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  • How to force "full range" flag on export

    28 janvier 2019, par Emanuele Vissani

    I have an ffmpeg command to remap audio tracks to descrete channels in a ProRes 4444 quicktime file. Even if the input video is copied to the output, the exported file is interpreted by a professional video player software as video range (16-234 values) instead of the original full range (0-255 values), making it look more contrasted. The content is actually correct, changing manually the range setting in the player software gives back the right light range, so I think the output file just lose some kind of range flag.

    I already tried the following options without results:

    -colorspace bt709 -movflags +write_colr

    -dst_range 1 -color_range 2

    -vf scale=out_range=full

    -vf scale=in_range=full:out_range=full

    Original command is:

    ffmpeg -i F:\_IMPORT\TST_ProRes4444_4k.mov -map 0:0 -c copy -map 0:1 -c copy -map_channel 0.2.0:0.2 -c:a pcm_s24le F:\_EXPORT\TEST\test.mov
    

    Thank you for your help.

  • How to change wave file frequency while retaining sample rate [duplicate]

    28 janvier 2019, par q1w2e3r4t5y6

    This question already has an answer here:

    I am wondering how to change a wave file's frequency while retaining its sample rate. I have tried:

    ffmpeg -i $1 -filter:a "asetrate=$2" -y $3
    

    The second argument here changes drastically, by the way. However, this changes the sample rate, making it impossible to concatenate with sox. How can I modify the above code so that it changes the frequency while still having the same sample rate?

  • Change wave file frequency while retaining sample rate bash

    28 janvier 2019, par q1w2e3r4t5y6

    I am wondering how to change a wave file's frequency while its retaining sample rate. I have tried:

    ffmpeg -i $1 -filter:a "asetrate=$2" -y $3
    

    The second argument here changes drastically, by the way. However, this changes the sample rate, making it impossible to concatenate with sox. How can I make the above code so that it changes the frequency while still having the same sample rate? Thanks!

  • ffmpeg, v4l, snd_aloop ... sound asyncron (alsa buffer xrun)

    28 janvier 2019, par Tobias

    I'm trying to create a stream that automatically reloads random inputs. I would like to extend this to a database later.

    Each time ffmpeg finishes and starts again, so the input changes, the connection to the rtmp is interrupted briefly causing the whole connection breaks down. I then tried to separate audio and video, to send them to virtual devices and read from there again. Split the stream on virtual devices, reassemble them directly and send them to rtmp. If the input is then exchanged, the sending to the devices interrupts what does not bother the second ffmpeg. As soon as I stop sending to the devices the fps go very slowly (10 - 20 sec) from 25 to 0. Only then does the transmitter ffmpeg break the connection to the rtmp. The script which exchanges the inputs needs only one second. A practical test showed that everything works as desired.

    I can quite comfortably change the input while the second ffmpeg maintains the stream ...

    The joy did not last long. The sound is good 1 sec delayed. But sporadically. Sometimes everything works great. Sometimes the sound is offset.

    I wrote several scripts for this.

    Background:

    1. File is selected by random
    2. Media file is split and written to / dev / video0 (v4l loopback) and alsa default (snd_aloop loopback)
    3. Put the splits together again and stream them to a rtmp server

    Code that selects the input and sends to / dev / video0 and alsa default

    #!/bin/bash
    
    cat /dev/null > log
    
    while true;
    do
    
      WATERMARK="watermark.png";
      dir='/homeXXXXXXXXXX/mix'
      file=`/bin/ls -1 "$dir" | sort --random-sort | head -1`
      DATEI=`readlink --canonicalize "$dir/$file"` # Converts to full path
    
    if [ -z $DATEI ]
    then
      echo "Keine Datei gefunden" >> log;
    else
      START=$(date +%s);
      echo "Sende $DATEI" >> log;
      ffmpeg -re -y -i "$DATEI" -c:v libx264 -vf "fps=25,scale=640:480,setdar=4:3" -async 1 -pix_fmt yuv420p -preset ultrafast -map 0:0 -f v4l2 -vcodec rawvideo /dev/video0 -f alsa default
    fi
    
    DOKILL=`cat kill`;
    if [ "$DOKILL" = "1"]
    then
      break;
    fi
    
    done
    

    The Output

    ./run.sh 
    ffmpeg version 3.2.12-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers
      built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
      configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
      libavutil      55. 34.101 / 55. 34.101
      libavcodec     57. 64.101 / 57. 64.101
      libavformat    57. 56.101 / 57. 56.101
      libavdevice    57.  1.100 / 57.  1.100
      libavfilter     6. 65.100 /  6. 65.100
      libavresample   3.  1.  0 /  3.  1.  0
      libswscale      4.  2.100 /  4.  2.100
      libswresample   2.  3.100 /  2.  3.100
      libpostproc    54.  1.100 / 54.  1.100
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/mix/XXXXXXXXXXXXX.mp4':
      Metadata:
        major_brand     : isom
        minor_version   : 512
        compatible_brands: isomiso2avc1mp41
        title           : XXXXXXXXXXXXXXX 
    
        encoder         : Lavf57.41.100
      Duration: 00:03:53.48, start: 0.000000, bitrate: 2705 kb/s
        Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 2573 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc (default)
        Metadata:
          handler_name    : VideoHandler
        Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 127 kb/s (default)
        Metadata:
          handler_name    : SoundHandler
    Codec AVOption preset (Configuration preset) specified for output file #0 (/dev/video0) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
    [Parsed_setdar_2 @ 0x5571234fe020] num:den syntax is deprecated, please use num/den or named options instead
    -async is forwarded to lavfi similarly to -af aresample=async=1:min_hard_comp=0.100000:first_pts=0.
    Output #0, v4l2, to '/dev/video0':
      Metadata:
        major_brand     : isom
        minor_version   : 512
        compatible_brands: isomiso2avc1mp41
        title           : XXXXXXXXXXX 
    
        encoder         : Lavf57.56.101
        Stream #0:0(und): Video: rawvideo (I420 / 0x30323449), yuv420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc (default)
        Metadata:
          handler_name    : VideoHandler
          encoder         : Lavc57.64.101 rawvideo
    Output #1, alsa, to 'default':
      Metadata:
        major_brand     : isom
        minor_version   : 512
        compatible_brands: isomiso2avc1mp41
        title           : XXXXXXXXXX 
    
        encoder         : Lavf57.56.101
        Stream #1:0(und): Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s (default)
        Metadata:
          handler_name    : SoundHandler
          encoder         : Lavc57.64.101 pcm_s16le
    Stream mapping:
      Stream #0:0 -> #0:0 (h264 (native) -> rawvideo (native))
      Stream #0:1 -> #1:0 (aac (native) -> pcm_s16le (native))
    Press [q] to stop, [?] for help
    frame=  736 fps= 24 q=-0.0 Lsize=N/A time=00:00:29.67 bitrate=N/A speed=0.979x    
    video:331200kB audio:5112kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
    Exiting normally, received signal 2.
    

    The send script

    #!/bin/bash
    
    IP="XXXXXXXXX";
    
    ffmpeg -f video4linux2 -i /dev/video0 -f alsa -acodec pcm_s16le -i default -f flv -async 1 -pix_fmt yuv420p -preset ultrafast -vcodec libx264 -r 25 -s 640x260 -acodec aac rtmp://$IP:1935/live/test
    

    The Output

    ./send_stream.sh 
    ffmpeg version 3.2.12-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers
      built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
      configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
      libavutil      55. 34.101 / 55. 34.101
      libavcodec     57. 64.101 / 57. 64.101
      libavformat    57. 56.101 / 57. 56.101
      libavdevice    57.  1.100 / 57.  1.100
      libavfilter     6. 65.100 /  6. 65.100
      libavresample   3.  1.  0 /  3.  1.  0
      libswscale      4.  2.100 /  4.  2.100
      libswresample   2.  3.100 /  2.  3.100
      libpostproc    54.  1.100 / 54.  1.100
    Input #0, video4linux2,v4l2, from '/dev/video0':
      Duration: N/A, start: 1548393682.674066, bitrate: 110592 kb/s
        Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 640x480, 110592 kb/s, 30 fps, 30 tbr, 1000k tbn, 1000k tbc
    Guessed Channel Layout for Input Stream #1.0 : stereo
    Input #1, alsa, from 'default':
      Duration: N/A, start: 1548393682.677901, bitrate: 1536 kb/s
        Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    -async is forwarded to lavfi similarly to -af aresample=async=1:min_hard_comp=0.100000:first_pts=0.
    [libx264 @ 0x55e22cfa4f00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
    [libx264 @ 0x55e22cfa4f00] profile Constrained Baseline, level 2.1
    [libx264 @ 0x55e22cfa4f00] 264 - core 148 r2748 97eaef2 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
    Output #0, flv, to 'rtmp://XXXXXXXXXXX:1935/live/test':
      Metadata:
        encoder         : Lavf57.56.101
        Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 640x260, q=-1--1, 25 fps, 1k tbn, 25 tbc
        Metadata:
          encoder         : Lavc57.64.101 libx264
        Side data:
          cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
        Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 48000 Hz, stereo, fltp, 128 kb/s
        Metadata:
          encoder         : Lavc57.64.101 aac
    Stream mapping:
      Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
      Stream #1:0 -> #0:1 (pcm_s16le (native) -> aac (native))
    Press [q] to stop, [?] for help
    [alsa @ 0x55e22cf87300] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
    [video4linux2,v4l2 @ 0x55e22cf84fe0] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
    Past duration 0.613319 too large    7344kB time=00:01:05.85 bitrate= 913.5kbits/s speed=1.04x    
    Past duration 0.614372 too large    7644kB time=00:01:08.39 bitrate= 915.6kbits/s speed=1.04x    
    Past duration 0.609749 too large    7834kB time=00:01:10.91 bitrate= 905.0kbits/s speed=1.04x    
    Past duration 0.604362 too large    8038kB time=00:01:12.92 bitrate= 903.0kbits/s speed=1.04x    
    Past duration 0.609489 too large    8070kB time=00:01:13.45 bitrate= 900.1kbits/s speed=1.04x    
    Past duration 0.615013 too large    8094kB time=00:01:13.94 bitrate= 896.8kbits/s speed=1.04x    
    Past duration 0.610893 too large    8179kB time=00:01:14.94 bitrate= 894.0kbits/s speed=1.04x    
    Past duration 0.664711 too large
    Past duration 0.639565 too large    8263kB time=00:01:15.47 bitrate= 896.8kbits/s speed=1.04x    
    Past duration 0.668999 too large    8339kB time=00:01:15.94 bitrate= 899.5kbits/s speed=1.04x    
    Past duration 0.605766 too large
    Past duration 0.633049 too large    8399kB time=00:01:16.48 bitrate= 899.6kbits/s speed=1.04x    
    Past duration 0.674599 too large
    Past duration 0.616035 too large    8451kB time=00:01:16.95 bitrate= 899.7kbits/s speed=1.04x    
    Past duration 0.656136 too large
    Past duration 0.604195 too large
    Past duration 0.601387 too large    8512kB time=00:01:17.46 bitrate= 900.2kbits/s speed=1.04x    
    Past duration 0.621895 too large    8565kB time=00:01:17.95 bitrate= 900.1kbits/s speed=1.04x    
    Past duration 0.670937 too large    8605kB time=00:01:18.46 bitrate= 898.4kbits/s speed=1.04x    
    Past duration 0.604500 too large    8642kB time=00:01:18.99 bitrate= 896.2kbits/s speed=1.04x    
    frame= 1913 fps= 25 q=-1.0 Lsize=    8670kB time=00:01:19.48 bitrate= 893.6kbits/s speed=1.04x    
    video:7290kB audio:1280kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.160292%
    [libx264 @ 0x55e22cfa4f00] frame I:8     Avg QP:18.25  size: 15502
    [libx264 @ 0x55e22cfa4f00] frame P:1905  Avg QP:20.95  size:  3853
    [libx264 @ 0x55e22cfa4f00] mb I  I16..4: 100.0%  0.0%  0.0%
    [libx264 @ 0x55e22cfa4f00] mb P  I16..4:  6.4%  0.0%  0.0%  P16..4: 38.1%  0.0%  0.0%  0.0%  0.0%    skip:55.5%
    [libx264 @ 0x55e22cfa4f00] coded y,uvDC,uvAC intra: 46.0% 30.3% 13.4% inter: 20.1% 9.8% 1.1%
    [libx264 @ 0x55e22cfa4f00] i16 v,h,dc,p: 47% 34% 10%  9%
    [libx264 @ 0x55e22cfa4f00] i8c dc,h,v,p: 45% 28% 22%  5%
    [libx264 @ 0x55e22cfa4f00] kb/s:750.98
    [aac @ 0x55e22cfa62a0] Qavg: 579.067
    Exiting normally, received signal 2.
    

    First everything is fine and then comes

    Past duration 0.616035 too large    8451kB time=00:01:16.95 bitrate= 899.7kbits/s speed=1.04x    
    Past duration 0.656136 too large
    Past duration 0.604195 too large
    Past duration 0.601387 too large    8512kB time=00:01:17.46 bitrate= 900.2kbits/s speed=1.04x 
    

    And then when that comes, dives in the first window, so in the ffmpeg sends the input:

    Stream mapping:
      Stream #0:0 -> #0:0 (h264 (native) -> rawvideo (native))
      Stream #0:1 -> #1:0 (aac (native) -> pcm_s16le (native))
    Press [q] to stop, [?] for help
    frame=    9 fps=0.0 q=-0.0 size=N/A time=00:00:00.36 bitrate=N/A dup=1 drop=0 spframe=   21 fps= 21 q=-0.0 size=N/A time=00:00:00.84 bitrate=N/A dup=1 drop=0 sp[alsa @ 0x5643b3293160] ALSA buffer xrun.
        Last message repeated 1 times
    frame=   33 fps= 22 q=-0.0 size=N/A time=00:00:01.32 bitrate=N/A dup=1 drop=0 sp[alsa @ 0x5643b3293160] ALSA buffer xrun.
        Last message repeated 1 times
    frame=   46 fps= 23 q=-0.0 size=N/A time=00:00:01.84 bitrate=N/A dup=1 drop=0 spframe=   58 fps= 23 q=-0.0 size=N/A time=00:00:02.32 bitrate=N/A dup=1 drop=0 spframe=   71 fps= 24 q=-0.0 size=N/A time=00:00:02.84 bitrate=N/A dup=1 drop=0 spframe=   83 fps= 24 q=-0.0 size=N/A time=00:00:03.32 bitrate=N/A dup=1 drop=0 sp[alsa @ 0x5643b3293160] ALSA buffer xrun.
    frame=   96 fps= 24 q=-0.0 size=N/A time=00:00:03.84 bitrate=N/A dup=1 drop=0 sp[alsa @ 0x5643b3293160] ALSA buffer xrun.
    

    The sound is then absolutely unsynchronized ...

    Does anyone have any advice and can help me?

  • FFmpeg doesn't use GPU

    28 janvier 2019, par hamidi

    I got the latest version binaries of ffmpeg from here. When I examine CPU and GPU usages when I play a video by its ffplay, I see that GPU is used during play. Not much using of CPU also indicates it. But when I get the latest version sources from the original site, I can't use GPU. To clarify, I include a player test program I wrote until now. When I uncomment the line which includes avcodec_find_decoder_by_name("h264_cuvid"), I get error -1. The error happens in avcodec_open2 with the description of Operation not permitted.

    CString format(const char *fmt, ...) 
    { 
        va_list ap;
        va_start(ap, fmt);
        char buf[512];
        vsnprintf(buf, sizeof(buf), fmt, ap);
        va_end(ap);
        return buf;
    }
    
    int CplayerDlg::play()
    {
        FILE *fp = fopen("video_files/1010.brf", "rb");
        if (!fp)
        {
            AfxMessageBox("can't open video file");
            return -1;
        }
        RecordFrame frame;
        RecordHeader hdr;
        fread(&frame, sizeof(frame), 1, fp);
        if (frame.frameType != FRAME_TYPE_HEADER)
        {
            AfxMessageBox("record file doesn't begin with header");
            return -1;
        }
        fread(&hdr, sizeof(hdr), 1, fp);
        GetDlgItem(IDC_DIM)->SetWindowText(format("%dx%d", hdr.width, hdr.height));
        GetDlgItem(IDC_CODEC_ID)->SetWindowText(format("%d", hdr.codecId));
        GetDlgItem(IDC_PIXEL_FORMAT)->SetWindowText(format("%d", hdr.pixelFormat));
        GetDlgItem(IDC_TIMEBASE)->SetWindowText(format("%d/%d", hdr.timebaseNum, hdr.timebaseDen));
        AVCodec *pCodec;
    
    #if 0
    #define CHECK(decoder)\
        pCodec = avcodec_find_decoder_by_name(#decoder);\
        AfxMessageBox(pCodec ? #decoder " found" : "can't find " #decoder);
    
        CHECK(h264_cuvid);
    
    #undef CHECK
    #endif
    
        pCodec = avcodec_find_decoder(AV_CODEC_ID_H264);
        //pCodec = avcodec_find_decoder_by_name("h264_cuvid");
        if (!pCodec)
        {
            AfxMessageBox("can't find h264 decoder");
            return -1;
        }
    
        AVCodecContext *pCodecContext = avcodec_alloc_context3(pCodec);
        if (!pCodecContext)
        {
            AfxMessageBox("can't allocate codec context");
            return -1;
        }
    
    #if 0
    // enumerating available codecs
        //av_register_all();
        avcodec_register_all();
    
        AVCodec *current_codec = av_codec_next(NULL);
        while (current_codec != NULL)
        {
            TRACE("%s\n", current_codec->name);
            current_codec = av_codec_next(current_codec);
        }
    #endif
    
        int err = avcodec_open2(pCodecContext, pCodec, NULL);
        if (err != 0)
        {
            char buf[AV_ERROR_MAX_STRING_SIZE];
            av_make_error_string(buf, AV_ERROR_MAX_STRING_SIZE, err);
            char buf2[AV_ERROR_MAX_STRING_SIZE];
            sprintf(buf2, "%d (%x): %s\n", err, err, buf);
            AfxMessageBox(buf2);
            return -1;
        }
        AfxMessageBox("operation completed successfully");
        fclose(fp);
        return 0;
    }