Merge commit 'd34f05cb6e67e28a4955332173950466edf0be3e' into audio-hal.lnx.5.1
Change-Id: I4656a1a78c2b1d4b4834b2fd538699f96e2aa90e
diff --git a/hal/Makefile.am b/hal/Makefile.am
index 995622d..aafc3e9 100644
--- a/hal/Makefile.am
+++ b/hal/Makefile.am
@@ -187,6 +187,7 @@
if AUDIO_HW_LOOPBACK
AM_CFLAGS += -DAUDIO_HW_LOOPBACK_ENABLED
+AM_CFLAGS += -DCOMPRESS_METADATA_NEEDED
c_sources += audio_extn/hw_loopback.c
endif
@@ -208,6 +209,10 @@
AM_CFLAGS += -DINSTANCE_ID_ENABLED
endif
+if LL_AS_PRIMARY_OUTPUT
+AM_CFLAGS += -DUSE_LL_AS_PRIMARY_OUTPUT
+endif
+
h_sources = audio_extn/audio_defs.h \
audio_extn/audio_extn.h \
audio_hw.h \
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
index c5edd42..e986a54 100644
--- a/hal/audio_extn/a2dp.c
+++ b/hal/audio_extn/a2dp.c
@@ -50,15 +50,18 @@
#ifdef SPLIT_A2DP_ENABLED
#define AUDIO_PARAMETER_A2DP_STARTED "A2dpStarted"
-#define BT_IPC_LIB_NAME "libbthost_if.so"
-#define ENC_MEDIA_FMT_NONE 0
-#define ENC_MEDIA_FMT_AAC 0x00010DA6
-#define ENC_MEDIA_FMT_APTX 0x000131ff
-#define ENC_MEDIA_FMT_APTX_HD 0x00013200
-#define ENC_MEDIA_FMT_APTX_AD 0x00013204
-#define ENC_MEDIA_FMT_SBC 0x00010BF2
-#define ENC_MEDIA_FMT_CELT 0x00013221
-#define ENC_MEDIA_FMT_LDAC 0x00013224
+#define BT_IPC_SOURCE_LIB_NAME "libbthost_if.so"
+#define BT_IPC_SINK_LIB_NAME "libbthost_if_sink.so"
+#define MEDIA_FMT_NONE 0
+#define MEDIA_FMT_AAC 0x00010DA6
+#define MEDIA_FMT_APTX 0x000131ff
+#define MEDIA_FMT_APTX_HD 0x00013200
+#define MEDIA_FMT_APTX_AD 0x00013204
+#define MEDIA_FMT_SBC 0x00010BF2
+#define MEDIA_FMT_CELT 0x00013221
+#define MEDIA_FMT_LDAC 0x00013224
+#define MEDIA_FMT_MP3 0x00010BE9
+#define MEDIA_FMT_APTX_ADAPTIVE 0x00013204
#define MEDIA_FMT_AAC_AOT_LC 2
#define MEDIA_FMT_AAC_AOT_SBR 5
#define MEDIA_FMT_AAC_AOT_PS 29
@@ -71,16 +74,20 @@
#define MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO 9
#define MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS 0
#define MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR 1
-#define MIXER_ENC_CONFIG_BLOCK "SLIM_7_RX Encoder Config"
-#define MIXER_DEC_CONFIG_BLOCK "SLIM_7_TX Decoder Config"
+#define MIXER_ENC_CONFIG_BLOCK "SLIM_7_RX Encoder Config"
+#define MIXER_SOURCE_DEC_CONFIG_BLOCK "SLIM_7_TX Decoder Config"
+#define MIXER_SINK_DEC_CONFIG_BLOCK "SLIM_9_TX Decoder Config"
#define MIXER_ENC_BIT_FORMAT "AFE Input Bit Format"
+#define MIXER_DEC_BIT_FORMAT "AFE Output Bit Format"
#define MIXER_SCRAMBLER_MODE "AFE Scrambler Mode"
#define MIXER_SAMPLE_RATE_RX "BT SampleRate RX"
-#define MIXER_SAMPLE_RATE_TX "BT SampleRate TX"
+#define MIXER_SOURCE_SAMPLE_RATE_TX "BT SampleRate TX"
#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_SET_FEEDBACK_CHANNEL "BT set feedback channel"
+#define MIXER_SINK_SAMPLE_RATE "BT_TX SampleRate"
+#define MIXER_AFE_SINK_CHANNELS "AFE Output Channels"
#define MIXER_ENC_FMT_SBC "SBC"
#define MIXER_ENC_FMT_AAC "AAC"
#define MIXER_ENC_FMT_APTX "APTX"
@@ -118,25 +125,28 @@
#define SAMPLING_RATE_441K 44100
#define CH_STEREO 2
#define CH_MONO 1
+#define SOURCE 0
+#define SINK 1
+
/*
* Below enum values are extended from audio_base.h to
- * to keep encoder codec type local to bthost_ipc
+ * to keep encoder and decoder type local to bthost_ipc
* and audio_hal as these are intended only for handshake
* between IPC lib and Audio HAL.
*/
typedef enum {
- ENC_CODEC_TYPE_INVALID = AUDIO_FORMAT_INVALID, // 0xFFFFFFFFUL
- ENC_CODEC_TYPE_AAC = AUDIO_FORMAT_AAC, // 0x04000000UL
- ENC_CODEC_TYPE_SBC = AUDIO_FORMAT_SBC, // 0x1F000000UL
- ENC_CODEC_TYPE_APTX = AUDIO_FORMAT_APTX, // 0x20000000UL
- ENC_CODEC_TYPE_APTX_HD = AUDIO_FORMAT_APTX_HD, // 0x21000000UL
+ CODEC_TYPE_INVALID = AUDIO_FORMAT_INVALID, // 0xFFFFFFFFUL
+ CODEC_TYPE_AAC = AUDIO_FORMAT_AAC, // 0x04000000UL
+ CODEC_TYPE_SBC = AUDIO_FORMAT_SBC, // 0x1F000000UL
+ CODEC_TYPE_APTX = AUDIO_FORMAT_APTX, // 0x20000000UL
+ CODEC_TYPE_APTX_HD = AUDIO_FORMAT_APTX_HD, // 0x21000000UL
#ifndef LINUX_ENABLED
- ENC_CODEC_TYPE_APTX_DUAL_MONO = 570425344u, // 0x22000000UL
+ CODEC_TYPE_APTX_DUAL_MONO = 570425344u, // 0x22000000UL
#endif
- ENC_CODEC_TYPE_LDAC = AUDIO_FORMAT_LDAC, // 0x23000000UL
- ENC_CODEC_TYPE_CELT = 603979776u, // 0x24000000UL
- ENC_CODEC_TYPE_APTX_AD = 620756992u, // 0x25000000UL
-}enc_codec_t;
+ CODEC_TYPE_LDAC = AUDIO_FORMAT_LDAC, // 0x23000000UL
+ CODEC_TYPE_CELT = 603979776u, // 0x24000000UL
+ CODEC_TYPE_APTX_AD = 620756992u, // 0x25000000UL
+}codec_t;
/*
* enums which describes the APTX Adaptive
@@ -162,19 +172,24 @@
APTX_AD_44_1 = 0x2, // 44.1kHz
} enc_aptx_ad_s_rate;
-typedef int (*audio_stream_open_t)(void);
-typedef int (*audio_stream_close_t)(void);
-typedef int (*audio_start_stream_t)(void);
-typedef int (*audio_stop_stream_t)(void);
-typedef int (*audio_suspend_stream_t)(void);
-typedef void (*audio_handoff_triggered_t)(void);
-typedef void (*clear_a2dpsuspend_flag_t)(void);
-typedef void * (*audio_get_codec_config_t)(uint8_t *multicast_status,uint8_t *num_dev,
- enc_codec_t *codec_type);
-typedef int (*audio_check_a2dp_ready_t)(void);
-typedef uint16_t (*audio_get_a2dp_sink_latency_t)(void);
-typedef int (*audio_is_scrambling_enabled_t)(void);
+typedef int (*audio_source_open_t)(void);
+typedef int (*audio_source_close_t)(void);
+typedef int (*audio_source_start_t)(void);
+typedef int (*audio_source_stop_t)(void);
+typedef int (*audio_source_suspend_t)(void);
+typedef void (*audio_source_handoff_triggered_t)(void);
+typedef void (*clear_source_a2dpsuspend_flag_t)(void);
+typedef void * (*audio_get_enc_config_t)(uint8_t *multicast_status,
+ uint8_t *num_dev, codec_t *codec_type);
+typedef int (*audio_source_check_a2dp_ready_t)(void);
+typedef int (*audio_is_source_scrambling_enabled_t)(void);
typedef bool (*audio_is_tws_mono_mode_enable_t)(void);
+typedef int (*audio_sink_start_t)(void);
+typedef int (*audio_sink_stop_t)(void);
+typedef void * (*audio_get_dec_config_t)(codec_t *codec_type);
+typedef void * (*audio_sink_session_setup_complete_t)(uint64_t system_latency);
+typedef int (*audio_sink_check_a2dp_ready_t)(void);
+typedef uint16_t (*audio_sink_get_a2dp_latency_t)(void);
enum A2DP_STATE {
A2DP_STATE_CONNECTED,
@@ -230,26 +245,25 @@
*/
struct a2dp_data {
struct audio_device *adev;
- void *bt_lib_handle;
- audio_stream_open_t audio_stream_open;
- audio_stream_close_t audio_stream_close;
- audio_start_stream_t audio_start_stream;
- audio_stop_stream_t audio_stop_stream;
- audio_suspend_stream_t audio_suspend_stream;
- audio_handoff_triggered_t audio_handoff_triggered;
- clear_a2dpsuspend_flag_t clear_a2dpsuspend_flag;
- audio_get_codec_config_t audio_get_codec_config;
- audio_check_a2dp_ready_t audio_check_a2dp_ready;
- audio_get_a2dp_sink_latency_t audio_get_a2dp_sink_latency;
- audio_is_scrambling_enabled_t audio_is_scrambling_enabled;
+ void *bt_lib_source_handle;
+ audio_source_open_t audio_source_open;
+ audio_source_close_t audio_source_close;
+ audio_source_start_t audio_source_start;
+ audio_source_stop_t audio_source_stop;
+ audio_source_suspend_t audio_source_suspend;
+ audio_source_handoff_triggered_t audio_source_handoff_triggered;
+ clear_source_a2dpsuspend_flag_t clear_source_a2dpsuspend_flag;
+ audio_get_enc_config_t audio_get_enc_config;
+ audio_source_check_a2dp_ready_t audio_source_check_a2dp_ready;
audio_is_tws_mono_mode_enable_t audio_is_tws_mono_mode_enable;
- enum A2DP_STATE bt_state;
- enc_codec_t bt_encoder_format;
+ audio_is_source_scrambling_enabled_t audio_is_source_scrambling_enabled;
+ enum A2DP_STATE bt_state_source;
+ codec_t bt_encoder_format;
uint32_t enc_sampling_rate;
uint32_t enc_channels;
- bool a2dp_started;
- bool a2dp_suspended;
- int a2dp_total_active_session_request;
+ bool a2dp_source_started;
+ bool a2dp_source_suspended;
+ int a2dp_source_total_active_session_requests;
bool is_a2dp_offload_supported;
bool is_handoff_in_progress;
bool is_aptx_dual_mono_supported;
@@ -258,6 +272,20 @@
bool is_aptx_adaptive;
/* Adaptive bitrate config for A2DP codecs */
struct a2dp_abr_config abr_config;
+
+ void *bt_lib_sink_handle;
+ audio_sink_start_t audio_sink_start;
+ audio_sink_stop_t audio_sink_stop;
+ audio_get_dec_config_t audio_get_dec_config;
+ audio_sink_session_setup_complete_t audio_sink_session_setup_complete;
+ audio_sink_check_a2dp_ready_t audio_sink_check_a2dp_ready;
+ audio_sink_get_a2dp_latency_t audio_sink_get_a2dp_latency;
+ enum A2DP_STATE bt_state_sink;
+ codec_t bt_decoder_format;
+ uint32_t dec_sampling_rate;
+ uint32_t dec_channels;
+ bool a2dp_sink_started;
+ int a2dp_sink_total_active_session_requests;
};
struct a2dp_data a2dp;
@@ -364,6 +392,37 @@
struct aac_frame_size_control_t frame_ctl;
} __attribute__ ((packed));
+typedef struct audio_aac_decoder_config_t audio_aac_decoder_config_t;
+struct audio_aac_decoder_config_t {
+ uint16_t aac_fmt_flag; /* LATM*/
+ uint16_t audio_object_type; /* LC */
+ uint16_t channels; /* Stereo */
+ uint16_t total_size_of_pce_bits; /* 0 - only for channel conf PCE */
+ uint32_t sampling_rate; /* 8k, 11.025k, 12k, 16k, 22.05k, 24k, 32k,
+ 44.1k, 48k, 64k, 88.2k, 96k */
+} __attribute__ ((packed));
+
+typedef struct audio_sbc_decoder_config_t audio_sbc_decoder_config_t;
+struct audio_sbc_decoder_config_t {
+ uint16_t channels; /* Mono, Stereo */
+ uint32_t sampling_rate; /* 8k, 11.025k, 12k, 16k, 22.05k, 24k, 32k,
+ 44.1k, 48k, 64k, 88.2k, 96k */
+} __attribute__ ((packed));
+
+/* AAC decoder configuration structure. */
+typedef struct aac_dec_cfg_t aac_dec_cfg_t;
+struct aac_dec_cfg_t {
+ uint32_t dec_format;
+ audio_aac_decoder_config_t data;
+} __attribute__ ((packed));
+
+/* SBC decoder configuration structure. */
+typedef struct sbc_dec_cfg_t sbc_dec_cfg_t;
+struct sbc_dec_cfg_t {
+ uint32_t dec_format;
+ audio_sbc_decoder_config_t data;
+} __attribute__ ((packed));
+
/* SBC encoder configuration structure. */
typedef struct sbc_enc_cfg_t sbc_enc_cfg_t;
@@ -581,6 +640,29 @@
uint32_t bits_per_sample;
} audio_ldac_encoder_config;
+/* Information about BT AAC decoder configuration
+ * This data is used between audio HAL module and
+ * BT IPC library to configure DSP decoder
+ */
+typedef struct {
+ uint16_t aac_fmt_flag; /* LATM*/
+ uint16_t audio_object_type; /* LC */
+ uint16_t channels; /* Stereo */
+ uint16_t total_size_of_pce_bits; /* 0 - only for channel conf PCE */
+ uint32_t sampling_rate; /* 8k, 11.025k, 12k, 16k, 22.05k, 24k, 32k,
+ 44.1k, 48k, 64k, 88.2k, 96k */
+} audio_aac_dec_config_t;
+
+/* Information about BT SBC decoder configuration
+ * This data is used between audio HAL module and
+ * BT IPC library to configure DSP decoder
+ */
+typedef struct {
+ uint16_t channels; /* Mono, Stereo */
+ uint32_t sampling_rate; /* 8k, 11.025k, 12k, 16k, 22.05k, 24k, 32k,
+ 44.1k, 48k, 64k, 88.2k, 96k */
+}audio_sbc_dec_config_t;
+
/*********** END of DSP configurable structures ********************/
/* API to identify DSP encoder captabilities */
@@ -750,98 +832,150 @@
return -ENOSYS;
}
-/* API to open BT IPC library to start IPC communication */
-static void open_a2dp_output()
+/* API to open BT IPC library to start IPC communication for BT Source*/
+static void open_a2dp_source()
{
int ret = 0;
- ALOGD(" Open A2DP output start ");
- if (a2dp.bt_lib_handle == NULL){
+ ALOGD(" Open A2DP source start ");
+ if (a2dp.bt_lib_source_handle == NULL){
ALOGD(" Requesting for BT lib handle");
- a2dp.bt_lib_handle = dlopen(BT_IPC_LIB_NAME, RTLD_NOW);
+ a2dp.bt_lib_source_handle = dlopen(BT_IPC_SOURCE_LIB_NAME, RTLD_NOW);
- if (a2dp.bt_lib_handle == NULL) {
- ALOGE("%s: DLOPEN failed for %s", __func__, BT_IPC_LIB_NAME);
+ if (a2dp.bt_lib_source_handle == NULL) {
+ ALOGE("%s: DLOPEN failed for %s", __func__, BT_IPC_SOURCE_LIB_NAME);
ret = -ENOSYS;
goto init_fail;
} else {
- a2dp.audio_stream_open = (audio_stream_open_t)
- dlsym(a2dp.bt_lib_handle, "audio_stream_open");
- a2dp.audio_start_stream = (audio_start_stream_t)
- dlsym(a2dp.bt_lib_handle, "audio_start_stream");
- a2dp.audio_get_codec_config = (audio_get_codec_config_t)
- dlsym(a2dp.bt_lib_handle, "audio_get_codec_config");
- a2dp.audio_suspend_stream = (audio_suspend_stream_t)
- dlsym(a2dp.bt_lib_handle, "audio_suspend_stream");
- a2dp.audio_handoff_triggered = (audio_handoff_triggered_t)
- dlsym(a2dp.bt_lib_handle, "audio_handoff_triggered");
- a2dp.clear_a2dpsuspend_flag = (clear_a2dpsuspend_flag_t)
- dlsym(a2dp.bt_lib_handle, "clear_a2dpsuspend_flag");
- a2dp.audio_stop_stream = (audio_stop_stream_t)
- dlsym(a2dp.bt_lib_handle, "audio_stop_stream");
- a2dp.audio_stream_close = (audio_stream_close_t)
- dlsym(a2dp.bt_lib_handle, "audio_stream_close");
- a2dp.audio_check_a2dp_ready = (audio_check_a2dp_ready_t)
- dlsym(a2dp.bt_lib_handle,"audio_check_a2dp_ready");
- a2dp.audio_get_a2dp_sink_latency = (audio_get_a2dp_sink_latency_t)
- dlsym(a2dp.bt_lib_handle,"audio_get_a2dp_sink_latency");
- a2dp.audio_is_scrambling_enabled = (audio_is_scrambling_enabled_t)
- dlsym(a2dp.bt_lib_handle,"audio_is_scrambling_enabled");
+ a2dp.audio_source_open = (audio_source_open_t)
+ dlsym(a2dp.bt_lib_source_handle, "audio_stream_open");
+ a2dp.audio_source_start = (audio_source_start_t)
+ dlsym(a2dp.bt_lib_source_handle, "audio_start_stream");
+ a2dp.audio_get_enc_config = (audio_get_enc_config_t)
+ dlsym(a2dp.bt_lib_source_handle, "audio_get_codec_config");
+ a2dp.audio_source_suspend = (audio_source_suspend_t)
+ dlsym(a2dp.bt_lib_source_handle, "audio_suspend_stream");
+ a2dp.audio_source_handoff_triggered = (audio_source_handoff_triggered_t)
+ dlsym(a2dp.bt_lib_source_handle, "audio_handoff_triggered");
+ a2dp.clear_source_a2dpsuspend_flag = (clear_source_a2dpsuspend_flag_t)
+ dlsym(a2dp.bt_lib_source_handle, "clear_a2dpsuspend_flag");
+ a2dp.audio_source_stop = (audio_source_stop_t)
+ dlsym(a2dp.bt_lib_source_handle, "audio_stop_stream");
+ a2dp.audio_source_close = (audio_source_close_t)
+ dlsym(a2dp.bt_lib_source_handle, "audio_stream_close");
+ a2dp.audio_source_check_a2dp_ready = (audio_source_check_a2dp_ready_t)
+ dlsym(a2dp.bt_lib_source_handle,"audio_check_a2dp_ready");
+ a2dp.audio_sink_get_a2dp_latency = (audio_sink_get_a2dp_latency_t)
+ dlsym(a2dp.bt_lib_source_handle,"audio_sink_get_a2dp_latency");
+ a2dp.audio_is_source_scrambling_enabled = (audio_is_source_scrambling_enabled_t)
+ dlsym(a2dp.bt_lib_source_handle,"audio_is_scrambling_enabled");
a2dp.audio_is_tws_mono_mode_enable = (audio_is_tws_mono_mode_enable_t)
- dlsym(a2dp.bt_lib_handle,"isTwsMonomodeEnable");
+ dlsym(a2dp.bt_lib_source_handle,"isTwsMonomodeEnable");
}
}
- if (a2dp.bt_lib_handle && a2dp.audio_stream_open) {
- if (a2dp.bt_state == A2DP_STATE_DISCONNECTED) {
+ if (a2dp.bt_lib_source_handle && a2dp.audio_source_open) {
+ if (a2dp.bt_state_source == A2DP_STATE_DISCONNECTED) {
ALOGD("calling BT stream open");
- ret = a2dp.audio_stream_open();
+ ret = a2dp.audio_source_open();
if(ret != 0) {
- ALOGE("Failed to open output stream for a2dp: status %d", ret);
+ ALOGE("Failed to open source stream for a2dp: status %d", ret);
goto init_fail;
}
- a2dp.bt_state = A2DP_STATE_CONNECTED;
+ a2dp.bt_state_source = A2DP_STATE_CONNECTED;
} else {
- ALOGD("Called a2dp open with improper state, Ignoring request state %d", a2dp.bt_state);
+ ALOGD("Called a2dp open with improper state, Ignoring request state %d", a2dp.bt_state_source);
}
} else {
ALOGE("a2dp handle is not identified, Ignoring open request");
- a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+ a2dp.bt_state_source = A2DP_STATE_DISCONNECTED;
goto init_fail;
}
init_fail:
- if(ret != 0 && (a2dp.bt_lib_handle != NULL)) {
- dlclose(a2dp.bt_lib_handle);
- a2dp.bt_lib_handle = NULL;
+ if(ret != 0 && (a2dp.bt_lib_source_handle != NULL)) {
+ dlclose(a2dp.bt_lib_source_handle);
+ a2dp.bt_lib_source_handle = NULL;
+ }
+}
+
+/* API to open BT IPC library to start IPC communication for BT Sink*/
+static void open_a2dp_sink()
+{
+ ALOGD(" Open A2DP input start ");
+ if (a2dp.bt_lib_sink_handle == NULL){
+ ALOGD(" Requesting for BT lib handle");
+ a2dp.bt_lib_sink_handle = dlopen(BT_IPC_SINK_LIB_NAME, RTLD_NOW);
+
+ if (a2dp.bt_lib_sink_handle == NULL) {
+ ALOGE("%s: DLOPEN failed for %s", __func__, BT_IPC_SINK_LIB_NAME);
+ } else {
+ a2dp.audio_sink_start = (audio_sink_start_t)
+ dlsym(a2dp.bt_lib_sink_handle, "audio_sink_start_capture");
+ a2dp.audio_get_dec_config = (audio_get_dec_config_t)
+ dlsym(a2dp.bt_lib_sink_handle, "audio_get_decoder_config");
+ a2dp.audio_sink_stop = (audio_sink_stop_t)
+ dlsym(a2dp.bt_lib_sink_handle, "audio_sink_stop_capture");
+ a2dp.audio_sink_check_a2dp_ready = (audio_sink_check_a2dp_ready_t)
+ dlsym(a2dp.bt_lib_sink_handle,"audio_sink_check_a2dp_ready");
+ a2dp.audio_sink_session_setup_complete = (audio_sink_session_setup_complete_t)
+ dlsym(a2dp.bt_lib_sink_handle, "audio_sink_session_setup_complete");
+ }
}
}
static int close_a2dp_output()
{
ALOGV("%s\n",__func__);
- if (!(a2dp.bt_lib_handle && a2dp.audio_stream_close)) {
- ALOGE("a2dp handle is not identified, Ignoring close request");
+
+ if (!(a2dp.bt_lib_source_handle && a2dp.audio_source_close)) {
+ ALOGE("a2dp source handle is not identified, Ignoring close request");
return -ENOSYS;
}
- if (a2dp.bt_state != A2DP_STATE_DISCONNECTED) {
- ALOGD("calling BT stream close");
- if(a2dp.audio_stream_close() == false)
- ALOGE("failed close a2dp control path from BT library");
+
+ if (a2dp.bt_state_source != A2DP_STATE_DISCONNECTED) {
+ ALOGD("calling BT source stream close");
+ if(a2dp.audio_source_close() == false)
+ ALOGE("failed close a2dp source control path from BT library");
}
- a2dp.a2dp_started = false;
- a2dp.a2dp_total_active_session_request = 0;
- a2dp.a2dp_suspended = false;
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_INVALID;
+ a2dp.a2dp_source_started = false;
+ a2dp.a2dp_source_total_active_session_requests = 0;
+ a2dp.a2dp_source_suspended = false;
+ a2dp.bt_encoder_format = CODEC_TYPE_INVALID;
a2dp.enc_sampling_rate = 48000;
a2dp.enc_channels = 2;
- a2dp.bt_state = A2DP_STATE_DISCONNECTED;
+ a2dp.bt_state_source = A2DP_STATE_DISCONNECTED;
if (a2dp.abr_config.is_abr_enabled && a2dp.abr_config.abr_started)
stop_abr();
a2dp.abr_config.is_abr_enabled = false;
a2dp.abr_config.abr_started = false;
a2dp.abr_config.imc_instance = 0;
a2dp.abr_config.abr_tx_handle = NULL;
+ a2dp.bt_state_source = A2DP_STATE_DISCONNECTED;
+
+ return 0;
+}
+
+static int close_a2dp_input()
+{
+ ALOGV("%s\n",__func__);
+
+ if (!(a2dp.bt_lib_sink_handle && a2dp.audio_source_close)) {
+ ALOGE("a2dp sink handle is not identified, Ignoring close request");
+ return -ENOSYS;
+ }
+
+ if (a2dp.bt_state_sink != A2DP_STATE_DISCONNECTED) {
+ ALOGD("calling BT sink stream close");
+ if(a2dp.audio_source_close() == false)
+ ALOGE("failed close a2dp sink control path from BT library");
+ }
+ a2dp.a2dp_sink_started = false;
+ a2dp.a2dp_sink_total_active_session_requests = 0;
+ a2dp.bt_decoder_format = CODEC_TYPE_INVALID;
+ a2dp.dec_sampling_rate = 48000;
+ a2dp.dec_channels = 2;
+ a2dp.bt_state_sink = A2DP_STATE_DISCONNECTED;
return 0;
}
@@ -850,15 +984,15 @@
{
bool scrambler_mode = false;
struct mixer_ctl *ctrl_scrambler_mode = NULL;
- if (a2dp.audio_is_scrambling_enabled && (a2dp.bt_state != A2DP_STATE_DISCONNECTED))
- scrambler_mode = a2dp.audio_is_scrambling_enabled();
+ if (a2dp.audio_is_source_scrambling_enabled && (a2dp.bt_state_source != A2DP_STATE_DISCONNECTED))
+ scrambler_mode = a2dp.audio_is_source_scrambling_enabled();
if (scrambler_mode) {
//enable scrambler in dsp
ctrl_scrambler_mode = mixer_get_ctl_by_name(a2dp.adev->mixer,
MIXER_SCRAMBLER_MODE);
if (!ctrl_scrambler_mode) {
- ALOGE(" ERROR scrambler mode mixer control not identifed");
+ ALOGE(" ERROR scrambler mode mixer control not identified");
return;
} else {
if (mixer_ctl_set_value(ctrl_scrambler_mode, 0, true) != 0) {
@@ -869,21 +1003,29 @@
}
}
-static int a2dp_set_backend_cfg()
+static bool a2dp_set_backend_cfg(uint8_t direction)
{
- char *rate_str = NULL, *in_channels = NULL;
- uint32_t sampling_rate_rx = a2dp.enc_sampling_rate;
- struct mixer_ctl *ctl_sample_rate = NULL, *ctrl_in_channels = NULL;
+ char *rate_str = NULL, *channels = NULL;
+ uint32_t sampling_rate;
+ struct mixer_ctl *ctl_sample_rate = NULL, *ctrl_channels = NULL;
+ bool is_configured = false;
- //For LDAC encoder open slimbus port at 96Khz for 48Khz input
- //and 88.2Khz for 44.1Khz input.
- if ((a2dp.bt_encoder_format == ENC_CODEC_TYPE_LDAC) &&
- (sampling_rate_rx == 48000 || sampling_rate_rx == 44100 )) {
- sampling_rate_rx *= 2;
+ if (direction == SINK) {
+ sampling_rate = a2dp.dec_sampling_rate;
+ } else {
+ sampling_rate = a2dp.enc_sampling_rate;
+ }
+ //For LDAC encoder and AAC decoder open slimbus port at
+ //96Khz for 48Khz input and 88.2Khz for 44.1Khz input.
+ if (((a2dp.bt_encoder_format == CODEC_TYPE_LDAC) ||
+ (a2dp.bt_decoder_format == CODEC_TYPE_SBC) ||
+ (a2dp.bt_decoder_format == AUDIO_FORMAT_AAC)) &&
+ (sampling_rate == 48000 || sampling_rate == 44100 )) {
+ sampling_rate = sampling_rate *2;
}
- // Set Rx backend sample rate
- switch (sampling_rate_rx) {
+ //Configure backend sampling rate
+ switch (sampling_rate) {
case 44100:
rate_str = "KHZ_44P1";
break;
@@ -899,31 +1041,42 @@
break;
}
- ALOGD("%s: set backend rx sample rate = %s", __func__, rate_str);
- ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
- MIXER_SAMPLE_RATE_RX);
+ if (direction == SINK) {
+ ALOGD("%s: set sink backend sample rate =%s", __func__, rate_str);
+ ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_SINK_SAMPLE_RATE);
+ } else {
+ ALOGD("%s: set source backend sample rate =%s", __func__, rate_str);
+ ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_SAMPLE_RATE_RX);
+ }
if (ctl_sample_rate) {
if (mixer_ctl_set_enum_by_string(ctl_sample_rate, rate_str) != 0) {
ALOGE("%s: Failed to set backend sample rate = %s", __func__, rate_str);
- return -ENOSYS;
+ is_configured = false;
+ goto fail;
}
- /* Set Tx backend sample rate */
- if (a2dp.abr_config.is_abr_enabled)
- rate_str = ABR_TX_SAMPLE_RATE;
+ if (direction == SOURCE) {
+ /* Set Tx backend sample rate */
+ if (a2dp.abr_config.is_abr_enabled)
+ 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_SAMPLE_RATE_TX);
- if (!ctl_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);
+ if (!ctl_sample_rate) {
ALOGE("%s: ERROR backend sample rate mixer control not identifed", __func__);
- return -ENOSYS;
- }
+ is_configured = false;
+ goto fail;
+ }
- if (mixer_ctl_set_enum_by_string(ctl_sample_rate, rate_str) != 0) {
- ALOGE("%s: Failed to set backend sample rate = %s", __func__, rate_str);
- return -ENOSYS;
+ if (mixer_ctl_set_enum_by_string(ctl_sample_rate, rate_str) != 0) {
+ ALOGE("%s: Failed to set backend sample rate = %s", __func__, rate_str);
+ is_configured = false;
+ goto fail;
+ }
}
} else {
/* Fallback to legacy approch if MIXER_SAMPLE_RATE_RX and
@@ -932,39 +1085,127 @@
MIXER_SAMPLE_RATE_DEFAULT);
if (!ctl_sample_rate) {
ALOGE("%s: ERROR backend sample rate mixer control not identifed", __func__);
- return -ENOSYS;
+ is_configured = false;
+ goto fail;
}
if (mixer_ctl_set_enum_by_string(ctl_sample_rate, rate_str) != 0) {
ALOGE("%s: Failed to set backend sample rate = %s", __func__, rate_str);
- return -ENOSYS;
+ is_configured = false;
+ goto fail;
}
}
- //Configure AFE input channels
- switch (a2dp.enc_channels) {
- case 1:
- in_channels = "One";
+ if (direction == SINK) {
+ switch (a2dp.dec_channels) {
+ case 1:
+ channels = "One";
+ break;
+ case 2:
+ default:
+ channels = "Two";
+ break;
+ }
+
+ ALOGD("%s: set afe dec channels =%d", __func__, channels);
+ ctrl_channels = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_AFE_SINK_CHANNELS);
+ } else {
+ //Configure AFE enc channels
+ switch (a2dp.enc_channels) {
+ case 1:
+ channels = "One";
+ break;
+ case 2:
+ default:
+ channels = "Two";
+ break;
+ }
+
+ ALOGD("%s: set afe enc channels =%d", __func__, channels);
+ ctrl_channels = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_AFE_IN_CHANNELS);
+ }
+
+ if (!ctrl_channels) {
+ ALOGE(" ERROR AFE channels mixer control not identified");
+ } else {
+ if (mixer_ctl_set_enum_by_string(ctrl_channels, channels) != 0) {
+ ALOGE("%s: Failed to set AFE channels =%d", __func__, channels);
+ is_configured = false;
+ goto fail;
+ }
+ }
+ is_configured = true;
+fail:
+ return is_configured;
+}
+
+bool configure_aac_dec_format(audio_aac_dec_config_t *aac_bt_cfg)
+{
+ struct mixer_ctl *ctl_dec_data = NULL, *ctrl_bit_format = NULL;
+ struct aac_dec_cfg_t aac_dsp_cfg;
+ bool is_configured = false;
+ int ret = 0;
+
+ if(aac_bt_cfg == NULL)
+ return false;
+
+ ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_SINK_DEC_CONFIG_BLOCK);
+ if (!ctl_dec_data) {
+ ALOGE(" ERROR a2dp decoder CONFIG data mixer control not identified");
+ is_configured = false;
+ goto fail;
+ }
+
+ memset(&aac_dsp_cfg, 0x0, sizeof(struct aac_dec_cfg_t));
+ aac_dsp_cfg.dec_format = MEDIA_FMT_AAC;
+ aac_dsp_cfg.data.aac_fmt_flag = aac_bt_cfg->aac_fmt_flag;
+ aac_dsp_cfg.data.channels = aac_bt_cfg->channels;
+ switch(aac_bt_cfg->audio_object_type) {
+ case 0:
+ aac_dsp_cfg.data.audio_object_type = MEDIA_FMT_AAC_AOT_LC;
break;
case 2:
+ aac_dsp_cfg.data.audio_object_type = MEDIA_FMT_AAC_AOT_PS;
+ break;
+ case 1:
default:
- in_channels = "Two";
+ aac_dsp_cfg.data.audio_object_type = MEDIA_FMT_AAC_AOT_SBR;
break;
}
-
- ALOGD("%s: set AFE input channels = %d", __func__, a2dp.enc_channels);
- ctrl_in_channels = mixer_get_ctl_by_name(a2dp.adev->mixer,
- MIXER_AFE_IN_CHANNELS);
- if (!ctrl_in_channels) {
- ALOGE("%s: ERROR AFE input channels mixer control not identifed", __func__);
- return -ENOSYS;
- }
- if (mixer_ctl_set_enum_by_string(ctrl_in_channels, in_channels) != 0) {
- ALOGE("%s: Failed to set AFE in channels = %d", __func__, a2dp.enc_channels);
- return -ENOSYS;
+ aac_dsp_cfg.data.total_size_of_pce_bits = aac_bt_cfg->total_size_of_pce_bits;
+ aac_dsp_cfg.data.sampling_rate = aac_bt_cfg->sampling_rate;
+ ret = mixer_ctl_set_array(ctl_dec_data, (void *)&aac_dsp_cfg,
+ sizeof(struct aac_dec_cfg_t));
+ if (ret != 0) {
+ ALOGE("%s: failed to set AAC decoder config", __func__);
+ is_configured = false;
+ goto fail;
}
- return 0;
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_DEC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ ALOGE(" ERROR Dec bit format mixer control not identified");
+ is_configured = false;
+ goto fail;
+ }
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set bit format to decoder", __func__);
+ is_configured = false;
+ goto fail;
+ }
+
+ is_configured = true;
+ a2dp.bt_decoder_format = CODEC_TYPE_AAC;
+ a2dp.dec_channels = aac_dsp_cfg.data.channels;
+ a2dp.dec_sampling_rate = aac_dsp_cfg.data.sampling_rate;
+ ALOGV("Successfully updated AAC dec format with sampling_rate: %d channels:%d",
+ aac_dsp_cfg.data.sampling_rate, aac_dsp_cfg.data.channels);
+fail:
+ return is_configured;
}
static int a2dp_set_bit_format(uint32_t enc_bit_format)
@@ -1000,25 +1241,31 @@
return 0;
}
-static int a2dp_reset_backend_cfg()
+static int a2dp_reset_backend_cfg(uint8_t direction)
{
- const char *rate_str = "KHZ_8", *in_channels = "Zero";
- struct mixer_ctl *ctl_sample_rate_rx = NULL, *ctl_sample_rate_tx = NULL;
- struct mixer_ctl *ctrl_in_channels = NULL;
+ const char *rate_str = "KHZ_8", *channels = "Zero";
+ struct mixer_ctl *ctl_sample_rate = NULL, *ctl_sample_rate_tx = NULL;
+ struct mixer_ctl *ctrl_channels = NULL;
// Reset backend sampling rate
- ALOGD("%s: reset backend sample rate = %s", __func__, rate_str);
- ctl_sample_rate_rx = mixer_get_ctl_by_name(a2dp.adev->mixer,
- MIXER_SAMPLE_RATE_RX);
- if (ctl_sample_rate_rx) {
+ if (direction == SINK) {
+ ALOGD("%s: reset sink backend sample rate =%s", __func__, rate_str);
+ ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_SINK_SAMPLE_RATE);
+ } else {
+ ALOGD("%s: reset source backend sample rate =%s", __func__, rate_str);
+ ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_SAMPLE_RATE_RX);
+ }
+ if (ctl_sample_rate) {
- if (mixer_ctl_set_enum_by_string(ctl_sample_rate_rx, rate_str) != 0) {
- ALOGE("%s: Failed to reset Rx backend sample rate = %s", __func__, rate_str);
+ if (mixer_ctl_set_enum_by_string(ctl_sample_rate, rate_str) != 0) {
+ 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_SAMPLE_RATE_TX);
+ MIXER_SOURCE_SAMPLE_RATE_TX);
if (!ctl_sample_rate_tx) {
ALOGE("%s: ERROR Tx backend sample rate mixer control not identifed", __func__);
return -ENOSYS;
@@ -1030,28 +1277,34 @@
}
} else {
- ctl_sample_rate_rx = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ ctl_sample_rate = mixer_get_ctl_by_name(a2dp.adev->mixer,
MIXER_SAMPLE_RATE_DEFAULT);
- if (!ctl_sample_rate_rx) {
+ if (!ctl_sample_rate) {
ALOGE("%s: ERROR backend sample rate mixer control not identifed", __func__);
return -ENOSYS;
}
- if (mixer_ctl_set_enum_by_string(ctl_sample_rate_rx, rate_str) != 0) {
+ if (mixer_ctl_set_enum_by_string(ctl_sample_rate, rate_str) != 0) {
ALOGE("%s: Failed to reset backend sample rate = %s", __func__, rate_str);
return -ENOSYS;
}
}
// Reset AFE input channels
- ALOGD("%s: reset AFE input channels = %s", __func__, in_channels);
- ctrl_in_channels = mixer_get_ctl_by_name(a2dp.adev->mixer,
- MIXER_AFE_IN_CHANNELS);
- if (!ctrl_in_channels) {
+ if (direction == SINK) {
+ ALOGD("%s: reset afe sink channels =%s", __func__, channels);
+ ctrl_channels = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_AFE_SINK_CHANNELS);
+ } else {
+ ALOGD("%s: reset afe source channels =%s", __func__, channels);
+ ctrl_channels = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_AFE_IN_CHANNELS);
+ }
+ if (!ctrl_channels) {
ALOGE("%s: ERROR AFE input channels mixer control not identifed", __func__);
return -ENOSYS;
}
- if (mixer_ctl_set_enum_by_string(ctrl_in_channels, in_channels) != 0) {
+ if (mixer_ctl_set_enum_by_string(ctrl_channels, channels) != 0) {
ALOGE("%s: Failed to reset AFE in channels = %d", __func__, a2dp.enc_channels);
return -ENOSYS;
}
@@ -1060,14 +1313,14 @@
}
/* API to configure AFE decoder in DSP */
-static bool configure_a2dp_decoder_format(int dec_format)
+static bool configure_a2dp_source_decoder_format(int dec_format)
{
struct mixer_ctl *ctl_dec_data = NULL;
struct abr_dec_cfg_t dec_cfg;
int ret = 0;
if (a2dp.abr_config.is_abr_enabled) {
- ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_DEC_CONFIG_BLOCK);
+ ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_SOURCE_DEC_CONFIG_BLOCK);
if (!ctl_dec_data) {
ALOGE("%s: ERROR A2DP codec config data mixer control not identifed", __func__);
return false;
@@ -1085,9 +1338,104 @@
ALOGE("%s: Failed to set decoder config", __func__);
return false;
}
+ }
+
+ return true;
+}
+
+bool configure_sbc_dec_format(audio_sbc_dec_config_t *sbc_bt_cfg)
+{
+ struct mixer_ctl *ctl_dec_data = NULL, *ctrl_bit_format = NULL;
+ struct sbc_dec_cfg_t sbc_dsp_cfg;
+ bool is_configured = false;
+ int ret = 0;
+
+ if(sbc_bt_cfg == NULL)
+ goto fail;
+
+ ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_SINK_DEC_CONFIG_BLOCK);
+ if (!ctl_dec_data) {
+ ALOGE(" ERROR a2dp decoder CONFIG data mixer control not identified");
+ is_configured = false;
+ goto fail;
}
- return true;
+ memset(&sbc_dsp_cfg, 0x0, sizeof(struct sbc_dec_cfg_t));
+ sbc_dsp_cfg.dec_format = MEDIA_FMT_SBC;
+ sbc_dsp_cfg.data.channels = sbc_bt_cfg->channels;
+ sbc_dsp_cfg.data.sampling_rate = sbc_bt_cfg->sampling_rate;
+ ret = mixer_ctl_set_array(ctl_dec_data, (void *)&sbc_dsp_cfg,
+ sizeof(struct sbc_dec_cfg_t));
+
+ if (ret != 0) {
+ ALOGE("%s: failed to set SBC decoder config", __func__);
+ is_configured = false;
+ goto fail;
+ }
+
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_DEC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ ALOGE(" ERROR Dec bit format mixer control not identified");
+ is_configured = false;
+ goto fail;
+ }
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set bit format to decoder", __func__);
+ is_configured = false;
+ goto fail;
+ }
+
+ is_configured = true;
+ a2dp.bt_decoder_format = CODEC_TYPE_SBC;
+ if (sbc_dsp_cfg.data.channels == MEDIA_FMT_SBC_CHANNEL_MODE_MONO)
+ a2dp.dec_channels = 1;
+ else
+ a2dp.dec_channels = 2;
+ a2dp.dec_sampling_rate = sbc_dsp_cfg.data.sampling_rate;
+ ALOGV("Successfully updated SBC dec format");
+fail:
+ return is_configured;
+}
+
+/* API to configure AFE decoder in DSP */
+static bool configure_a2dp_sink_decoder_format()
+{
+ void *codec_info = NULL;
+ codec_t codec_type = CODEC_TYPE_INVALID;
+ bool is_configured = false;
+ struct mixer_ctl *ctl_dec_data = NULL;
+ int ret = 0;
+
+ if (!a2dp.audio_get_dec_config) {
+ ALOGE(" a2dp handle is not identified, ignoring a2dp decoder config");
+ return false;
+ }
+
+ ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_SINK_DEC_CONFIG_BLOCK);
+ if (!ctl_dec_data) {
+ ALOGE(" ERROR a2dp decoder CONFIG data mixer control not identified");
+ is_configured = false;
+ return false;
+ }
+ codec_info = a2dp.audio_get_dec_config(&codec_type);
+ switch(codec_type) {
+ case CODEC_TYPE_SBC:
+ ALOGD(" SBC decoder supported BT device");
+ is_configured = configure_sbc_dec_format((audio_sbc_dec_config_t *)codec_info);
+ break;
+ case CODEC_TYPE_AAC:
+ ALOGD(" AAC decoder supported BT device");
+ is_configured =
+ configure_aac_dec_format((audio_aac_dec_config_t *)codec_info);
+ break;
+ default:
+ ALOGD(" Received Unsupported decoder format");
+ is_configured = false;
+ break;
+ }
+ return is_configured;
}
/* API to configure SBC DSP encoder */
@@ -1103,12 +1451,12 @@
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");
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identified");
is_configured = false;
goto fail;
}
memset(&sbc_dsp_cfg, 0x0, sizeof(struct sbc_enc_cfg_t));
- sbc_dsp_cfg.enc_format = ENC_MEDIA_FMT_SBC;
+ sbc_dsp_cfg.enc_format = MEDIA_FMT_SBC;
sbc_dsp_cfg.num_subbands = sbc_bt_cfg->subband;
sbc_dsp_cfg.blk_len = sbc_bt_cfg->blk_len;
switch(sbc_bt_cfg->channels) {
@@ -1145,7 +1493,7 @@
goto fail;
}
is_configured = true;
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_SBC;
+ a2dp.bt_encoder_format = CODEC_TYPE_SBC;
a2dp.enc_sampling_rate = sbc_bt_cfg->sampling_rate;
if (sbc_dsp_cfg.channel_mode == MEDIA_FMT_SBC_CHANNEL_MODE_MONO)
@@ -1259,7 +1607,7 @@
}
memset(aptx_dsp_cfg, 0x0, sizeof(struct aptx_enc_cfg_t));
- aptx_dsp_cfg->custom_cfg.enc_format = ENC_MEDIA_FMT_APTX;
+ aptx_dsp_cfg->custom_cfg.enc_format = MEDIA_FMT_APTX;
if (!a2dp.is_aptx_dual_mono_supported) {
aptx_dsp_cfg->custom_cfg.sample_rate = aptx_bt_cfg->default_cfg->sampling_rate;
@@ -1316,7 +1664,7 @@
}
memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_t));
- aptx_dsp_cfg->enc_format = ENC_MEDIA_FMT_APTX;
+ aptx_dsp_cfg->enc_format = MEDIA_FMT_APTX;
aptx_dsp_cfg->sample_rate = aptx_bt_cfg->sampling_rate;
aptx_dsp_cfg->num_channels = aptx_bt_cfg->channels;
switch(aptx_dsp_cfg->num_channels) {
@@ -1410,9 +1758,9 @@
}
is_configured = true;
if (a2dp.is_aptx_adaptive)
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_APTX_AD;
+ a2dp.bt_encoder_format = CODEC_TYPE_APTX_AD;
else
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_APTX;
+ a2dp.bt_encoder_format = CODEC_TYPE_APTX;
fail:
/*restore sample rate */
if(!is_configured)
@@ -1438,13 +1786,13 @@
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");
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identified");
is_configured = false;
goto fail;
}
memset(&aptx_dsp_cfg, 0x0, sizeof(struct custom_enc_cfg_t));
- aptx_dsp_cfg.enc_format = ENC_MEDIA_FMT_APTX_HD;
+ aptx_dsp_cfg.enc_format = MEDIA_FMT_APTX_HD;
aptx_dsp_cfg.sample_rate = aptx_bt_cfg->sampling_rate;
aptx_dsp_cfg.num_channels = aptx_bt_cfg->channels;
switch(aptx_dsp_cfg.num_channels) {
@@ -1470,7 +1818,7 @@
goto fail;
}
is_configured = true;
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_APTX_HD;
+ a2dp.bt_encoder_format = CODEC_TYPE_APTX_HD;
a2dp.enc_sampling_rate = aptx_bt_cfg->sampling_rate;
a2dp.enc_channels = aptx_bt_cfg->channels;
ALOGV("Successfully updated APTX HD encformat with samplingrate: %d channels:%d",
@@ -1492,12 +1840,12 @@
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");
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identified");
is_configured = false;
goto fail;
}
memset(&aac_dsp_cfg, 0x0, sizeof(struct aac_enc_cfg_t));
- aac_dsp_cfg.enc_format = ENC_MEDIA_FMT_AAC;
+ aac_dsp_cfg.enc_format = MEDIA_FMT_AAC;
aac_dsp_cfg.bit_rate = aac_bt_cfg->bitrate;
aac_dsp_cfg.sample_rate = aac_bt_cfg->sampling_rate;
switch (aac_bt_cfg->enc_mode) {
@@ -1528,7 +1876,7 @@
goto fail;
}
is_configured = true;
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_AAC;
+ a2dp.bt_encoder_format = CODEC_TYPE_AAC;
a2dp.enc_sampling_rate = aac_bt_cfg->sampling_rate;
a2dp.enc_channels = aac_bt_cfg->channels;
ALOGV("%s: Successfully updated AAC enc format with sampling rate: %d channels:%d",
@@ -1607,13 +1955,13 @@
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");
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identified");
is_configured = false;
goto fail;
}
memset(&celt_dsp_cfg, 0x0, sizeof(struct celt_enc_cfg_t));
- celt_dsp_cfg.custom_cfg.enc_format = ENC_MEDIA_FMT_CELT;
+ celt_dsp_cfg.custom_cfg.enc_format = MEDIA_FMT_CELT;
celt_dsp_cfg.custom_cfg.sample_rate = celt_bt_cfg->sampling_rate;
celt_dsp_cfg.custom_cfg.num_channels = celt_bt_cfg->channels;
switch(celt_dsp_cfg.custom_cfg.num_channels) {
@@ -1648,7 +1996,7 @@
goto fail;
}
is_configured = true;
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_CELT;
+ a2dp.bt_encoder_format = CODEC_TYPE_CELT;
a2dp.enc_sampling_rate = celt_bt_cfg->sampling_rate;
a2dp.enc_channels = celt_bt_cfg->channels;
ALOGV("Successfully updated CELT encformat with samplingrate: %d channels:%d",
@@ -1668,13 +2016,13 @@
ldac_enc_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_ENC_CONFIG_BLOCK);
if (!ldac_enc_data) {
- ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identifed");
+ ALOGE(" ERROR a2dp encoder CONFIG data mixer control not identified");
is_configured = false;
goto fail;
}
memset(&ldac_dsp_cfg, 0x0, sizeof(struct ldac_enc_cfg_t));
- ldac_dsp_cfg.custom_cfg.enc_format = ENC_MEDIA_FMT_LDAC;
+ ldac_dsp_cfg.custom_cfg.enc_format = MEDIA_FMT_LDAC;
ldac_dsp_cfg.custom_cfg.sample_rate = ldac_bt_cfg->sampling_rate;
ldac_dsp_cfg.ldac_cfg.channel_mode = ldac_bt_cfg->channel_mode;
switch(ldac_dsp_cfg.ldac_cfg.channel_mode) {
@@ -1716,7 +2064,7 @@
goto fail;
}
is_configured = true;
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_LDAC;
+ a2dp.bt_encoder_format = CODEC_TYPE_LDAC;
a2dp.enc_sampling_rate = ldac_bt_cfg->sampling_rate;
a2dp.enc_channels = ldac_dsp_cfg.custom_cfg.num_channels;
a2dp.abr_config.is_abr_enabled = ldac_bt_cfg->is_abr_enabled;
@@ -1730,16 +2078,16 @@
{
void *codec_info = NULL;
uint8_t multi_cast = 0, num_dev = 1;
- enc_codec_t codec_type = ENC_CODEC_TYPE_INVALID;
+ codec_t codec_type = CODEC_TYPE_INVALID;
bool is_configured = false;
audio_aptx_encoder_config aptx_encoder_cfg;
- if (!a2dp.audio_get_codec_config) {
+ if (!a2dp.audio_get_enc_config) {
ALOGE(" a2dp handle is not identified, ignoring a2dp encoder config");
return false;
}
ALOGD("configure_a2dp_encoder_format start");
- codec_info = a2dp.audio_get_codec_config(&multi_cast, &num_dev,
+ codec_info = a2dp.audio_get_enc_config(&multi_cast, &num_dev,
&codec_type);
// ABR disabled by default for all codecs
@@ -1747,12 +2095,12 @@
a2dp.is_aptx_adaptive = false;
switch(codec_type) {
- case ENC_CODEC_TYPE_SBC:
+ case CODEC_TYPE_SBC:
ALOGD(" Received SBC encoder supported BT device");
is_configured =
configure_sbc_enc_format((audio_sbc_encoder_config *)codec_info);
break;
- case ENC_CODEC_TYPE_APTX:
+ case CODEC_TYPE_APTX:
ALOGD(" Received APTX encoder supported BT device");
#ifndef LINUX_ENABLED
a2dp.is_aptx_dual_mono_supported = false;
@@ -1761,7 +2109,7 @@
is_configured =
configure_aptx_enc_format(&aptx_encoder_cfg);
break;
- case ENC_CODEC_TYPE_APTX_HD:
+ case CODEC_TYPE_APTX_HD:
ALOGD(" Received APTX HD encoder supported BT device");
#ifndef LINUX_ENABLED
is_configured =
@@ -1772,7 +2120,7 @@
#endif
break;
#ifndef LINUX_ENABLED
- case ENC_CODEC_TYPE_APTX_DUAL_MONO:
+ case CODEC_TYPE_APTX_DUAL_MONO:
ALOGD(" Received APTX dual mono encoder supported BT device");
a2dp.is_aptx_dual_mono_supported = true;
if (a2dp.audio_is_tws_mono_mode_enable != NULL)
@@ -1782,7 +2130,7 @@
configure_aptx_enc_format(&aptx_encoder_cfg);
break;
#endif
- case ENC_CODEC_TYPE_AAC:
+ case CODEC_TYPE_AAC:
ALOGD(" Received AAC encoder supported BT device");
bool is_aac_frame_ctl_enabled =
property_get_bool("persist.vendor.bt.aac_frm_ctl.enabled", false);
@@ -1790,22 +2138,22 @@
configure_aac_enc_format_v2((audio_aac_encoder_config_v2 *) codec_info) :
configure_aac_enc_format((audio_aac_encoder_config *) codec_info);
break;
- case ENC_CODEC_TYPE_CELT:
+ case CODEC_TYPE_CELT:
ALOGD(" Received CELT encoder supported BT device");
is_configured =
configure_celt_enc_format((audio_celt_encoder_config *)codec_info);
break;
- case ENC_CODEC_TYPE_LDAC:
+ case CODEC_TYPE_LDAC:
ALOGD(" Received LDAC encoder supported BT device");
if (!instance_id || instance_id > MAX_INSTANCE_ID)
instance_id = MAX_INSTANCE_ID;
a2dp.abr_config.imc_instance = instance_id--;
is_configured =
(configure_ldac_enc_format((audio_ldac_encoder_config *)codec_info) &&
- configure_a2dp_decoder_format(ENC_CODEC_TYPE_LDAC));
+ configure_a2dp_source_decoder_format(CODEC_TYPE_LDAC));
break;
#ifndef LINUX_ENABLED //Temporarily disabled for LE, need to take care while doing VT FR
- case ENC_CODEC_TYPE_APTX_AD:
+ case CODEC_TYPE_APTX_AD:
ALOGD(" Received APTX AD encoder supported BT device");
if (!instance_id || instance_id > MAX_INSTANCE_ID)
instance_id = MAX_INSTANCE_ID;
@@ -1815,7 +2163,7 @@
aptx_encoder_cfg.ad_cfg = (audio_aptx_ad_config *)codec_info;
is_configured =
(configure_aptx_enc_format(&aptx_encoder_cfg) &&
- configure_a2dp_decoder_format(ENC_MEDIA_FMT_APTX_AD));
+ configure_a2dp_source_decoder_format(ENC_MEDIA_FMT_APTX_AD));
break;
#endif
default:
@@ -1832,49 +2180,135 @@
ALOGD("audio_extn_a2dp_start_playback start");
- if(!(a2dp.bt_lib_handle && a2dp.audio_start_stream
- && a2dp.audio_get_codec_config)) {
- ALOGE("a2dp handle is not identified, Ignoring start request");
+ if(!(a2dp.bt_lib_source_handle && a2dp.audio_source_start
+ && a2dp.audio_get_enc_config)) {
+ ALOGE("a2dp handle is not identified, Ignoring start playback request");
return -ENOSYS;
}
- if(a2dp.a2dp_suspended == true) {
+ if(a2dp.a2dp_source_suspended == true) {
//session will be restarted after suspend completion
ALOGD("a2dp start requested during suspend state");
return -ENOSYS;
}
- if (!a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
+ if (!a2dp.a2dp_source_started && !a2dp.a2dp_source_total_active_session_requests) {
ALOGD("calling BT module stream start");
/* This call indicates BT IPC lib to start playback */
- ret = a2dp.audio_start_stream();
+ ret = a2dp.audio_source_start();
ALOGE("BT controller start return = %d",ret);
if (ret != 0 ) {
ALOGE("BT controller start failed");
- a2dp.a2dp_started = false;
+ a2dp.a2dp_source_started = false;
} else {
if(configure_a2dp_encoder_format() == true) {
- a2dp.a2dp_started = true;
+ a2dp.a2dp_source_started = true;
ret = 0;
ALOGD("Start playback successful to BT library");
} else {
ALOGD(" unable to configure DSP encoder");
- a2dp.a2dp_started = false;
+ a2dp.a2dp_source_started = false;
ret = -ETIMEDOUT;
}
}
}
- if (a2dp.a2dp_started) {
- a2dp.a2dp_total_active_session_request++;
+ if (a2dp.a2dp_source_started) {
+ a2dp.a2dp_source_total_active_session_requests++;
a2dp_check_and_set_scrambler();
- a2dp_set_backend_cfg();
+ a2dp_set_backend_cfg(SOURCE);
if (a2dp.abr_config.is_abr_enabled)
start_abr();
}
ALOGD("start A2DP playback total active sessions :%d",
- a2dp.a2dp_total_active_session_request);
+ a2dp.a2dp_source_total_active_session_requests);
+ return ret;
+}
+
+uint64_t audio_extn_a2dp_get_decoder_latency()
+{
+ uint32_t latency = 0;
+
+ switch(a2dp.bt_decoder_format) {
+ case CODEC_TYPE_SBC:
+ latency = DEFAULT_SINK_LATENCY_SBC;
+ break;
+ case CODEC_TYPE_AAC:
+ latency = DEFAULT_SINK_LATENCY_AAC;
+ break;
+ default:
+ latency = 200;
+ ALOGD("No valid decoder defined, setting latency to %dms", latency);
+ break;
+ }
+ return (uint64_t)latency;
+}
+
+bool a2dp_send_sink_setup_complete(void) {
+ uint64_t system_latency = 0;
+ bool is_complete = false;
+
+ system_latency = audio_extn_a2dp_get_decoder_latency();
+
+ if (a2dp.audio_sink_session_setup_complete(system_latency) == 0) {
+ is_complete = true;
+ }
+ return is_complete;
+}
+
+int audio_extn_a2dp_start_capture()
+{
+ int ret = 0;
+
+ ALOGD("audio_extn_a2dp_start_capture start");
+
+ if(!(a2dp.bt_lib_sink_handle && a2dp.audio_sink_start
+ && a2dp.audio_get_dec_config)) {
+ ALOGE("a2dp handle is not identified, Ignoring start capture request");
+ return -ENOSYS;
+ }
+
+ if (!a2dp.a2dp_sink_started && !a2dp.a2dp_sink_total_active_session_requests) {
+ ALOGD("calling BT module stream start");
+ /* This call indicates BT IPC lib to start capture */
+ ret = a2dp.audio_sink_start();
+ ALOGE("BT controller start capture return = %d",ret);
+ if (ret != 0 ) {
+ ALOGE("BT controller start capture failed");
+ a2dp.a2dp_sink_started = false;
+ } else {
+
+ if(!audio_extn_a2dp_sink_is_ready()) {
+ ALOGD("Wait for capture ready not successful");
+ ret = -ETIMEDOUT;
+ }
+
+ if(configure_a2dp_sink_decoder_format() == true) {
+ a2dp.a2dp_sink_started = true;
+ ret = 0;
+ ALOGD("Start capture successful to BT library");
+ } else {
+ ALOGD(" unable to configure DSP decoder");
+ a2dp.a2dp_sink_started = false;
+ ret = -ETIMEDOUT;
+ }
+
+ if (!a2dp_send_sink_setup_complete()) {
+ ALOGD("sink_setup_complete not successful");
+ ret = -ETIMEDOUT;
+ }
+ }
+ }
+
+ if (a2dp.a2dp_sink_started) {
+ if (a2dp_set_backend_cfg(SINK) == true) {
+ a2dp.a2dp_sink_total_active_session_requests++;
+ }
+ }
+
+ ALOGD("start A2DP sink total active sessions :%d",
+ a2dp.a2dp_sink_total_active_session_requests);
return ret;
}
@@ -1890,16 +2324,16 @@
ctl_enc_config = mixer_get_ctl_by_name(a2dp.adev->mixer,
MIXER_ENC_CONFIG_BLOCK);
if (!ctl_enc_config) {
- ALOGE(" ERROR a2dp encoder format mixer control not identifed");
+ ALOGE(" ERROR a2dp encoder format mixer control not identified");
} else {
ret = mixer_ctl_set_array(ctl_enc_config, (void *)&dummy_reset_config,
sizeof(struct sbc_enc_cfg_t));
- a2dp.bt_encoder_format = ENC_MEDIA_FMT_NONE;
+ a2dp.bt_encoder_format = MEDIA_FMT_NONE;
}
ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
MIXER_ENC_BIT_FORMAT);
if (!ctrl_bit_format) {
- ALOGE(" ERROR bit format CONFIG data mixer control not identifed");
+ ALOGE(" ERROR bit format CONFIG data mixer control not identified");
} else {
ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
if (ret != 0) {
@@ -1919,14 +2353,14 @@
}
}
-static int reset_a2dp_dec_config_params()
+static int reset_a2dp_source_dec_config_params()
{
struct mixer_ctl *ctl_dec_data = NULL;
struct abr_dec_cfg_t dummy_reset_cfg;
int ret = 0;
if (a2dp.abr_config.is_abr_enabled) {
- ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_DEC_CONFIG_BLOCK);
+ ctl_dec_data = mixer_get_ctl_by_name(a2dp.adev->mixer, MIXER_SOURCE_DEC_CONFIG_BLOCK);
if (!ctl_dec_data) {
ALOGE("%s: ERROR A2DP decoder config mixer control not identifed", __func__);
return -EINVAL;
@@ -1943,38 +2377,98 @@
return ret;
}
+static void reset_a2dp_sink_dec_config_params()
+{
+ int ret =0;
+
+ struct mixer_ctl *ctl_dec_config, *ctrl_bit_format;
+ struct aac_dec_cfg_t dummy_reset_config;
+
+ memset(&dummy_reset_config, 0x0, sizeof(struct aac_dec_cfg_t));
+ ctl_dec_config = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_SINK_DEC_CONFIG_BLOCK);
+ if (!ctl_dec_config) {
+ ALOGE(" ERROR a2dp decoder format mixer control not identified");
+ } else {
+ ret = mixer_ctl_set_array(ctl_dec_config, (void *)&dummy_reset_config,
+ sizeof(struct aac_dec_cfg_t));
+ a2dp.bt_decoder_format = MEDIA_FMT_NONE;
+ }
+ ctrl_bit_format = mixer_get_ctl_by_name(a2dp.adev->mixer,
+ MIXER_DEC_BIT_FORMAT);
+ if (!ctrl_bit_format) {
+ ALOGE(" ERROR bit format CONFIG data mixer control not identified");
+ } else {
+ ret = mixer_ctl_set_enum_by_string(ctrl_bit_format, "S16_LE");
+ if (ret != 0) {
+ ALOGE("%s: Failed to set bit format to decoder", __func__);
+ }
+ }
+}
+
int audio_extn_a2dp_stop_playback()
{
int ret =0;
ALOGV("audio_extn_a2dp_stop_playback start");
- if(!(a2dp.bt_lib_handle && a2dp.audio_stop_stream)) {
- ALOGE("a2dp handle is not identified, Ignoring start request");
+ if(!(a2dp.bt_lib_source_handle && a2dp.audio_source_stop)) {
+ ALOGE("a2dp handle is not identified, Ignoring stop request");
return -ENOSYS;
}
- if (a2dp.a2dp_total_active_session_request > 0)
- a2dp.a2dp_total_active_session_request--;
+ if (a2dp.a2dp_source_total_active_session_requests > 0)
+ a2dp.a2dp_source_total_active_session_requests--;
- if ( a2dp.a2dp_started && !a2dp.a2dp_total_active_session_request) {
+ if ( a2dp.a2dp_source_started && !a2dp.a2dp_source_total_active_session_requests) {
ALOGV("calling BT module stream stop");
- ret = a2dp.audio_stop_stream();
+ ret = a2dp.audio_source_stop();
if (ret < 0)
ALOGE("stop stream to BT IPC lib failed");
else
ALOGV("stop steam to BT IPC lib successful");
reset_a2dp_enc_config_params();
- reset_a2dp_dec_config_params();
- a2dp_reset_backend_cfg();
+ 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;
- a2dp.a2dp_started = false;
+ a2dp.a2dp_source_started = false;
+ a2dp_reset_backend_cfg(SOURCE);
}
- if(!a2dp.a2dp_total_active_session_request)
- a2dp.a2dp_started = false;
- ALOGD("Stop A2DP playback total active sessions :%d",
- a2dp.a2dp_total_active_session_request);
+ if(!a2dp.a2dp_source_total_active_session_requests)
+ a2dp.a2dp_source_started = false;
+ ALOGD("Stop A2DP playback, total active sessions :%d",
+ a2dp.a2dp_source_total_active_session_requests);
+ return 0;
+}
+
+int audio_extn_a2dp_stop_capture()
+{
+ int ret =0;
+
+ ALOGV("audio_extn_a2dp_stop_capture start");
+ if(!(a2dp.bt_lib_sink_handle && a2dp.audio_sink_stop)) {
+ ALOGE("a2dp handle is not identified, Ignoring stop request");
+ return -ENOSYS;
+ }
+
+ if (a2dp.a2dp_sink_total_active_session_requests > 0)
+ a2dp.a2dp_sink_total_active_session_requests--;
+
+ if ( a2dp.a2dp_sink_started && !a2dp.a2dp_sink_total_active_session_requests) {
+ ALOGV("calling BT module stream stop");
+ ret = a2dp.audio_sink_stop();
+ if (ret < 0)
+ ALOGE("stop stream to BT IPC lib failed");
+ else
+ ALOGV("stop steam to BT IPC lib successful");
+ reset_a2dp_sink_dec_config_params();
+ a2dp_reset_backend_cfg(SINK);
+ }
+ if(!a2dp.a2dp_sink_total_active_session_requests)
+ a2dp.a2dp_source_started = false;
+ ALOGD("Stop A2DP capture, total active sessions :%d",
+ a2dp.a2dp_sink_total_active_session_requests);
return 0;
}
@@ -1986,7 +2480,7 @@
struct listnode *node;
if(a2dp.is_a2dp_offload_supported == false) {
- ALOGV("no supported encoders identified,ignoring a2dp setparam");
+ ALOGV("no supported codecs identified,ignoring a2dp setparam");
return;
}
@@ -1995,8 +2489,8 @@
if (ret >= 0) {
val = atoi(value);
if (audio_is_a2dp_out_device(val)) {
- ALOGV("Received device connect request for A2DP");
- open_a2dp_output();
+ ALOGV("Received device connect request for A2DP source");
+ open_a2dp_source();
}
goto param_handled;
}
@@ -2007,15 +2501,20 @@
if (ret >= 0) {
val = atoi(value);
if (audio_is_a2dp_out_device(val)) {
- ALOGV("Received device dis- connect request");
- reset_a2dp_enc_config_params();
- reset_a2dp_dec_config_params();
+ ALOGV("Received source device dis- connect request");
close_a2dp_output();
- a2dp_reset_backend_cfg();
+ reset_a2dp_enc_config_params();
+ reset_a2dp_source_dec_config_params();
+ a2dp_reset_backend_cfg(SOURCE);
+ } else if (audio_is_a2dp_in_device(val)) {
+ ALOGV("Received sink device dis- connect request");
+ close_a2dp_input();
+ reset_a2dp_sink_dec_config_params();
+ a2dp_reset_backend_cfg(SINK);
}
goto param_handled;
}
-
+#ifndef LINUX_ENABLED
ret = str_parms_get_str(parms, "TwsChannelConfig", value, sizeof(value));
if (ret>=0) {
ALOGD("Setting tws channel mode to %s",value);
@@ -2026,14 +2525,14 @@
audio_a2dp_update_tws_channel_mode();
goto param_handled;
}
-
+#endif
ret = str_parms_get_str(parms, "A2dpSuspended", value, sizeof(value));
if (ret >= 0) {
- if (a2dp.bt_lib_handle) {
+ if (a2dp.bt_lib_source_handle) {
if ((!strncmp(value,"true",sizeof(value)))) {
ALOGD("Setting a2dp to suspend state");
- a2dp.a2dp_suspended = true;
- if (a2dp.bt_state == A2DP_STATE_DISCONNECTED)
+ a2dp.a2dp_source_suspended = true;
+ if (a2dp.bt_state_source == A2DP_STATE_DISCONNECTED)
goto param_handled;
list_for_each(node, &a2dp.adev->usecase_list) {
uc_info = node_to_item(node, struct audio_usecase, list);
@@ -2045,16 +2544,16 @@
}
}
reset_a2dp_enc_config_params();
- reset_a2dp_dec_config_params();
- if(a2dp.audio_suspend_stream)
- a2dp.audio_suspend_stream();
- } else if (a2dp.a2dp_suspended == true) {
+ reset_a2dp_source_dec_config_params();
+ if(a2dp.audio_source_suspend)
+ a2dp.audio_source_suspend();
+ } else if (a2dp.a2dp_source_suspended == true) {
ALOGD("Resetting a2dp suspend state");
struct audio_usecase *uc_info;
struct listnode *node;
- if(a2dp.clear_a2dpsuspend_flag)
- a2dp.clear_a2dpsuspend_flag();
- a2dp.a2dp_suspended = false;
+ if(a2dp.clear_source_a2dpsuspend_flag)
+ a2dp.clear_source_a2dpsuspend_flag();
+ a2dp.a2dp_source_suspended = false;
/*
* It is possible that before suspend,a2dp sessions can be active
* for example during music + voice activation concurrency
@@ -2066,13 +2565,13 @@
* Fix is to call a2dp start for IPC library post suspend
* based on number of active session count
*/
- if (a2dp.a2dp_total_active_session_request > 0) {
+ if (a2dp.a2dp_source_total_active_session_requests > 0) {
ALOGD(" Calling IPC lib start post suspend state");
- if(a2dp.audio_start_stream) {
- ret = a2dp.audio_start_stream();
+ if(a2dp.audio_source_start) {
+ ret = a2dp.audio_source_start();
if (ret != 0) {
ALOGE("BT controller start failed");
- a2dp.a2dp_started = false;
+ a2dp.a2dp_source_started = false;
}
}
}
@@ -2103,44 +2602,59 @@
//During encoder reconfiguration mode, force a2dp device switch
// Or if a2dp device is selected but earlier start failed ( as a2dp
// was suspended, force retry.
- return a2dp.is_handoff_in_progress || !a2dp.a2dp_started;
+ return a2dp.is_handoff_in_progress || !a2dp.a2dp_source_started;
}
-void audio_extn_a2dp_get_sample_rate(int *sample_rate)
+void audio_extn_a2dp_get_enc_sample_rate(int *sample_rate)
{
*sample_rate = a2dp.enc_sampling_rate;
}
-bool audio_extn_a2dp_is_ready()
+void audio_extn_a2dp_get_dec_sample_rate(int *sample_rate)
+{
+ *sample_rate = a2dp.dec_sampling_rate;
+}
+
+bool audio_extn_a2dp_source_is_ready()
{
bool ret = false;
- if (a2dp.a2dp_suspended)
+ if (a2dp.a2dp_source_suspended)
return ret;
- if ((a2dp.bt_state != A2DP_STATE_DISCONNECTED) &&
+ if ((a2dp.bt_state_source != A2DP_STATE_DISCONNECTED) &&
(a2dp.is_a2dp_offload_supported) &&
- (a2dp.audio_check_a2dp_ready))
- ret = a2dp.audio_check_a2dp_ready();
+ (a2dp.audio_source_check_a2dp_ready))
+ ret = a2dp.audio_source_check_a2dp_ready();
return ret;
}
-bool audio_extn_a2dp_is_suspended()
+bool audio_extn_a2dp_sink_is_ready()
{
- return a2dp.a2dp_suspended;
+ bool ret = false;
+
+ if ((a2dp.bt_state_sink != A2DP_STATE_DISCONNECTED) &&
+ (a2dp.is_a2dp_offload_supported) &&
+ (a2dp.audio_sink_check_a2dp_ready))
+ ret = a2dp.audio_sink_check_a2dp_ready();
+ return ret;
+}
+
+bool audio_extn_a2dp_source_is_suspended()
+{
+ return a2dp.a2dp_source_suspended;
}
void audio_extn_a2dp_init (void *adev)
{
a2dp.adev = (struct audio_device*)adev;
- a2dp.bt_lib_handle = NULL;
- a2dp.a2dp_started = false;
- a2dp.bt_state = A2DP_STATE_DISCONNECTED;
- a2dp.a2dp_total_active_session_request = 0;
- a2dp.a2dp_suspended = false;
- a2dp.bt_encoder_format = ENC_CODEC_TYPE_INVALID;
+ a2dp.bt_lib_source_handle = NULL;
+ a2dp.a2dp_source_started = false;
+ a2dp.bt_state_source = A2DP_STATE_DISCONNECTED;
+ a2dp.a2dp_source_total_active_session_requests = 0;
+ a2dp.a2dp_source_suspended = false;
+ a2dp.bt_encoder_format = CODEC_TYPE_INVALID;
a2dp.enc_sampling_rate = 48000;
- a2dp.is_a2dp_offload_supported = false;
a2dp.is_handoff_in_progress = false;
a2dp.is_aptx_dual_mono_supported = false;
a2dp.is_aptx_adaptive = false;
@@ -2150,7 +2664,16 @@
a2dp.abr_config.abr_tx_handle = NULL;
a2dp.is_tws_mono_mode_on = false;
reset_a2dp_enc_config_params();
- reset_a2dp_dec_config_params();
+ reset_a2dp_source_dec_config_params();
+ reset_a2dp_sink_dec_config_params();
+
+ a2dp.bt_lib_sink_handle = NULL;
+ a2dp.a2dp_sink_started = false;
+ a2dp.bt_state_sink = A2DP_STATE_DISCONNECTED;
+ a2dp.a2dp_sink_total_active_session_requests = 0;
+ open_a2dp_sink();
+
+ a2dp.is_a2dp_offload_supported = false;
update_offload_codec_capabilities();
}
@@ -2173,32 +2696,32 @@
}
uint32_t slatency = 0;
- if (a2dp.audio_get_a2dp_sink_latency && a2dp.bt_state != A2DP_STATE_DISCONNECTED) {
- slatency = a2dp.audio_get_a2dp_sink_latency();
+ if (a2dp.audio_sink_get_a2dp_latency && a2dp.bt_state_source != A2DP_STATE_DISCONNECTED) {
+ slatency = a2dp.audio_sink_get_a2dp_latency();
}
switch(a2dp.bt_encoder_format) {
- case ENC_CODEC_TYPE_SBC:
+ case CODEC_TYPE_SBC:
latency = (avsync_runtime_prop > 0) ? sbc_offset : ENCODER_LATENCY_SBC;
latency += (slatency <= 0) ? DEFAULT_SINK_LATENCY_SBC : slatency;
break;
- case ENC_CODEC_TYPE_APTX:
+ case CODEC_TYPE_APTX:
latency = (avsync_runtime_prop > 0) ? aptx_offset : ENCODER_LATENCY_APTX;
latency += (slatency <= 0) ? DEFAULT_SINK_LATENCY_APTX : slatency;
break;
- case ENC_CODEC_TYPE_APTX_HD:
+ case CODEC_TYPE_APTX_HD:
latency = (avsync_runtime_prop > 0) ? aptxhd_offset : ENCODER_LATENCY_APTX_HD;
latency += (slatency <= 0) ? DEFAULT_SINK_LATENCY_APTX_HD : slatency;
break;
- case ENC_CODEC_TYPE_AAC:
+ case CODEC_TYPE_AAC:
latency = (avsync_runtime_prop > 0) ? aac_offset : ENCODER_LATENCY_AAC;
latency += (slatency <= 0) ? DEFAULT_SINK_LATENCY_AAC : slatency;
break;
- case ENC_CODEC_TYPE_CELT:
+ case CODEC_TYPE_CELT:
latency = (avsync_runtime_prop > 0) ? celt_offset : ENCODER_LATENCY_CELT;
latency += (slatency <= 0) ? DEFAULT_SINK_LATENCY_CELT : slatency;
break;
- case ENC_CODEC_TYPE_LDAC:
+ case CODEC_TYPE_LDAC:
latency = (avsync_runtime_prop > 0) ? ldac_offset : ENCODER_LATENCY_LDAC;
latency += (slatency <= 0) ? DEFAULT_SINK_LATENCY_LDAC : slatency;
break;
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 9047cb3..9eb3e9e 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -293,10 +293,14 @@
#define audio_extn_a2dp_set_parameters(parms) (0)
#define audio_extn_a2dp_is_force_device_switch() (0)
#define audio_extn_a2dp_set_handoff_mode(is_on) (0)
-#define audio_extn_a2dp_get_sample_rate(sample_rate) (0)
+#define audio_extn_a2dp_get_enc_sample_rate(sample_rate) (0)
+#define audio_extn_a2dp_get_dec_sample_rate(sample_rate) (0)
#define audio_extn_a2dp_get_encoder_latency() (0)
-#define audio_extn_a2dp_is_ready() (0)
-#define audio_extn_a2dp_is_suspended() (0)
+#define audio_extn_a2dp_sink_is_ready() (0)
+#define audio_extn_a2dp_source_is_ready() (0)
+#define audio_extn_a2dp_source_is_suspended() (0)
+#define audio_extn_a2dp_start_capture() (0)
+#define audio_extn_a2dp_stop_capture() (0)
#else
void audio_extn_a2dp_init(void *adev);
int audio_extn_a2dp_start_playback();
@@ -304,10 +308,14 @@
void audio_extn_a2dp_set_parameters(struct str_parms *parms);
bool audio_extn_a2dp_is_force_device_switch();
void audio_extn_a2dp_set_handoff_mode(bool is_on);
-void audio_extn_a2dp_get_sample_rate(int *sample_rate);
+void audio_extn_a2dp_get_enc_sample_rate(int *sample_rate);
+void audio_extn_a2dp_get_dec_sample_rate(int *sample_rate);
uint32_t audio_extn_a2dp_get_encoder_latency();
-bool audio_extn_a2dp_is_ready();
-bool audio_extn_a2dp_is_suspended();
+bool audio_extn_a2dp_sink_is_ready();
+bool audio_extn_a2dp_source_is_ready();
+bool audio_extn_a2dp_source_is_suspended();
+int audio_extn_a2dp_start_capture();
+int audio_extn_a2dp_stop_capture();
#endif
#ifndef SSR_ENABLED
diff --git a/hal/audio_extn/hw_loopback.c b/hal/audio_extn/hw_loopback.c
index c76319f..77e5ab2 100644
--- a/hal/audio_extn/hw_loopback.c
+++ b/hal/audio_extn/hw_loopback.c
@@ -77,9 +77,9 @@
typedef struct loopback_patch {
audio_patch_handle_t patch_handle_id; /* patch unique ID */
struct audio_port_config loopback_source; /* Source port config */
- struct audio_port_config loopback_sink; /* Source port config */
+ struct audio_port_config loopback_sink; /* Sink port config */
struct compress *source_stream; /* Source stream */
- struct compress *sink_stream; /* Source stream */
+ struct compress *sink_stream; /* Sink stream */
struct stream_inout patch_stream; /* InOut type stream */
patch_state_t patch_state; /* Patch operation state */
} loopback_patch_t;
@@ -196,7 +196,9 @@
case AUDIO_PORT_TYPE_DEVICE :
if ((loopback_patch->loopback_source.config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
if ((loopback_patch->loopback_source.ext.device.type & AUDIO_DEVICE_IN_HDMI) ||
- (loopback_patch->loopback_source.ext.device.type & AUDIO_DEVICE_IN_SPDIF)) {
+ (loopback_patch->loopback_source.ext.device.type & AUDIO_DEVICE_IN_SPDIF) ||
+ (loopback_patch->loopback_source.ext.device.type & AUDIO_DEVICE_IN_BLUETOOTH_A2DP)) {
+
switch (loopback_patch->loopback_source.format) {
case AUDIO_FORMAT_PCM:
case AUDIO_FORMAT_PCM_16_BIT:
@@ -205,6 +207,10 @@
case AUDIO_FORMAT_IEC61937:
case AUDIO_FORMAT_AC3:
case AUDIO_FORMAT_E_AC3:
+ case AUDIO_FORMAT_AAC_LATM_LC:
+ case AUDIO_FORMAT_AAC_LATM_HE_V1:
+ case AUDIO_FORMAT_AAC_LATM_HE_V2:
+ case AUDIO_FORMAT_SBC:
is_source_supported = true;
break;
}
@@ -214,8 +220,8 @@
}
break;
default :
- break;
- //Unsupported as of now, need to extend for other source types
+ //Unsupported as of now, need to extend for other source types
+ break;
}
}
@@ -241,14 +247,13 @@
}
break;
default :
- break;
//Unsupported as of now, need to extend for other sink types
+ break;
}
}
if (is_source_supported && is_sink_supported) {
return source_device | sink_device;
}
-
ALOGE("%s, Unsupported source or sink port config", __func__);
return loopback_patch->patch_handle_id;
}
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index f084396..4067055 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -97,7 +97,11 @@
#define MAX_RESISTANCE_SPKR_Q24 (40 * (1 << 24))
/*Path where the calibration file will be stored*/
-#define CALIB_FILE "/data/vendor/audio/audio.cal"
+#ifdef LINUX_ENABLED
+#define CALIB_FILE "/data/audio/audio.cal"
+#else
+#define CALIB_FILE "/data/vendor/misc/audio/audio.cal"
+#endif
/*Time between retries for calibartion or intial wait time
after boot up*/
@@ -982,6 +986,7 @@
struct audio_device *adev = handle.adev_handle;
unsigned long min_idle_time = MIN_SPKR_IDLE_SEC;
char value[PROPERTY_VALUE_MAX];
+ char afe_version_value[PROPERTY_VALUE_MAX];
char wsa_path[MAX_PATH] = {0};
int spk_1_tzn, spk_2_tzn;
char buf[32] = {0};
@@ -1022,7 +1027,10 @@
}
spv3_enable = property_get_bool("persist.vendor.audio.spv3.enable", false);
- afe_api_version = property_get_int32("persist.vendor.audio.avs.afe_api_version", 0);
+ property_get("persist.vendor.audio.avs.afe_api_version", afe_version_value,
+ "0");
+ if (atoi(afe_version_value) > 0)
+ afe_api_version = atoi(afe_version_value);
fp = fopen(CALIB_FILE,"rb");
if (fp) {
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 1af22c2..7c5756b 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -994,7 +994,14 @@
if (usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
usecase->stream.out->app_type_cfg.sample_rate = usecase->stream.out->sample_rate;
} else if (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
- usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ if (platform_spkr_use_default_sample_rate(adev->platform)) {
+ usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ } else {
+ platform_check_and_update_copp_sample_rate(adev->platform, snd_device,
+ usecase->stream.out->sample_rate,
+ &usecase->stream.out->app_type_cfg.sample_rate);
+ }
+
} else if ((snd_device == SND_DEVICE_OUT_HDMI ||
snd_device == SND_DEVICE_OUT_USB_HEADSET ||
snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
@@ -1018,7 +1025,7 @@
* For a2dp playback get encoder sampling rate and set copp sampling rate,
* for bit width use the stream param only.
*/
- audio_extn_a2dp_get_sample_rate(&usecase->stream.out->app_type_cfg.sample_rate);
+ audio_extn_a2dp_get_enc_sample_rate(&usecase->stream.out->app_type_cfg.sample_rate);
ALOGI("%s using %d sample rate rate for A2DP CoPP",
__func__, usecase->stream.out->app_type_cfg.sample_rate);
}
@@ -1070,6 +1077,11 @@
} else {
audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
}
+ if (usecase->stream.in->device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP & ~AUDIO_DEVICE_BIT_IN) {
+ audio_extn_a2dp_get_dec_sample_rate(&usecase->stream.in->app_type_cfg.sample_rate);
+ ALOGI("%s using %d sample rate rate for A2DP dec CoPP",
+ __func__, usecase->stream.in->app_type_cfg.sample_rate);
+ }
sample_rate = usecase->stream.in->app_type_cfg.sample_rate;
app_type_cfg[len++] = sample_rate;
if (snd_device_be_idx > 0)
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 7f41763..49145ec 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -1091,7 +1091,13 @@
if ((SND_DEVICE_OUT_BT_A2DP == snd_device) &&
(audio_extn_a2dp_start_playback() < 0)) {
- ALOGE(" fail to configure A2dp control path ");
+ ALOGE(" fail to configure A2dp Source control path ");
+ return -EINVAL;
+ }
+
+ if ((SND_DEVICE_IN_BT_A2DP == snd_device) &&
+ (audio_extn_a2dp_start_capture() < 0)) {
+ ALOGE(" fail to configure A2dp Sink control path ");
return -EINVAL;
}
@@ -1179,6 +1185,9 @@
if (SND_DEVICE_OUT_BT_A2DP == snd_device)
audio_extn_a2dp_stop_playback();
+ if (SND_DEVICE_IN_BT_A2DP == snd_device)
+ audio_extn_a2dp_stop_capture();
+
if (snd_device == SND_DEVICE_OUT_HDMI || snd_device == SND_DEVICE_OUT_DISPLAY_PORT)
adev->is_channel_status_set = false;
else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
@@ -1987,12 +1996,12 @@
audio_extn_a2dp_is_force_device_switch()) {
ALOGD("Force a2dp device switch to update new encoder config");
ret = true;
- }
+ }
- if (usecase->stream.out->stream_config_changed) {
+ if (usecase->stream.out->stream_config_changed) {
ALOGD("Force stream_config_changed to update iec61937 transmission config");
return true;
- }
+ }
return ret;
}
@@ -2199,7 +2208,7 @@
}
if ((is_btsco_device(out_snd_device,in_snd_device) && !adev->bt_sco_on) ||
- (is_a2dp_device(out_snd_device) && !audio_extn_a2dp_is_ready())) {
+ (is_a2dp_device(out_snd_device) && !audio_extn_a2dp_source_is_ready())) {
ALOGD("SCO/A2DP is selected but they are not connected/ready hence dont route");
return 0;
}
@@ -2233,7 +2242,7 @@
}
if ((out_snd_device == SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP) &&
- (!audio_extn_a2dp_is_ready())) {
+ (!audio_extn_a2dp_source_is_ready())) {
ALOGW("%s: A2DP profile is not ready, routing to speaker only", __func__);
out_snd_device = SND_DEVICE_OUT_SPEAKER;
}
@@ -3037,7 +3046,7 @@
}
if (out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
- if (!audio_extn_a2dp_is_ready()) {
+ if (!audio_extn_a2dp_source_is_ready()) {
if (out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
a2dp_combo = true;
} else {
@@ -3107,7 +3116,7 @@
}
if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
- (!audio_extn_a2dp_is_ready())) {
+ (!audio_extn_a2dp_source_is_ready())) {
if (!a2dp_combo) {
check_a2dp_restore_l(adev, out, false);
} else {
@@ -3473,6 +3482,7 @@
int channel_count,
bool is_low_latency)
{
+ int i = 0;
size_t size = 0;
uint32_t bytes_per_period_sample = 0;
@@ -3486,7 +3496,8 @@
bytes_per_period_sample = audio_bytes_per_sample(format) * channel_count;
size *= bytes_per_period_sample;
- /* make sure the size is multiple of 32 bytes
+ /* make sure the size is multiple of 32 bytes and additionally multiple of
+ * the frame_size (required for 24bit samples and non-power-of-2 channel counts)
* At 48 kHz mono 16-bit PCM:
* 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
* 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
@@ -3873,13 +3884,13 @@
/*
* When A2DP is disconnected the
* music playback is paused and the policy manager sends routing=0
- * But the audioflingercontinues to write data until standby time
+ * But the audioflinger continues to write data until standby time
* (3sec). As BT is turned off, the write gets blocked.
* Avoid this by routing audio to speaker until standby.
*/
if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
(val == AUDIO_DEVICE_NONE) &&
- !audio_extn_a2dp_is_ready()) {
+ !audio_extn_a2dp_source_is_ready()) {
val = AUDIO_DEVICE_OUT_SPEAKER;
}
/*
@@ -3898,7 +3909,7 @@
* check with BT lib about a2dp streaming support before routing
*/
if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
- if (!audio_extn_a2dp_is_ready()) {
+ if (!audio_extn_a2dp_source_is_ready()) {
if (val & AUDIO_DEVICE_OUT_SPEAKER) {
//combo usecase just by pass a2dp
ALOGW("%s: A2DP profile is not ready,routing to speaker only", __func__);
@@ -4000,7 +4011,7 @@
}
if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
out->a2dp_compress_mute &&
- (!(out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) || audio_extn_a2dp_is_ready())) {
+ (!(out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) || audio_extn_a2dp_source_is_ready())) {
pthread_mutex_lock(&out->compr_mute_lock);
out->a2dp_compress_mute = false;
out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
@@ -4617,7 +4628,7 @@
}
if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
- (audio_extn_a2dp_is_suspended())) {
+ (audio_extn_a2dp_source_is_suspended())) {
if (!(out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
ret = -EIO;
@@ -4687,6 +4698,15 @@
audio_format_t dst_format = out->hal_op_format;
audio_format_t src_format = out->hal_ip_format;
+ /* prevent division-by-zero */
+ uint32_t bitwidth_src = format_to_bitwidth_table[src_format];
+ uint32_t bitwidth_dst = format_to_bitwidth_table[dst_format];
+ if ((bitwidth_src == 0) || (bitwidth_dst == 0)) {
+ ALOGE("%s: Error bitwidth == 0", __func__);
+ ATRACE_END();
+ return -EINVAL;
+ }
+
uint32_t frames = bytes / format_to_bitwidth_table[src_format];
uint32_t bytes_to_write = frames * format_to_bitwidth_table[dst_format];
@@ -4853,10 +4873,18 @@
out->standby = true;
}
out_on_error(&out->stream.common);
- if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
- usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
- out_get_sample_rate(&out->stream.common));
+ if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
+ /* prevent division-by-zero */
+ uint32_t stream_size = audio_stream_out_frame_size(stream);
+ uint32_t srate = out_get_sample_rate(&out->stream.common);
+ if ((stream_size == 0) || (srate == 0)) {
+ ALOGE("%s: stream_size= %d, srate = %d", __func__, stream_size, srate);
+ ATRACE_END();
+ return -EINVAL;
+ }
+ usleep((uint64_t)bytes * 1000000 / stream_size / srate);
+ }
if (audio_extn_passthru_is_passthrough_stream(out)) {
//ALOGE("%s: write error, ret = %zd", __func__, ret);
ATRACE_END();
@@ -7399,6 +7427,13 @@
config->format,
channel_count,
is_low_latency);
+ /* prevent division-by-zero */
+ if (frame_size == 0) {
+ ALOGE("%s: Error frame_size==0", __func__);
+ ret = -EINVAL;
+ goto err_open;
+ }
+
in->config.period_size = buffer_size / frame_size;
if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
diff --git a/hal/edid.h b/hal/edid.h
index f97d0e3..a7578e6 100644
--- a/hal/edid.h
+++ b/hal/edid.h
@@ -57,15 +57,17 @@
#define PCM_CHANNEL_FRC 14 /* Front right of center. */
#define PCM_CHANNEL_RLC 15 /* Rear left of center. */
#define PCM_CHANNEL_RRC 16 /* Rear right of center. */
-#define PCM_CHANNEL_LFE2 17 /* Rear right of center. */
+#define PCM_CHANNEL_LFE2 17 /* Second low frequency channel. */
#define PCM_CHANNEL_SL 18 /* Side left channel. */
-#define PCM_CHANNEL_SR 19 /* Side right channel */
+#define PCM_CHANNEL_SR 19 /* Side right channel. */
#define PCM_CHANNEL_TFL 20 /* Top front left channel. */
+#define PCM_CHANNEL_LVH 20 /* Left vertical height channel. */
#define PCM_CHANNEL_TFR 21 /* Top front right channel. */
+#define PCM_CHANNEL_RVH 21 /* Right vertical height channel. */
#define PCM_CHANNEL_TC 22 /* Top center channel. */
#define PCM_CHANNEL_TBL 23 /* Top back left channel. */
#define PCM_CHANNEL_TBR 24 /* Top back right channel. */
-#define PCM_CHANNEL_TSL 25 /* Top side left channel */
+#define PCM_CHANNEL_TSL 25 /* Top side left channel. */
#define PCM_CHANNEL_TSR 26 /* Top side right channel. */
#define PCM_CHANNEL_TBC 27 /* Top back center channel. */
#define PCM_CHANNEL_BFC 28 /* Bottom front center channel. */
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 3ea991b..eb2cecc 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -308,6 +308,7 @@
struct audio_microphone_characteristic_t microphones[AUDIO_MICROPHONE_MAX_COUNT];
struct snd_device_to_mic_map mic_map[SND_DEVICE_MAX];
struct spkr_device_chmap *spkr_ch_map;
+ bool use_sprk_default_sample_rate;
};
struct spkr_device_chmap {
@@ -2342,6 +2343,7 @@
my_data->voice_speaker_stereo = false;
my_data->declared_mic_count = 0;
my_data->spkr_ch_map = NULL;
+ my_data->use_sprk_default_sample_rate = true;
be_dai_name_table = NULL;
@@ -2885,6 +2887,9 @@
/* free acdb_meta_key_list */
platform_release_acdb_metainfo_key(platform);
+ if (my_data->acdb_deallocate)
+ my_data->acdb_deallocate();
+
free(platform);
/* deinit usb */
audio_extn_usb_deinit();
@@ -7488,6 +7493,11 @@
platform_get_edid_info(platform);
}
+bool platform_spkr_use_default_sample_rate(void *platform) {
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->use_sprk_default_sample_rate;
+}
+
void platform_invalidate_backend_config(void * platform,snd_device_t snd_device)
{
struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index ad8c9aa..e337870 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -188,6 +188,7 @@
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_A2DP,
SND_DEVICE_IN_CAMCORDER_MIC,
SND_DEVICE_IN_VOICE_DMIC,
SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index e10a955..59fad85 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -67,6 +67,7 @@
#define MIXER_XML_PATH_I2S "/etc/mixer_paths_i2s.xml"
#define PLATFORM_INFO_XML_PATH_I2S "/etc/audio_platform_info_extcodec.xml"
#define PLATFORM_INFO_XML_PATH_WSA "/etc/audio_platform_info_wsa.xml"
+#define PLATFORM_INFO_XML_PATH_TDM "/etc/audio_platform_info_tdm.xml"
#else
#define MIXER_XML_BASE_STRING "/vendor/etc/mixer_paths"
#define MIXER_XML_DEFAULT_PATH "/vendor/etc/mixer_paths.xml"
@@ -79,6 +80,7 @@
#define MIXER_XML_PATH_I2S "/vendor/etc/mixer_paths_i2s.xml"
#define PLATFORM_INFO_XML_PATH_I2S "/vendor/etc/audio_platform_info_i2s.xml"
#define PLATFORM_INFO_XML_PATH_WSA "/vendor/etc/audio_platform_info_wsa.xml"
+#define PLATFORM_INFO_XML_PATH_TDM "/vendor/etc/audio_platform_info_tdm.xml"
#endif
#include <linux/msm_audio.h>
@@ -124,7 +126,6 @@
/* Mixer path names */
#define AFE_SIDETONE_MIXER_PATH "afe-sidetone"
-#define AUDIO_PARAMETER_KEY_FLUENCE_TYPE "fluence"
#define AUDIO_PARAMETER_KEY_SLOWTALK "st_enable"
#define AUDIO_PARAMETER_KEY_HD_VOICE "hd_voice"
#define AUDIO_PARAMETER_KEY_VOLUME_BOOST "volume_boost"
@@ -133,6 +134,15 @@
#define AUDIO_PARAMETER_KEY_MONO_SPEAKER "mono_speaker"
+#define AUDIO_PARAMETER_KEY_FLUENCE_TYPE "fluence_type"
+#define AUDIO_PARAMETER_KEY_FLUENCE_VOICE_CALL "fluence_voice"
+#define AUDIO_PARAMETER_KEY_FLUENCE_VOICE_REC "fluence_voice_rec"
+#define AUDIO_PARAMETER_KEY_FLUENCE_AUDIO_REC "fluence_audio_rec"
+#define AUDIO_PARAMETER_KEY_FLUENCE_SPEAKER "fluence_speaker"
+#define AUDIO_PARAMETER_KEY_FLUENCE_MODE "fluence_mode"
+#define AUDIO_PARAMETER_KEY_FLUENCE_HFPCALL "fluence_hfp"
+#define AUDIO_PARAMETER_KEY_FLUENCE_TRI_MIC "fluence_tri_mic"
+
#define AUDIO_PARAMETER_KEY_PERF_LOCK_OPTS "perf_lock_opts"
/* Reload ACDB files from specified path */
@@ -290,6 +300,7 @@
struct audio_microphone_characteristic_t microphones[AUDIO_MICROPHONE_MAX_COUNT];
struct snd_device_to_mic_map mic_map[SND_DEVICE_MAX];
struct spkr_device_chmap *spkr_ch_map;
+ bool use_sprk_default_sample_rate;
};
struct spkr_device_chmap {
@@ -507,6 +518,7 @@
[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_A2DP] = "bt-a2dp-cap",
[SND_DEVICE_IN_CAMCORDER_MIC] = "camcorder-mic",
[SND_DEVICE_IN_VOICE_DMIC] = "voice-dmic-ef",
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC] = "voice-speaker-dmic-ef",
@@ -693,6 +705,7 @@
[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_A2DP] = 21,
[SND_DEVICE_IN_CAMCORDER_MIC] = 4,
[SND_DEVICE_IN_VOICE_DMIC] = 41,
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC] = 43,
@@ -853,6 +866,7 @@
{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_A2DP)},
{TO_NAME_INDEX(SND_DEVICE_IN_CAMCORDER_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC)},
@@ -1511,6 +1525,7 @@
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("voice-speaker-2-vbat");
backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
+ backend_tag_table[SND_DEVICE_IN_BT_A2DP] = strdup("bt-a2dp-cap");
backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_AND_VOICE_HEADPHONES] = strdup("speaker-and-headphones");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_AND_VOICE_ANC_HEADSET] = strdup("speaker-and-headphones");
@@ -1612,6 +1627,7 @@
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_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");
hw_interface_table[SND_DEVICE_IN_VOICE_SPEAKER_DMIC] = strdup("SLIMBUS_0_TX");
@@ -2204,7 +2220,7 @@
my_data->voice_speaker_stereo = false;
my_data->declared_mic_count = 0;
my_data->spkr_ch_map = NULL;
-
+ my_data->use_sprk_default_sample_rate = true;
be_dai_name_table = NULL;
property_get("ro.vendor.audio.sdk.fluencetype", my_data->fluence_cap, "");
@@ -2295,11 +2311,23 @@
else if (!strncmp(snd_card_name, "qcs405-wsa-snd-card",
sizeof("qcs405-wsa-snd-card")))
platform_info_init(PLATFORM_INFO_XML_PATH_WSA, my_data, PLATFORM);
+ else if (!strncmp(snd_card_name, "qcs405-tdm-snd-card",
+ sizeof("qcs405-tdm-snd-card")))
+ platform_info_init(PLATFORM_INFO_XML_PATH_TDM, my_data, PLATFORM);
else if (my_data->is_internal_codec)
platform_info_init(PLATFORM_INFO_XML_PATH_INTCODEC, my_data, PLATFORM);
else
platform_info_init(PLATFORM_INFO_XML_PATH, my_data, PLATFORM);
+ /* CSRA devices support multiple sample rates via I2S at spkr out */
+ if (!strncmp(snd_card_name, "qcs405-csra", strlen("qcs405-csra"))) {
+ ALOGE("%s: soundcard: %s supports multiple sample rates", __func__, snd_card_name);
+ my_data->use_sprk_default_sample_rate = false;
+ } else {
+ my_data->use_sprk_default_sample_rate = true;
+ ALOGE("%s: soundcard: %s supports only default sample rate", __func__, snd_card_name);
+ }
+
my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
if (my_data->acdb_handle == NULL) {
@@ -2576,11 +2604,18 @@
}
} else {
if (!strncmp(snd_card_name, "qcs405", strlen("qcs405"))) {
- my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
- strdup("WSA_CDC_DMA_RX_0 Format");
- my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
- strdup("WSA_CDC_DMA_RX_0 SampleRate");
+ if (!strncmp(snd_card_name, "qcs405-csra", strlen("qcs405-csra"))) {
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ strdup("PRIM_MI2S_RX Format");
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ strdup("PRIM_MI2S_RX SampleRate");
+ } else {
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ strdup("WSA_CDC_DMA_RX_0 Format");
+ my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ strdup("WSA_CDC_DMA_RX_0 SampleRate");
+ }
my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
strdup("VA_CDC_DMA_TX_0 Format");
my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
@@ -4811,6 +4846,8 @@
}
} else if (in_device & AUDIO_DEVICE_IN_SPDIF) {
snd_device = SND_DEVICE_IN_SPDIF;
+ } else if (in_device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
+ snd_device = SND_DEVICE_IN_BT_A2DP;
} else if (in_device & AUDIO_DEVICE_IN_AUX_DIGITAL) {
snd_device = SND_DEVICE_IN_HDMI_MIC;
} else if (in_device & AUDIO_DEVICE_IN_HDMI_ARC) {
@@ -5273,13 +5310,106 @@
platform->spkr_ch_map->num_ch = num_ch;
for (i = 0; i < num_ch; i++) {
opts = strtok_r(NULL, ", ", &test_r);
- platform->spkr_ch_map->chmap[i] = strtoul(opts, NULL, 16);
+ if (opts == NULL) {
+ ALOGE("%s: incorrect ch_map\n", __func__);
+ free(platform->spkr_ch_map);
+ platform->spkr_ch_map = NULL;
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_SPKR_DEVICE_CHMAP);
+ return;
+ } else {
+ platform->spkr_ch_map->chmap[i] = strtoul(opts, NULL, 16);
+ }
}
}
str_parms_del(parms, AUDIO_PARAMETER_KEY_SPKR_DEVICE_CHMAP);
}
}
+static void platform_set_fluence_params(void *platform, struct str_parms *parms, char *value, int len)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ int err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE_TYPE, value, len);
+
+ if (err >= 0) {
+ if (!strncmp("fluence", value, sizeof("fluence")))
+ my_data->fluence_type = FLUENCE_DUAL_MIC;
+ else if (!strncmp("fluencepro", value, sizeof("fluencepro")))
+ my_data->fluence_type = FLUENCE_QUAD_MIC | FLUENCE_DUAL_MIC;
+ else if (!strncmp("none", value, sizeof("none")))
+ my_data->fluence_type = FLUENCE_NONE;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_FLUENCE_TYPE);
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE_TRI_MIC, value, len);
+ if (err >= 0) {
+ if (!strncmp("true", value, sizeof("true")))
+ my_data->fluence_type |= FLUENCE_TRI_MIC;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_FLUENCE_TRI_MIC);
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE_VOICE_CALL, value, len);
+ if (err >= 0) {
+ if (!strncmp("true", value, sizeof("true")))
+ my_data->fluence_in_voice_call = true;
+ else
+ my_data->fluence_in_voice_call = false;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_FLUENCE_VOICE_CALL);
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE_VOICE_REC, value, len);
+ if (err >= 0) {
+ if (!strncmp("true", value, sizeof("true")))
+ my_data->fluence_in_voice_rec = true;
+ else
+ my_data->fluence_in_voice_rec = false;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_FLUENCE_VOICE_REC);
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE_AUDIO_REC, value, len);
+ if (err >= 0) {
+ if (!strncmp("true", value, sizeof("true")))
+ my_data->fluence_in_audio_rec = true;
+ else
+ my_data->fluence_in_audio_rec = false;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_FLUENCE_AUDIO_REC);
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE_SPEAKER, value, len);
+ if (err >= 0) {
+ if (!strncmp("true", value, sizeof("true")))
+ my_data->fluence_in_spkr_mode = true;
+ else
+ my_data->fluence_in_spkr_mode = false;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_FLUENCE_SPEAKER);
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE_MODE, value, len);
+ if (err >= 0) {
+ if (!strncmp("broadside", value, sizeof("broadside")))
+ my_data->fluence_mode = FLUENCE_BROADSIDE;
+ else if (!strncmp("endfire", value, sizeof("endfire")))
+ my_data->fluence_mode = FLUENCE_ENDFIRE;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_FLUENCE_MODE);
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_FLUENCE_HFPCALL, value, len);
+ if (err >= 0) {
+ if (!strncmp("true", value, sizeof("true")))
+ my_data->fluence_in_hfp_call = true;
+ else
+ my_data->fluence_in_hfp_call = false;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_FLUENCE_HFPCALL);
+ }
+}
+
int platform_set_parameters(void *platform, struct str_parms *parms)
{
struct platform_data *my_data = (struct platform_data *)platform;
@@ -5413,6 +5543,8 @@
ALOGV("%s: max_mic_count %d", __func__, my_data->max_mic_count);
}
+ platform_set_fluence_params(platform, parms, value, len);
+
/* handle audio calibration parameters */
set_audiocal(platform, parms, value, len);
native_audio_set_params(platform, parms, value, len);
@@ -6372,6 +6504,7 @@
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_IN_BT_A2DP ||
snd_device == SND_DEVICE_OUT_AFE_PROXY) {
backend_change = false;
return backend_change;
@@ -6515,9 +6648,15 @@
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
ALOGD("%s:becf: afe: reset to default bitwidth %d", __func__, bit_width);
}
- sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
+ /*
+ * In case of CSRA speaker out, all sample rates are supported, so
+ * check platform here
+ */
+ if (platform_spkr_use_default_sample_rate(adev->platform)) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ ALOGD("%s:becf: afe: playback on codec device not supporting native playback set "
"default Sample Rate(48k)", __func__);
+ }
}
if (backend_idx == USB_AUDIO_RX_BACKEND) {
@@ -7219,6 +7358,40 @@
channel_map[7] = PCM_CHANNEL_RS;
}
break;
+ case 12:
+ /* AUDIO_CHANNEL_OUT_7POINT1POINT4 */
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_FC;
+ channel_map[3] = PCM_CHANNEL_LFE;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ channel_map[8] = PCM_CHANNEL_TFL;
+ channel_map[9] = PCM_CHANNEL_TFR;
+ channel_map[10] = PCM_CHANNEL_TSL;
+ channel_map[11] = PCM_CHANNEL_TSR;
+ break;
+ case 16:
+ /* 16 channels */
+ channel_map[0] = PCM_CHANNEL_FL;
+ channel_map[1] = PCM_CHANNEL_FR;
+ channel_map[2] = PCM_CHANNEL_FC;
+ channel_map[3] = PCM_CHANNEL_LFE;
+ channel_map[4] = PCM_CHANNEL_LB;
+ channel_map[5] = PCM_CHANNEL_RB;
+ channel_map[6] = PCM_CHANNEL_LS;
+ channel_map[7] = PCM_CHANNEL_RS;
+ channel_map[8] = PCM_CHANNEL_TFL;
+ channel_map[9] = PCM_CHANNEL_TFR;
+ channel_map[10] = PCM_CHANNEL_TSL;
+ channel_map[11] = PCM_CHANNEL_TSR;
+ channel_map[12] = PCM_CHANNEL_FLC;
+ channel_map[13] = PCM_CHANNEL_FRC;
+ channel_map[14] = PCM_CHANNEL_RLC;
+ channel_map[15] = PCM_CHANNEL_RRC;
+ break;
default:
ALOGE("unsupported channels %d for setting channel map", channels);
return -1;
@@ -7343,12 +7516,21 @@
struct mixer_ctl *ctl;
char mixer_ctl_name[44] = {0}; // max length of name is 44 as defined
int ret;
- unsigned int i;
- long set_values[FCC_8] = {0};
+ unsigned int i=0, n=0;
+ long set_values[AUDIO_MAX_DSP_CHANNELS];
struct platform_data *my_data = (struct platform_data *)platform;
struct audio_device *adev = my_data->adev;
ALOGV("%s channel_count:%d",__func__, ch_count);
- if (NULL == ch_map || (ch_count < 1) || (ch_count > FCC_8)) {
+
+ /*
+ * FIXME:
+ * Currently the channel mask in audio.h is limited to 30 channels,
+ * (=AUDIO_CHANNEL_COUNT_MAX), whereas the mixer controls already
+ * allow up to AUDIO_MAX_DSP_CHANNELS channels as per final requirement.
+ * Until channel mask definition is not changed from a uint32_t value
+ * to something else, a sanity check is needed here.
+ */
+ if (NULL == ch_map || (ch_count < 1) || (ch_count > AUDIO_CHANNEL_COUNT_MAX)) {
ALOGE("%s: Invalid channel mapping or channel count value", __func__);
return -EINVAL;
}
@@ -7366,12 +7548,34 @@
ALOGD("%s mixer_ctl_name:%s", __func__, mixer_ctl_name);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
- for (i = 0; i < (unsigned int)ch_count; i++) {
+
+ /* find out how many values the control can set */
+ n = mixer_ctl_get_num_values(ctl);
+
+ if (n != ch_count)
+ ALOGV("%s chcnt %d != mixerctl elem size %d",__func__, ch_count, n);
+
+ if (n < ch_count) {
+ ALOGE("%s chcnt %d > mixerctl elem size %d",__func__, ch_count, n);
+ return -EINVAL;
+ }
+
+ if (n > AUDIO_MAX_DSP_CHANNELS) {
+ ALOGE("%s mixerctl elem size %d > AUDIO_MAX_DSP_CHANNELS %d",__func__, n, AUDIO_MAX_DSP_CHANNELS);
+ return -EINVAL;
+ }
+
+ /* initialize all set_values to zero */
+ memset (set_values, 0, sizeof(set_values));
+
+ /* copy only as many values as corresponding mixer_ctrl allows */
+ for (i = 0; i < ch_count; i++) {
set_values[i] = ch_map[i];
}
@@ -7379,7 +7583,8 @@
set_values[0], set_values[1], set_values[2], set_values[3], set_values[4],
set_values[5], set_values[6], set_values[7], ch_count);
- ret = mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+ ret = mixer_ctl_set_array(ctl, set_values, n);
+
if (ret < 0) {
ALOGE("%s: Could not set ctl, error:%d ch_count:%d",
__func__, ret, ch_count);
@@ -7544,6 +7749,11 @@
return 0;
}
+bool platform_spkr_use_default_sample_rate(void *platform) {
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->use_sprk_default_sample_rate;
+}
+
int platform_set_edid_channels_configuration(void *platform, int channels, int backend_idx) {
struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index eb6edb0..60e6581 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -186,6 +186,7 @@
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_A2DP,
SND_DEVICE_IN_CAMCORDER_MIC,
SND_DEVICE_IN_VOICE_DMIC,
SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
@@ -293,6 +294,8 @@
#define AUDIO_PARAMETER_KEY_TRUE_32_BIT "true_32_bit"
+#define AUDIO_MAX_DSP_CHANNELS 32
+
#define ALL_SESSION_VSID 0xFFFFFFFF
#define DEFAULT_MUTE_RAMP_DURATION_MS 20
#define DEFAULT_VOLUME_RAMP_DURATION_MS 20
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 6474b61..21f3e72 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -225,6 +225,7 @@
snd_device_t snd_device,
struct mix_matrix_params mm_params);
int platform_set_edid_channels_configuration(void *platform, int channels, int backend_idx);
+bool platform_spkr_use_default_sample_rate(void *platform);
unsigned char platform_map_to_edid_format(int format);
bool platform_is_edid_supported_format(void *platform, int format);
bool platform_is_edid_supported_sample_rate(void *platform, int sample_rate);