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• How can I limit the CPU usage when encoding h264 with libavcodec ?

  17 septembre 2021, par oarfish

  I encode raw images to h264 video and set up my encoder parameters like this before calling avocdec_open2() :

  


  void set_codec_params(AVFormatContext *&fctx, AVCodecContext *&codec_ctx,
                      double width, double height, int fps) {
  const AVRational dst_fps = {fps, 1};

  codec_ctx->codec_tag = 0;
  codec_ctx->bit_rate = target_bitrate;
  codec_ctx->thread_count = 1; // <----- does nothing
  codec_ctx->codec_id = AV_CODEC_ID_H264;
  codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;
  codec_ctx->width = width;
  codec_ctx->height = height;
  codec_ctx->gop_size = 12;
  codec_ctx->pix_fmt = AV_PIX_FMT_YUV420P;
  codec_ctx->framerate = dst_fps;
  codec_ctx->time_base = av_inv_q(dst_fps);
  if (fctx->oformat->flags & AVFMT_GLOBALHEADER) {
    codec_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
  }
}


  


  I set profile parameters like this

  


    AVDictionary *codec_options = nullptr;
  av_dict_set(&codec_options, "profile", "high", 0);
  av_dict_set(&codec_options, "preset", "ultrafast", 0);
  av_dict_set(&codec_options, "tune", "zerolatency", 0);


  


  No matter what I do, all cores are maxed out during encoding, but I would like to limit this to some number of threads. The thread_count structure member seems to be ignored.

  


  Generally, what steps can be taken to limit the number of threads used for encoding ? Do some settings conflict with a user-defined thread count ?

  


• ffmpeg : How to limit/decrease the size of PCM files ?

  7 août 2021, par a_hayler

  I am playing around with the Shazam API and trying to let it identify a song from a chosen sound-bite. For this I need as the body of the request the following :

  


  "Encoded base64 string of byte[] that generated from raw data less than 500KB (3-5 seconds sample are good enough for detection). The raw sound data must be 44100Hz, 1 channel (Mono), signed 16 bit PCM little endian."

  


  I extracted a ten-second interval in the file slice.mp3 and (hopefully) converted it to the right format by using :

  


  ffmpeg -i song_mono.mp3 -f s16le -ac 1 -ar 44100 -b:a 128k result.raw

  


  The problem now is that the resulting file is about 21MB, just 20.5MB over the API's limit. I am sure that there has to be a way to decrease the size of the audio file to under 500KB. The first thing that I have noticed is that the bitrate of the output file is again at 700+ kb/s even though I changed it in the slicing process to 128kb/s. Additionally adding -b:a 128k doesn't seem to do anything.

  


  Thus I am asking myself (and you now) : How do I bring the size of the file under control (in this case 500KB) whilst still maintaining the specified requirements.

  


  Any help is greatly appreciated !

  


  Here is the output of the following commands :

  


  ffmpeg -i slice.mp3
ffmpeg -i song_mono.mp3 -f s16le -ac 1 -ar 44100 -b:a 128k result.raw


  


  ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
  configuration: --prefix=/usr --extra-version=0ubuntu0.2 --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-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --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-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Input #0, mp3, from 'slice.mp3':
  Metadata:
    encoder         : Lavf57.83.100
  Duration: 00:00:10.06, start: 0.023021, bitrate: 128 kb/s
    Stream #0:0: Audio: mp3, 48000 Hz, mono, s16p, 128 kb/s
At least one output file must be specified
ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
  configuration: --prefix=/usr --extra-version=0ubuntu0.2 --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-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --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-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Input #0, mp3, from 'song_mono.mp3':
  Metadata:
    encoder         : Lavf57.83.100
  Duration: 00:04:04.87, start: 0.023021, bitrate: 128 kb/s
    Stream #0:0: Audio: mp3, 48000 Hz, mono, s16p, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, s16le, to 'result.raw':
  Metadata:
    encoder         : Lavf57.83.100
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
    Metadata:
      encoder         : Lavc57.107.100 pcm_s16le
size=   21089kB time=00:04:04.83 bitrate= 705.6kbits/s speed= 460x    
video:0kB audio:21089kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%


  


• avformat/mov : Limit nb_chapter_tracks to input size

  19 avril 2021, par Michael Niedermayer

  avformat/mov : Limit nb_chapter_tracks to input size
  Fixes : Timeout (15k loop iterations instead of 400m)
  
  Fixes : 31368/clusterfuzz-testcase-minimized-ffmpeg_dem_MOV_fuzzer-6601583174483968
  Found-by : continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/mov.c