On Thu, 30 Oct 2014 00:31:25 +0100
Marton Balint <c...@passwd.hu> wrote:
> Signed-off-by: Marton Balint <c...@passwd.hu>
> ---
> 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)
Is there any actual advantage to this? Also, ffmpeg does support
multithreaded audio decoding; only for some codecs though.
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