ffplay.c

ffmpeg源码分析

            break;

        SDL_LockMutex(is->subpq_mutex);
        while (is->subpq_size >= SUBPICTURE_QUEUE_SIZE &&
               !is->subtitleq.abort_request) {
            SDL_CondWait(is->subpq_cond, is->subpq_mutex);
        }
        SDL_UnlockMutex(is->subpq_mutex);

        if (is->subtitleq.abort_request)
            goto the_end;

        sp = &is->subpq[is->subpq_windex];

       /* NOTE: ipts is the PTS of the _first_ picture beginning in
           this packet, if any */
        pts = 0;
        if (pkt->pts != AV_NOPTS_VALUE)
            pts = av_q2d(is->subtitle_st->time_base)*pkt->pts;

        SDL_LockMutex(is->subtitle_decoder_mutex);
        len1 = avcodec_decode_subtitle(is->subtitle_st->codec,
                                    &sp->sub, &got_subtitle,
                                    pkt->data, pkt->size);
        SDL_UnlockMutex(is->subtitle_decoder_mutex);
//            if (len1 < 0)
//                break;
        if (got_subtitle && sp->sub.format == 0) {
            sp->pts = pts;

            for (i = 0; i < sp->sub.num_rects; i++)
            {
                for (j = 0; j < sp->sub.rects[i].nb_colors; j++)
                {
                    RGBA_IN(r, g, b, a, sp->sub.rects[i].rgba_palette + j);
                    y = RGB_TO_Y_CCIR(r, g, b);
                    u = RGB_TO_U_CCIR(r, g, b, 0);
                    v = RGB_TO_V_CCIR(r, g, b, 0);
                    YUVA_OUT(sp->sub.rects[i].rgba_palette + j, y, u, v, a);
                }
            }

            /* now we can update the picture count */
            if (++is->subpq_windex == SUBPICTURE_QUEUE_SIZE)
                is->subpq_windex = 0;
            SDL_LockMutex(is->subpq_mutex);
            is->subpq_size++;
            SDL_UnlockMutex(is->subpq_mutex);
        }
        av_free_packet(pkt);
//        if (step)
//            if (cur_stream)
//                stream_pause(cur_stream);
    }
 the_end:
    return 0;
}

/* copy samples for viewing in editor window */
static void update_sample_display(VideoState *is, short *samples, int samples_size)
{
    int size, len, channels;

    channels = is->audio_st->codec->channels;

    size = samples_size / sizeof(short);
    while (size > 0) {
        len = SAMPLE_ARRAY_SIZE - is->sample_array_index;
        if (len > size)
            len = size;
        memcpy(is->sample_array + is->sample_array_index, samples, len * sizeof(short));
        samples += len;
        is->sample_array_index += len;
        if (is->sample_array_index >= SAMPLE_ARRAY_SIZE)
            is->sample_array_index = 0;
        size -= len;
    }
}

/* return the new audio buffer size (samples can be added or deleted
   to get better sync if video or external master clock) */
static int synchronize_audio(VideoState *is, short *samples,
                             int samples_size1, double pts)
{
    int n, samples_size;
    double ref_clock;

    n = 2 * is->audio_st->codec->channels;
    samples_size = samples_size1;

    /* if not master, then we try to remove or add samples to correct the clock */
    if (((is->av_sync_type == AV_SYNC_VIDEO_MASTER && is->video_st) ||
         is->av_sync_type == AV_SYNC_EXTERNAL_CLOCK)) {
        double diff, avg_diff;
        int wanted_size, min_size, max_size, nb_samples;

        ref_clock = get_master_clock(is);
        diff = get_audio_clock(is) - ref_clock;

        if (diff < AV_NOSYNC_THRESHOLD) {
            is->audio_diff_cum = diff + is->audio_diff_avg_coef * is->audio_diff_cum;
            if (is->audio_diff_avg_count < AUDIO_DIFF_AVG_NB) {
                /* not enough measures to have a correct estimate */
                is->audio_diff_avg_count++;
            } else {
                /* estimate the A-V difference */
                avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef);

                if (fabs(avg_diff) >= is->audio_diff_threshold) {
                    wanted_size = samples_size + ((int)(diff * is->audio_st->codec->sample_rate) * n);
                    nb_samples = samples_size / n;

                    min_size = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
                    max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
                    if (wanted_size < min_size)
                        wanted_size = min_size;
                    else if (wanted_size > max_size)
                        wanted_size = max_size;

                    /* add or remove samples to correction the synchro */
                    if (wanted_size < samples_size) {
                        /* remove samples */
                        samples_size = wanted_size;
                    } else if (wanted_size > samples_size) {
                        uint8_t *samples_end, *q;
                        int nb;

                        /* add samples */
                        nb = (samples_size - wanted_size);
                        samples_end = (uint8_t *)samples + samples_size - n;
                        q = samples_end + n;
                        while (nb > 0) {
                            memcpy(q, samples_end, n);
                            q += n;
                            nb -= n;
                        }
                        samples_size = wanted_size;
                    }
                }
#if 0
                printf("diff=%f adiff=%f sample_diff=%d apts=%0.3f vpts=%0.3f %f\n",
                       diff, avg_diff, samples_size - samples_size1,
                       is->audio_clock, is->video_clock, is->audio_diff_threshold);
#endif
            }
        } else {
            /* too big difference : may be initial PTS errors, so
               reset A-V filter */
            is->audio_diff_avg_count = 0;
            is->audio_diff_cum = 0;
        }
    }

    return samples_size;
}

/* decode one audio frame and returns its uncompressed size */
static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, double *pts_ptr)
{
    AVPacket *pkt = &is->audio_pkt;
    int n, len1, data_size;
    double pts;

    for(;;) {
        /* NOTE: the audio packet can contain several frames */
        while (is->audio_pkt_size > 0) {
            SDL_LockMutex(is->audio_decoder_mutex);
            len1 = avcodec_decode_audio(is->audio_st->codec,
                                        (int16_t *)audio_buf, &data_size,
                                        is->audio_pkt_data, is->audio_pkt_size);
            SDL_UnlockMutex(is->audio_decoder_mutex);
            if (len1 < 0) {
                /* if error, we skip the frame */
                is->audio_pkt_size = 0;
                break;
            }

            is->audio_pkt_data += len1;
            is->audio_pkt_size -= len1;
            if (data_size <= 0)
                continue;
            /* if no pts, then compute it */
            pts = is->audio_clock;
            *pts_ptr = pts;
            n = 2 * is->audio_st->codec->channels;
            is->audio_clock += (double)data_size /
                (double)(n * is->audio_st->codec->sample_rate);
#if defined(DEBUG_SYNC)
            {
                static double last_clock;
                printf("audio: delay=%0.3f clock=%0.3f pts=%0.3f\n",
                       is->audio_clock - last_clock,
                       is->audio_clock, pts);
                last_clock = is->audio_clock;
            }
#endif
            return data_size;
        }

        /* free the current packet */
        if (pkt->data)
            av_free_packet(pkt);

        if (is->paused || is->audioq.abort_request) {
            return -1;
        }

        /* read next packet */
        if (packet_queue_get(&is->audioq, pkt, 1) < 0)
            return -1;
        is->audio_pkt_data = pkt->data;
        is->audio_pkt_size = pkt->size;

        /* if update the audio clock with the pts */
        if (pkt->pts != AV_NOPTS_VALUE) {
            is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts;
        }
    }
}

/* get the current audio output buffer size, in samples. With SDL, we
   cannot have a precise information */
static int audio_write_get_buf_size(VideoState *is)
{
    return is->audio_hw_buf_size - is->audio_buf_index;
}


/* prepare a new audio buffer */
void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
{
    VideoState *is = opaque;
    int audio_size, len1;
    double pts;

    audio_callback_time = av_gettime();

    while (len > 0) {
        if (is->audio_buf_index >= is->audio_buf_size) {
           audio_size = audio_decode_frame(is, is->audio_buf, &pts);
           if (audio_size < 0) {
                /* if error, just output silence */
               is->audio_buf_size = 1024;
               memset(is->audio_buf, 0, is->audio_buf_size);
           } else {
               if (is->show_audio)
                   update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
               audio_size = synchronize_audio(is, (int16_t *)is->audio_buf, audio_size,
                                              pts);
               is->audio_buf_size = audio_size;
           }
           is->audio_buf_index = 0;
        }
        len1 = is->audio_buf_size - is->audio_buf_index;
        if (len1 > len)
            len1 = len;
        memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);
        len -= len1;
        stream += len1;
        is->audio_buf_index += len1;
    }
}


/* open a given stream. Return 0 if OK */
static int stream_component_open(VideoState *is, int stream_index)
{
    AVFormatContext *ic = is->ic;
    AVCodecContext *enc;
    AVCodec *codec;
    SDL_AudioSpec wanted_spec, spec;

    if (stream_index < 0 || stream_index >= ic->nb_streams)
        return -1;
    enc = ic->streams[stream_index]->codec;

    /* prepare audio output */
    if (enc->codec_type == CODEC_TYPE_AUDIO) {
        wanted_spec.freq = enc->sample_rate;
        wanted_spec.format = AUDIO_S16SYS;
        /* hack for AC3. XXX: suppress that */
        if (enc->channels > 2)
            enc->channels = 2;
        wanted_spec.channels = enc->channels;
        wanted_spec.silence = 0;
        wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
        wanted_spec.callback = sdl_audio_callback;
        wanted_spec.userdata = is;
        if (SDL_OpenAudio(&wanted_spec, &spec) < 0) {
            fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());
            return -1;
        }
        is->audio_hw_buf_size = spec.size;
    }

    codec = avcodec_find_decoder(enc->codec_id);
    enc->debug_mv = debug_mv;
    enc->debug = debug;
    if(debug)
        av_log_set_level(AV_LOG_DEBUG);
    enc->workaround_bugs = workaround_bugs;
    enc->lowres = lowres;
    if(lowres) enc->flags |= CODEC_FLAG_EMU_EDGE;
    enc->idct_algo= idct;
    if(fast) enc->flags2 |= CODEC_FLAG2_FAST;
    enc->skip_frame= skip_frame;
    enc->skip_idct= skip_idct;
    enc->skip_loop_filter= skip_loop_filter;
    enc->error_resilience= error_resilience;
    enc->error_concealment= error_concealment;
    if (!codec ||
        avcodec_open(enc, codec) < 0)
        return -1;
#if defined(HAVE_THREADS)
    if(thread_count>1)
        avcodec_thread_init(enc, thread_count);
#endif
    enc->thread_count= thread_count;
    switch(enc->codec_type) {
    case CODEC_TYPE_AUDIO:
        is->audio_stream = stream_index;
        is->audio_st = ic->streams[stream_index];
        is->audio_buf_size = 0;
        is->audio_buf_index = 0;

        /* init averaging filter */
        is->audio_diff_avg_coef = exp(log(0.01) / AUDIO_DIFF_AVG_NB);
        is->audio_diff_avg_count = 0;
        /* since we do not have a precise anough audio fifo fullness,
           we correct audio sync only if larger than this threshold */
        is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / enc->sample_rate;

        memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
        packet_queue_init(&is->audioq);
        SDL_PauseAudio(0);
        break;
    case CODEC_TYPE_VIDEO:
        is->video_stream = stream_index;
        is->video_st = ic->streams[stream_index];

        is->frame_last_delay = 40e-3;
        is->frame_timer = (double)av_gettime() / 1000000.0;
        is->video_current_pts_time = av_gettime();

        packet_queue_init(&is->videoq);
        is->video_tid = SDL_CreateThread(video_thread, is);
        break;
    case CODEC_TYPE_SUBTITLE:
        is->subtitle_stream = stream_index;
        is->subtitle_st = ic->streams[stream_index];
        packet_queue_init(&is->subtitleq);

        is->subtitle_tid = SDL_CreateThread(subtitle_thread, is);
        break;
    default:
        break;
    }
    return 0;
}

static void stream_component_close(VideoState *is, int stream_index)
{
    AVFormatContext *ic = is->ic;
    AVCodecContext *enc;

    if (stream_index < 0 || stream_index >= ic->nb_streams)
        return;
    enc = ic->streams[stream_index]->codec;

    switch(enc->codec_type) {
    case CODEC_TYPE_AUDIO:
        packet_queue_abort(&is->audioq);

        SDL_CloseAudio();

        packet_queue_end(&is->audioq);
        break;
    case CODEC_TYPE_VIDEO:
        packet_queue_abort(&is->videoq);

        /* note: we also signal this mutex to make sure we deblock the
           video thread in all cases */
        SDL_LockMutex(is->pictq_mutex);
        SDL_CondSignal(is->pictq_cond);
        SDL_UnlockMutex(is->pictq_mutex);

        SDL_WaitThread(is->video_tid, NULL);

        packet_queue_end(&is->videoq);
        break;
    case CODEC_TYPE_SUBTITLE:
        packet_queue_abort(&is->subtitleq);

        /* note: we also signal this mutex to make sure we deblock the
           video thread in all cases */
        SDL_LockMutex(is->subpq_mutex);
        is->subtitle_stream_changed = 1;

        SDL_CondSignal(is->subpq_cond);
        SDL_UnlockMutex(is->subpq_mutex);

        SDL_WaitThread(is->subtitle_tid, NULL);

        packet_queue_end(&is->subtitleq);
        break;
    default:
        break;
    }

    avcodec_close(enc);
    switch(enc->codec_type) {
    case CODEC_TYPE_AUDIO:
        is->audio_st = NULL;
        is->audio_stream = -1;
        break;
    case CODEC_TYPE_VIDEO:
        is->video_st = NULL;
        is->video_stream = -1;
        break;
    case CODEC_TYPE_SUBTITLE:
        is->subtitle_st = NULL;
        is->subtitle_stream = -1;
        break;
    default:
        break;
    }
}

void dump_stream_info(AVFormatContext *s)
{
    if (s->track != 0)
        fprintf(stderr, "Track: %d\n", s->track);
    if (s->title[0] != '\0')
        fprintf(stderr, "Title: %s\n", s->title);
    if (s->author[0] != '\0')
        fprintf(stderr, "Author: %s\n", s->author);
    if (s->album[0] != '\0')
        fprintf(stderr, "Album: %s\n", s->album);
    if (s->year != 0)