Merge "hal: codec calibration data using hwdep nodes"
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 4c1416d..dd6063f 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -309,6 +309,7 @@
struct listnode *streams_output_cfg_list);
void audio_extn_utils_update_stream_app_type_cfg(void *platform,
struct listnode *streams_output_cfg_list,
+ audio_devices_t devices,
audio_output_flags_t flags,
audio_format_t format,
uint32_t sample_rate,
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index e148ead..5f4c6ba 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -139,12 +139,12 @@
pthread_mutex_lock(&st_dev->lock);
st_ses_info = get_sound_trigger_info(in->capture_handle);
+ pthread_mutex_unlock(&st_dev->lock);
if (st_ses_info) {
event.u.ses_info = st_ses_info->st_ses;
ALOGV("%s: AUDIO_EVENT_STOP_LAB pcm %p", __func__, st_ses_info->st_ses.pcm);
st_dev->st_callback(AUDIO_EVENT_STOP_LAB, &event);
}
- pthread_mutex_unlock(&st_dev->lock);
}
void audio_extn_sound_trigger_check_and_get_session(struct stream_in *in)
{
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 5287ae8..6818161 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -422,6 +422,7 @@
void audio_extn_utils_update_stream_app_type_cfg(void *platform,
struct listnode *streams_output_cfg_list,
+ audio_devices_t devices,
audio_output_flags_t flags,
audio_format_t format,
uint32_t sample_rate,
@@ -433,7 +434,14 @@
struct stream_format *sf_info;
struct stream_sample_rate *ss_info;
- ALOGV("%s: flags: %x, format: %x", __func__, flags, format);
+ if ((24 == bit_width) &&
+ (devices & AUDIO_DEVICE_OUT_SPEAKER)) {
+ sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ ALOGI("%s Allowing 24-bit playback on speaker ONLY at default sampling rate", __func__);
+ }
+
+ ALOGV("%s: flags: %x, format: %x sample_rate %d",
+ __func__, flags, format, sample_rate);
list_for_each(node_i, streams_output_cfg_list) {
so_info = node_to_item(node_i, struct streams_output_cfg, list);
if (so_info->flags == flags) {
@@ -516,7 +524,7 @@
}
if ((24 == usecase->stream.out->bit_width) &&
- (AUDIO_DEVICE_OUT_SPEAKER == snd_device)) {
+ (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
} else {
sample_rate = out->app_type_cfg.sample_rate;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 627900b..390fb46 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -60,9 +60,14 @@
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+#define PROXY_OPEN_RETRY_COUNT 100
+#define PROXY_OPEN_WAIT_TIME 20
#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER
+static unsigned int configured_low_latency_capture_period_size =
+ LOW_LATENCY_CAPTURE_PERIOD_SIZE;
+
struct pcm_config pcm_config_deep_buffer = {
.channels = 2,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
@@ -102,6 +107,37 @@
.format = PCM_FORMAT_S16_LE,
};
+#define AFE_PROXY_CHANNEL_COUNT 2
+#define AFE_PROXY_SAMPLING_RATE 48000
+
+#define AFE_PROXY_PLAYBACK_PERIOD_SIZE 768
+#define AFE_PROXY_PLAYBACK_PERIOD_COUNT 4
+
+struct pcm_config pcm_config_afe_proxy_playback = {
+ .channels = AFE_PROXY_CHANNEL_COUNT,
+ .rate = AFE_PROXY_SAMPLING_RATE,
+ .period_size = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+ .period_count = AFE_PROXY_PLAYBACK_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+ .stop_threshold = INT_MAX,
+ .avail_min = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+};
+
+#define AFE_PROXY_RECORD_PERIOD_SIZE 768
+#define AFE_PROXY_RECORD_PERIOD_COUNT 4
+
+struct pcm_config pcm_config_afe_proxy_record = {
+ .channels = AFE_PROXY_CHANNEL_COUNT,
+ .rate = AFE_PROXY_SAMPLING_RATE,
+ .period_size = AFE_PROXY_RECORD_PERIOD_SIZE,
+ .period_count = AFE_PROXY_RECORD_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = AFE_PROXY_RECORD_PERIOD_SIZE,
+ .stop_threshold = INT_MAX,
+ .avail_min = AFE_PROXY_RECORD_PERIOD_SIZE,
+};
+
const char * const use_case_table[AUDIO_USECASE_MAX] = {
[USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback",
[USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback",
@@ -142,6 +178,9 @@
[USECASE_INCALL_MUSIC_UPLINK2] = "incall_music_uplink2",
[USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib",
[USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record",
+
+ [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback",
+ [USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record",
};
static const audio_usecase_t offload_usecases[] = {
@@ -288,6 +327,19 @@
return 0;
}
+int pcm_ioctl(struct pcm *pcm, int request, ...)
+{
+ va_list ap;
+ void * arg;
+ int pcm_fd = *(int*)pcm;
+
+ va_start(ap, request);
+ arg = va_arg(ap, void *);
+ va_end(ap);
+
+ return ioctl(pcm_fd, request, arg);
+}
+
int enable_audio_route(struct audio_device *adev,
struct audio_usecase *usecase)
{
@@ -704,7 +756,7 @@
* usecase. This is to avoid switching devices for voice call when
* check_usecases_codec_backend() is called below.
*/
- if (adev->voice.in_call && adev->mode == AUDIO_MODE_IN_CALL) {
+ if (voice_is_in_call(adev) && adev->mode == AUDIO_MODE_IN_CALL) {
vc_usecase = get_usecase_from_list(adev,
get_voice_usecase_id_from_list(adev));
if ((vc_usecase) && ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
@@ -744,14 +796,14 @@
usecase->devices = usecase->stream.in->device;
out_snd_device = SND_DEVICE_NONE;
if (in_snd_device == SND_DEVICE_NONE) {
+ audio_devices_t out_device = AUDIO_DEVICE_NONE;
if (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
adev->primary_output && !adev->primary_output->standby) {
- in_snd_device = platform_get_input_snd_device(adev->platform,
- adev->primary_output->devices);
- } else {
- in_snd_device = platform_get_input_snd_device(adev->platform,
- AUDIO_DEVICE_NONE);
+ out_device = adev->primary_output->devices;
+ } else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
+ out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
}
+ in_snd_device = platform_get_input_snd_device(adev->platform, out_device);
}
}
}
@@ -822,26 +874,17 @@
usecase->out_snd_device = out_snd_device;
if (usecase->type == PCM_PLAYBACK) {
- if ((24 == usecase->stream.out->bit_width) &&
- (AUDIO_DEVICE_OUT_SPEAKER == usecase->stream.out->devices)) {
- audio_extn_utils_update_stream_app_type_cfg(adev->platform,
+ audio_extn_utils_update_stream_app_type_cfg(adev->platform,
&adev->streams_output_cfg_list,
- usecase->stream.out->flags,
- usecase->stream.out->format,
- DEFAULT_OUTPUT_SAMPLING_RATE,
- usecase->stream.out->bit_width,
- &usecase->stream.out->app_type_cfg);
- } else {
- audio_extn_utils_update_stream_app_type_cfg(adev->platform,
- &adev->streams_output_cfg_list,
+ usecase->stream.out->devices,
usecase->stream.out->flags,
usecase->stream.out->format,
usecase->stream.out->sample_rate,
usecase->stream.out->bit_width,
&usecase->stream.out->app_type_cfg);
- }
- ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
+ ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
}
+
enable_audio_route(adev, usecase);
/* Applicable only on the targets that has external modem.
@@ -943,16 +986,33 @@
select_devices(adev, in->usecase);
ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
- __func__, adev->snd_card,
- in->pcm_device_id, in->config.channels);
- in->pcm = pcm_open(adev->snd_card,
- in->pcm_device_id, PCM_IN, &in->config);
- if (in->pcm && !pcm_is_ready(in->pcm)) {
- ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
- pcm_close(in->pcm);
- in->pcm = NULL;
- ret = -EIO;
- goto error_open;
+ __func__, adev->snd_card, in->pcm_device_id, in->config.channels);
+
+ unsigned int flags = PCM_IN;
+ unsigned int pcm_open_retry_count = 0;
+
+ if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
+ flags |= PCM_MMAP | PCM_NOIRQ;
+ pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+ }
+
+ while (1) {
+ in->pcm = pcm_open(adev->snd_card, in->pcm_device_id,
+ flags, &in->config);
+ if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
+ ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
+ if (in->pcm != NULL) {
+ pcm_close(in->pcm);
+ in->pcm = NULL;
+ }
+ if (pcm_open_retry_count-- == 0) {
+ ret = -EIO;
+ goto error_open;
+ }
+ usleep(PROXY_OPEN_WAIT_TIME * 1000);
+ continue;
+ }
+ break;
}
ALOGV("%s: exit", __func__);
@@ -1354,18 +1414,34 @@
select_devices(adev, out->usecase);
- ALOGV("%s: Opening PCM device card_id(%d) device_id(%d)",
- __func__, 0, out->pcm_device_id);
+ ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
+ __func__, adev->snd_card, out->pcm_device_id, out->config.format);
if (!is_offload_usecase(out->usecase)) {
- out->pcm = pcm_open(adev->snd_card,
- out->pcm_device_id,
- PCM_OUT | PCM_MONOTONIC, &out->config);
- if (out->pcm && !pcm_is_ready(out->pcm)) {
- ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
- pcm_close(out->pcm);
- out->pcm = NULL;
- ret = -EIO;
- goto error_open;
+ unsigned int flags = PCM_OUT;
+ unsigned int pcm_open_retry_count = 0;
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
+ flags |= PCM_MMAP | PCM_NOIRQ;
+ pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+ } else
+ flags |= PCM_MONOTONIC;
+
+ while (1) {
+ out->pcm = pcm_open(adev->snd_card, out->pcm_device_id,
+ flags, &out->config);
+ if (out->pcm == NULL || !pcm_is_ready(out->pcm)) {
+ ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
+ if (out->pcm != NULL) {
+ pcm_close(out->pcm);
+ out->pcm = NULL;
+ }
+ if (pcm_open_retry_count-- == 0) {
+ ret = -EIO;
+ goto error_open;
+ }
+ usleep(PROXY_OPEN_WAIT_TIME * 1000);
+ continue;
+ }
+ break;
}
} else {
out->pcm = NULL;
@@ -1439,7 +1515,8 @@
static size_t get_input_buffer_size(uint32_t sample_rate,
audio_format_t format,
- int channel_count)
+ int channel_count,
+ bool is_low_latency)
{
size_t size = 0;
@@ -1447,13 +1524,19 @@
return 0;
size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
+ if (is_low_latency)
+ size = configured_low_latency_capture_period_size;
/* ToDo: should use frame_size computed based on the format and
channel_count here. */
size *= sizeof(short) * channel_count;
- /* make sure the size is multiple of 64 */
- size += 0x3f;
- size &= ~0x3f;
+ /* make sure the size is multiple of 32 bytes
+ * 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)
+ */
+ size += 0x1f;
+ size &= ~0x1f;
return size;
}
@@ -1630,6 +1713,10 @@
return 0;
}
+static bool output_drives_call(struct audio_device *adev, struct stream_out *out)
+{
+ return out == adev->primary_output || out == adev->voice_tx_output;
+}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
@@ -1691,22 +1778,15 @@
select_devices(adev, out->usecase);
if ((adev->mode == AUDIO_MODE_IN_CALL) &&
- !adev->voice.in_call &&
- (out == adev->primary_output)) {
- ret = voice_start_call(adev);
- } else if ((adev->mode == AUDIO_MODE_IN_CALL) &&
- adev->voice.in_call &&
- (out == adev->primary_output)) {
- voice_update_devices_for_all_voice_usecases(adev);
+ output_drives_call(adev, out)) {
+ adev->current_call_output = out;
+ if (!voice_is_in_call(adev))
+ ret = voice_start_call(adev);
+ else
+ voice_update_devices_for_all_voice_usecases(adev);
}
}
- if ((adev->mode == AUDIO_MODE_NORMAL) &&
- adev->voice.in_call &&
- (out == adev->primary_output)) {
- ret = voice_stop_call(adev);
- }
-
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&out->lock);
}
@@ -1895,7 +1975,10 @@
if (out->muted)
memset((void *)buffer, 0, bytes);
ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
- ret = pcm_write(out->pcm, (void *)buffer, bytes);
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY)
+ ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
+ else
+ ret = pcm_write(out->pcm, (void *)buffer, bytes);
if (ret < 0)
ret = -errno;
else if (ret == 0)
@@ -2243,7 +2326,7 @@
err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
if (err >= 0) {
val = atoi(value);
- if ((in->device != val) && (val != 0)) {
+ if (((int)in->device != val) && (val != 0)) {
in->device = val;
/* If recording is in progress, change the tx device to new device */
if (!in->standby && !in->is_st_session)
@@ -2332,6 +2415,8 @@
ret = audio_extn_ssr_read(stream, buffer, bytes);
else if (audio_extn_compr_cap_usecase_supported(in->usecase))
ret = audio_extn_compr_cap_read(in, buffer, bytes);
+ else if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY)
+ ret = pcm_mmap_read(in->pcm, buffer, bytes);
else
ret = pcm_read(in->pcm, buffer, bytes);
if (ret < 0)
@@ -2433,7 +2518,6 @@
struct stream_out *out;
int i, ret = 0;
audio_format_t format;
- int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
*stream_out = NULL;
@@ -2577,11 +2661,6 @@
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
}
- if (out->bit_width == 24 && !platform_check_24_bit_support()) {
- ALOGW("24 bit support is not enabled, using 16 bit backend");
- out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
- }
-
if (config->offload_info.format == AUDIO_FORMAT_FLAC)
out->compr_config.codec->options.flac_dec.sample_size = PCM_OUTPUT_BIT_WIDTH;
@@ -2606,6 +2685,28 @@
__func__, ret);
goto error_open;
}
+ } else if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
+ if (config->sample_rate == 0)
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
+ config->sample_rate != 8000) {
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ ret = -EINVAL;
+ goto error_open;
+ }
+ out->sample_rate = config->sample_rate;
+ out->config.rate = config->sample_rate;
+ if (config->format == AUDIO_FORMAT_DEFAULT)
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ ret = -EINVAL;
+ goto error_open;
+ }
+ out->format = config->format;
+ out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
+ out->config = pcm_config_afe_proxy_playback;
+ adev->voice_tx_output = out;
} else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
format = AUDIO_FORMAT_PCM_16_BIT;
out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
@@ -2619,20 +2720,11 @@
out->sample_rate = out->config.rate;
}
- if ((24 == out->bit_width) &&
- (devices == AUDIO_DEVICE_OUT_SPEAKER)) {
- sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
- ALOGI("%s 24-bit playback on Speaker is allowed ONLY at 48khz. Hence changing sample rate to: %d",
- __func__, sample_rate);
- } else {
- sample_rate = out->sample_rate;
- }
-
- ALOGV("%s flags %x, format %x, sample_rate %d, out->bit_width %d",
- __func__, flags, format, sample_rate, out->bit_width);
+ ALOGV("%s devices %d,flags %x, format %x, out->sample_rate %d, out->bit_width %d",
+ __func__, devices, flags, format, out->sample_rate, out->bit_width);
audio_extn_utils_update_stream_app_type_cfg(adev->platform,
&adev->streams_output_cfg_list,
- flags, format, sample_rate,
+ devices, flags, format, out->sample_rate,
out->bit_width, &out->app_type_cfg);
if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) ||
(flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
@@ -2725,6 +2817,10 @@
if (out->compr_config.codec != NULL)
free(out->compr_config.codec);
}
+
+ if (adev->voice_tx_output == out)
+ adev->voice_tx_output = NULL;
+
pthread_cond_destroy(&out->cond);
pthread_mutex_destroy(&out->lock);
free(stream);
@@ -2938,10 +3034,16 @@
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
struct audio_device *adev = (struct audio_device *)dev;
+
pthread_mutex_lock(&adev->lock);
if (adev->mode != mode) {
- ALOGD("%s mode %d\n", __func__, mode);
+ ALOGD("%s: mode %d\n", __func__, mode);
adev->mode = mode;
+ if ((mode == AUDIO_MODE_NORMAL || mode == AUDIO_MODE_IN_COMMUNICATION) &&
+ voice_is_in_call(adev)) {
+ voice_stop_call(adev);
+ adev->current_call_output = NULL;
+ }
}
pthread_mutex_unlock(&adev->lock);
return 0;
@@ -2970,7 +3072,8 @@
{
int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
- return get_input_buffer_size(config->sample_rate, config->format, channel_count);
+ return get_input_buffer_size(config->sample_rate, config->format, channel_count,
+ false /* is_low_latency: since we don't know, be conservative */);
}
static int adev_open_input_stream(struct audio_hw_device *dev,
@@ -2986,7 +3089,7 @@
struct stream_in *in;
int ret = 0, buffer_size, frame_size;
int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
-
+ bool is_low_latency = false;
*stream_in = NULL;
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
@@ -3030,11 +3133,39 @@
/* Update config params with the requested sample rate and channels */
in->usecase = USECASE_AUDIO_RECORD;
+ if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
+ (flags & AUDIO_INPUT_FLAG_FAST) != 0) {
+ is_low_latency = true;
+#if LOW_LATENCY_CAPTURE_USE_CASE
+ in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
+#endif
+ }
in->config = pcm_config_audio_capture;
in->config.rate = config->sample_rate;
in->format = config->format;
- if (channel_count == 6) {
+ if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
+ if (config->sample_rate == 0)
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
+ config->sample_rate != 8000) {
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ ret = -EINVAL;
+ goto err_open;
+ }
+ if (config->format == AUDIO_FORMAT_DEFAULT)
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ ret = -EINVAL;
+ goto err_open;
+ }
+
+ in->usecase = USECASE_AUDIO_RECORD_AFE_PROXY;
+ in->config = pcm_config_afe_proxy_record;
+ in->config.channels = channel_count;
+ in->config.rate = config->sample_rate;
+ } else if (channel_count == 6) {
if(audio_extn_ssr_get_enabled()) {
if(audio_extn_ssr_init(in)) {
ALOGE("%s: audio_extn_ssr_init failed", __func__);
@@ -3053,7 +3184,8 @@
frame_size = audio_stream_in_frame_size(&in->stream);
buffer_size = get_input_buffer_size(config->sample_rate,
config->format,
- channel_count);
+ channel_count,
+ is_low_latency);
in->config.period_size = buffer_size / frame_size;
}
@@ -3135,6 +3267,23 @@
return 0;
}
+/* This returns 1 if the input parameter looks at all plausible as a low latency period size,
+ * or 0 otherwise. A return value of 1 doesn't mean the value is guaranteed to work,
+ * just that it _might_ work.
+ */
+static int period_size_is_plausible_for_low_latency(int period_size)
+{
+ switch (period_size) {
+ case 160:
+ case 240:
+ case 320:
+ case 480:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
static int adev_open(const hw_module_t *module, const char *name,
hw_device_t **device)
{
@@ -3258,25 +3407,31 @@
&adev->streams_output_cfg_list);
audio_device_ref_count++;
+
+ char value[PROPERTY_VALUE_MAX];
+ int trial;
+ if (property_get("audio_hal.period_size", value, NULL) > 0) {
+ trial = atoi(value);
+ if (period_size_is_plausible_for_low_latency(trial)) {
+ pcm_config_low_latency.period_size = trial;
+ pcm_config_low_latency.start_threshold = trial / 4;
+ pcm_config_low_latency.avail_min = trial / 4;
+ configured_low_latency_capture_period_size = trial;
+ }
+ }
+ if (property_get("audio_hal.in_period_size", value, NULL) > 0) {
+ trial = atoi(value);
+ if (period_size_is_plausible_for_low_latency(trial)) {
+ configured_low_latency_capture_period_size = trial;
+ }
+ }
+
pthread_mutex_unlock(&adev_init_lock);
ALOGV("%s: exit", __func__);
return 0;
}
-int pcm_ioctl(struct pcm *pcm, int request, ...)
-{
- va_list ap;
- void * arg;
- int pcm_fd = *(int*)pcm;
-
- va_start(ap, request);
- arg = va_arg(ap, void *);
- va_end(ap);
-
- return ioctl(pcm_fd, request, arg);
-}
-
static struct hw_module_methods_t hal_module_methods = {
.open = adev_open,
};
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index bdb9aae..d05f743 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -32,9 +32,6 @@
#define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"
-#define BT_SCO_SAMPLE_RATE "bt-sco-samplerate"
-#define BT_SCO_WB_SAMPLE_RATE "bt-sco-wb-samplerate"
-
/* Flags used to initialize acdb_settings variable that goes to ACDB library */
#define NONE_FLAG 0x00000000
#define ANC_FLAG 0x00000001
@@ -114,6 +111,10 @@
USECASE_AUDIO_SPKR_CALIB_RX,
USECASE_AUDIO_SPKR_CALIB_TX,
+
+ USECASE_AUDIO_PLAYBACK_AFE_PROXY,
+ USECASE_AUDIO_RECORD_AFE_PROXY,
+
AUDIO_USECASE_MAX
};
@@ -203,7 +204,7 @@
int standby;
int source;
int pcm_device_id;
- int device;
+ audio_devices_t device;
audio_channel_mask_t channel_mask;
audio_usecase_t usecase;
bool enable_aec;
@@ -269,6 +270,8 @@
audio_devices_t out_device;
struct stream_in *active_input;
struct stream_out *primary_output;
+ struct stream_out *voice_tx_output;
+ struct stream_out *current_call_output;
bool bluetooth_nrec;
bool screen_off;
int *snd_dev_ref_cnt;
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 005c0b8..b978ae3 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -282,15 +282,15 @@
static int acdb_device_table[SND_DEVICE_MAX] = {
[SND_DEVICE_NONE] = -1,
[SND_DEVICE_OUT_HANDSET] = 7,
- [SND_DEVICE_OUT_SPEAKER] = 131,
- [SND_DEVICE_OUT_SPEAKER_REVERSE] = 131,
+ [SND_DEVICE_OUT_SPEAKER] = 14,
+ [SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
[SND_DEVICE_OUT_HEADPHONES] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
[SND_DEVICE_OUT_VOICE_HANDSET] = 7,
[SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
[SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
[SND_DEVICE_OUT_HDMI] = 18,
- [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 131,
+ [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
[SND_DEVICE_OUT_BT_SCO] = 22,
[SND_DEVICE_OUT_BT_SCO_WB] = 39,
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
@@ -298,7 +298,7 @@
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
[SND_DEVICE_OUT_AFE_PROXY] = 0,
[SND_DEVICE_OUT_USB_HEADSET] = 45,
- [SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 131,
+ [SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 14,
[SND_DEVICE_OUT_TRANSMISSION_FM] = 0,
[SND_DEVICE_OUT_ANC_HEADSET] = 26,
[SND_DEVICE_OUT_ANC_FB_HEADSET] = 27,
@@ -765,7 +765,7 @@
struct platform_data *my_data = NULL;
int retry_num = 0, snd_card_num = 0;
const char *snd_card_name;
- char mixer_xml_path[100];
+ char mixer_xml_path[100],ffspEnable[PROPERTY_VALUE_MAX];
char *cvd_version = NULL;
my_data = calloc(1, sizeof(struct platform_data));
@@ -867,7 +867,13 @@
my_data->fluence_mode = FLUENCE_BROADSIDE;
}
}
-
+ property_get("persist.audio.FFSP.enable", ffspEnable, "");
+ if (!strncmp("true", ffspEnable, sizeof("true"))) {
+ acdb_device_table[SND_DEVICE_OUT_SPEAKER] = 131;
+ acdb_device_table[SND_DEVICE_OUT_SPEAKER_REVERSE] = 131;
+ acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 131;
+ acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 131;
+ }
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) {
@@ -2375,10 +2381,6 @@
return fragment_size;
}
-bool platform_check_24_bit_support() {
- return false;
-}
-
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev __unused,
struct audio_usecase *usecase __unused)
{
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index dec9eda..e38503f 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -164,6 +164,10 @@
#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
+#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
+#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
+#define LOW_LATENCY_CAPTURE_USE_CASE 0
+
#define HDMI_MULTI_PERIOD_SIZE 336
#define HDMI_MULTI_PERIOD_COUNT 8
#define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index e2c30f4..eea10da 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1030,10 +1030,6 @@
return false;
}
-bool platform_check_24_bit_support() {
- return false;
-}
-
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev __unused,
struct audio_usecase *usecase __unused)
{
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index 20984d8..950ea84 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -138,4 +138,11 @@
/* Define macro for Internal FM volume mixer */
#define FM_RX_VOLUME "Internal FM RX Volume"
+#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
+#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
+#define LOW_LATENCY_CAPTURE_USE_CASE 0
+
+#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
+#define AFE_PROXY_RECORD_PCM_DEVICE 8
+
#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index de1c04d..f034ee1 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -220,6 +220,12 @@
INCALL_MUSIC_UPLINK2_PCM_DEVICE},
[USECASE_AUDIO_SPKR_CALIB_RX] = {SPKR_PROT_CALIB_RX_PCM_DEVICE, -1},
[USECASE_AUDIO_SPKR_CALIB_TX] = {-1, SPKR_PROT_CALIB_TX_PCM_DEVICE},
+
+ [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = {AFE_PROXY_PLAYBACK_PCM_DEVICE,
+ AFE_PROXY_RECORD_PCM_DEVICE},
+ [USECASE_AUDIO_RECORD_AFE_PROXY] = {AFE_PROXY_PLAYBACK_PCM_DEVICE,
+ AFE_PROXY_RECORD_PCM_DEVICE},
+
};
/* Array to store sound devices */
@@ -245,6 +251,7 @@
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
+ [SND_DEVICE_OUT_VOICE_TX] = "voice-tx",
[SND_DEVICE_OUT_AFE_PROXY] = "afe-proxy",
[SND_DEVICE_OUT_USB_HEADSET] = "usb-headphones",
[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = "speaker-and-usb-headphones",
@@ -292,6 +299,8 @@
[SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC] = "voice-tty-full-headset-mic",
[SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC] = "voice-tty-vco-handset-mic",
[SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC] = "voice-tty-hco-headset-mic",
+ [SND_DEVICE_IN_VOICE_RX] = "voice-rx",
+
[SND_DEVICE_IN_VOICE_REC_MIC] = "voice-rec-mic",
[SND_DEVICE_IN_VOICE_REC_MIC_NS] = "voice-rec-mic",
[SND_DEVICE_IN_VOICE_REC_DMIC_STEREO] = "voice-rec-dmic-ef",
@@ -332,6 +341,7 @@
[SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
[SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
[SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
+ [SND_DEVICE_OUT_VOICE_TX] = 45,
[SND_DEVICE_OUT_AFE_PROXY] = 0,
[SND_DEVICE_OUT_USB_HEADSET] = 45,
[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 14,
@@ -378,6 +388,8 @@
[SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC] = 16,
[SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC] = 36,
[SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC] = 16,
+ [SND_DEVICE_IN_VOICE_RX] = 44,
+
[SND_DEVICE_IN_VOICE_REC_MIC] = 4,
[SND_DEVICE_IN_VOICE_REC_MIC_NS] = 107,
[SND_DEVICE_IN_VOICE_REC_DMIC_STEREO] = 34,
@@ -716,6 +728,9 @@
backend_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("bt-sco-wb");
backend_table[SND_DEVICE_OUT_HDMI] = strdup("hdmi");
backend_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("speaker-and-hdmi");
+ backend_table[SND_DEVICE_OUT_VOICE_TX] = strdup("afe-proxy");
+ backend_table[SND_DEVICE_IN_VOICE_RX] = strdup("afe-proxy");
+
backend_table[SND_DEVICE_OUT_AFE_PROXY] = strdup("afe-proxy");
backend_table[SND_DEVICE_OUT_USB_HEADSET] = strdup("usb-headphones");
backend_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] =
@@ -1648,7 +1663,9 @@
snd_device = SND_DEVICE_OUT_ANC_HANDSET;
else
snd_device = SND_DEVICE_OUT_VOICE_HANDSET;
- }
+ } else if (devices & AUDIO_DEVICE_OUT_TELEPHONY_TX)
+ snd_device = SND_DEVICE_OUT_VOICE_TX;
+
if (snd_device != SND_DEVICE_NONE) {
goto exit;
}
@@ -1805,7 +1822,8 @@
if (audio_extn_hfp_is_active(adev))
platform_set_echo_reference(adev->platform, true);
}
- }
+ } else if (out_device & AUDIO_DEVICE_OUT_TELEPHONY_TX)
+ snd_device = SND_DEVICE_IN_VOICE_RX;
} else if (source == AUDIO_SOURCE_CAMCORDER) {
if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC ||
in_device & AUDIO_DEVICE_IN_BACK_MIC) {
@@ -2520,17 +2538,6 @@
return fragment_size;
}
-bool platform_check_24_bit_support() {
-
- char value[PROPERTY_VALUE_MAX] = {0};
- property_get("audio.offload.24bit.enable", value, "0");
- if (atoi(value)) {
- ALOGW("Property audio.offload.24bit.enable is set");
- return true;
- }
- return false;
-}
-
int platform_set_codec_backend_cfg(struct audio_device* adev,
unsigned int bit_width, unsigned int sample_rate)
{
@@ -2659,9 +2666,11 @@
}
}
- // 16 bit playbacks is allowed through 16 bit/48 khz backend only
+ // 24 bit playback on speakers and all 16 bit playbacks is allowed through
+ // 16 bit/48 khz backend only
if ((16 == bit_width) ||
- ((24 == bit_width) && (SND_DEVICE_OUT_SPEAKER == usecase->devices))) {
+ ((24 == bit_width) &&
+ (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER))) {
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
// Force routing if the expected bitwdith or samplerate
@@ -2680,12 +2689,6 @@
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev, struct audio_usecase *usecase)
{
- // check if 24bit configuration is enabled first
- if (!platform_check_24_bit_support()) {
- ALOGW("24bit not enable, no need to check for backend change");
- return false;
- }
-
ALOGV("platform_check_and_set_codec_backend_cfg usecase = %d",usecase->id );
unsigned int new_bit_width, old_bit_width;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 62aa40a..831ee58 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -70,6 +70,7 @@
SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
+ SND_DEVICE_OUT_VOICE_TX,
SND_DEVICE_OUT_AFE_PROXY,
SND_DEVICE_OUT_USB_HEADSET,
SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET,
@@ -127,6 +128,7 @@
SND_DEVICE_IN_VOICE_REC_MIC_NS,
SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
+ SND_DEVICE_IN_VOICE_RX,
SND_DEVICE_IN_USB_HEADSET_MIC,
SND_DEVICE_IN_CAPTURE_FM,
SND_DEVICE_IN_AANC_HANDSET_MIC,
@@ -178,6 +180,10 @@
#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
#define AUDIO_CAPTURE_PERIOD_COUNT 2
+#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
+#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
+#define LOW_LATENCY_CAPTURE_USE_CASE 1
+
#define DEVICE_NAME_MAX_SIZE 128
#define HW_INFO_ARRAY_MAX_SIZE 32
@@ -278,6 +284,9 @@
#define VOWLAN_CALL_PCM_DEVICE 36
#endif
+#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
+#define AFE_PROXY_RECORD_PCM_DEVICE 8
+
#ifdef PLATFORM_MSM8x26
#define HFP_SCO_RX 28
#define HFP_ASM_RX_TX 29
diff --git a/hal/platform_api.h b/hal/platform_api.h
index b11a0c7..520b27b 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -89,7 +89,6 @@
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info);
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev, struct audio_usecase *usecase);
-bool platform_check_24_bit_support();
int platform_get_usecase_index(const char * usecase);
int platform_set_usecase_pcm_id(audio_usecase_t usecase, int32_t type, int32_t pcm_id);
void platform_set_echo_reference(void *platform, bool enable);
diff --git a/hal/voice.c b/hal/voice.c
index ecad7a7..c4fa163 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -131,8 +131,8 @@
uc_info->id = usecase_id;
uc_info->type = VOICE_CALL;
- uc_info->stream.out = adev->primary_output;
- uc_info->devices = adev->primary_output->devices;
+ uc_info->stream.out = adev->current_call_output ;
+ uc_info->devices = adev->current_call_output ->devices;
uc_info->in_snd_device = SND_DEVICE_NONE;
uc_info->out_snd_device = SND_DEVICE_NONE;
@@ -213,6 +213,11 @@
return call_state;
}
+bool voice_is_in_call(struct audio_device *adev)
+{
+ return adev->voice.in_call;
+}
+
bool voice_is_in_call_rec_stream(struct stream_in *in)
{
bool in_call_rec = false;
@@ -494,6 +499,7 @@
if (usecase->type == VOICE_CALL) {
ALOGV("%s: updating device for usecase:%s", __func__,
use_case_table[usecase->id]);
+ usecase->stream.out = adev->current_call_output;
select_devices(adev, usecase->id);
}
}
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index 8952522..e5b979f 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -299,7 +299,8 @@
* occured, otherwise voice calls will be started unintendedly on
* speaker.
*/
- if (is_call_active || adev->voice.in_call) {
+ if (is_call_active ||
+ (adev->voice.in_call && adev->mode == AUDIO_MODE_IN_CALL)) {
/* Device routing is not triggered for voice calls on the subsequent
* subs, Hence update the call states if voice call is already
* active on other sub.