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  • how to fix errors while converting yuv to mp4 using ffmpeg, with out compression

    10 janvier 2020, par johanna

    I am trying to convert a yuv file to mp4, without further compression, using :
    ffmpeg -f rawvideo -s 768x432 -r 50 -i test.yuv -c:a acc -strict experimental test.mp4 but I am getting the following error :

    ffmpeg version N-94000-g78e1d7f Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-36)
    
  configuration:
    
  libavutil      56. 28.100 / 56. 28.100
    
  libavcodec     58. 52.102 / 58. 52.102
    
  libavformat    58. 27.103 / 58. 27.103
    
  libavdevice    58.  7.100 / 58.  7.100
    
  libavfilter     7. 55.100 /  7. 55.100
    
  libswscale      5.  4.101 /  5.  4.101
    
  libswresample   3.  4.100 /  3.  4.100
    
[rawvideo @ 0x22f0700] Estimating duration from bitrate, this may be inaccurate
    
Input #0, rawvideo, from 'test.yuv':
    
  Duration: 00:00:00.38, start: 0.000000, bitrate: 198476 kb/s
    
    Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 768x432, 199065 kb/s, 50 tbr, 50 tbn, 50 tbc
    
File 'test.mp4' already exists. Overwrite ? [y/N] y
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg4 (native))
    
Press [q] to stop, [?] for help
    
Output #0, mp4, to 'test.mp4':
    
  Metadata:
    
    encoder         : Lavf58.27.103
    
    Stream #0:0: Video: mpeg4 (mp4v / 0x7634706D), yuv420p, 768x432, q=2-31, 200 kb/s, 50 fps, 12800 tbn, 50 tbc
    
    Metadata:
    
      encoder         : Lavc58.52.102 mpeg4
    
    Side data:
    
      cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
    
test.yuv: corrupt input packet in stream 0
    
[rawvideo @ 0x22fa780] Invalid buffer size, packet size 469698 < expected frame_size 497664
    
Error while decoding stream #0:0: Invalid argument
    
frame=   18 fps=0.0 q=24.8 Lsize=    3340kB time=00:00:00.34 bitrate=80445.8kbits/s speed=3.51x
    
video:3339kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.028110%

    I am also trying : ffmpeg -f rawvideo -s 768x432 -r 50 -i test.yuv -c:a acc -strict experimental test.mp4 but I am getting the following error :

    ffmpeg version N-94000-g78e1d7f Copyright (c) 2000-2019 the FFmpeg developers built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-36) configuration: libavutil      56. 28.100 / 56. 28.100 libavcodec     58. 52.102 / 58. 52.102 libavformat    58. 27.103 / 58. 27.103 libavdevice    58.  7.100 / 58.  7.100 libavfilter     7. 55.100 /  7. 55.100 libswscale      5.  4.101 /  5.  4.101 libswresample   3.  4.100 /  3.  4.100 [rawvideo @ 0x22f0700] Estimating duration from bitrate, this may be inaccurate Input #0, rawvideo, from 'test.yuv': Duration: 00:00:00.38, start: 0.000000, bitrate: 198476 kb/s Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 768x432, 199065 kb/s, 50 tbr, 50 tbn, 50 tbc File 'test.mp4' already exists. Overwrite ? [y/N] y Stream mapping: Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg4 (native)) Press [q] to stop, [?] for help Output #0, mp4, to 'test.mp4': Metadata: encoder         : Lavf58.27.103 Stream #0:0: Video: mpeg4 (mp4v / 0x7634706D), yuv420p, 768x432, q=2-31, 200 kb/s, 50 fps, 12800 tbn, 50 tbc Metadata: encoder         : Lavc58.52.102 mpeg4 Side data: cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1 pr_768x432_50_QP22.yuv: corrupt input packet in stream 0 [rawvideo @ 0x22fa780] Invalid buffer size, packet size 469698 < expected frame_size 497664 Error while decoding stream #0:0: Invalid argument frame=   18 fps=0.0 q=24.8 Lsize=    3340kB time=00:00:00.34 bitrate=80445.8kbits/s speed=3.51x video:3339kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.028110%

    how can I solve these errors ?

  • Command line ffmpeg, to insert part of a mp3 audio to another file

    14 janvier 2020, par tgkprog

    Want to copy from 3rd second of b.mp3 till 2:05 seconds of b.mp3 and append to the end of a.mp3 (its 3:23 long).

    Command :

    ffmpeg -i a.mp3 -i b.mp3

    Output :

    ffmpeg version 4.0.4-0ubuntu1 Copyright (c) 2000-2019 the FFmpeg developers   built with gcc 8 (Ubuntu 8.2.0-7ubuntu1)
    
  configuration: --prefix=/usr --extra-version=0ubuntu1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --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-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
    
  libavutil      56. 14.100 / 56. 14.100
    
  libavcodec     58. 18.100 / 58. 18.100
    
  libavformat    58. 12.100 / 58. 12.100
    
  libavdevice    58.  3.100 / 58.  3.100
    
  libavfilter     7. 16.100 /  7. 16.100
    
  libavresample   4.  0.  0 /  4.  0.  0
    
  libswscale      5.  1.100 /  5.  1.100
    
  libswresample   3.  1.100 /  3.  1.100
    
  libpostproc    55.  1.100 / 55.  1.100
    
Input #0, mp3, from 'a.mp3':
    
  Metadata:
    
    encoder         : Lavf58.12.100
    
  Duration: 00:02:48.00, start: 0.011021, bitrate: 144 kb/s
    
    Stream #0:0: Audio: mp3, 48000 Hz, stereo, fltp, 144 kb/s
    
    Metadata:
    
      encoder         : Lavc58.18
    
Input #1, mp3, from 'b.mp3':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2mp41
    
    encoder         : Lavf58.12.100
    
  Duration: 00:01:33.02, start: 0.011995, bitrate: 117 kb/s
    
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 117 kb/s
    
    Metadata:
    
      encoder         : Lavc58.18
    
At least one output file must be specified

  • PyQt-thread. Get dynamically output

    1er janvier 2020, par ZPro

    I use PyQt-thread for parallel conversion of mp3 files to aac via ffmpeg.
    Here is my code :

    class SubprocessThread(QThread):
    
    signal = pyqtSignal('PyQt_PyObject')
    

    
    def __init__(self, command, args):
    
        QThread.__init__(self)
    
        self.command = command
    
        self.args = args
    

    
    def __del__(self):
    
        self.wait()
    

    
    def run(self):
    
        output = subprocess.check_output('{0} {1}'.format(self.command, self.args), shell=True).split()
    
        self.signal.emit(output)

    And here is example of usage :

    threads = []
    

    
for part in parts.keys():
    
    args = "-i \'{0}.mp3\' -c:a aac -b:a {1}k \'{2}.m4a\'".format(
    
        os.path.join(tmp_dir, str(part)),
    
        int(self.bitrate_cbx.currentText()),
    
        os.path.join(tmp_dir, str(part)))
    
    print(args)  # debug
    
    ffmpeg_thread = SubprocessThread('ffmpeg', args)
    
    ffmpeg_thread.signal.connect(self.on_data_ready)
    
    threads.append(ffmpeg_thread)
    
    ffmpeg_thread.start()
    
    self.threads_count += 1

    I want to make progress bar, based on conversion, but ffmpeg always updates last string in his output (when conversion in progress).
    Here is an example of ffmpeg output while files are converting :

    user@host$ ffmpeg -i '/home/user/001.mp3' -c:a aac -b:a 128k -vn '/home/user/test.m4a' 
    
ffmpeg version n4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 9.2.0 (GCC)
    
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
    
  libavutil      56. 31.100 / 56. 31.100
    
  libavcodec     58. 54.100 / 58. 54.100
    
  libavformat    58. 29.100 / 58. 29.100
    
  libavdevice    58.  8.100 / 58.  8.100
    
  libavfilter     7. 57.100 /  7. 57.100
    
  libswscale      5.  5.100 /  5.  5.100
    
  libswresample   3.  5.100 /  3.  5.100
    
  libpostproc    55.  5.100 / 55.  5.100
    
Input #0, mp3, from '/home/user/001.mp3':
    
  Metadata:
    
    encoder         : Lavf57.41.100
    
    title           : test
    
    artist          : test
    
    album_artist    : test
    
    album           : test
    
    composer        : test
    
    genre           : test
    
    date            : 2018
    
  Duration: 00:12:38.02, start: 0.025056, bitrate: 192 kb/s
    
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
    
    Metadata:
    
      encoder         : Lavc57.48
    
    Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, bt470bg/unknown/unknown), 500x500 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
    
    Metadata:
    
      comment         : Cover (front)
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (mp3 (mp3float) -> aac (native))
    
Press [q] to stop, [?] for help
    
Output #0, ipod, to '/home/user/test.m4a':
    
  Metadata:
    
    date            : test
    
    title           : test
    
    artist          : test
    
    album_artist    : test
    
    album           : test
    
    composer        : test
    
    genre           : test
    
    encoder         : Lavf58.29.100
    
    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
    
    Metadata:
    
      encoder         : Lavc58.54.100 aac
    
size=   12107kB time=00:12:38.01 bitrate= 130.8kbits/s speed=79.2x

    How can I receive this data (string, that begins from "size=...") from my parallel QThreads to calculate overall progress ?

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