audio: Enable 24 bit packed direct pcm support.
-Add support for 24 bit packed audio in audio hal.
-Disable gapless for AV playback and direct pcm usecase,
this ensures that the buffering in DSP is not more.
-Simulate rendered time stamp for direct pcm usecase
based on the number of frames written to the compress
driver, bufferring in the driver and DSP latency.
-Pass mixer instance to offload effects library to avoid
an unnecessary mixer_open call, this optimizes audio
start delay in compress playback.
Change-Id: I422a53af5632eaf6cc362a6c44f62ff8412965f7
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 8486e18..c53f124 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -74,6 +74,8 @@
#include "sound/asound.h"
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+/*DIRECT PCM has same buffer sizes as DEEP Buffer*/
+#define DIRECT_PCM_NUM_FRAGMENTS 2
/* ToDo: Check and update a proper value in msec */
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
@@ -287,17 +289,17 @@
return ret_val;
}
-static int check_and_set_gapless_mode(struct audio_device *adev) {
-
-
- char value[PROPERTY_VALUE_MAX] = {0};
+static int check_and_set_gapless_mode(struct audio_device *adev, bool enable_gapless)
+{
bool gapless_enabled = false;
const char *mixer_ctl_name = "Compress Gapless Playback";
struct mixer_ctl *ctl;
ALOGV("%s:", __func__);
- property_get("audio.offload.gapless.enabled", value, NULL);
- gapless_enabled = atoi(value) || !strncmp("true", value, 4);
+ gapless_enabled = property_get_bool("audio.offload.gapless.enabled", false);
+
+ /*Disable gapless if its AV playback*/
+ gapless_enabled = gapless_enabled && enable_gapless;
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
@@ -323,8 +325,8 @@
format == AUDIO_FORMAT_AAC_ADTS_LC ||
format == AUDIO_FORMAT_AAC_ADTS_HE_V1 ||
format == AUDIO_FORMAT_AAC_ADTS_HE_V2 ||
- format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD ||
- format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD ||
+ format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+ format == AUDIO_FORMAT_PCM_8_24_BIT ||
format == AUDIO_FORMAT_PCM_16_BIT ||
format == AUDIO_FORMAT_FLAC ||
format == AUDIO_FORMAT_ALAC ||
@@ -351,7 +353,6 @@
case AUDIO_FORMAT_AAC_ADTS:
id = SND_AUDIOCODEC_AAC;
break;
- case AUDIO_FORMAT_PCM_OFFLOAD:
case AUDIO_FORMAT_PCM:
id = SND_AUDIOCODEC_PCM;
break;
@@ -1956,7 +1957,7 @@
for the default max poll time (20s) in the event of an SSR.
Reduce the poll time to observe and deal with SSR faster.
*/
- if (out->use_small_bufs) {
+ if (!out->non_blocking) {
compress_set_max_poll_wait(out->compr, 1000);
}
@@ -1973,7 +1974,7 @@
if (adev->visualizer_start_output != NULL)
adev->visualizer_start_output(out->handle, out->pcm_device_id);
if (adev->offload_effects_start_output != NULL)
- adev->offload_effects_start_output(out->handle, out->pcm_device_id);
+ adev->offload_effects_start_output(out->handle, out->pcm_device_id, adev->mixer);
audio_extn_check_and_set_dts_hpx_state(adev);
}
}
@@ -2061,6 +2062,37 @@
return size;
}
+static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out)
+{
+ uint64_t actual_frames_rendered = 0;
+ size_t kernel_buffer_size = out->compr_config.fragment_size * out->compr_config.fragments;
+
+ /* This adjustment accounts for buffering after app processor.
+ * It is based on estimated DSP latency per use case, rather than exact.
+ */
+ int64_t platform_latency = platform_render_latency(out->usecase) *
+ out->sample_rate / 1000000LL;
+
+ /* not querying actual state of buffering in kernel as it would involve an ioctl call
+ * which then needs protection, this causes delay in TS query for pcm_offload usecase
+ * hence only estimate.
+ */
+ int64_t signed_frames = out->written - kernel_buffer_size;
+
+ signed_frames = signed_frames / (audio_bytes_per_sample(out->format) * popcount(out->channel_mask)) - platform_latency;
+
+ if (signed_frames > 0)
+ actual_frames_rendered = signed_frames;
+
+ ALOGVV("%s signed frames %lld out_written %lld kernel_buffer_size %d"
+ "bytes/sample %zu channel count %d", __func__,(long long int)signed_frames,
+ (long long int)out->written, (int)kernel_buffer_size,
+ audio_bytes_per_sample(out->compr_config.codec->format),
+ popcount(out->channel_mask));
+
+ return actual_frames_rendered;
+}
+
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
@@ -2553,6 +2585,9 @@
out_standby(&out->stream.common);
return ret;
}
+ if ( ret == (ssize_t)bytes && !out->non_blocking)
+ out->written += bytes;
+
if (!out->playback_started && ret >= 0) {
compress_start(out->compr);
audio_extn_dts_eagle_fade(adev, true, out);
@@ -2627,14 +2662,24 @@
*dsp_frames = 0;
if (is_offload_usecase(out->usecase)) {
ssize_t ret = 0;
+
+ /* Below piece of code is not guarded against any lock beacuse audioFliner serializes
+ * this operation and adev_close_output_stream(where out gets reset).
+ */
+ if (!out->non_blocking && (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
+ *dsp_frames = get_actual_pcm_frames_rendered(out);
+ ALOGVV("dsp_frames %d sampleRate %d",(int)*dsp_frames,out->sample_rate);
+ return 0;
+ }
+
lock_output_stream(out);
- if (out->compr != NULL) {
+ if (out->compr != NULL && out->non_blocking) {
ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
&out->sample_rate);
if (ret < 0)
ret = -errno;
ALOGVV("%s rendered frames %d sample_rate %d",
- __func__, *dsp_frames, out->sample_rate);
+ __func__, *dsp_frames, out->sample_rate);
}
pthread_mutex_unlock(&out->lock);
if (-ENETRESET == ret) {
@@ -2686,27 +2731,37 @@
int ret = -1;
unsigned long dsp_frames;
+ /* below piece of code is not guarded against any lock because audioFliner serializes
+ * this operation and adev_close_output_stream( where out gets reset).
+ */
+ if (is_offload_usecase(out->usecase) && !out->non_blocking &&
+ (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
+ *frames = get_actual_pcm_frames_rendered(out);
+ /* this is the best we can do */
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ ALOGVV("frames %lld playedat %lld",(long long int)*frames,
+ timestamp->tv_sec * 1000000LL + timestamp->tv_nsec / 1000);
+ return 0;
+ }
+
lock_output_stream(out);
- if (is_offload_usecase(out->usecase)) {
- if (out->compr != NULL) {
- ret = compress_get_tstamp(out->compr, &dsp_frames,
- &out->sample_rate);
- ALOGVV("%s rendered frames %ld sample_rate %d",
- __func__, dsp_frames, out->sample_rate);
- *frames = dsp_frames;
- if (ret < 0)
- ret = -errno;
- if (-ENETRESET == ret) {
- ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
- set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
- ret = -EINVAL;
- } else
- ret = 0;
-
- /* this is the best we can do */
- clock_gettime(CLOCK_MONOTONIC, timestamp);
- }
+ if (is_offload_usecase(out->usecase) && out->compr != NULL && out->non_blocking) {
+ ret = compress_get_tstamp(out->compr, &dsp_frames,
+ &out->sample_rate);
+ ALOGVV("%s rendered frames %ld sample_rate %d",
+ __func__, dsp_frames, out->sample_rate);
+ *frames = dsp_frames;
+ if (ret < 0)
+ ret = -errno;
+ if (-ENETRESET == ret) {
+ ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+ ret = -EINVAL;
+ } else
+ ret = 0;
+ /* this is the best we can do */
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
} else {
if (out->pcm) {
unsigned int avail;
@@ -2837,6 +2892,7 @@
ALOGD("copl(%p):calling compress flush", out);
lock_output_stream(out);
stop_compressed_output_l(out);
+ out->written = 0;
pthread_mutex_unlock(&out->lock);
ALOGD("copl(%p):out of compress flush", out);
return 0;
@@ -3238,7 +3294,6 @@
out->handle = handle;
out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
out->non_blocking = 0;
- out->use_small_bufs = false;
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
(flags & AUDIO_OUTPUT_FLAG_DIRECT)) {
@@ -3337,6 +3392,9 @@
}
if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ out->stream.pause = out_pause;
+ out->stream.flush = out_flush;
+ out->stream.resume = out_resume;
out->usecase = get_offload_usecase(adev, true);
ALOGV("DIRECT_PCM usecase ... usecase selected %d ", out->usecase);
} else {
@@ -3381,18 +3439,19 @@
out->compr_config.codec->id =
get_snd_codec_id(config->offload_info.format);
- if (((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD)||
- ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM)) {
+ if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
out->compr_config.fragment_size =
platform_get_pcm_offload_buffer_size(&config->offload_info);
+ out->compr_config.fragments = DIRECT_PCM_NUM_FRAGMENTS;
} else if (audio_extn_dolby_is_passthrough_stream(out)) {
out->compr_config.fragment_size =
audio_extn_dolby_get_passt_buffer_size(&config->offload_info);
+ out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
} else {
out->compr_config.fragment_size =
platform_get_compress_offload_buffer_size(&config->offload_info);
+ out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
}
- out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
out->compr_config.codec->sample_rate =
config->offload_info.sample_rate;
out->compr_config.codec->bit_rate =
@@ -3408,16 +3467,12 @@
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS)
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
- if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD)
- out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
- if (config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD)
- out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT)
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
-
- if (out->bit_width == 24) {
+ if (config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_3LE;
+ if (config->offload_info.format == AUDIO_FORMAT_PCM_8_24_BIT)
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
- }
if (config->offload_info.format == AUDIO_FORMAT_FLAC)
out->compr_config.codec->options.flac_dec.sample_size = AUDIO_OUTPUT_BIT_WIDTH;
@@ -3425,14 +3480,6 @@
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
- if (platform_use_small_buffer(&config->offload_info)) {
- //this flag is set from framework only if its for PCM formats
- //no need to check for PCM format again
- out->non_blocking = 0;
- out->use_small_bufs = true;
- ALOGI("Keep write blocking for small buff: non_blockling %d",
- out->non_blocking);
- }
out->send_new_metadata = 1;
out->send_next_track_params = false;
@@ -3446,11 +3493,18 @@
ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
__func__, config->offload_info.version,
config->offload_info.bit_rate);
- //Decide if we need to use gapless mode by default
- if (!audio_extn_dolby_is_passthrough_stream(out)) {
- ALOGV("%s: don't enable gapless for passthrough", __func__);
- check_and_set_gapless_mode(adev);
- }
+
+ /* Disable gapless if any of the following is true
+ * passthrough playback
+ * AV playback
+ * Direct PCM playback
+ */
+ if (audio_extn_dolby_is_passthrough_stream(out) ||
+ config->offload_info.has_video ||
+ out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ check_and_set_gapless_mode(adev, false);
+ } else
+ check_and_set_gapless_mode(adev, true);
if (audio_extn_dolby_is_passthrough_stream(out)) {
out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
@@ -4288,7 +4342,7 @@
ALOGV("%s: DLOPEN successful for %s", __func__,
OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
adev->offload_effects_start_output =
- (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
+ (int (*)(audio_io_handle_t, int, struct mixer *))dlsym(adev->offload_effects_lib,
"offload_effects_bundle_hal_start_output");
adev->offload_effects_stop_output =
(int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,