[FFmpeg-cvslog] ffplay: implement separete audio decoder thread
Marton Balint
git at videolan.org
Sun Nov 9 16:43:14 CET 2014
ffmpeg | branch: master | Marton Balint <cus at passwd.hu> | Sat Oct 19 14:34:28 2013 +0200| [631ac655c00e978e19d05dab572bc1ffd6078c63] | committer: Marton Balint
ffplay: implement separete audio decoder thread
Signed-off-by: Marton Balint <cus at passwd.hu>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=631ac655c00e978e19d05dab572bc1ffd6078c63
---
ffplay.c | 265 ++++++++++++++++++++++++++++++++++++--------------------------
1 file changed, 153 insertions(+), 112 deletions(-)
diff --git a/ffplay.c b/ffplay.c
index a979164..24bcae2 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -121,7 +121,8 @@ typedef struct PacketQueue {
#define VIDEO_PICTURE_QUEUE_SIZE 3
#define SUBPICTURE_QUEUE_SIZE 16
-#define FRAME_QUEUE_SIZE FFMAX(VIDEO_PICTURE_QUEUE_SIZE, SUBPICTURE_QUEUE_SIZE)
+#define SAMPLE_QUEUE_SIZE 9
+#define FRAME_QUEUE_SIZE FFMAX(SAMPLE_QUEUE_SIZE, FFMAX(VIDEO_PICTURE_QUEUE_SIZE, SUBPICTURE_QUEUE_SIZE))
typedef struct AudioParams {
int freq;
@@ -196,6 +197,7 @@ typedef struct Decoder {
typedef struct VideoState {
SDL_Thread *read_tid;
SDL_Thread *video_tid;
+ SDL_Thread *audio_tid;
AVInputFormat *iformat;
int no_background;
int abort_request;
@@ -217,6 +219,7 @@ typedef struct VideoState {
FrameQueue pictq;
FrameQueue subpq;
+ FrameQueue sampq;
Decoder auddec;
Decoder viddec;
@@ -242,8 +245,6 @@ typedef struct VideoState {
unsigned int audio_buf1_size;
int audio_buf_index; /* in bytes */
int audio_write_buf_size;
- int audio_buf_frames_pending;
- int audio_last_serial;
struct AudioParams audio_src;
#if CONFIG_AVFILTER
struct AudioParams audio_filter_src;
@@ -252,7 +253,6 @@ typedef struct VideoState {
struct SwrContext *swr_ctx;
int frame_drops_early;
int frame_drops_late;
- AVFrame *frame;
enum ShowMode {
SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
@@ -712,12 +712,29 @@ static Frame *frame_queue_peek_writable(FrameQueue *f)
return &f->queue[f->windex];
}
+static Frame *frame_queue_peek_readable(FrameQueue *f)
+{
+ /* wait until we have a readable a new frame */
+ SDL_LockMutex(f->mutex);
+ while (f->size - f->rindex_shown <= 0 &&
+ !f->pktq->abort_request) {
+ SDL_CondWait(f->cond, f->mutex);
+ }
+ SDL_UnlockMutex(f->mutex);
+
+ if (f->pktq->abort_request)
+ return NULL;
+
+ return &f->queue[(f->rindex + f->rindex_shown) % f->max_size];
+}
+
static void frame_queue_push(FrameQueue *f)
{
if (++f->windex == f->max_size)
f->windex = 0;
SDL_LockMutex(f->mutex);
f->size++;
+ SDL_CondSignal(f->cond);
SDL_UnlockMutex(f->mutex);
}
@@ -1280,6 +1297,7 @@ static void stream_close(VideoState *is)
/* free all pictures */
frame_queue_destory(&is->pictq);
+ frame_queue_destory(&is->sampq);
frame_queue_destory(&is->subpq);
SDL_DestroyCond(is->continue_read_thread);
#if !CONFIG_AVFILTER
@@ -2100,6 +2118,93 @@ end:
}
#endif /* CONFIG_AVFILTER */
+static int audio_thread(void *arg)
+{
+ VideoState *is = arg;
+ AVFrame *frame = av_frame_alloc();
+ Frame *af;
+#if CONFIG_AVFILTER
+ int last_serial = -1;
+ int64_t dec_channel_layout;
+ int reconfigure;
+#endif
+ int got_frame = 0;
+ AVRational tb;
+ int ret = 0;
+
+ if (!frame)
+ return AVERROR(ENOMEM);
+
+ do {
+ if ((got_frame = decoder_decode_frame(&is->auddec, frame, NULL)) < 0)
+ goto the_end;
+
+ if (got_frame) {
+ tb = (AVRational){1, frame->sample_rate};
+
+#if CONFIG_AVFILTER
+ dec_channel_layout = get_valid_channel_layout(frame->channel_layout, av_frame_get_channels(frame));
+
+ reconfigure =
+ cmp_audio_fmts(is->audio_filter_src.fmt, is->audio_filter_src.channels,
+ frame->format, av_frame_get_channels(frame)) ||
+ is->audio_filter_src.channel_layout != dec_channel_layout ||
+ is->audio_filter_src.freq != frame->sample_rate ||
+ is->auddec.pkt_serial != last_serial;
+
+ if (reconfigure) {
+ char buf1[1024], buf2[1024];
+ av_get_channel_layout_string(buf1, sizeof(buf1), -1, is->audio_filter_src.channel_layout);
+ av_get_channel_layout_string(buf2, sizeof(buf2), -1, dec_channel_layout);
+ av_log(NULL, AV_LOG_DEBUG,
+ "Audio frame changed from rate:%d ch:%d fmt:%s layout:%s serial:%d to rate:%d ch:%d fmt:%s layout:%s serial:%d\n",
+ is->audio_filter_src.freq, is->audio_filter_src.channels, av_get_sample_fmt_name(is->audio_filter_src.fmt), buf1, last_serial,
+ frame->sample_rate, av_frame_get_channels(frame), av_get_sample_fmt_name(frame->format), buf2, is->auddec.pkt_serial);
+
+ is->audio_filter_src.fmt = frame->format;
+ is->audio_filter_src.channels = av_frame_get_channels(frame);
+ is->audio_filter_src.channel_layout = dec_channel_layout;
+ is->audio_filter_src.freq = frame->sample_rate;
+ last_serial = is->auddec.pkt_serial;
+
+ if ((ret = configure_audio_filters(is, afilters, 1)) < 0)
+ goto the_end;
+ }
+
+ if ((ret = av_buffersrc_add_frame(is->in_audio_filter, frame)) < 0)
+ goto the_end;
+
+ while ((ret = av_buffersink_get_frame_flags(is->out_audio_filter, frame, 0)) >= 0) {
+ tb = is->out_audio_filter->inputs[0]->time_base;
+#endif
+ if (!(af = frame_queue_peek_writable(&is->sampq)))
+ goto the_end;
+
+ af->pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(tb);
+ af->pos = av_frame_get_pkt_pos(frame);
+ af->serial = is->auddec.pkt_serial;
+ af->duration = av_q2d((AVRational){frame->nb_samples, frame->sample_rate});
+
+ av_frame_move_ref(af->frame, frame);
+ frame_queue_push(&is->sampq);
+
+#if CONFIG_AVFILTER
+ if (is->audioq.serial != is->auddec.pkt_serial)
+ break;
+ }
+ if (ret == AVERROR_EOF)
+ is->auddec.finished = is->auddec.pkt_serial;
+#endif
+ }
+ } while (ret >= 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF);
+ the_end:
+#if CONFIG_AVFILTER
+ avfilter_graph_free(&is->agraph);
+#endif
+ av_frame_free(&frame);
+ return ret;
+}
+
static int video_thread(void *arg)
{
VideoState *is = arg;
@@ -2315,135 +2420,77 @@ static int audio_decode_frame(VideoState *is)
{
int data_size, resampled_data_size;
int64_t dec_channel_layout;
- int got_frame = 0;
av_unused double audio_clock0;
int wanted_nb_samples;
- AVRational tb;
- int ret;
- int reconfigure;
-
- if (!is->frame)
- if (!(is->frame = av_frame_alloc()))
- return AVERROR(ENOMEM);
-
- for (;;) {
- if (is->audioq.serial != is->auddec.pkt_serial)
- is->audio_buf_frames_pending = got_frame = 0;
-
- if (!got_frame)
- av_frame_unref(is->frame);
+ Frame *af;
+ {
if (is->paused)
return -1;
- while (is->audio_buf_frames_pending || got_frame) {
- if (!is->audio_buf_frames_pending) {
- got_frame = 0;
- tb = (AVRational){1, is->frame->sample_rate};
-
-#if CONFIG_AVFILTER
- dec_channel_layout = get_valid_channel_layout(is->frame->channel_layout, av_frame_get_channels(is->frame));
-
- reconfigure =
- cmp_audio_fmts(is->audio_filter_src.fmt, is->audio_filter_src.channels,
- is->frame->format, av_frame_get_channels(is->frame)) ||
- is->audio_filter_src.channel_layout != dec_channel_layout ||
- is->audio_filter_src.freq != is->frame->sample_rate ||
- is->auddec.pkt_serial != is->audio_last_serial;
-
- if (reconfigure) {
- char buf1[1024], buf2[1024];
- av_get_channel_layout_string(buf1, sizeof(buf1), -1, is->audio_filter_src.channel_layout);
- av_get_channel_layout_string(buf2, sizeof(buf2), -1, dec_channel_layout);
- av_log(NULL, AV_LOG_DEBUG,
- "Audio frame changed from rate:%d ch:%d fmt:%s layout:%s serial:%d to rate:%d ch:%d fmt:%s layout:%s serial:%d\n",
- is->audio_filter_src.freq, is->audio_filter_src.channels, av_get_sample_fmt_name(is->audio_filter_src.fmt), buf1, is->audio_last_serial,
- is->frame->sample_rate, av_frame_get_channels(is->frame), av_get_sample_fmt_name(is->frame->format), buf2, is->auddec.pkt_serial);
-
- is->audio_filter_src.fmt = is->frame->format;
- is->audio_filter_src.channels = av_frame_get_channels(is->frame);
- is->audio_filter_src.channel_layout = dec_channel_layout;
- is->audio_filter_src.freq = is->frame->sample_rate;
- is->audio_last_serial = is->auddec.pkt_serial;
-
- if ((ret = configure_audio_filters(is, afilters, 1)) < 0)
- return ret;
- }
-
- if ((ret = av_buffersrc_add_frame(is->in_audio_filter, is->frame)) < 0)
- return ret;
-#endif
- }
-#if CONFIG_AVFILTER
- if ((ret = av_buffersink_get_frame_flags(is->out_audio_filter, is->frame, 0)) < 0) {
- if (ret == AVERROR(EAGAIN)) {
- is->audio_buf_frames_pending = 0;
- continue;
- }
- if (ret == AVERROR_EOF)
- is->auddec.finished = is->auddec.pkt_serial;
- return ret;
- }
- is->audio_buf_frames_pending = 1;
- tb = is->out_audio_filter->inputs[0]->time_base;
-#endif
+ do {
+ if (!(af = frame_queue_peek_readable(&is->sampq)))
+ return -1;
+ frame_queue_next(&is->sampq);
+ } while (af->serial != is->audioq.serial);
- data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(is->frame),
- is->frame->nb_samples,
- is->frame->format, 1);
+ {
+ data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(af->frame),
+ af->frame->nb_samples,
+ af->frame->format, 1);
dec_channel_layout =
- (is->frame->channel_layout && av_frame_get_channels(is->frame) == av_get_channel_layout_nb_channels(is->frame->channel_layout)) ?
- is->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(is->frame));
- wanted_nb_samples = synchronize_audio(is, is->frame->nb_samples);
+ (af->frame->channel_layout && av_frame_get_channels(af->frame) == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
+ af->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af->frame));
+ wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
- if (is->frame->format != is->audio_src.fmt ||
+ if (af->frame->format != is->audio_src.fmt ||
dec_channel_layout != is->audio_src.channel_layout ||
- is->frame->sample_rate != is->audio_src.freq ||
- (wanted_nb_samples != is->frame->nb_samples && !is->swr_ctx)) {
+ af->frame->sample_rate != is->audio_src.freq ||
+ (wanted_nb_samples != af->frame->nb_samples && !is->swr_ctx)) {
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc_set_opts(NULL,
is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
- dec_channel_layout, is->frame->format, is->frame->sample_rate,
+ dec_channel_layout, af->frame->format, af->frame->sample_rate,
0, NULL);
if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
av_log(NULL, AV_LOG_ERROR,
"Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
- is->frame->sample_rate, av_get_sample_fmt_name(is->frame->format), av_frame_get_channels(is->frame),
+ af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), av_frame_get_channels(af->frame),
is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
swr_free(&is->swr_ctx);
- break;
+ return -1;
}
is->audio_src.channel_layout = dec_channel_layout;
- is->audio_src.channels = av_frame_get_channels(is->frame);
- is->audio_src.freq = is->frame->sample_rate;
- is->audio_src.fmt = is->frame->format;
+ is->audio_src.channels = av_frame_get_channels(af->frame);
+ is->audio_src.freq = af->frame->sample_rate;
+ is->audio_src.fmt = af->frame->format;
}
if (is->swr_ctx) {
- const uint8_t **in = (const uint8_t **)is->frame->extended_data;
+ const uint8_t **in = (const uint8_t **)af->frame->extended_data;
uint8_t **out = &is->audio_buf1;
- int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / is->frame->sample_rate + 256;
+ int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
int out_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
int len2;
if (out_size < 0) {
av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
- break;
+ return -1;
}
- if (wanted_nb_samples != is->frame->nb_samples) {
- if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt.freq / is->frame->sample_rate,
- wanted_nb_samples * is->audio_tgt.freq / is->frame->sample_rate) < 0) {
+ if (wanted_nb_samples != af->frame->nb_samples) {
+ if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
+ wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {
av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
- break;
+ return -1;
}
}
av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
if (!is->audio_buf1)
return AVERROR(ENOMEM);
- len2 = swr_convert(is->swr_ctx, out, out_count, in, is->frame->nb_samples);
+ len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);
if (len2 < 0) {
av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");
- break;
+ return -1;
}
if (len2 == out_count) {
av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
@@ -2453,17 +2500,17 @@ static int audio_decode_frame(VideoState *is)
is->audio_buf = is->audio_buf1;
resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
} else {
- is->audio_buf = is->frame->data[0];
+ is->audio_buf = af->frame->data[0];
resampled_data_size = data_size;
}
audio_clock0 = is->audio_clock;
/* update the audio clock with the pts */
- if (is->frame->pts != AV_NOPTS_VALUE)
- is->audio_clock = is->frame->pts * av_q2d(tb) + (double) is->frame->nb_samples / is->frame->sample_rate;
+ if (!isnan(af->pts))
+ is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;
else
is->audio_clock = NAN;
- is->audio_clock_serial = is->auddec.pkt_serial;
+ is->audio_clock_serial = af->serial;
#ifdef DEBUG
{
static double last_clock;
@@ -2475,12 +2522,6 @@ static int audio_decode_frame(VideoState *is)
#endif
return resampled_data_size;
}
-
- if ((got_frame = decoder_decode_frame(&is->auddec, is->frame, NULL)) < 0)
- return -1;
-
- if (is->auddec.flushed)
- is->audio_buf_frames_pending = 0;
}
}
@@ -2707,6 +2748,7 @@ static int stream_component_open(VideoState *is, int stream_index)
is->auddec.start_pts = is->audio_st->start_time;
is->auddec.start_pts_tb = is->audio_st->time_base;
}
+ is->audio_tid = SDL_CreateThread(audio_thread, is);
SDL_PauseAudio(0);
break;
case AVMEDIA_TYPE_VIDEO:
@@ -2750,6 +2792,8 @@ static void stream_component_close(VideoState *is, int stream_index)
packet_queue_abort(&is->audioq);
SDL_CloseAudio();
+ frame_queue_signal(&is->sampq);
+ SDL_WaitThread(is->audio_tid, NULL);
decoder_destroy(&is->auddec);
packet_queue_flush(&is->audioq);
@@ -2757,7 +2801,6 @@ static void stream_component_close(VideoState *is, int stream_index)
av_freep(&is->audio_buf1);
is->audio_buf1_size = 0;
is->audio_buf = NULL;
- av_frame_free(&is->frame);
if (is->rdft) {
av_rdft_end(is->rdft);
@@ -2765,9 +2808,6 @@ static void stream_component_close(VideoState *is, int stream_index)
is->rdft = NULL;
is->rdft_bits = 0;
}
-#if CONFIG_AVFILTER
- avfilter_graph_free(&is->agraph);
-#endif
break;
case AVMEDIA_TYPE_VIDEO:
packet_queue_abort(&is->videoq);
@@ -3065,7 +3105,7 @@ static int read_thread(void *arg)
continue;
}
if (!is->paused &&
- (!is->audio_st || is->auddec.finished == is->audioq.serial) &&
+ (!is->audio_st || (is->auddec.finished == is->audioq.serial && frame_queue_nb_remaining(&is->sampq) == 0)) &&
(!is->video_st || (is->viddec.finished == is->videoq.serial && frame_queue_nb_remaining(&is->pictq) == 0))) {
if (loop != 1 && (!loop || --loop)) {
stream_seek(is, start_time != AV_NOPTS_VALUE ? start_time : 0, 0, 0);
@@ -3160,6 +3200,8 @@ static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
goto fail;
if (frame_queue_init(&is->subpq, &is->subtitleq, SUBPICTURE_QUEUE_SIZE, 0) < 0)
goto fail;
+ if (frame_queue_init(&is->sampq, &is->audioq, SAMPLE_QUEUE_SIZE, 1) < 0)
+ goto fail;
packet_queue_init(&is->videoq);
packet_queue_init(&is->audioq);
@@ -3171,7 +3213,6 @@ static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
init_clock(&is->audclk, &is->audioq.serial);
init_clock(&is->extclk, &is->extclk.serial);
is->audio_clock_serial = -1;
- is->audio_last_serial = -1;
is->av_sync_type = av_sync_type;
is->read_tid = SDL_CreateThread(read_thread, is);
if (!is->read_tid) {
@@ -3421,8 +3462,8 @@ static void event_loop(VideoState *cur_stream)
pos = -1;
if (pos < 0 && cur_stream->video_stream >= 0)
pos = frame_queue_last_pos(&cur_stream->pictq);
- if (pos < 0 && cur_stream->audio_stream >= 0 && cur_stream->frame)
- pos = av_frame_get_pkt_pos(cur_stream->frame);
+ if (pos < 0 && cur_stream->audio_stream >= 0)
+ pos = frame_queue_last_pos(&cur_stream->sampq);
if (pos < 0)
pos = avio_tell(cur_stream->ic->pb);
if (cur_stream->ic->bit_rate)
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