxxaudio_deinit;
+
+typedef bool (*maxxaudio_set_state_t)(struct audio_device*, int,
+ float, bool);
+static maxxaudio_set_state_t maxxaudio_set_state;
+
+typedef void (*maxxaudio_set_device_t)(struct audio_usecase *);
+static maxxaudio_set_device_t maxxaudio_set_device;
+
+typedef void (*maxxaudio_set_parameters_t)(struct audio_device *,
+ struct str_parms *);
+static maxxaudio_set_parameters_t maxxaudio_set_parameters;
+
+typedef bool (*maxxaudio_supported_usb_t)();
+static maxxaudio_supported_usb_t maxxaudio_supported_usb;
+
+int maxx_audio_feature_init(bool is_feature_enabled)
+{
+ audio_extn_maxx_audio_enabled = is_feature_enabled;
+ ALOGD("%s: Called with feature %s", __func__,
+ is_feature_enabled ? "Enabled" : "NOT Enabled");
+ if (is_feature_enabled) {
+ // dlopen lib
+ maxxaudio_lib_handle = dlopen(MAXX_AUDIO_LIB_PATH, RTLD_NOW);
+
+ if (!maxxaudio_lib_handle) {
+ ALOGE("%s: dlopen failed", __func__);
+ goto feature_disabled;
+ }
+
+ if (!(maxxaudio_init =
+ (maxxaudio_init_t)dlsym(maxxaudio_lib_handle, "ma_init")) ||
+ !(maxxaudio_deinit =
+ (maxxaudio_deinit_t)dlsym(maxxaudio_lib_handle, "ma_deinit")) ||
+ !(maxxaudio_set_state =
+ (maxxaudio_set_state_t)dlsym(maxxaudio_lib_handle, "ma_set_state")) ||
+ !(maxxaudio_set_device =
+ (maxxaudio_set_device_t)dlsym(maxxaudio_lib_handle, "ma_set_device")) ||
+ !(maxxaudio_set_parameters =
+ (maxxaudio_set_parameters_t)dlsym(maxxaudio_lib_handle, "ma_set_parameters")) ||
+ !(maxxaudio_supported_usb =
+ (maxxaudio_supported_usb_t)dlsym(
+ maxxaudio_lib_handle, "ma_supported_usb"))) {
+ ALOGE("%s: dlsym failed", __func__);
+ goto feature_disabled;
+ }
+ ALOGD("%s:: ---- Feature MAXX_AUDIO is Enabled ----", __func__);
+ return 0;
+ }
+
+feature_disabled:
+ if (maxxaudio_lib_handle) {
+ dlclose(maxxaudio_lib_handle);
+ maxxaudio_lib_handle = NULL;
+ }
+
+ maxxaudio_init = NULL;
+ maxxaudio_deinit = NULL;
+ maxxaudio_set_state = NULL;
+ maxxaudio_set_device = NULL;
+ maxxaudio_set_parameters = NULL;
+ maxxaudio_supported_usb = NULL;
+ ALOGW(":: %s: ---- Feature MAXX_AUDIO is disabled ----", __func__);
+ return -ENOSYS;
+}
+
+bool audio_extn_is_maxx_audio_enabled()
+{
+ return audio_extn_maxx_audio_enabled;
+}
+
+void audio_extn_ma_init(void *platform)
+{
+
+ if (maxxaudio_init) {
+ maxx_audio_init_config_t init_config;
+ init_config.fp_platform_set_parameters = platform_set_parameters;
+ init_config.fp_audio_extn_get_snd_card_split = audio_extn_get_snd_card_split;
+ maxxaudio_init(platform, init_config);
+ }
+}
+
+void audio_extn_ma_deinit()
+{
+ if (maxxaudio_deinit)
+ maxxaudio_deinit();
+}
+
+bool audio_extn_ma_set_state(struct audio_device *adev, int stream_type,
+ float vol, bool active)
+{
+ return (maxxaudio_set_state ?
+ maxxaudio_set_state(adev, stream_type, vol, active): false);
+}
+
+void audio_extn_ma_set_device(struct audio_usecase *usecase)
+{
+ if (maxxaudio_set_device)
+ maxxaudio_set_device(usecase);
+}
+
+void audio_extn_ma_set_parameters(struct audio_device *adev,
+ struct str_parms *parms)
+{
+ if (maxxaudio_set_parameters)
+ maxxaudio_set_parameters(adev, parms);
+}
+
+bool audio_extn_ma_supported_usb()
+{
+ return (maxxaudio_supported_usb ? maxxaudio_supported_usb(): false);
+}
+// END: MAXX_AUDIO =====================================================================
void audio_extn_feature_init(int is_running_with_enhanced_fwk)
{
@@ -4831,6 +4986,9 @@
case BATTERY_LISTENER:
battery_listener_feature_init(enable);
break;
+ case MAXX_AUDIO:
+ maxx_audio_feature_init(enable);
+ break;
default:
break;
}
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 51e3ba2..e1e3ca0 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -313,6 +313,9 @@
bool audio_extn_a2dp_source_is_suspended();
int audio_extn_a2dp_start_capture();
int audio_extn_a2dp_stop_capture();
+int audio_extn_sco_start_configuration();
+void audio_extn_sco_reset_configuration();
+
// --- Function pointers from audio_extn needed by A2DP_OFFLOAD
typedef int (*fp_check_a2dp_restore_t)(struct audio_device *,
@@ -323,7 +326,24 @@
};
typedef struct a2dp_offload_init_config a2dp_offload_init_config_t;
// END: A2DP_OFFLOAD FEATURE ====================================================
-
+// START: MAXX_AUDIO FEATURE ==================================================
+void audio_extn_ma_init(void *platform);
+void audio_extn_ma_deinit();
+bool audio_extn_ma_set_state(struct audio_device *adev, int stream_type,
+ float vol, bool active);
+void audio_extn_ma_set_device(struct audio_usecase *usecase);
+void audio_extn_ma_set_parameters(struct audio_device *adev,
+ 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;
+ fp_audio_extn_get_snd_card_split_t fp_audio_extn_get_snd_card_split;
+};
+typedef struct maxx_audio_init_config maxx_audio_init_config_t;
+// START: MAXX_AUDIO FEATURE ==================================================
//START: SSRRC_FEATURE ==========================================================
bool audio_extn_ssr_check_usecase(struct stream_in *in);
int audio_extn_ssr_set_usecase(struct stream_in *in,
diff --git a/hal/audio_extn/audio_feature_manager.c b/hal/audio_extn/audio_feature_manager.c
index a3120df..e121426 100644
--- a/hal/audio_extn/audio_feature_manager.c
+++ b/hal/audio_extn/audio_feature_manager.c
@@ -168,6 +168,8 @@
return confValues->compress_in_enabled;
case BATTERY_LISTENER:
return confValues->battery_listener_enabled;
+ case MAXX_AUDIO:
+ return confValues->maxx_audio_enabled;
case COMPRESS_METADATA_NEEDED:
return confValues->compress_metadata_needed;
case INCALL_MUSIC:
diff --git a/hal/audio_extn/audio_feature_manager.h b/hal/audio_extn/audio_feature_manager.h
index 9e3c541..8df076c 100644
--- a/hal/audio_extn/audio_feature_manager.h
+++ b/hal/audio_extn/audio_feature_manager.h
@@ -70,6 +70,7 @@
COMPRESS_IN_CAPTURE,
BATTERY_LISTENER,
COMPRESS_METADATA_NEEDED,
+ MAXX_AUDIO,
COMPRESS_VOIP,
VOICE_START = COMPRESS_VOIP,
DYNAMIC_ECNS,
diff --git a/hal/audio_extn/hw_loopback.c b/hal/audio_extn/hw_loopback.c
index 5366066..76c8873 100644
--- a/hal/audio_extn/hw_loopback.c
+++ b/hal/audio_extn/hw_loopback.c
@@ -1,5 +1,5 @@
/*
-* Copyright (c) 2017-2018, The Linux Foundation. All rights reserved.
+* Copyright (c) 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
@@ -553,7 +553,8 @@
inout->adsp_hdlr_stream_handle = NULL;
goto exit;
}
- if (audio_extn_ip_hdlr_intf_supported(source_patch_config->format,false, true)) {
+ if (audio_extn_ip_hdlr_intf_supported(source_patch_config->format,false, true) ||
+ audio_extn_ip_hdlr_intf_supported_for_copp(adev->platform)) {
ret = audio_extn_ip_hdlr_intf_init(&inout->ip_hdlr_handle, NULL, NULL, adev,
USECASE_AUDIO_TRANSCODE_LOOPBACK_RX);
if (ret < 0) {
diff --git a/hal/audio_extn/ip_hdlr_intf.c b/hal/audio_extn/ip_hdlr_intf.c
old mode 100755
new mode 100644
index 3d5e1fe..0afc705
--- a/hal/audio_extn/ip_hdlr_intf.c
+++ b/hal/audio_extn/ip_hdlr_intf.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2017-2018, The Linux Foundation. All rights reserved.
+ * Copyright (c) 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
@@ -56,12 +56,19 @@
#include "audio_extn.h"
#include "platform_api.h"
#include "adsp_hdlr.h"
+#include "audio_hw.h"
/* These values defined by ADSP */
#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
+struct lib_fd_info {
+ int32_t fd;
+ int32_t flag;
+};
struct ip_hdlr_stream {
struct listnode list;
void *stream;
@@ -74,35 +81,109 @@
int (*deinit)(void *handle);
int (*open)(void *handle, bool is_dsp_decode, void *aud_sess_handle);
int (*shm_info)(void *handle, int *fd);
+ int (*shm_pp_info)(void *handle, int *fd);
+ int (*shm_pp)(void *handle, bool is_adm_event);
int (*get_lib_fd)(void *handle, int *lib_fd);
int (*close)(void *handle);
int (*event)(void *handle, void *payload);
int (*reg_cb)(void *handle, void *ack_cb, void *fail_cb);
+ int (*event_adm)(void *handle, void *payload);
+ int (*deinit_lib)(void *handle);
+
struct listnode stream_list;
pthread_mutex_t stream_list_lock;
int ref_cnt;
+ bool lib_fd_created;
+ void *ip_lib_handle; /*handle for dlclose of adsp lib*/
+ bool adm_event;
+ bool asm_event;
+ void *ip_dev_handle;
};
static struct ip_hdlr_intf *ip_hdlr = NULL;
-
+static bool adm_event_enable;
+static bool asm_event_enable;
+struct copp_cal_info {
+ uint32_t persist;
+ uint32_t snd_dev_id;
+ audio_devices_t dev_id;
+ int32_t acdb_dev_id;
+ uint32_t app_type;
+ uint32_t topo_id;
+ uint32_t sampling_rate;
+ uint32_t cal_type;
+ uint32_t module_id;
+#ifdef INSTANCE_ID_ENABLED
+ uint16_t instance_id;
+ uint16_t reserved;
+#endif
+ uint32_t param_id;
+};
+static struct copp_cal_info trumpet_data;
/* RTIC ack information */
struct rtic_ack_info {
uint32_t token;
uint32_t status;
};
+struct module_info {
+ uint32_t module_id;
+ uint32_t instance_id;
+ uint32_t token_coppidx;
+};
+
+struct rtic_ack_info_adm {
+ uint32_t token;
+ uint32_t status;
+ struct module_info mod_data;
+};
+
/* RTIC ack format sent to ADSP */
struct rtic_ack_param {
uint32_t param_size;
struct rtic_ack_info rtic_ack;
};
+struct rtic_ack_param_adm {
+ uint32_t param_size;
+ struct rtic_ack_info rtic_ack;
+ struct module_info mod_data;
+};
+
/* each event payload format */
struct reg_ev_pl {
uint32_t event_id;
uint32_t cfg_mask;
};
+struct reg_ev_pl_adm {
+ uint32_t event_id;
+ uint32_t module_id;
+ uint16_t instance_id;
+ uint16_t reserved;
+ uint32_t cfg_mask;
+};
+
+struct adm_pp_module {
+ uint32_t module_id;
+ uint32_t instance_id;
+ uint32_t be_id;
+ uint32_t fe_id;
+};
+
+struct adm_fd_info {
+ uint32_t fd;
+ struct adm_pp_module adm_pp_pl_fd;
+};
+/* adm event registration format */
+struct adm_reg_event {
+ struct adm_pp_module adm_info;
+ uint16_t version;
+ uint32_t num_reg_events;
+ struct reg_ev_pl_adm rtic;
+ struct reg_ev_pl_adm rtic_fail;
+};
+
/* event registration format */
struct reg_event {
uint16_t version;
@@ -121,23 +202,50 @@
uint8_t payload[0];
};
+int audio_extn_ip_hdlr_copp_update_cal_info(void *cfg, void *data)
+{
+ int ret = 0;
+ acdb_audio_cal_cfg_t *cal = (acdb_audio_cal_cfg_t*) cfg;
+ adm_event_enable = true; /* default enable with trumpet cal */
+
+ memcpy(&trumpet_data, cal, sizeof(struct copp_cal_info));
+ 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)
{
- if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DOLBY_TRUEHD)
+ if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DOLBY_TRUEHD) {
+ asm_event_enable = true;
return true;
- else if (!is_direct_passthrough && !audio_extn_qaf_is_enabled() &&
+ } 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)))
+ ((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)))
+ ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AC3))) {
+ asm_event_enable = true;
return true;
- else
+ } 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);
+
+ 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 )
@@ -147,7 +255,82 @@
return ip_hdlr->event(ip_hdlr_handle, payload);
}
-int audio_extn_ip_hdlr_intf_rtic_ack(void *aud_sess_handle, struct rtic_ack_info *info)
+int audio_extn_ip_hdlr_intf_rtic_ack_adm(void *aud_sess_handle,
+ struct rtic_ack_info_adm *info)
+{
+ char mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = {0};
+ int ret = 0;
+ int pcm_device_id = 0;
+ struct mixer_ctl *ctl = NULL;
+ struct rtic_ack_param_adm param;
+ struct listnode *node = NULL, *tempnode = NULL;
+ struct ip_hdlr_stream *stream_info = NULL;
+ struct audio_device *adev = NULL;
+ audio_usecase_t usecase = 0;
+
+ memset(¶m, 0, sizeof(struct rtic_ack_param_adm));
+ pthread_mutex_lock(&ip_hdlr->stream_list_lock);
+ list_for_each_safe(node, tempnode, &ip_hdlr->stream_list) {
+ stream_info = node_to_item(node, struct ip_hdlr_stream, list);
+ /* send the error if rtic failure notifcation is received */
+ if ((stream_info->stream == aud_sess_handle) &&
+ (stream_info->usecase == USECASE_AUDIO_TRANSCODE_LOOPBACK_RX)) {
+ struct stream_inout *inout = (struct stream_inout *)aud_sess_handle;
+ usecase = stream_info->usecase;
+ adev = inout->dev;
+ break;
+ } else if (stream_info->stream == aud_sess_handle) {
+ struct stream_out *out = (struct stream_out *)aud_sess_handle;
+ usecase = stream_info->usecase;
+ adev = out->dev;
+ break;
+ }
+ }
+ pthread_mutex_unlock(&ip_hdlr->stream_list_lock);
+
+ if (adev == NULL) {
+ ALOGE("%s:[%d] Invalid adev", __func__, ip_hdlr->ref_cnt);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ pcm_device_id = platform_get_pcm_device_id(usecase, PCM_PLAYBACK);
+
+ ALOGVV("%s:[%d] token = %d, info->status = %d, pcm_id = %d\n",__func__,
+ ip_hdlr->ref_cnt, info->token, info->status, pcm_device_id);
+
+ /* set mixer control to send RTIC done information */
+ ret = snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "COPP Event Ack");
+ if (ret < 0) {
+ ALOGE("%s:[%d] snprintf failed",__func__, ip_hdlr->ref_cnt);
+ ret = -EINVAL;
+ goto done;
+ }
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s:[%d] Could not get ctl for mixer cmd - %s", __func__,
+ ip_hdlr->ref_cnt, mixer_ctl_name);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ param.param_size = sizeof(struct rtic_ack_info);
+ memcpy(¶m.rtic_ack, info, sizeof(struct rtic_ack_info));
+ memcpy(¶m.mod_data, &info->mod_data, sizeof(struct module_info));
+ ret = mixer_ctl_set_array(ctl, (void *)¶m, sizeof(param));
+ if (ret < 0) {
+ ALOGE("%s:[%d] Could not set ctl for mixer cmd - %s, ret %d",
+ __func__, ip_hdlr->ref_cnt, mixer_ctl_name, ret);
+ goto done;
+ }
+
+done:
+ return ret;
+}
+
+int audio_extn_ip_hdlr_intf_rtic_ack(void *aud_sess_handle,
+ struct rtic_ack_info *info)
{
char mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = {0};
int ret = 0;
@@ -257,6 +440,139 @@
return 0;
}
+static int audio_extn_ip_hdlr_intf_open_adm_event(void *handle,
+ void *stream_handle, audio_usecase_t usecase)
+{
+ int ret = 0, fd = 0, pcm_device_id = 0;
+ struct adm_fd_info *fd_param_data = NULL;
+ struct audio_adsp_event *param = NULL;
+ struct adm_reg_event *reg_ev = NULL;
+ size_t shm_size = 0;
+ void *shm_buf = NULL;
+ struct stream_out *out = NULL;
+ struct stream_inout *inout = NULL;
+ void *adsp_hdlr_stream_handle = NULL;
+ struct audio_device *dev = NULL;
+ struct mixer_ctl *ctl = NULL;
+ struct audio_usecase *uc = NULL;
+ char mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = {0};
+
+ param = (struct audio_adsp_event *)calloc(1,
+ sizeof(struct audio_adsp_event));
+
+ if (param == NULL) {
+ ret = -ENOMEM;
+ goto done;
+ }
+
+ reg_ev = (struct adm_reg_event *)calloc(1, sizeof(struct adm_reg_event));
+
+ if (reg_ev == NULL) {
+ ret = -ENOMEM;
+ goto done;
+ }
+
+ fd_param_data = (struct adm_fd_info *)calloc(1, sizeof(struct adm_fd_info));
+
+ if (fd_param_data == NULL) {
+ ret = -ENOMEM;
+ goto done;
+ }
+
+ uc = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+
+ if (uc == NULL) {
+ ret = -ENOMEM;
+ goto done;
+ }
+
+ if (usecase == USECASE_AUDIO_TRANSCODE_LOOPBACK_RX) {
+ inout = (struct stream_inout *)stream_handle;
+ adsp_hdlr_stream_handle = inout->adsp_hdlr_stream_handle;
+ dev = inout->dev;
+ } else {
+ out = (struct stream_out *)stream_handle;
+ adsp_hdlr_stream_handle = out->adsp_hdlr_stream_handle;
+ dev = out->dev;
+ }
+ uc = get_usecase_from_list(dev, usecase);
+
+ reg_ev->adm_info.module_id = TRUMPET_MODULE;
+ reg_ev->adm_info.instance_id = 0;
+ reg_ev->adm_info.be_id = platform_get_snd_device_backend_index(
+ uc->out_snd_device);
+ reg_ev->adm_info.fe_id = platform_get_pcm_device_id(usecase, PCM_PLAYBACK);
+ reg_ev->num_reg_events = 2;
+ reg_ev->rtic.event_id = ADSP_EVENT_ID_RTIC;
+ reg_ev->rtic.module_id = TRUMPET_MODULE;
+ reg_ev->rtic.instance_id = 0;
+ reg_ev->rtic.cfg_mask = 1; /* event enabled */
+ reg_ev->rtic_fail.event_id = ADSP_EVENT_ID_RTIC_FAIL;
+ reg_ev->rtic_fail.module_id = TRUMPET_MODULE;
+ reg_ev->rtic_fail.instance_id = 0;
+ reg_ev->rtic_fail.cfg_mask = 1; /* event enabled */
+
+ param->event_type = AUDIO_COPP_EVENT;
+ param->payload_length = sizeof(struct adm_reg_event);
+ param->payload = reg_ev;
+ bool is_adm_event = true;
+
+ /* Register for event and its callback */
+ ret = audio_extn_adsp_hdlr_stream_register_event(adsp_hdlr_stream_handle,
+ param, audio_extn_ip_hdlr_intf_event_adm, handle, is_adm_event);
+ if (ret < 0) {
+ ALOGE("%s:[%d] failed to register event",__func__,
+ ip_hdlr->ref_cnt, ret);
+ goto done;
+ }
+
+ ip_hdlr->reg_cb(handle, &audio_extn_ip_hdlr_intf_rtic_ack_adm,
+ &audio_extn_ip_hdlr_intf_rtic_fail);
+ ip_hdlr->shm_pp_info(handle, &fd);
+ fd_param_data->fd = fd;
+ memcpy(&fd_param_data->adm_pp_pl_fd, ®_ev->adm_info,
+ sizeof(struct adm_pp_module));
+
+ ret = snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "COPP ION FD");
+
+ if (ret < 0) {
+ ALOGE("%s:[%d] snprintf failed",__func__, ip_hdlr->ref_cnt, ret);
+ goto done;
+ }
+ ALOGV("%s: fd = %d\n", __func__, fd);
+
+ ctl = mixer_get_ctl_by_name(dev->mixer, mixer_ctl_name);
+
+ if (!ctl) {
+ ALOGE("%s:[%d] Could not get ctl for mixer cmd - %s", __func__,
+ ip_hdlr->ref_cnt, mixer_ctl_name);
+ ret = -EINVAL;
+ goto done;
+ }
+ ret = mixer_ctl_set_array(ctl, fd_param_data, sizeof(fd_param_data));
+
+ if (ret < 0) {
+ ALOGE("%s:[%d] Could not set ctl for mixer cmd - %s, ret %d"
+ , __func__, ip_hdlr->ref_cnt, mixer_ctl_name, ret);
+ goto done;
+ }
+
+done:
+ if (param)
+ free(param);
+
+ if (reg_ev)
+ free(reg_ev);
+
+ if (fd_param_data)
+ free(fd_param_data);
+
+ if (uc)
+ free(uc);
+
+ return ret;
+}
+
static int audio_extn_ip_hdlr_intf_open_dsp(void *handle, void *stream_handle, audio_usecase_t usecase)
{
int ret = 0, fd = 0, pcm_device_id = 0;
@@ -268,7 +584,7 @@
struct audio_device *dev = NULL;
struct mixer_ctl *ctl = NULL;
char mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = {0};
-
+ bool is_adm_event = false;
param = (struct audio_adsp_event *)calloc(1, sizeof(struct audio_adsp_event));
if (!param)
return -ENOMEM;
@@ -301,7 +617,7 @@
/* Register for event and its callback */
ret = audio_extn_adsp_hdlr_stream_register_event(adsp_hdlr_stream_handle, param,
audio_extn_ip_hdlr_intf_event,
- handle);
+ handle, is_adm_event);
if (ret < 0) {
ALOGE("%s:[%d] failed to register event %d",__func__, ip_hdlr->ref_cnt, ret);
goto done;
@@ -363,12 +679,18 @@
}
ALOGD("%s:[%d] handle = %p, sess_handle = %p, is_dsp_decode = %d, usecase = %d",
__func__, ip_hdlr->ref_cnt, handle, aud_sess_handle, is_dsp_decode, usecase);
- if (is_dsp_decode) {
+ if (is_dsp_decode && ip_hdlr->asm_event) {
ret = audio_extn_ip_hdlr_intf_open_dsp(handle, aud_sess_handle, usecase);
if (ret < 0)
ip_hdlr->close(handle);
}
+ if (is_dsp_decode && ip_hdlr->adm_event) {
+ ret = audio_extn_ip_hdlr_intf_open_adm_event(handle, aud_sess_handle, usecase);
+ if (ret < 0)
+ ip_hdlr->close(handle);
+ }
+
pthread_mutex_lock(&ip_hdlr->stream_list_lock);
list_add_tail(&ip_hdlr->stream_list, &stream_info->list);
pthread_mutex_unlock(&ip_hdlr->stream_list_lock);
@@ -430,10 +752,12 @@
struct audio_device *dev, audio_usecase_t usecase)
{
int ret = 0, pcm_device_id;
- int lib_fd;
+ struct lib_fd_info lib_fd;
struct mixer_ctl *ctl = NULL;
char mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = {0};
+ memset(&lib_fd, 0, sizeof(struct lib_fd_info));
+
if (!ip_hdlr) {
ip_hdlr = (struct ip_hdlr_intf *)calloc(1, sizeof(struct ip_hdlr_intf));
if (!ip_hdlr)
@@ -451,44 +775,72 @@
ip_hdlr->init =(int (*)(void **handle, char *lib_path,
void **lib_handle))dlsym(ip_hdlr->lib_hdl, "audio_ip_hdlr_init");
ip_hdlr->deinit = (int (*)(void *handle))dlsym(ip_hdlr->lib_hdl, "audio_ip_hdlr_deinit");
+ ip_hdlr->deinit_lib = (int (*)(void *handle))dlsym(ip_hdlr->lib_hdl,
+ "audio_ip_hdlr_deinit_lib");
ip_hdlr->open = (int (*)(void *handle, bool is_dsp_decode,
void *sess_handle))dlsym(ip_hdlr->lib_hdl, "audio_ip_hdlr_open");
ip_hdlr->close =(int (*)(void *handle))dlsym(ip_hdlr->lib_hdl, "audio_ip_hdlr_close");
ip_hdlr->reg_cb =(int (*)(void *handle, void *ack_cb,
void *fail_cb))dlsym(ip_hdlr->lib_hdl, "audio_ip_hdlr_reg_cb");
- ip_hdlr->shm_info =(int (*)(void *handle, int *fd))dlsym(ip_hdlr->lib_hdl,
- "audio_ip_hdlr_shm_info");
+ ip_hdlr->shm_info =(int (*)(void *handle, int *fd))dlsym(
+ ip_hdlr->lib_hdl, "audio_ip_hdlr_shm_info");
+ ip_hdlr->shm_pp_info =(int (*)(void *handle, int *fd))dlsym(ip_hdlr->lib_hdl,
+ "audio_ip_hdlr_shm_pp_info");
ip_hdlr->get_lib_fd =(int (*)(void *handle, int *fd))dlsym(ip_hdlr->lib_hdl,
"audio_ip_hdlr_lib_fd");
ip_hdlr->event =(int (*)(void *handle, void *payload))dlsym(ip_hdlr->lib_hdl,
"audio_ip_hdlr_event");
+ ip_hdlr->event_adm =(int (*)(void *handle, void *payload))dlsym(
+ ip_hdlr->lib_hdl, "audio_ip_hdlr_event_adm");
+ ip_hdlr->shm_pp = (int (*)(void *handle, bool is_adm_event))dlsym(
+ ip_hdlr->lib_hdl, "audio_ip_hdlr_create_shm_pp");
if (!ip_hdlr->init || !ip_hdlr->deinit || !ip_hdlr->open ||
!ip_hdlr->close || !ip_hdlr->reg_cb || !ip_hdlr->shm_info ||
- !ip_hdlr->event || !ip_hdlr->get_lib_fd) {
+ !ip_hdlr->event || !ip_hdlr->get_lib_fd || !ip_hdlr->deinit_lib ||
+ !ip_hdlr->shm_pp || !ip_hdlr->shm_pp_info || !ip_hdlr->event_adm) {
ALOGE("%s: failed to get symbols", __func__);
ret = -EINVAL;
goto dlclose;
-
}
}
+ ip_hdlr->ip_dev_handle = dev;
ret = ip_hdlr->init(handle, lib_path, lib_handle);
if (ret < 0) {
ALOGE("%s:[%d] init failed ret = %d", __func__, ip_hdlr->ref_cnt, ret);
ret = -EINVAL;
goto dlclose;
}
- if (!lib_path) {
- ip_hdlr->get_lib_fd(*handle, &lib_fd);
- pcm_device_id = platform_get_pcm_device_id(usecase, PCM_PLAYBACK);
+ if (adm_event_enable) {
+ ret = ip_hdlr->shm_pp(*handle, adm_event_enable);
+
+ if (ret < 0) {
+ ALOGE("%s:[%d] init failed ret = %d", __func__, ip_hdlr->ref_cnt, ret);
+ ret = -EINVAL;
+ goto dlclose;
+ }
+ ip_hdlr->adm_event = adm_event_enable;
+ adm_event_enable = false;
+ }
+
+ if (asm_event_enable) {
+ ip_hdlr->asm_event = asm_event_enable;
+ asm_event_enable = false;
+ }
+
+ if (!lib_path && !(ip_hdlr->lib_fd_created)) {
+ /* save handle to dlcose of lib fd */
+ ip_hdlr->ip_lib_handle = handle;
+ ip_hdlr->get_lib_fd(*handle, &lib_fd.fd);
+ lib_fd.flag = 1;
+ /* sending lib ion fd to routing driver */
ret = snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
- "Playback ION LIB FD %d", pcm_device_id);
+ "ADSP ION LIB FD");
if (ret < 0) {
ALOGE("%s:[%d] snprintf failed %d", __func__, ip_hdlr->ref_cnt, ret);
goto dlclose;
}
- ALOGV("%s: fd = %d pcm_id = %d", __func__, lib_fd, pcm_device_id);
ctl = mixer_get_ctl_by_name(dev->mixer, mixer_ctl_name);
if (!ctl) {
@@ -497,12 +849,14 @@
ret = -EINVAL;
goto dlclose;
}
- ret = mixer_ctl_set_array(ctl, &lib_fd, sizeof(lib_fd));
+
+ ret = mixer_ctl_set_array(ctl, &lib_fd, sizeof(struct lib_fd_info));
if (ret < 0) {
ALOGE("%s:[%d] Could not set ctl for mixer cmd - %s, ret %d", __func__, ip_hdlr->ref_cnt,
mixer_ctl_name, ret);
goto dlclose;
}
+ ip_hdlr->lib_fd_created = true;
}
ip_hdlr->ref_cnt++;
ALOGD("%s:[%d] init done", __func__, ip_hdlr->ref_cnt);
@@ -521,6 +875,12 @@
int audio_extn_ip_hdlr_intf_deinit(void *handle)
{
int ret = 0;
+ struct lib_fd_info lib_fd;
+ struct mixer_ctl *ctl = NULL;
+ char mixer_ctl_name[MIXER_PATH_MAX_LENGTH] = {0};
+ struct audio_device *ip_dev = (struct audio_device *) (ip_hdlr->ip_dev_handle);
+
+ memset(&lib_fd, 0, sizeof(struct lib_fd_info));
if (!handle) {
ALOGE("%s:[%d] handle is NULL", __func__, ip_hdlr->ref_cnt);
@@ -532,12 +892,41 @@
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);
+ lib_fd.flag = 0;
+ /* sending lib ion fd to routing driver */
+ ret = snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "ADSP ION LIB FD");
+ if (ret < 0) {
+ ALOGE("%s:[%d] snprintf failed %d"
+ , __func__, ip_hdlr->ref_cnt, ret);
+ goto dlclose;
+ }
+
+ ctl = mixer_get_ctl_by_name(ip_dev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s:[%d] Could not get ctl for mixer cmd - %s", __func__,
+ ip_hdlr->ref_cnt, mixer_ctl_name);
+ ret = -EINVAL;
+ goto dlclose;
+ }
+ ret = mixer_ctl_set_array(ctl, &lib_fd, sizeof(struct lib_fd_info));
+ if (ret < 0) {
+ ALOGE("%s:[%d] Could not set ctl for mixer cmd - %s, ret %d"
+ , __func__, ip_hdlr->ref_cnt, mixer_ctl_name, ret);
+ goto dlclose;
+ }
+
+ ret = ip_hdlr->deinit_lib(ip_hdlr->ip_lib_handle);
+ ip_hdlr->lib_fd_created = false;
if (ip_hdlr->lib_hdl)
dlclose(ip_hdlr->lib_hdl);
-
+dlclose:
pthread_mutex_destroy(&ip_hdlr->stream_list_lock);
free(ip_hdlr);
ip_hdlr = NULL;
}
+ ALOGD("%s: done\n", __func__);
return ret;
+
}
diff --git a/hal/audio_extn/ip_hdlr_intf.h b/hal/audio_extn/ip_hdlr_intf.h
index 6040620..5cfbc24 100644
--- a/hal/audio_extn/ip_hdlr_intf.h
+++ b/hal/audio_extn/ip_hdlr_intf.h
@@ -41,6 +41,8 @@
bool audio_extn_ip_hdlr_intf_supported(audio_format_t format,
bool is_direct_passthru,
bool is_transcode_loopback);
+bool audio_extn_ip_hdlr_intf_supported_for_copp(void *platform);
+int audio_extn_ip_hdlr_copp_update_cal_info(void *cfg, void *data);
#else
@@ -49,6 +51,8 @@
#define audio_extn_ip_hdlr_intf_init(handle, lib_path, lib_handlei, adev, usecase) (0)
#define audio_extn_ip_hdlr_intf_deinit(handle) (0)
#define audio_extn_ip_hdlr_intf_supported(format, is_direct_passthru, is_loopback) (0)
+#define audio_extn_ip_hdlr_intf_supported_for_copp(platform) (0)
+#define audio_extn_ip_hdlr_copp_update_cal_info(cfg, data) (0)
#endif
diff --git a/hal/audio_extn/maxxaudio.c b/hal/audio_extn/maxxaudio.c
index 5f2a7f0..8830486 100644
--- a/hal/audio_extn/maxxaudio.c
+++ b/hal/audio_extn/maxxaudio.c
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2018 The Android Open Source Project
+ * Copyright (C) 2018-2019 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.
@@ -156,12 +156,15 @@
static uint16_t g_supported_dev = 0;
static struct ma_state ma_cur_state_table[STREAM_MAX_TYPES];
static struct ma_platform_data *my_data = NULL;
+// --- external function dependency ---
+fp_platform_set_parameters_t fp_platform_set_parameters;
+fp_audio_extn_get_snd_card_split_t fp_audio_extn_get_snd_card_split;
static int set_audio_cal(const char *audio_cal)
{
ALOGV("set_audio_cal: %s", audio_cal);
- return platform_set_parameters(my_data->platform,
+ return fp_platform_set_parameters(my_data->platform,
str_parms_create_str(audio_cal));
}
@@ -200,7 +203,7 @@
((usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) ||
(usecase->devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) ||
(audio_is_usb_out_device(usecase->devices) &&
- audio_extn_ma_supported_usb())))
+ ma_supported_usb())))
/* TODO: enable A2DP when it is ready */
return true;
@@ -390,7 +393,7 @@
}
// adev_init lock held
-void audio_extn_ma_init(void *platform)
+void ma_init(void *platform, maxx_audio_init_config_t init_config)
{
ma_stream_type_t i = 0;
int ret = 0;
@@ -398,7 +401,11 @@
char mps_path[128] = {0};
char cnf_path[128] = {0};
struct snd_card_split *snd_split_handle = NULL;
- snd_split_handle = audio_extn_get_snd_card_split();
+
+ fp_platform_set_parameters = init_config.fp_platform_set_parameters;
+ fp_audio_extn_get_snd_card_split = init_config.fp_audio_extn_get_snd_card_split;
+
+ snd_split_handle = fp_audio_extn_get_snd_card_split();
if (platform == NULL) {
ALOGE("%s: platform is NULL", __func__);
@@ -549,7 +556,7 @@
}
//adev_init lock held
-void audio_extn_ma_deinit()
+void ma_deinit()
{
if (my_data) {
/* deinit ma parameter */
@@ -566,7 +573,7 @@
}
// adev_init and adev lock held
-bool audio_extn_ma_set_state(struct audio_device *adev, int stream_type,
+bool ma_set_state(struct audio_device *adev, int stream_type,
float vol, bool active)
{
bool ret = false;
@@ -613,7 +620,7 @@
return ret;
}
-void audio_extn_ma_set_device(struct audio_usecase *usecase)
+void ma_set_device(struct audio_usecase *usecase)
{
int i = 0;
struct ma_audio_cal_settings ma_cal;
@@ -663,7 +670,7 @@
pthread_mutex_unlock(&my_data->lock);
}
-void audio_extn_ma_set_parameters(struct audio_device *adev,
+void ma_set_parameters(struct audio_device *adev,
struct str_parms *parms)
{
int ret;
@@ -720,7 +727,7 @@
}
}
-bool audio_extn_ma_supported_usb()
+bool ma_supported_usb()
{
ALOGV("%s: current support 0x%x", __func__, g_supported_dev);
return (g_supported_dev & SUPPORTED_USB) ? true : false;
diff --git a/hal/audio_extn/maxxaudio.h b/hal/audio_extn/maxxaudio.h
index 4c91107..1ab7f80 100644
--- a/hal/audio_extn/maxxaudio.h
+++ b/hal/audio_extn/maxxaudio.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2018 The Android Open Source Project
+ * Copyright (C) 2018-2019 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.
@@ -17,23 +17,12 @@
#ifndef MAXXAUDIO_H_
#define MAXXAUDIO_H_
-#ifndef MAXXAUDIO_QDSP_ENABLED
-#define audio_extn_ma_init(platform) (0)
-#define audio_extn_ma_deinit() (0)
-#define audio_extn_ma_set_state(adev, type, vol, active) (false)
-#define audio_extn_ma_set_device(usecase) (0)
-#define audio_extn_ma_set_parameters(adev, param) (0)
-#define audio_extn_ma_supported_usb() (false)
-#else
-void audio_extn_ma_init(void *platform);
-void audio_extn_ma_deinit();
-bool audio_extn_ma_set_state(struct audio_device *adev, int stream_type,
+void ma_init(void *platform, maxx_audio_init_config_t init_config);
+void ma_deinit();
+bool ma_set_state(struct audio_device *adev, int stream_type,
float vol, bool active);
-void audio_extn_ma_set_device(struct audio_usecase *usecase);
-void audio_extn_ma_set_parameters(struct audio_device *adev,
+void ma_set_device(struct audio_usecase *usecase);
+void ma_set_parameters(struct audio_device *adev,
struct str_parms *parms);
-bool audio_extn_ma_supported_usb();
-#endif /* MAXXAUDIO_QDSP_ENABLED */
-
-#endif /* MAXXAUDIO_H_ */
-
+bool ma_supported_usb();
+#endif /* MAXXAUDIO_H_ */
\ No newline at end of file
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index c53fa31..4319174 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -966,7 +966,7 @@
}
static int enable_disable_effect(struct audio_device *adev, int effect_type, bool enable)
-{
+{
struct audio_effect_config effect_config;
struct audio_usecase *usecase = NULL;
int ret = 0;
@@ -1189,6 +1189,14 @@
goto err;
}
+ if (((SND_DEVICE_OUT_BT_SCO_SWB == snd_device) ||
+ (SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC == snd_device) ||
+ (SND_DEVICE_IN_BT_SCO_MIC_SWB == snd_device)) &&
+ (audio_extn_sco_start_configuration() < 0)) {
+ ALOGE(" fail to configure sco control path ");
+ goto err;
+ }
+
/* due to the possibility of calibration overwrite between listen
and audio, notify listen hal before audio calibration is sent */
audio_extn_sound_trigger_update_device_status(snd_device,
@@ -1281,7 +1289,14 @@
audio_extn_a2dp_stop_playback();
else if (snd_device == SND_DEVICE_IN_BT_A2DP)
audio_extn_a2dp_stop_capture();
- else if ((snd_device == SND_DEVICE_OUT_HDMI) ||
+ else if ((snd_device == SND_DEVICE_OUT_BT_SCO_SWB) ||
+ (snd_device == SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC) ||
+ (snd_device == SND_DEVICE_IN_BT_SCO_MIC_SWB)) {
+ if ((adev->snd_dev_ref_cnt[SND_DEVICE_OUT_BT_SCO_SWB] == 0) &&
+ (adev->snd_dev_ref_cnt[SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] == 0) &&
+ (adev->snd_dev_ref_cnt[SND_DEVICE_IN_BT_SCO_MIC_SWB] == 0))
+ audio_extn_sco_reset_configuration();
+ } else if ((snd_device == SND_DEVICE_OUT_HDMI) ||
(snd_device == SND_DEVICE_OUT_DISPLAY_PORT))
adev->is_channel_status_set = false;
else if ((snd_device == SND_DEVICE_OUT_HEADPHONES) &&
@@ -6698,7 +6713,7 @@
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_out *out;
- int ret = 0;
+ int ret = 0, ip_hdlr_stream = 0, ip_hdlr_dev = 0;
audio_format_t format;
struct adsp_hdlr_stream_cfg hdlr_stream_cfg;
bool is_direct_passthough = false;
@@ -7447,7 +7462,8 @@
is_direct_passthough = audio_extn_passthru_is_direct_passthrough(out);
if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) ||
(out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) ||
- (audio_extn_ip_hdlr_intf_supported(config->format, is_direct_passthough, false))) {
+ audio_extn_ip_hdlr_intf_supported_for_copp(adev->platform) ||
+ (audio_extn_ip_hdlr_intf_supported(config->format, is_direct_passthough, false))) {
hdlr_stream_cfg.pcm_device_id = platform_get_pcm_device_id(
out->usecase, PCM_PLAYBACK);
hdlr_stream_cfg.flags = out->flags;
@@ -7459,7 +7475,10 @@
out->adsp_hdlr_stream_handle = NULL;
}
}
- if (audio_extn_ip_hdlr_intf_supported(config->format, is_direct_passthough, false)) {
+ ip_hdlr_stream = audio_extn_ip_hdlr_intf_supported(config->format,
+ is_direct_passthough, false);
+ ip_hdlr_dev = audio_extn_ip_hdlr_intf_supported_for_copp(adev->platform);
+ if (ip_hdlr_stream || ip_hdlr_dev ) {
ret = audio_extn_ip_hdlr_intf_init(&out->ip_hdlr_handle, NULL, NULL, adev, out->usecase);
if (ret < 0) {
ALOGE("%s: audio_extn_ip_hdlr_intf_init failed %d",__func__, ret);
@@ -7623,33 +7642,34 @@
adev->screen_off = true;
}
-#ifndef MAXXAUDIO_QDSP_ENABLED
- ret = str_parms_get_int(parms, "rotation", &val);
- if (ret >= 0) {
- bool reverse_speakers = false;
- switch(val) {
- // FIXME: note that the code below assumes that the speakers are in the correct placement
- // relative to the user when the device is rotated 90deg from its default rotation. This
- // assumption is device-specific, not platform-specific like this code.
- case 270:
- reverse_speakers = true;
- break;
- case 0:
- case 90:
- case 180:
- break;
- default:
- ALOGE("%s: unexpected rotation of %d", __func__, val);
- status = -EINVAL;
+ if(!audio_extn_is_maxx_audio_enabled()) {
+ ret = str_parms_get_int(parms, "rotation", &val);
+ if (ret >= 0) {
+ bool reverse_speakers = false;
+ switch(val) {
+ // FIXME: note that the code below assumes that the speakers are
+ // in the correct placement relative to the user when the device
+ // is rotated 90deg from its default rotation. This assumption
+ // is device-specific, not platform-specific like this code.
+ case 270:
+ reverse_speakers = true;
+ break;
+ case 0:
+ case 90:
+ case 180:
+ break;
+ default:
+ ALOGE("%s: unexpected rotation of %d", __func__, val);
+ status = -EINVAL;
+ }
+ if (status == 0) {
+ // check and set swap
+ // - check if orientation changed and speaker active
+ // - set rotation and cache the rotation value
+ platform_check_and_set_swap_lr_channels(adev, reverse_speakers);
+ }
}
- if (status == 0) {
- // check and set swap
- // - check if orientation changed and speaker active
- // - set rotation and cache the rotation value
- platform_check_and_set_swap_lr_channels(adev, reverse_speakers);
- }
- }
-#endif
+ }
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value));
if (ret >= 0) {
@@ -7659,6 +7679,12 @@
adev->bt_wb_speech_enabled = false;
}
+ ret = str_parms_get_str(parms, "bt_swb", value, sizeof(value));
+ if (ret >= 0) {
+ val = atoi(value);
+ adev->swb_speech_mode = val;
+ }
+
ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
@@ -9104,6 +9130,7 @@
adev->enable_voicerx = false;
adev->bt_wb_speech_enabled = false;
+ adev->swb_speech_mode = SPEECH_MODE_INVALID;
//initialize this to false for now,
//this will be set to true through set param
adev->vr_audio_mode_enabled = false;
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 1e7e1c5..1a04cff 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -85,6 +85,9 @@
#define ACDB_DEV_TYPE_OUT 1
#define ACDB_DEV_TYPE_IN 2
+/* SCO SWB codec mode */
+#define SPEECH_MODE_INVALID 0xFFFF
+
/* support positional and index masks to 8ch */
#define MAX_SUPPORTED_CHANNEL_MASKS (2 * FCC_8)
#define MAX_SUPPORTED_FORMATS 15
@@ -527,6 +530,7 @@
unsigned int cur_hdmi_bit_width;
unsigned int cur_wfd_channels;
bool bt_wb_speech_enabled;
+ unsigned int swb_speech_mode;
bool allow_afe_proxy_usage;
bool is_charging; // from battery listener
bool mic_break_enabled;
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 45fadf5..7b15f53 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -6159,7 +6159,6 @@
ALOGV("%s: Format doesnt have to be set", __func__);
}
- format = format & AUDIO_FORMAT_MAIN_MASK;
/* Set data format only if there is a change from PCM to compressed
and vice versa */
if (set_mi2s_tx_data_format && (format ^ my_data->current_backend_cfg[backend_idx].format)) {
@@ -6169,7 +6168,7 @@
__func__, ext_disp_format);
return -EINVAL;
}
- if (format == AUDIO_FORMAT_PCM) {
+ if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
ALOGE("%s:MI2S data format LPCM", __func__);
mixer_ctl_set_enum_by_string(ctl, "LPCM");
} else {
@@ -6724,18 +6723,25 @@
__func__, bit_width, sample_rate, channels);
}
- ALOGI("%s:txbecf: afe: Codec selected backend: %d updated bit width: %d and "
- "sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
+ ALOGI("%s:txbecf: afe: current backend bit_width %d sample_rate %d channels %d, format %x",
+ __func__,
+ my_data->current_backend_cfg[backend_idx].bit_width,
+ my_data->current_backend_cfg[backend_idx].sample_rate,
+ my_data->current_backend_cfg[backend_idx].channels,
+ my_data->current_backend_cfg[backend_idx].format);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
+ // Note that below if block would be entered even if main format is same
+ // but subformat is different for e.g. current backend is configured for 16 bit PCM
+ // as compared to 24 bit PCM backend requested
if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
(sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
(channels != my_data->current_backend_cfg[backend_idx].channels) ||
- ((format & AUDIO_FORMAT_MAIN_MASK) != my_data->current_backend_cfg[backend_idx].format)) {
+ (format != my_data->current_backend_cfg[backend_idx].format)) {
backend_cfg->bit_width = bit_width;
backend_cfg->sample_rate= sample_rate;
backend_cfg->channels = channels;
- backend_cfg->format = format & AUDIO_FORMAT_MAIN_MASK;
+ backend_cfg->format = format;
backend_change = true;
ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
"new sample rate: %d new channel: %d new format: %d",
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 01df5bf..1c46ed7 100755
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -684,6 +684,10 @@
sizeof("sm6150-qrd-snd-card"))) {
hw_info->is_stereo_spkr = false;
strlcpy(hw_info->name, "sm6150", sizeof(hw_info->name));
+ } else if (!strncmp(snd_card_name, "sm6150-wcd9375qrd-snd-card",
+ sizeof("sm6150-wcd9375qrd-snd-card"))) {
+ strlcpy(hw_info->name, "sm6150", sizeof(hw_info->name));
+ hw_info->is_stereo_spkr = false;
} else if (!strncmp(snd_card_name, "sm6150-tavil-snd-card",
sizeof("sm6150-tavil-snd-card"))) {
strlcpy(hw_info->name, "sm6150", sizeof(hw_info->name));
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index a428f9a..acd237e 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -498,6 +498,7 @@
[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = "speaker-and-display-port",
[SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
[SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+ [SND_DEVICE_OUT_BT_SCO_SWB] = "bt-sco-headset-swb",
[SND_DEVICE_OUT_BT_A2DP] = "bt-a2dp",
[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = "speaker-and-bt-a2dp",
[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP] = "speaker-safe-and-bt-a2dp",
@@ -544,9 +545,12 @@
[SND_DEVICE_OUT_SPEAKER_AND_BT_SCO] = "speaker-and-bt-sco",
[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO] = "speaker-safe-and-bt-sco",
[SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB] = "speaker-and-bt-sco-wb",
+ [SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB] = "speaker-and-bt-sco-swb",
[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB] = "speaker-safe-and-bt-sco-wb",
+ [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB] = "speaker-safe-and-bt-sco-swb",
[SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO] = "wsa-speaker-and-bt-sco",
[SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB] = "wsa-speaker-and-bt-sco-wb",
+ [SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB] = "wsa-speaker-and-bt-sco-wb",
/* Capture sound devices */
[SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
@@ -593,6 +597,8 @@
[SND_DEVICE_IN_BT_SCO_MIC_NREC] = "bt-sco-mic",
[SND_DEVICE_IN_BT_SCO_MIC_WB] = "bt-sco-mic-wb",
[SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = "bt-sco-mic-wb",
+ [SND_DEVICE_IN_BT_SCO_MIC_SWB] = "bt-sco-mic-swb",
+ [SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] = "bt-sco-mic-swb",
[SND_DEVICE_IN_BT_A2DP] = "bt-a2dp-cap",
[SND_DEVICE_IN_CAMCORDER_MIC] = "camcorder-mic",
[SND_DEVICE_IN_VOICE_DMIC] = "voice-dmic-ef",
@@ -751,7 +757,9 @@
[SND_DEVICE_OUT_BT_SCO] = 22,
[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO] = 14,
[SND_DEVICE_OUT_BT_SCO_WB] = 39,
+ [SND_DEVICE_OUT_BT_SCO_SWB] = 39,
[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB] = 14,
+ [SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB] = 14,
[SND_DEVICE_OUT_BT_A2DP] = 20,
[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = 14,
[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP] = 14,
@@ -834,6 +842,8 @@
[SND_DEVICE_IN_BT_SCO_MIC_NREC] = 122,
[SND_DEVICE_IN_BT_SCO_MIC_WB] = 38,
[SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = 123,
+ [SND_DEVICE_IN_BT_SCO_MIC_SWB] = 38,
+ [SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] = 123,
[SND_DEVICE_IN_BT_A2DP] = 21,
[SND_DEVICE_IN_CAMCORDER_MIC] = 4,
[SND_DEVICE_IN_VOICE_DMIC] = 41,
@@ -954,7 +964,9 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO)},
{TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_SWB)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB)},
{TO_NAME_INDEX(SND_DEVICE_OUT_BT_A2DP)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP)},
@@ -965,8 +977,10 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_SCO)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_USB)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_USB)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TX)},
@@ -1044,6 +1058,8 @@
{TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_NREC)},
{TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_WB)},
{TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_WB_NREC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_SWB)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC)},
{TO_NAME_INDEX(SND_DEVICE_IN_BT_A2DP)},
{TO_NAME_INDEX(SND_DEVICE_IN_CAMCORDER_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_DMIC)},
@@ -1515,7 +1531,9 @@
!strncmp(snd_card_name, "sdm439-sku1-snd-card",
sizeof("sdm439-sku1-snd-card")) ||
!strncmp(snd_card_name, "sdm439-snd-card-mtp",
- sizeof("sdm439-snd-card-mtp"))) {
+ sizeof("sdm439-snd-card-mtp")) ||
+ !strncmp(snd_card_name, "sm6150-wcd9375qrd-snd-card",
+ sizeof("sm6150-wcd9375qrd-snd-card"))) {
ALOGI("%s: snd_card_name: %s", __func__, snd_card_name);
my_data->is_internal_codec = true;
my_data->is_slimbus_interface = false;
@@ -1658,9 +1676,9 @@
if (enable) {
if (!voice_extn_is_compress_voip_supported()) {
- if (adev->mode == AUDIO_MODE_IN_COMMUNICATION)
- strlcat(ec_ref_mixer_path, "-voip", MIXER_PATH_MAX_LENGTH);
- }
+ if (adev->mode == AUDIO_MODE_IN_COMMUNICATION)
+ strlcat(ec_ref_mixer_path, "-voip", MIXER_PATH_MAX_LENGTH);
+ }
strlcpy(my_data->ec_ref_mixer_path, ec_ref_mixer_path,
MIXER_PATH_MAX_LENGTH);
/*
@@ -1894,11 +1912,14 @@
backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_WB] = strdup("bt-sco-wb");
backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_NREC] = strdup("bt-sco");
backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = strdup("bt-sco-wb");
+ backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_SWB] = strdup("bt-sco-swb");
+ backend_tag_table[SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] = strdup("bt-sco-swb");
backend_tag_table[SND_DEVICE_IN_SPDIF] = strdup("spdif-in");
backend_tag_table[SND_DEVICE_IN_HDMI_MIC] = strdup("hdmi-in");
backend_tag_table[SND_DEVICE_IN_HDMI_ARC] = strdup("hdmi-arc-in");
backend_tag_table[SND_DEVICE_OUT_BT_SCO] = strdup("bt-sco");
backend_tag_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("bt-sco-wb");
+ backend_tag_table[SND_DEVICE_OUT_BT_SCO_SWB] = strdup("bt-sco-swb");
backend_tag_table[SND_DEVICE_OUT_HDMI] = strdup("hdmi");
backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("speaker-and-hdmi");
backend_tag_table[SND_DEVICE_OUT_DISPLAY_PORT] = strdup("display-port");
@@ -1920,6 +1941,8 @@
strdup("speaker-safe-and-bt-sco");
backend_tag_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB] =
strdup("speaker-safe-and-bt-sco-wb");
+ backend_tag_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB] =
+ strdup("speaker-safe-and-bt-sco-swb");
backend_tag_table[SND_DEVICE_IN_VOICE_TTY_FULL_USB_MIC] = strdup("usb-headset-mic");
backend_tag_table[SND_DEVICE_IN_VOICE_TTY_HCO_USB_MIC] = strdup("usb-headset-mic");
backend_tag_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
@@ -1987,6 +2010,7 @@
hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT] = strdup("SLIMBUS_0_RX-and-DISPLAY_PORT");
hw_interface_table[SND_DEVICE_OUT_BT_SCO] = strdup("SLIMBUS_7_RX");
hw_interface_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("SLIMBUS_7_RX");
+ hw_interface_table[SND_DEVICE_OUT_BT_SCO_SWB] = strdup("SLIMBUS_7_RX");
hw_interface_table[SND_DEVICE_OUT_BT_A2DP] = strdup("SLIMBUS_7_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("SLIMBUS_0_RX-and-SLIMBUS_7_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP] =
@@ -2020,8 +2044,10 @@
hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_HFP] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_BT_SCO] = strdup("SLIMBUS_0_RX-and-SEC_AUX_PCM_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB] = strdup("SLIMBUS_0_RX-and-SEC_AUX_PCM_RX");
+ hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB] = strdup("SLIMBUS_0_RX-and-SEC_AUX_PCM_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO] = strdup("QUAT_TDM_RX_0-and-SLIMBUS_7_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB] = strdup("QUAT_TDM_RX_0-and-SLIMBUS_7_RX");
+ hw_interface_table[SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB] = strdup("QUAT_TDM_RX_0-and-SLIMBUS_7_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_VOICE_SPEAKER_STEREO_PROTECTED] = strdup("SLIMBUS_0_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = strdup("SLIMBUS_0_RX");
@@ -2076,6 +2102,8 @@
hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_NREC] = strdup("SLIMBUS_7_TX");
hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_WB] = strdup("SLIMBUS_7_TX");
hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_WB_NREC] = strdup("SLIMBUS_7_TX");
+ hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_SWB] = strdup("SLIMBUS_7_TX");
+ hw_interface_table[SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC] = strdup("SLIMBUS_7_TX");
hw_interface_table[SND_DEVICE_IN_BT_A2DP] = strdup("SLIMBUS_7_TX");
hw_interface_table[SND_DEVICE_IN_CAMCORDER_MIC] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_VOICE_DMIC] = strdup("SLIMBUS_0_TX");
@@ -2987,6 +3015,9 @@
else if (!strncmp(snd_card_name, "sm6150-qrd-snd-card",
sizeof("sm6150-qrd-snd-card")))
platform_info_init(PLATFORM_INFO_XML_PATH_QRD, my_data, PLATFORM);
+ else if (!strncmp(snd_card_name, "sm6150-wcd9375qrd-snd-card",
+ sizeof("sm6150-wcd9375qrd-snd-card")))
+ platform_info_init(PLATFORM_INFO_XML_PATH_QRD, my_data, PLATFORM);
else if (!strncmp(snd_card_name, "kona-qrd-snd-card",
sizeof("kona-qrd-snd-card")))
platform_info_init(PLATFORM_INFO_XML_PATH_QRD, my_data, PLATFORM);
@@ -4940,6 +4971,19 @@
new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_SAFE;
new_snd_devices[1] = SND_DEVICE_OUT_BT_SCO_WB;
ret = 0;
+ } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB &&
+ !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_BT_SCO_SWB)) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
+ new_snd_devices[1] = SND_DEVICE_OUT_BT_SCO_SWB;
+ ret = 0;
+ } else if (snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB &&
+ !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER_SAFE,
+ SND_DEVICE_OUT_BT_SCO_SWB)) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_SAFE;
+ new_snd_devices[1] = SND_DEVICE_OUT_BT_SCO_SWB;
+ ret = 0;
} else if (snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_USB_HEADSET &&
!platform_check_backends_match(SND_DEVICE_OUT_SPEAKER_SAFE, SND_DEVICE_OUT_USB_HEADSET)) {
*num_devices = 2;
@@ -5131,19 +5175,29 @@
snd_device = SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP;
} else if ((devices & AUDIO_DEVICE_OUT_ALL_SCO) &&
((devices & ~AUDIO_DEVICE_OUT_ALL_SCO) == AUDIO_DEVICE_OUT_SPEAKER)) {
- if (my_data->is_wsa_speaker)
- snd_device = adev->bt_wb_speech_enabled ?
- SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB :
- SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO;
- else
- snd_device = adev->bt_wb_speech_enabled ?
- SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB :
- SND_DEVICE_OUT_SPEAKER_AND_BT_SCO;
+ if (my_data->is_wsa_speaker) {
+ if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+ snd_device = SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB;
+ else
+ snd_device = adev->bt_wb_speech_enabled ?
+ SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB :
+ SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO;
+ } else {
+ if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+ snd_device = SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB;
+ else
+ snd_device = adev->bt_wb_speech_enabled ?
+ SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB :
+ SND_DEVICE_OUT_SPEAKER_AND_BT_SCO;
+ }
} else if ((devices & AUDIO_DEVICE_OUT_ALL_SCO) &&
((devices & ~AUDIO_DEVICE_OUT_ALL_SCO) == AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
- snd_device = adev->bt_wb_speech_enabled ?
- SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB :
- SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO;
+ if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+ snd_device = SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB;
+ else
+ snd_device = adev->bt_wb_speech_enabled ?
+ SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB :
+ SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO;
} else if ((devices == (AUDIO_DEVICE_OUT_USB_DEVICE |
AUDIO_DEVICE_OUT_SPEAKER_SAFE)) ||
(devices == (AUDIO_DEVICE_OUT_USB_HEADSET |
@@ -5225,7 +5279,9 @@
SND_DEVICE_OUT_VOICE_USB_HEADPHONES;
}
} else if (devices & AUDIO_DEVICE_OUT_ALL_SCO) {
- if (adev->bt_wb_speech_enabled)
+ if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+ snd_device = SND_DEVICE_OUT_BT_SCO_SWB;
+ else if (adev->bt_wb_speech_enabled)
snd_device = SND_DEVICE_OUT_BT_SCO_WB;
else
snd_device = SND_DEVICE_OUT_BT_SCO;
@@ -5351,7 +5407,9 @@
else
snd_device = SND_DEVICE_OUT_SPEAKER;
} else if (devices & AUDIO_DEVICE_OUT_ALL_SCO) {
- if (adev->bt_wb_speech_enabled)
+ if (adev->swb_speech_mode != SPEECH_MODE_INVALID)
+ snd_device = SND_DEVICE_OUT_BT_SCO_SWB;
+ else if (adev->bt_wb_speech_enabled)
snd_device = SND_DEVICE_OUT_BT_SCO_WB;
else
snd_device = SND_DEVICE_OUT_BT_SCO;
@@ -5740,7 +5798,12 @@
if (audio_extn_hfp_is_active(adev))
platform_set_echo_reference(adev, true, out_device);
} else if (out_device & AUDIO_DEVICE_OUT_ALL_SCO) {
- if (adev->bt_wb_speech_enabled) {
+ if (adev->swb_speech_mode != SPEECH_MODE_INVALID) {
+ if (adev->bluetooth_nrec)
+ snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC;
+ else
+ snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB;
+ } else if (adev->bt_wb_speech_enabled) {
if (adev->bluetooth_nrec)
snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB_NREC;
else
@@ -6050,7 +6113,12 @@
} else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_HEADSET_MIC;
} else if (in_device & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- if (adev->bt_wb_speech_enabled) {
+ if (adev->swb_speech_mode != SPEECH_MODE_INVALID) {
+ if (adev->bluetooth_nrec)
+ snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC;
+ else
+ snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB;
+ } else if (adev->bt_wb_speech_enabled) {
if (adev->bluetooth_nrec)
snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB_NREC;
else
@@ -6113,7 +6181,12 @@
snd_device = my_data->fluence_sb_enabled ? SND_DEVICE_IN_HANDSET_MIC_SB
: SND_DEVICE_IN_HANDSET_MIC;
} else if (out_device & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET) {
- if (adev->bt_wb_speech_enabled) {
+ if (adev->swb_speech_mode != SPEECH_MODE_INVALID) {
+ if (adev->bluetooth_nrec)
+ snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC;
+ else
+ snd_device = SND_DEVICE_IN_BT_SCO_MIC_SWB;
+ } else if (adev->bt_wb_speech_enabled) {
if (adev->bluetooth_nrec)
snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB_NREC;
else
@@ -7389,7 +7462,6 @@
int ret = -EINVAL;
int backend_idx = platform_get_backend_index(snd_device);
struct platform_data *my_data = (struct platform_data *)adev->platform;
- backend_idx = platform_get_backend_index(snd_device);
unsigned int bit_width = backend_cfg.bit_width;
unsigned int sample_rate = backend_cfg.sample_rate;
unsigned int channels = backend_cfg.channels;
@@ -7611,7 +7683,6 @@
ALOGV("%s: Format doesnt have to be set", __func__);
}
- format = format & AUDIO_FORMAT_MAIN_MASK;
/* Set data format only if there is a change from PCM to compressed
and vice versa */
if (set_mi2s_tx_data_format && (format ^ my_data->current_backend_cfg[backend_idx].format)) {
@@ -7621,7 +7692,7 @@
__func__, ext_disp_format);
return -EINVAL;
}
- if (format == AUDIO_FORMAT_PCM) {
+ if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
ALOGE("%s:Set %s to LPCM", __func__, ext_disp_format);
mixer_ctl_set_enum_by_string(ctl, "LPCM");
} else {
@@ -7790,6 +7861,7 @@
if (snd_device == SND_DEVICE_OUT_BT_A2DP ||
snd_device == SND_DEVICE_OUT_BT_SCO ||
snd_device == SND_DEVICE_OUT_BT_SCO_WB ||
+ snd_device == SND_DEVICE_OUT_BT_SCO_SWB ||
snd_device == SND_DEVICE_IN_BT_A2DP ||
snd_device == SND_DEVICE_OUT_AFE_PROXY) {
backend_change = false;
@@ -8193,18 +8265,25 @@
__func__, bit_width, sample_rate, channels);
}
- ALOGI("%s:txbecf: afe: Codec selected backend: %d updated bit width: %d and "
- "sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
+ ALOGI("%s:txbecf: afe: current backend bit_width %d sample_rate %d channels %d, format %x",
+ __func__,
+ my_data->current_backend_cfg[backend_idx].bit_width,
+ my_data->current_backend_cfg[backend_idx].sample_rate,
+ my_data->current_backend_cfg[backend_idx].channels,
+ my_data->current_backend_cfg[backend_idx].format);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
+ // Note that below if block would be entered even if main format is same
+ // but subformat is different for e.g. current backend is configured for 16 bit PCM
+ // as compared to 24 bit PCM backend requested
if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
(sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
(channels != my_data->current_backend_cfg[backend_idx].channels) ||
- ((format & AUDIO_FORMAT_MAIN_MASK) != my_data->current_backend_cfg[backend_idx].format)) {
+ (format != my_data->current_backend_cfg[backend_idx].format)) {
backend_cfg->bit_width = bit_width;
backend_cfg->sample_rate= sample_rate;
backend_cfg->channels = channels;
- backend_cfg->format = format & AUDIO_FORMAT_MAIN_MASK;
+ backend_cfg->format = format;
backend_change = true;
ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
"new sample rate: %d new channel: %d new format: %d",
@@ -9571,6 +9650,9 @@
ret = -EINVAL;
goto ERROR_RETURN;
}
+ if ((cal->acdb_dev_id == ACDB_ID_STEREO_SPEAKER_DEVICE) &&
+ (cal->topo_id == TRUMPET_TOPOLOGY))
+ audio_extn_ip_hdlr_copp_update_cal_info((void*)cal, data);
if (my_data->acdb_set_audio_cal) {
// persist audio cal in local cache
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 2cb1741..6203cf8 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -21,6 +21,9 @@
#define QCOM_AUDIO_PLATFORM_H
#include <sound/voice_params.h>
+#define TRUMPET_TOPOLOGY 0x11000099
+#define ACDB_ID_STEREO_SPEAKER_DEVICE 15
+
enum {
FLUENCE_NONE,
FLUENCE_DUAL_MIC = 0x1,
@@ -110,6 +113,7 @@
SND_DEVICE_OUT_SPEAKER_AND_DISPLAY_PORT,
SND_DEVICE_OUT_BT_SCO,
SND_DEVICE_OUT_BT_SCO_WB,
+ SND_DEVICE_OUT_BT_SCO_SWB,
SND_DEVICE_OUT_BT_A2DP,
SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP,
SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP,
@@ -118,8 +122,11 @@
SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO,
SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_WB,
SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_WB,
+ SND_DEVICE_OUT_SPEAKER_AND_BT_SCO_SWB,
+ SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_SCO_SWB,
SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO,
SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_WB,
+ SND_DEVICE_OUT_SPEAKER_WSA_AND_BT_SCO_SWB,
SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
@@ -219,6 +226,8 @@
SND_DEVICE_IN_BT_SCO_MIC_NREC,
SND_DEVICE_IN_BT_SCO_MIC_WB,
SND_DEVICE_IN_BT_SCO_MIC_WB_NREC,
+ SND_DEVICE_IN_BT_SCO_MIC_SWB,
+ SND_DEVICE_IN_BT_SCO_MIC_SWB_NREC,
SND_DEVICE_IN_BT_A2DP,
SND_DEVICE_IN_CAMCORDER_MIC,
SND_DEVICE_IN_VOICE_DMIC,