avconv.c

    av_free(audio_buf);
    allocated_audio_buf_size = 0;
    av_free(async_buf);
    allocated_async_buf_size = 0;

    avfilter_uninit();
    avformat_network_deinit();

    if (received_sigterm) {
        av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
               (int) received_sigterm);
        exit (255);
    }

    exit(ret);
}

static void assert_avoptions(AVDictionary *m)
{
    AVDictionaryEntry *t;
    if ((t = av_dict_get(m, "", NULL, AV_DICT_IGNORE_SUFFIX))) {
        av_log(NULL, AV_LOG_FATAL, "Option %s not found.\n", t->key);
        exit_program(1);
    }
}

static void assert_codec_experimental(AVCodecContext *c, int encoder)
{
    const char *codec_string = encoder ? "encoder" : "decoder";
    AVCodec *codec;
    if (c->codec->capabilities & CODEC_CAP_EXPERIMENTAL &&
        c->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
        av_log(NULL, AV_LOG_FATAL, "%s '%s' is experimental and might produce bad "
                "results.\nAdd '-strict experimental' if you want to use it.\n",
                codec_string, c->codec->name);
        codec = encoder ? avcodec_find_encoder(c->codec->id) : avcodec_find_decoder(c->codec->id);
        if (!(codec->capabilities & CODEC_CAP_EXPERIMENTAL))
            av_log(NULL, AV_LOG_FATAL, "Or use the non experimental %s '%s'.\n",
                   codec_string, codec->name);
        exit_program(1);
    }
}

static void choose_sample_fmt(AVStream *st, AVCodec *codec)
{
    if (codec && codec->sample_fmts) {
        const enum AVSampleFormat *p = codec->sample_fmts;
        for (; *p != -1; p++) {
            if (*p == st->codec->sample_fmt)
                break;
        }
        if (*p == -1) {
            av_log(NULL, AV_LOG_WARNING,
                   "Incompatible sample format '%s' for codec '%s', auto-selecting format '%s'\n",
                   av_get_sample_fmt_name(st->codec->sample_fmt),
                   codec->name,
                   av_get_sample_fmt_name(codec->sample_fmts[0]));
            st->codec->sample_fmt = codec->sample_fmts[0];
        }
    }
}

/**
 * Update the requested input sample format based on the output sample format.
 * This is currently only used to request float output from decoders which
 * support multiple sample formats, one of which is AV_SAMPLE_FMT_FLT.
 * Ideally this will be removed in the future when decoders do not do format
 * conversion and only output in their native format.
 */
static void update_sample_fmt(AVCodecContext *dec, AVCodec *dec_codec,
                              AVCodecContext *enc)
{
    /* if sample formats match or a decoder sample format has already been
       requested, just return */
    if (enc->sample_fmt == dec->sample_fmt ||
        dec->request_sample_fmt > AV_SAMPLE_FMT_NONE)
        return;

    /* if decoder supports more than one output format */
    if (dec_codec && dec_codec->sample_fmts &&
        dec_codec->sample_fmts[0] != AV_SAMPLE_FMT_NONE &&
        dec_codec->sample_fmts[1] != AV_SAMPLE_FMT_NONE) {
        const enum AVSampleFormat *p;
        int min_dec = -1, min_inc = -1;

        /* find a matching sample format in the encoder */
        for (p = dec_codec->sample_fmts; *p != AV_SAMPLE_FMT_NONE; p++) {
            if (*p == enc->sample_fmt) {
                dec->request_sample_fmt = *p;
                return;
            } else if (*p > enc->sample_fmt) {
                min_inc = FFMIN(min_inc, *p - enc->sample_fmt);
            } else
                min_dec = FFMIN(min_dec, enc->sample_fmt - *p);
        }

        /* if none match, provide the one that matches quality closest */
        dec->request_sample_fmt = min_inc > 0 ? enc->sample_fmt + min_inc :
                                  enc->sample_fmt - min_dec;
    }
}

static void choose_sample_rate(AVStream *st, AVCodec *codec)
{
    if (codec && codec->supported_samplerates) {
        const int *p  = codec->supported_samplerates;
        int best      = 0;
        int best_dist = INT_MAX;
        for (; *p; p++) {
            int dist = abs(st->codec->sample_rate - *p);
            if (dist < best_dist) {
                best_dist = dist;
                best      = *p;
            }
        }
        if (best_dist) {
            av_log(st->codec, AV_LOG_WARNING, "Requested sampling rate unsupported using closest supported (%d)\n", best);
        }
        st->codec->sample_rate = best;
    }
}

static double
get_sync_ipts(const OutputStream *ost, int64_t pts)
{
    OutputFile *of = output_files[ost->file_index];
    return (double)(pts - of->start_time) / AV_TIME_BASE;
}

static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
{
    AVBitStreamFilterContext *bsfc = ost->bitstream_filters;
    AVCodecContext          *avctx = ost->st->codec;
    int ret;

    /*
     * Audio encoders may split the packets --  #frames in != #packets out.
     * But there is no reordering, so we can limit the number of output packets
     * by simply dropping them here.
     * Counting encoded video frames needs to be done separately because of
     * reordering, see do_video_out()
     */
    if (!(avctx->codec_type == AVMEDIA_TYPE_VIDEO && avctx->codec)) {
        if (ost->frame_number >= ost->max_frames) {
            av_free_packet(pkt);
            return;
        }
        ost->frame_number++;
    }

    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 & AV_PKT_FLAG_KEY);
        if (a > 0) {
            av_free_packet(pkt);
            new_pkt.destruct = av_destruct_packet;
        } else if (a < 0) {
            av_log(NULL, AV_LOG_ERROR, "%s failed for stream %d, codec %s",
                   bsfc->filter->name, pkt->stream_index,
                   avctx->codec ? avctx->codec->name : "copy");
            print_error("", a);
            if (exit_on_error)
                exit_program(1);
        }
        *pkt = new_pkt;

        bsfc = bsfc->next;
    }

    pkt->stream_index = ost->index;
    ret = av_interleaved_write_frame(s, pkt);
    if (ret < 0) {
        print_error("av_interleaved_write_frame()", ret);
        exit_program(1);
    }
}

static int check_recording_time(OutputStream *ost)
{
    OutputFile *of = output_files[ost->file_index];

    if (of->recording_time != INT64_MAX &&
        av_compare_ts(ost->sync_opts - ost->first_pts, ost->st->codec->time_base, of->recording_time,
                      AV_TIME_BASE_Q) >= 0) {
        ost->is_past_recording_time = 1;
        return 0;
    }
    return 1;
}

static void get_default_channel_layouts(OutputStream *ost, InputStream *ist)
{
    char layout_name[256];
    AVCodecContext *enc = ost->st->codec;
    AVCodecContext *dec = ist->st->codec;

    if (dec->channel_layout &&
        av_get_channel_layout_nb_channels(dec->channel_layout) != dec->channels) {
        av_get_channel_layout_string(layout_name, sizeof(layout_name),
                                     dec->channels, dec->channel_layout);
        av_log(NULL, AV_LOG_ERROR, "New channel layout (%s) is invalid\n",
               layout_name);
        dec->channel_layout = 0;
    }
    if (!dec->channel_layout) {
        if (enc->channel_layout && dec->channels == enc->channels) {
            dec->channel_layout = enc->channel_layout;
        } else {
            dec->channel_layout = av_get_default_channel_layout(dec->channels);

            if (!dec->channel_layout) {
                av_log(NULL, AV_LOG_FATAL, "Unable to find default channel "
                       "layout for Input Stream #%d.%d\n", ist->file_index,
                       ist->st->index);
                exit_program(1);
            }
        }
        av_get_channel_layout_string(layout_name, sizeof(layout_name),
                                     dec->channels, dec->channel_layout);
        av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for  Input Stream "
               "#%d.%d : %s\n", ist->file_index, ist->st->index, layout_name);
    }
    if (!enc->channel_layout) {
        if (dec->channels == enc->channels) {
            enc->channel_layout = dec->channel_layout;
            return;
        } else {
            enc->channel_layout = av_get_default_channel_layout(enc->channels);
        }
        if (!enc->channel_layout) {
            av_log(NULL, AV_LOG_FATAL, "Unable to find default channel layout "
                   "for Output Stream #%d.%d\n", ost->file_index,
                   ost->st->index);
            exit_program(1);
        }
        av_get_channel_layout_string(layout_name, sizeof(layout_name),
                                     enc->channels, enc->channel_layout);
        av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for Output Stream "
               "#%d.%d : %s\n", ost->file_index, ost->st->index, layout_name);
    }
}

static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_t size)
{
    int fill_char = 0x00;
    if (sample_fmt == AV_SAMPLE_FMT_U8)
        fill_char = 0x80;
    memset(buf, fill_char, size);
}

static int encode_audio_frame(AVFormatContext *s, OutputStream *ost,
                              const uint8_t *buf, int buf_size)
{
    AVCodecContext *enc = ost->st->codec;
    AVFrame *frame = NULL;
    AVPacket pkt;
    int ret, got_packet;

    av_init_packet(&pkt);
    pkt.data = NULL;
    pkt.size = 0;

    if (buf) {
        if (!ost->output_frame) {
            ost->output_frame = avcodec_alloc_frame();
            if (!ost->output_frame) {
                av_log(NULL, AV_LOG_FATAL, "out-of-memory in encode_audio_frame()\n");
                exit_program(1);
            }
        }
        frame = ost->output_frame;
        if (frame->extended_data != frame->data)
            av_freep(&frame->extended_data);
        avcodec_get_frame_defaults(frame);

        frame->nb_samples  = buf_size /
                             (enc->channels * av_get_bytes_per_sample(enc->sample_fmt));
        if ((ret = avcodec_fill_audio_frame(frame, enc->channels, enc->sample_fmt,
                                            buf, buf_size, 1)) < 0) {
            av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
            exit_program(1);
        }

        if (!check_recording_time(ost))
            return 0;

        frame->pts = ost->sync_opts;
        ost->sync_opts += frame->nb_samples;
    }

    got_packet = 0;
    if (avcodec_encode_audio2(enc, &pkt, frame, &got_packet) < 0) {
        av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
        exit_program(1);
    }

    if (got_packet) {
        if (pkt.pts != AV_NOPTS_VALUE)
            pkt.pts      = av_rescale_q(pkt.pts,      enc->time_base, ost->st->time_base);
        if (pkt.dts != AV_NOPTS_VALUE)
            pkt.dts      = av_rescale_q(pkt.dts,      enc->time_base, ost->st->time_base);
        if (pkt.duration > 0)
            pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base);

        write_frame(s, &pkt, ost);

        audio_size += pkt.size;
    }

    return pkt.size;
}

static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc,
                                  int nb_samples)
{
    int64_t audio_buf_samples;
    int audio_buf_size;

    /* calculate required number of samples to allocate */
    audio_buf_samples = ((int64_t)nb_samples * enc->sample_rate + dec->sample_rate) /
                        dec->sample_rate;
    audio_buf_samples = 4 * audio_buf_samples + 16; // safety factors for resampling
    audio_buf_samples = FFMAX(audio_buf_samples, enc->frame_size);
    if (audio_buf_samples > INT_MAX)
        return AVERROR(EINVAL);

    audio_buf_size = av_samples_get_buffer_size(NULL, enc->channels,
                                                audio_buf_samples,
                                                enc->sample_fmt, 0);
    if (audio_buf_size < 0)
        return audio_buf_size;

    av_fast_malloc(&audio_buf, &allocated_audio_buf_size, audio_buf_size);
    if (!audio_buf)
        return AVERROR(ENOMEM);

    return 0;
}

static void do_audio_out(AVFormatContext *s, OutputStream *ost,
                         InputStream *ist, AVFrame *decoded_frame)
{
    uint8_t *buftmp;

    int size_out, frame_bytes, resample_changed;
    AVCodecContext *enc = ost->st->codec;
    AVCodecContext *dec = ist->st->codec;
    int osize = av_get_bytes_per_sample(enc->sample_fmt);
    int isize = av_get_bytes_per_sample(dec->sample_fmt);
    uint8_t *buf = decoded_frame->data[0];
    int size     = decoded_frame->nb_samples * dec->channels * isize;

    get_default_channel_layouts(ost, ist);

    if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples) < 0) {
        av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
        exit_program(1);
    }

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

    resample_changed = ost->resample_sample_fmt  != dec->sample_fmt ||
                       ost->resample_channels    != dec->channels   ||
                       ost->resample_sample_rate != dec->sample_rate;

    if ((ost->audio_resample && !ost->resample) || resample_changed) {
        if (resample_changed) {
            av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n",
                   ist->file_index, ist->st->index,
                   ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt), ost->resample_channels,
                   dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels);
            ost->resample_sample_fmt  = dec->sample_fmt;
            ost->resample_channels    = dec->channels;
            ost->resample_sample_rate = dec->sample_rate;
            if (ost->resample)
                audio_resample_close(ost->resample);
        }
        /* if audio_sync_method is >1 the resampler is needed for audio drift compensation */
        if (audio_sync_method <= 1 &&
            ost->resample_sample_fmt  == enc->sample_fmt &&
            ost->resample_channels    == enc->channels   &&
            ost->resample_sample_rate == enc->sample_rate) {
            ost->resample = NULL;
            ost->audio_resample = 0;
        } else if (ost->audio_resample) {
            if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
                av_log(NULL, AV_LOG_WARNING, "Using s16 intermediate sample format for resampling\n");
            ost->resample = av_audio_resample_init(enc->channels,    dec->channels,
                                                   enc->sample_rate, dec->sample_rate,
                                                   enc->sample_fmt,  dec->sample_fmt,
                                                   16, 10, 0, 0.8);
            if (!ost->resample) {
                av_log(NULL, AV_LOG_FATAL, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n",
                       dec->channels, dec->sample_rate,
                       enc->channels, enc->sample_rate);
                exit_program(1);
            }
        }
    }

#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b))
    if (!ost->audio_resample && dec->sample_fmt != enc->sample_fmt &&
        MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt) != ost->reformat_pair) {
        if (ost->reformat_ctx)
            av_audio_convert_free(ost->reformat_ctx);
        ost->reformat_ctx = av_audio_convert_alloc(enc->sample_fmt, 1,
                                                   dec->sample_fmt, 1, NULL, 0);
        if (!ost->reformat_ctx) {
            av_log(NULL, AV_LOG_FATAL, "Cannot convert %s sample format to %s sample format\n",
                   av_get_sample_fmt_name(dec->sample_fmt),
                   av_get_sample_fmt_name(enc->sample_fmt));
            exit_program(1);
        }
        ost->reformat_pair = MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt);
    }

    if (audio_sync_method > 0) {
        double delta = get_sync_ipts(ost, ist->last_dts) * enc->sample_rate - ost->sync_opts -
                       av_fifo_size(ost->fifo) / (enc->channels * osize);
        int idelta = delta * dec->sample_rate / enc->sample_rate;
        int byte_delta = idelta * isize * dec->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;
                    av_log(NULL, AV_LOG_VERBOSE, "discarding %d audio samples\n",
                           -byte_delta / (isize * dec->channels));
                    if (!size)
                        return;
                    ist->is_start = 0;
                } else {
                    av_fast_malloc(&async_buf, &allocated_async_buf_size,
                                   byte_delta + size);
                    if (!async_buf) {
                        av_log(NULL, AV_LOG_FATAL, "Out of memory in do_audio_out\n");
                        exit_program(1);
                    }

                    if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta) < 0) {
                        av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
                        exit_program(1);
                    }
                    ist->is_start = 0;

                    generate_silence(async_buf, dec->sample_fmt, byte_delta);
                    memcpy(async_buf + byte_delta, buf, size);
                    buf = async_buf;
                    size += byte_delta;
                    av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", idelta);
                }
            } else if (audio_sync_method > 1) {
                int comp = av_clip(delta, -audio_sync_method, audio_sync_method);
                av_assert0(ost->audio_resample);
                av_log(NULL, AV_LOG_VERBOSE, "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 if (audio_sync_method == 0)
        ost->sync_opts = lrintf(get_sync_ipts(ost, ist->last_dts) * enc->sample_rate) -
                                av_fifo_size(ost->fifo) / (enc->channels * osize); // FIXME wrong

    if (ost->audio_resample) {
        buftmp = audio_buf;
        size_out = audio_resample(ost->resample,
                                  (short *)buftmp, (short *)buf,
                                  size / (dec->channels * isize));
        size_out = size_out * enc->channels * osize;
    } else {
        buftmp = buf;
        size_out = size;
    }

    if (!ost->audio_resample && dec->sample_fmt != enc->sample_fmt) {
        const void *ibuf[6] = { buftmp };
        void *obuf[6]  = { audio_buf };
        int istride[6] = { isize };
        int ostride[6] = { osize };
        int len = size_out / istride[0];
        if (av_audio_convert(ost->reformat_ctx, obuf, ostride, ibuf, istride, len) < 0) {
            printf("av_audio_convert() failed\n");
            if (exit_on_error)
                exit_program(1);
            return;
        }
        buftmp = audio_buf;
        size_out = len * osize;
    }

    /* now encode as many frames as possible */
    if (!(enc->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) {
        /* output resampled raw samples */
        if (av_fifo_realloc2(ost->fifo, av_fifo_size(ost->fifo) + size_out) < 0) {
            av_log(NULL, AV_LOG_FATAL, "av_fifo_realloc2() failed\n");
            exit_program(1);
        }
        av_fifo_generic_write(ost->fifo, buftmp, size_out, NULL);

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

        while (av_fifo_size(ost->fifo) >= frame_bytes) {
            av_fifo_generic_read(ost->fifo, audio_buf, frame_bytes, NULL);
            encode_audio_frame(s, ost, audio_buf, frame_bytes);
        }
    } else {
        encode_audio_frame(s, ost, buftmp, size_out);
    }
}

static void pre_process_video_frame(InputStream *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) {
        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 (avpicture_deinterlace(picture2, picture,
                                 dec->pix_fmt, dec->width, dec->height) < 0) {
            /* if error, do not deinterlace */
            av_log(NULL, AV_LOG_WARNING, "Deinterlacing failed\n");
            av_free(buf);
            buf = NULL;
            picture2 = picture;
        }
    } else {
        picture2 = picture;
    }

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

static void do_subtitle_out(AVFormatContext *s,
                            OutputStream *ost,
                            InputStream *ist,
                            AVSubtitle *sub,
                            int64_t pts)
{
    static uint8_t *subtitle_out = NULL;
    int subtitle_out_max_size = 1024 * 1024;
    int subtitle_out_size, nb, i;
    AVCodecContext *enc;
    AVPacket pkt;

    if (pts == AV_NOPTS_VALUE) {
        av_log(NULL, AV_LOG_ERROR, "Subtitle packets must have a pts\n");
        if (exit_on_error)
            exit_program(1);
        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++) {
        ost->sync_opts = av_rescale_q(pts, ist->st->time_base, enc->time_base);
        if (!check_recording_time(ost))
            return;

        sub->pts = av_rescale_q(pts, ist->st->time_base, AV_TIME_BASE_Q);
        // start_display_time is required to be 0
        sub->pts               += av_rescale_q(sub->start_display_time, (AVRational){ 1, 1000 }, AV_TIME_BASE_Q);
        sub->end_display_time  -= sub->start_display_time;
        sub->start_display_time = 0;
        subtitle_out_size = avcodec_encode_subtitle(enc, subtitle_out,
                                                    subtitle_out_max_size, sub);
        if (subtitle_out_size < 0) {
            av_log(NULL, AV_LOG_FATAL, "Subtitle encoding failed\n");
            exit_program(1);
        }

        av_init_packet(&pkt);
        pkt.data = subtitle_out;
        pkt.size = subtitle_out_size;
        pkt.pts  = av_rescale_q(sub->pts, AV_TIME_BASE_Q, 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);
    }
}

static void do_video_out(AVFormatContext *s,
                         OutputStream *ost,
                         AVFrame *in_picture,
                         int *frame_size, float quality)
{
    int nb_frames, i, ret, format_video_sync;
    AVCodecContext *enc;
    double sync_ipts, delta;

    enc = ost->st->codec;

    sync_ipts = get_sync_ipts(ost, in_picture->pts) / av_q2d(enc->time_base);
    delta = sync_ipts - ost->sync_opts;

    /* by default, we output a single frame */
    nb_frames = 1;

    *frame_size = 0;

    format_video_sync = video_sync_method;
    if (format_video_sync == VSYNC_AUTO)
        format_video_sync = (s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH :
                            (s->oformat->flags & AVFMT_VARIABLE_FPS) ? VSYNC_VFR : VSYNC_CFR;

    switch (format_video_sync) {
    case VSYNC_CFR:
        // FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
        if (delta < -1.1)
            nb_frames = 0;
        else if (delta > 1.1)
            nb_frames = lrintf(delta);
        break;
    case VSYNC_VFR:
        if (delta <= -0.6)
            nb_frames = 0;
        else if (delta > 0.6)
            ost->sync_opts = lrintf(sync_ipts);
        break;
    case VSYNC_PASSTHROUGH:
        ost->sync_opts = lrintf(sync_ipts);
        break;
    default:
        av_assert0(0);
    }

    nb_frames = FFMIN(nb_frames, ost->max_frames - ost->frame_number);
    if (nb_frames == 0) {
        nb_frames_drop++;
        av_log(NULL, AV_LOG_VERBOSE, "*** drop!\n");
        return;
    } else if (nb_frames > 1) {
        nb_frames_dup += nb_frames - 1;
        av_log(NULL, AV_LOG_VERBOSE, "*** %d dup!\n", nb_frames - 1);
    }

    if (!ost->frame_number)
        ost->first_pts = ost->sync_opts;

    /* duplicates frame if needed */
    for (i = 0; i < nb_frames; i++) {
        AVPacket pkt;
        av_init_packet(&pkt);
        pkt.data = NULL;
        pkt.size = 0;

        if (!check_recording_time(ost))
            return;

        if (s->oformat->flags & AVFMT_RAWPICTURE &&
            enc->codec->id == CODEC_ID_RAWVIDEO) {
            /* raw pictures are written as AVPicture structure to
               avoid any copies. We support temporarily the older
               method. */
            enc->coded_frame->interlaced_frame = in_picture->interlaced_frame;
            enc->coded_frame->top_field_first  = in_picture->top_field_first;
            pkt.data   = (uint8_t *)in_picture;
            pkt.size   =  sizeof(AVPicture);
            pkt.pts    = av_rescale_q(ost->sync_opts, enc->time_base, ost->st->time_base);
            pkt.flags |= AV_PKT_FLAG_KEY;

            write_frame(s, &pkt, ost);
        } else {
            int got_packet;
            AVFrame big_picture;

            big_picture = *in_picture;
            /* better than nothing: use input picture interlaced
               settings */
            big_picture.interlaced_frame = in_picture->interlaced_frame;
            if (ost->st->codec->flags & (CODEC_FLAG_INTERLACED_DCT|CODEC_FLAG_INTERLACED_ME)) {
                if (ost->top_field_first == -1)
                    big_picture.top_field_first = in_picture->top_field_first;
                else
                    big_picture.top_field_first = !!ost->top_field_first;
            }

            /* handles same_quant here. This is not correct because it may
               not be a global option */
            big_picture.quality = quality;
            if (!enc->me_threshold)
                big_picture.pict_type = 0;
            big_picture.pts = ost->sync_opts;
            if (ost->forced_kf_index < ost->forced_kf_count &&
                big_picture.pts >= ost->forced_kf_pts[ost->forced_kf_index]) {
                big_picture.pict_type = AV_PICTURE_TYPE_I;
                ost->forced_kf_index++;
            }
            ret = avcodec_encode_video2(enc, &pkt, &big_picture, &got_packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_FATAL, "Video encoding failed\n");
                exit_program(1);
            }

            if (got_packet) {
                if (pkt.pts != AV_NOPTS_VALUE)
                    pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base);
                if (pkt.dts != AV_NOPTS_VALUE)
                    pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base);

                write_frame(s, &pkt, ost);
                *frame_size = pkt.size;
                video_size += pkt.size;

                /* if two pass, output log */
                if (ost->logfile && enc->stats_out) {
                    fprintf(ost->logfile, "%s", enc->stats_out);
                }
            }
        }
        ost->sync_opts++;
        /*
         * For video, number of frames in == number of packets out.
         * But there may be reordering, so we can't throw away frames on encoder
         * flush, we need to limit them here, before they go into encoder.
         */
        ost->frame_number++;
    }
}

static double psnr(double d)
{
    return -10.0 * log(d) / log(10.0);
}

static void do_video_stats(AVFormatContext *os, OutputStream *ost,
                           int frame_size)
{
    AVCodecContext *enc;
    int frame_number;
    double ti1, bitrate, avg_bitrate;

    /* this is executed just the first time do_video_stats is called */
    if (!vstats_file) {
        vstats_file = fopen(vstats_filename, "w");
        if (!vstats_file) {
            perror("fopen");
            exit_program(1);
        }
    }

    enc = ost->st->codec;
    if (enc->codec_type == AVMEDIA_TYPE_VIDEO) {
        frame_number = ost->frame_number;
        fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number, enc->coded_frame->quality / (float)FF_QP2LAMBDA);
        if (enc->flags&CODEC_FLAG_PSNR)
            fprintf(vstats_file, "PSNR= %6.2f ", psnr(enc->coded_frame->error[0] / (enc->width * enc->height * 255.0 * 255.0)));

        fprintf(vstats_file,"f_size= %6d ", frame_size);
        /* compute pts value */
        ti1 = ost->sync_opts * av_q2d(enc->time_base);
        if (ti1 < 0.01)
            ti1 = 0.01;

        bitrate     = (frame_size * 8) / av_q2d(enc->time_base) / 1000.0;
        avg_bitrate = (double)(video_size * 8) / ti1 / 1000.0;
        fprintf(vstats_file, "s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s ",
               (double)video_size / 1024, ti1, bitrate, avg_bitrate);
        fprintf(vstats_file, "type= %c\n", av_get_picture_type_char(enc->coded_frame->pict_type));
    }
}

/* check for new output on any of the filtergraphs */
static int poll_filters(void)
{
    AVFilterBufferRef *picref;
    AVFrame *filtered_frame = NULL;
    int i, frame_size, ret;

    for (i = 0; i < nb_output_streams; i++) {
        OutputStream *ost = output_streams[i];
        OutputFile    *of = output_files[ost->file_index];

        if (!ost->filter || ost->is_past_recording_time)
            continue;

        if (!ost->filtered_frame && !(ost->filtered_frame = avcodec_alloc_frame())) {
            return AVERROR(ENOMEM);
        } else
            avcodec_get_frame_defaults(ost->filtered_frame);
        filtered_frame = ost->filtered_frame;

        while (avfilter_poll_frame(ost->filter->filter->inputs[0])) {
            AVRational ist_pts_tb;
            if ((ret = get_filtered_video_frame(ost->filter->filter,
                                                filtered_frame, &picref,
                                                &ist_pts_tb)) < 0)
                return ret;
            filtered_frame->pts = av_rescale_q(picref->pts, ist_pts_tb, AV_TIME_BASE_Q);

            if (of->start_time && filtered_frame->pts < of->start_time)
                return 0;

            switch (ost->filter->filter->inputs[0]->type) {
            case AVMEDIA_TYPE_VIDEO:
                if (!ost->frame_aspect_ratio)
                    ost->st->codec->sample_aspect_ratio = picref->video->pixel_aspect;

                do_video_out(of->ctx, ost, filtered_frame, &frame_size,
                             same_quant ? ost->last_quality :
                                          ost->st->codec->global_quality);
                if (vstats_filename && frame_size)
                    do_video_stats(of->ctx, ost, frame_size);
                break;
            default:
                // TODO support audio/subtitle filters
                av_assert0(0);
            }

            avfilter_unref_buffer(picref);
        }
    }
    return 0;
}

static void print_report(int is_last_report, int64_t timer_start)
{
    char buf[1024];
    OutputStream *ost;
    AVFormatContext *oc;
    int64_t total_size;
    AVCodecContext *enc;
    int frame_number, vid, i;
    double bitrate, ti1, pts;
    static int64_t last_time = -1;
    static int qp_histogram[52];

    if (!print_stats && !is_last_report)
        return;

    if (!is_last_report) {
        int64_t cur_time;
        /* display the report every 0.5 seconds */
        cur_time = av_gettime();
        if (last_time == -1) {
            last_time = cur_time;
            return;
        }
        if ((cur_time - last_time) < 500000)
            return;
        last_time = cur_time;
    }


    oc = output_files[0]->ctx;

    total_size = avio_size(oc->pb);
    if (total_size < 0) // FIXME improve avio_size() so it works with non seekable output too
        total_size = avio_tell(oc->pb);

    buf[0] = '\0';
    ti1 = 1e10;
    vid = 0;
    for (i = 0; i < nb_output_streams; i++) {
        float q = -1;
        ost = output_streams[i];
        enc = ost->st->codec;
        if (!ost->stream_copy && enc->coded_frame)
            q = enc->coded_frame->quality / (float)FF_QP2LAMBDA;
        if (vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) {
            snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "q=%2.1f ", q);
        }
        if (!vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) {
            float t = (av_gettime() - timer_start) / 1000000.0;

            frame_number = ost->frame_number;
            snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "frame=%5d fps=%3d q=%3.1f ",
                     frame_number, (t > 1) ? (int)(frame_number / t + 0.5) : 0, q);
            if (is_last_report)
                snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "L");
            if (qp_hist) {
                int j;
                int qp = lrintf(q);
                if (qp >= 0 && qp < FF_ARRAY_ELEMS(qp_histogram))
                    qp_histogram[qp]++;
                for (j = 0; j < 32; j++)
                    snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%X", (int)lrintf(log(qp_histogram[j] + 1) / log(2)));
            }
            if (enc->flags&CODEC_FLAG_PSNR) {
                int j;
                double error, error_sum = 0;
                double scale, scale_sum = 0;
                char type[3] = { 'Y','U','V' };
                snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "PSNR=");
                for (j = 0; j < 3; j++) {
                    if (is_last_report) {
                        error = enc->error[j];
                        scale = enc->width * enc->height * 255.0 * 255.0 * frame_number;
                    } else {
                        error = enc->coded_frame->error[j];
                        scale = enc->width * enc->height * 255.0 * 255.0;
                    }
                    if (j)
                        scale /= 4;
                    error_sum += error;
                    scale_sum += scale;
                    snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%c:%2.2f ", type[j], psnr(error / scale));
                }
                snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "*:%2.2f ", psnr(error_sum / scale_sum));
            }
            vid = 1;
        }
        /* compute min output value */
        pts = (double)ost->st->pts.val * av_q2d(ost->st->time_base);
        if ((pts < ti1) && (pts > 0))
            ti1 = pts;
    }
    if (ti1 < 0.01)
        ti1 = 0.01;

    bitrate = (double)(total_size * 8) / ti1 / 1000.0;

    snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf),
            "size=%8.0fkB time=%0.2f bitrate=%6.1fkbits/s",
            (double)total_size / 1024, ti1, bitrate);

    if (nb_frames_dup || nb_frames_drop)
        snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), " dup=%d drop=%d",
                nb_frames_dup, nb_frames_drop);

    av_log(NULL, AV_LOG_INFO, "%s    \r", buf);

    fflush(stderr);

    if (is_last_report) {
        int64_t raw= audio_size + video_size + extra_size;
        av_log(NULL, AV_LOG_INFO, "\n");
        av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB global headers:%1.0fkB muxing overhead %f%%\n",
               video_size / 1024.0,
               audio_size / 1024.0,
               extra_size / 1024.0,
               100.0 * (total_size - raw) / raw
        );
    }
}

static void flush_encoders(void)
{
    int i, ret;

    for (i = 0; i < nb_output_streams; i++) {
        OutputStream   *ost = output_streams[i];
        AVCodecContext *enc = ost->st->codec;
        AVFormatContext *os = output_files[ost->file_index]->ctx;
        int stop_encoding = 0;

        if (!ost->encoding_needed)
            continue;

        if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <= 1)
            continue;
        if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE) && enc->codec->id == CODEC_ID_RAWVIDEO)
            continue;

        for (;;) {
            AVPacket pkt;
            int fifo_bytes, got_packet;
            av_init_packet(&pkt);