Newest 'ffmpeg' Questions - Stack Overflow

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  • FFmpeg Audio Merge Delays

    26 novembre 2016, par Matt Wall

    I'm trying to run this kind if FFmpeg command to stitch two videos together and merge the audio. 

     ffmpeg -i C:\Santa\Staging\santa_2016.11.21.12.01.33_Komo_Kwigly.flv -i C:\Santa\Staging\door_2016.11.21.12.01.33_Komo_Kwigly.flv -loop 1 -i C:\watermark.png -filter_complex "[0]setpts=PTS-STARTPTS,transpose=1[a];[1]setpts=PTS-STARTPTS,transpose=1[b];[a][b]hstack[c];[c][2]overlay=W-w-5:H-h-5:shortest=1[v];[0][1]amerge,pan=stereo:c0code>

    And I'm getting this strange error and audio sync issues:

    FFmpeg Error

    Does anyone have any tips for what to do? I've tried a lot and cannot seem to change the behavior. The sources are fine, too.

    The only other relevant log seems to be:

    [Parsed_pan_7 @ 00000000006a8440] This syntax is deprecated. Use '|' to separate the list items.
[Parsed_pan_7 @ 00000000006a8ac0] This syntax is deprecated. Use '|' to separate the list items.
[Parsed_amerge_6 @ 00000000006a8440] No channel layout for input 1
[Parsed_amerge_6 @ 00000000006a8440] Input channel layouts overlap: output layout will be determined by the number of di    stinct input channels

  • FFMPEG play in HTML while converting

    26 novembre 2016, par OwenHoward

    Hello :)

    I need to be able to display any popular video format (MKV, FLV, MP4,...) inside a browser WHILE converting. To do this I've turned to FFMPEG to convert any video to .ovg and spent hours on Google, forums and in documentation but I can't find a way to play the converted video real-time.

    Does anyone know how you can play a video in HTML while FFMPEG is still converting it?

  • Encoding AAC with ffmpeg (c++)

    26 novembre 2016, par Mockarutan

    I'm working on video encoding that will be used in a Unity plugin. I have made image encoding work, but now I'm at the audio. So trying only with the audio in to a mp4 file with AAC encoding. And I'm stuck. The resulting file does not contain anything. Also, from what I understand, AAC in ffmpeg only supports AV_SAMPLE_FMT_FLTP, that's why I use it. Here's my code:

    Setup:

    int initialize_encoding_audio(const char *filename)
{
    int ret;
    AVCodecID aud_codec_id = AV_CODEC_ID_AAC;
    AVSampleFormat sample_fmt = AV_SAMPLE_FMT_FLTP;

    avcodec_register_all();
    av_register_all();

    aud_codec = avcodec_find_encoder(aud_codec_id);
    avcodec_register(aud_codec);

    if (!aud_codec)
        return COULD_NOT_FIND_AUD_CODEC;

    aud_codec_context = avcodec_alloc_context3(aud_codec);
    if (!aud_codec_context)
        return CONTEXT_CREATION_ERROR;

    aud_codec_context->bit_rate = 192000;
    aud_codec_context->sample_rate = select_sample_rate(aud_codec);
    aud_codec_context->sample_fmt = sample_fmt;
    aud_codec_context->channel_layout = AV_CH_LAYOUT_STEREO;
    aud_codec_context->channels = av_get_channel_layout_nb_channels(aud_codec_context->channel_layout);

    aud_codec_context->codec = aud_codec;
    aud_codec_context->codec_id = aud_codec_id;

    ret = avcodec_open2(aud_codec_context, aud_codec, NULL);

    if (ret < 0)
        return COULD_NOT_OPEN_AUD_CODEC;

    outctx = avformat_alloc_context();
    ret = avformat_alloc_output_context2(&outctx, NULL, "mp4", filename);

    outctx->audio_codec = aud_codec;
    outctx->audio_codec_id = aud_codec_id;

    audio_st = avformat_new_stream(outctx, aud_codec);

    audio_st->codecpar->bit_rate = aud_codec_context->bit_rate;
    audio_st->codecpar->sample_rate = aud_codec_context->sample_rate;
    audio_st->codecpar->channels = aud_codec_context->channels;
    audio_st->codecpar->channel_layout = aud_codec_context->channel_layout;
    audio_st->codecpar->codec_id = aud_codec_id;
    audio_st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
    audio_st->codecpar->format = sample_fmt;
    audio_st->codecpar->frame_size = aud_codec_context->frame_size;
    audio_st->codecpar->block_align = aud_codec_context->block_align;
    audio_st->codecpar->initial_padding = aud_codec_context->initial_padding;

    outctx->streams = new AVStream*[1];
    outctx->streams[0] = audio_st;

    av_dump_format(outctx, 0, filename, 1);

    if (!(outctx->oformat->flags & AVFMT_NOFILE))
    {
        if (avio_open(&outctx->pb, filename, AVIO_FLAG_WRITE) < 0)
            return COULD_NOT_OPEN_FILE;
    }

    ret = avformat_write_header(outctx, NULL);

    aud_frame = av_frame_alloc();
    aud_frame->nb_samples = aud_codec_context->frame_size;
    aud_frame->format = aud_codec_context->sample_fmt;
    aud_frame->channel_layout = aud_codec_context->channel_layout;

    int buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
        aud_codec_context->sample_fmt, 0);

    av_frame_get_buffer(aud_frame, buffer_size / aud_codec_context->channels);

    if (!aud_frame)
        return COULD_NOT_ALLOCATE_FRAME;

    aud_frame_counter = 0;

    return 0;
}

    Encoding:

    int encode_audio_samples(uint8_t **aud_samples)
{
    int ret;

    int buffer_size = av_samples_get_buffer_size(NULL, aud_codec_context->channels, aud_codec_context->frame_size,
        aud_codec_context->sample_fmt, 0);

    for (size_t i = 0; i < buffer_size / aud_codec_context->channels; i++)
    {
        aud_frame->data[0][i] = aud_samples[0][i];
        aud_frame->data[1][i] = aud_samples[1][i];
    }

    aud_frame->pts = aud_frame_counter++;

    ret = avcodec_send_frame(aud_codec_context, aud_frame);
    if (ret < 0)
        return ERROR_ENCODING_SAMPLES_SEND;

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

    fflush(stdout);

    while (true)
    {
        ret = avcodec_receive_packet(aud_codec_context, &pkt);
        if (!ret)
        {
            av_packet_rescale_ts(&pkt, aud_codec_context->time_base, audio_st->time_base);

            pkt.stream_index = audio_st->index;
            av_write_frame(outctx, &pkt);
            av_packet_unref(&pkt);
        }
        if (ret == AVERROR(EAGAIN))
            break;
        else if (ret < 0)
            return ERROR_ENCODING_SAMPLES_RECEIVE;
        else
            break;
    }

    return 0;
}

    Finish encoding:

    int finish_audio_encoding()
{
    AVPacket pkt;
    av_init_packet(&pkt);
    pkt.data = NULL;
    pkt.size = 0;

    fflush(stdout);

    int ret = avcodec_send_frame(aud_codec_context, NULL);
    if (ret < 0)
        return ERROR_ENCODING_FRAME_SEND;

    while (true)
    {
        ret = avcodec_receive_packet(aud_codec_context, &pkt);
        if (!ret)
        {
            if (pkt.pts != AV_NOPTS_VALUE)
                pkt.pts = av_rescale_q(pkt.pts, aud_codec_context->time_base, audio_st->time_base);
            if (pkt.dts != AV_NOPTS_VALUE)
                pkt.dts = av_rescale_q(pkt.dts, aud_codec_context->time_base, audio_st->time_base);

            av_write_frame(outctx, &pkt);
            av_packet_unref(&pkt);
        }
        if (ret == -AVERROR(AVERROR_EOF))
            break;
        else if (ret < 0)
            return ERROR_ENCODING_FRAME_RECEIVE;
    }

    av_write_trailer(outctx);
}

    Main:

    void get_audio_frame(float_t *left_samples, float_t *right_samples, int frame_size, float* t, float* tincr, float* tincr2)
{
    int j, i;
    float v;
    for (j = 0; j < frame_size; j++)
    {
        v = sin(*t);
        *left_samples = v;
        *right_samples = v;

        left_samples++;
        right_samples++;

        *t += *tincr;
        *tincr += *tincr2;
    }
}

int main()
{
    int frame_rate = 30;  // this should be like 96000 / 1024 or somthing i guess?
    float t, tincr, tincr2;

    initialize_encoding_audio("audio.mp4");

    int sec = 50;

    float_t** aud_samples;
    int src_samples_linesize;
    int src_nb_samples = 1024;
    int src_channels = 2;

    int ret = av_samples_alloc_array_and_samples((uint8_t***)&aud_samples, &src_samples_linesize, src_channels,
        src_nb_samples, AV_SAMPLE_FMT_FLTP, 0);


    t = 0;
    tincr = 0;
    tincr2 = 0;

    for (size_t i = 0; i < frame_rate * sec; i++)
    {
        get_audio_frame(aud_samples[0], aud_samples[1], src_nb_samples, &t, &tincr, &tincr2);

        encode_audio_samples((uint8_t **)aud_samples);

    }

    finish_audio_encoding();
    //cleanup();

    return 0;
}

    I guess the first thing that I would want to make sure I got right is the synthetic sound generation and how I transfer that to the AVFrame. Are my conversions correct? But feel free to point out anything that might be wrong.

    Thanks in advance!

    Edit: the whole source: http://pastebin.com/jYtmkhek

    Edit2: Added initialization of tincr & tincr2

  • Backslash in Ruby String and Special FFmpeg Characters

    26 novembre 2016, par valii2334

    I enter a string in my xcode project and I send it via api to my rails application where I need to put the string over a picture.

    Let's say that in xcode (iOS app) I type the string /\%" ' . The string it's set in rails ( a field of an object of type string ) like this /\\%\" ' . But the problem is that in convert % and ' are special characters and I need to prefix them with \ to work (like ruby does with " and \). The convert command it's this:

    command = "convert image.jpg -gravity North -pointsize 40 -fill '#FFFFFF' -annotate +0+50 '#{@text}' image2.jpg"

    @text should contain a string such image2.jpg will contain /\%" '

    This is just an example with this special characters. It should display any string entered in iOS no metter what language.

  • No sound in output clips. RTSP to mp4 ffmpeg

    26 novembre 2016, par Hamad

    I was using the following command for recording input streams to mp4 into 10 seconds clip. But since few days there is no voice in the output mp4 clips. Command is

     ffmpeg -i rtsp://localhost:8554/mystream -c copy -bsf:a aac_adtstoasc -f segment -segment_time 10 -reset_timestamps 1 -map 0 output%d.mp4

    ffmpeg output

    I think its because of the format of input stream but i don't know for sure and also, i don't know how to solve it.

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