ffmpeg.c

mediastreamer2是开源的网络传输媒体流的库

#endif
    if (n > 0) {
        n = read(0, &ch, 1);
        if (n == 1)
            return ch;

        return n;
    }
#elif defined(HAVE_CONIO_H)
    if(kbhit())
        return(getch());
#endif
    return -1;
}

static int decode_interrupt_cb(void)
{
    return q_pressed || (q_pressed = read_key() == 'q');
}

static int read_ffserver_streams(AVFormatContext *s, const char *filename)
{
    int i, err;
    AVFormatContext *ic;

    err = av_open_input_file(&ic, filename, NULL, FFM_PACKET_SIZE, NULL);
    if (err < 0)
        return err;
    /* copy stream format */
    s->nb_streams = ic->nb_streams;
    for(i=0;i<ic->nb_streams;i++) {
        AVStream *st;

        // FIXME: a more elegant solution is needed
        st = av_mallocz(sizeof(AVStream));
        memcpy(st, ic->streams[i], sizeof(AVStream));
        st->codec = avcodec_alloc_context();
        memcpy(st->codec, ic->streams[i]->codec, sizeof(AVCodecContext));
        s->streams[i] = st;
    }

    av_close_input_file(ic);
    return 0;
}

static double
get_sync_ipts(const AVOutputStream *ost)
{
    const AVInputStream *ist = ost->sync_ist;
    return (double)(ist->pts - start_time)/AV_TIME_BASE;
}

static void write_frame(AVFormatContext *s, AVPacket *pkt, AVCodecContext *avctx, AVBitStreamFilterContext *bsfc){
    int ret;

    while(bsfc){
        AVPacket new_pkt= *pkt;
        int a= av_bitstream_filter_filter(bsfc, avctx, NULL,
                                          &new_pkt.data, &new_pkt.size,
                                          pkt->data, pkt->size,
                                          pkt->flags & PKT_FLAG_KEY);
        if(a>0){
            av_free_packet(pkt);
            new_pkt.destruct= av_destruct_packet;
        } else if(a<0){
            fprintf(stderr, "%s failed for stream %d, codec %s",
                    bsfc->filter->name, pkt->stream_index,
                    avctx->codec ? avctx->codec->name : "copy");
            print_error("", a);
        }
        *pkt= new_pkt;

        bsfc= bsfc->next;
    }

    ret= av_interleaved_write_frame(s, pkt);
    if(ret < 0){
        print_error("av_interleaved_write_frame()", ret);
        exit(1);
    }
}

#define MAX_AUDIO_PACKET_SIZE (128 * 1024)

static void do_audio_out(AVFormatContext *s,
                         AVOutputStream *ost,
                         AVInputStream *ist,
                         unsigned char *buf, int size)
{
    uint8_t *buftmp;
    static uint8_t *audio_buf = NULL;
    static uint8_t *audio_out = NULL;
    const int audio_out_size= 4*MAX_AUDIO_PACKET_SIZE;

    int size_out, frame_bytes, ret;
    AVCodecContext *enc= ost->st->codec;
    AVCodecContext *dec= ist->st->codec;

    /* SC: dynamic allocation of buffers */
    if (!audio_buf)
        audio_buf = av_malloc(2*MAX_AUDIO_PACKET_SIZE);
    if (!audio_out)
        audio_out = av_malloc(audio_out_size);
    if (!audio_buf || !audio_out)
        return;               /* Should signal an error ! */

    if (enc->channels != dec->channels)
        ost->audio_resample = 1;

    if (ost->audio_resample && !ost->resample) {
        ost->resample = audio_resample_init(enc->channels,    dec->channels,
                                            enc->sample_rate, dec->sample_rate);
        if (!ost->resample) {
            fprintf(stderr, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n",
                    dec->channels, dec->sample_rate,
                    enc->channels, enc->sample_rate);
            exit(1);
        }
    }

    if(audio_sync_method){
        double delta = get_sync_ipts(ost) * enc->sample_rate - ost->sync_opts
                - av_fifo_size(&ost->fifo)/(ost->st->codec->channels * 2);
        double idelta= delta*ist->st->codec->sample_rate / enc->sample_rate;
        int byte_delta= ((int)idelta)*2*ist->st->codec->channels;

        //FIXME resample delay
        if(fabs(delta) > 50){
            if(ist->is_start || fabs(delta) > audio_drift_threshold*enc->sample_rate){
                if(byte_delta < 0){
                    byte_delta= FFMAX(byte_delta, -size);
                    size += byte_delta;
                    buf  -= byte_delta;
                    if(verbose > 2)
                        fprintf(stderr, "discarding %d audio samples\n", (int)-delta);
                    if(!size)
                        return;
                    ist->is_start=0;
                }else{
                    static uint8_t *input_tmp= NULL;
                    input_tmp= av_realloc(input_tmp, byte_delta + size);

                    if(byte_delta + size <= MAX_AUDIO_PACKET_SIZE)
                        ist->is_start=0;
                    else
                        byte_delta= MAX_AUDIO_PACKET_SIZE - size;

                    memset(input_tmp, 0, byte_delta);
                    memcpy(input_tmp + byte_delta, buf, size);
                    buf= input_tmp;
                    size += byte_delta;
                    if(verbose > 2)
                        fprintf(stderr, "adding %d audio samples of silence\n", (int)delta);
                }
            }else if(audio_sync_method>1){
                int comp= av_clip(delta, -audio_sync_method, audio_sync_method);
                assert(ost->audio_resample);
                if(verbose > 2)
                    fprintf(stderr, "compensating audio timestamp drift:%f compensation:%d in:%d\n", delta, comp, enc->sample_rate);
//                fprintf(stderr, "drift:%f len:%d opts:%"PRId64" ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(get_sync_ipts(ost) * enc->sample_rate), av_fifo_size(&ost->fifo)/(ost->st->codec->channels * 2));
                av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate);
            }
        }
    }else
        ost->sync_opts= lrintf(get_sync_ipts(ost) * enc->sample_rate)
                        - av_fifo_size(&ost->fifo)/(ost->st->codec->channels * 2); //FIXME wrong

    if (ost->audio_resample) {
        buftmp = audio_buf;
        size_out = audio_resample(ost->resample,
                                  (short *)buftmp, (short *)buf,
                                  size / (ist->st->codec->channels * 2));
        size_out = size_out * enc->channels * 2;
    } else {
        buftmp = buf;
        size_out = size;
    }

    /* now encode as many frames as possible */
    if (enc->frame_size > 1) {
        /* output resampled raw samples */
        av_fifo_write(&ost->fifo, buftmp, size_out);

        frame_bytes = enc->frame_size * 2 * enc->channels;

        while (av_fifo_read(&ost->fifo, audio_buf, frame_bytes) == 0) {
            AVPacket pkt;
            av_init_packet(&pkt);

            ret = avcodec_encode_audio(enc, audio_out, audio_out_size,
                                       (short *)audio_buf);
            audio_size += ret;
            pkt.stream_index= ost->index;
            pkt.data= audio_out;
            pkt.size= ret;
            if(enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
                pkt.pts= av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
            pkt.flags |= PKT_FLAG_KEY;
            write_frame(s, &pkt, ost->st->codec, bitstream_filters[ost->file_index][pkt.stream_index]);

            ost->sync_opts += enc->frame_size;
        }
    } else {
        AVPacket pkt;
        av_init_packet(&pkt);

        ost->sync_opts += size_out / (2 * enc->channels);

        /* output a pcm frame */
        /* XXX: change encoding codec API to avoid this ? */
        switch(enc->codec->id) {
        case CODEC_ID_PCM_S32LE:
        case CODEC_ID_PCM_S32BE:
        case CODEC_ID_PCM_U32LE:
        case CODEC_ID_PCM_U32BE:
            size_out = size_out << 1;
            break;
        case CODEC_ID_PCM_S24LE:
        case CODEC_ID_PCM_S24BE:
        case CODEC_ID_PCM_U24LE:
        case CODEC_ID_PCM_U24BE:
        case CODEC_ID_PCM_S24DAUD:
            size_out = size_out / 2 * 3;
            break;
        case CODEC_ID_PCM_S16LE:
        case CODEC_ID_PCM_S16BE:
        case CODEC_ID_PCM_U16LE:
        case CODEC_ID_PCM_U16BE:
            break;
        default:
            size_out = size_out >> 1;
            break;
        }
        ret = avcodec_encode_audio(enc, audio_out, size_out,
                                   (short *)buftmp);
        audio_size += ret;
        pkt.stream_index= ost->index;
        pkt.data= audio_out;
        pkt.size= ret;
        if(enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
            pkt.pts= av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
        pkt.flags |= PKT_FLAG_KEY;
        write_frame(s, &pkt, ost->st->codec, bitstream_filters[ost->file_index][pkt.stream_index]);
    }
}

static void pre_process_video_frame(AVInputStream *ist, AVPicture *picture, void **bufp)
{
    AVCodecContext *dec;
    AVPicture *picture2;
    AVPicture picture_tmp;
    uint8_t *buf = 0;

    dec = ist->st->codec;

    /* deinterlace : must be done before any resize */
    if (do_deinterlace || using_vhook) {
        int size;

        /* create temporary picture */
        size = avpicture_get_size(dec->pix_fmt, dec->width, dec->height);
        buf = av_malloc(size);
        if (!buf)
            return;

        picture2 = &picture_tmp;
        avpicture_fill(picture2, buf, dec->pix_fmt, dec->width, dec->height);

        if (do_deinterlace){
            if(avpicture_deinterlace(picture2, picture,
                                     dec->pix_fmt, dec->width, dec->height) < 0) {
                /* if error, do not deinterlace */
                av_free(buf);
                buf = NULL;
                picture2 = picture;
            }
        } else {
            av_picture_copy(picture2, picture, dec->pix_fmt, dec->width, dec->height);
        }
    } else {
        picture2 = picture;
    }

    if (ENABLE_VHOOK)
        frame_hook_process(picture2, dec->pix_fmt, dec->width, dec->height,
                           1000000 * ist->pts / AV_TIME_BASE);

    if (picture != picture2)
        *picture = *picture2;
    *bufp = buf;
}

/* we begin to correct av delay at this threshold */
#define AV_DELAY_MAX 0.100

static void do_subtitle_out(AVFormatContext *s,
                            AVOutputStream *ost,
                            AVInputStream *ist,
                            AVSubtitle *sub,
                            int64_t pts)
{
    static uint8_t *subtitle_out = NULL;
    int subtitle_out_max_size = 65536;
    int subtitle_out_size, nb, i;
    AVCodecContext *enc;
    AVPacket pkt;

    if (pts == AV_NOPTS_VALUE) {
        fprintf(stderr, "Subtitle packets must have a pts\n");
        return;
    }

    enc = ost->st->codec;

    if (!subtitle_out) {
        subtitle_out = av_malloc(subtitle_out_max_size);
    }

    /* Note: DVB subtitle need one packet to draw them and one other
       packet to clear them */
    /* XXX: signal it in the codec context ? */
    if (enc->codec_id == CODEC_ID_DVB_SUBTITLE)
        nb = 2;
    else
        nb = 1;

    for(i = 0; i < nb; i++) {
        subtitle_out_size = avcodec_encode_subtitle(enc, subtitle_out,
                                                    subtitle_out_max_size, sub);

        av_init_packet(&pkt);
        pkt.stream_index = ost->index;
        pkt.data = subtitle_out;
        pkt.size = subtitle_out_size;
        pkt.pts = av_rescale_q(pts, ist->st->time_base, ost->st->time_base);
        if (enc->codec_id == CODEC_ID_DVB_SUBTITLE) {
            /* XXX: the pts correction is handled here. Maybe handling
               it in the codec would be better */
            if (i == 0)
                pkt.pts += 90 * sub->start_display_time;
            else
                pkt.pts += 90 * sub->end_display_time;
        }
        write_frame(s, &pkt, ost->st->codec, bitstream_filters[ost->file_index][pkt.stream_index]);
    }
}

static int bit_buffer_size= 1024*256;
static uint8_t *bit_buffer= NULL;

static void do_video_out(AVFormatContext *s,
                         AVOutputStream *ost,
                         AVInputStream *ist,
                         AVFrame *in_picture,
                         int *frame_size)
{
    int nb_frames, i, ret;