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• Using ffmpeg to convert sound files for use in an android app

  10 janvier 2012, par stefs

  short : i'm trying to simply play a sound file converted with ffmpeg in my android app, but happen to have problems getting it to work.

  long : we have an iphone app and an android app doing the same thing, and i have to port the feature playing a sound on an user interaction. i have the source file in the aiff format, and tried to convert it to mp3 for android. but the app keeps crashing when it tries to load the file

  AssetFileDescriptor fileDescriptor = context.getResources().openRawResourceFd(resid);
  
  final MediaPlayer mp = new MediaPlayer();
  
  mp.setDataSource(fileDescriptor.getFileDescriptor(), fileDescriptor.getStartOffset(), fileDescriptor.getLength());
  
  fileDescriptor.close();
  
  mp.prepare();

  more specifically, mp.setDataSource crashes. some digging around led me to believe that something's wrong with the encoding. the sound file itself resides in res/raw.

  11-29 17:11:48.012: ERROR/SoundManager(15580): java.io.IOException: setDataSourceFD failed.: status=0x80000000
  
  11-29 17:11:48.012: ERROR/SoundManager(15580):     at android.media.MediaPlayer.setDataSource(Native Method)
  
  ...

  what i tried :

  • using a different mp3 that's already used with the same code in a different place. this works.
  • converted it to wav file. this didn't cause the app to crash, but it neither played a sound. that might be a different problem.
  • converted it to ogg ; crashed

  so, the the ffmpeg conversion parameters are as follows :

  $ ffmpeg -i click_24db.aif -f mp3 ~/foobar/wheel_click.mp3
  
  ffmpeg version 0.7.8, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Nov 24 2011 14:31:00 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3)
  
    configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libopenjpeg --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm
  
    libavutil    50. 43. 0 / 50. 43. 0
  
    libavcodec   52.123. 0 / 52.123. 0
  
    libavformat  52.111. 0 / 52.111. 0
  
    libavdevice  52.  5. 0 / 52.  5. 0
  
    libavfilter   1. 80. 0 /  1. 80. 0
  
    libswscale    0. 14. 1 /  0. 14. 1
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  Input #0, aiff, from 'click_24db.aif':
  
    Duration: 00:00:00.01, start: 0.000000, bitrate: 1570 kb/s
  
      Stream #0.0: Audio: pcm_s16be, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Output #0, mp3, to '/Users/xyz/foobar/wheel_click.mp3':
  
    Metadata:
  
      TSSE            : Lavf52.111.0
  
      Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Press [q] to stop, [?] for help
  
  size=       1kB time=00:00:00.05 bitrate=  92.9kbits/s    
  
  video:0kB audio:0kB global headers:0kB muxing overhead 45.563549%

  the resulting file plays nice in itunes, does not play in vlc and crashes when loaded with the android.media.MediaPlayer (note : i first tried it with the SoundPool lib, with both mp3 and ogg, but that didn't work either).

  i also tried the following paramters, which didn't work :

  ffmpeg -i inputfile.aif -f mp3 -acodec libmp3lame -ab 192000 -ar 44100 outputfile.mp3

  i'm working on osx, built ffmpeg with macports today, android api level is 7 (google api, 2.1-update1). looking at the "supported formats" table on dev.android didn't indicate my file to be out of the spec, but i may be mistaken in that.

  i don't have the slightest clue regarding bitrates and so on, so could anybody please point me to the right combination of ffmpeg parameters to get a working mp3 for android ? i don't care if the resulting file would be mp3, ogg or 3gp or whatever.

• Invalid data when creating mkv container with h264 stream because extradata is null

  18 février 2020, par pixelou

  I need to encode raw frames into an h264 stream inside an mkv container using ffmpeg’s c api.
  All the examples I could find copy existing codec parameters from a decoder so I had to modify it a bit.

  #include <cstdio>
  
  #include <cstring>
  
  #include <iostream>
  
  
  
  extern "C" {
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libavutil></libavutil>avutil.h>
  
  #include <libavutil></libavutil>imgutils.h>
  
  #include <libavutil></libavutil>opt.h>
  
  #include <libswscale></libswscale>swscale.h>
  
  }
  
  
  
  static char error_msg[4096] = "";
  
  
  
  
  
  int main(int argc, char **argv) {
  
    int err;
  
    char *filename = (char *)av_malloc(4096);
  
    AVCodec *codec = nullptr;
  
    SwsContext *color_converter = nullptr;
  
    AVCodecContext *encoder = nullptr;
  
    AVFormatContext *container = nullptr;
  
    AVStream *stream = nullptr;
  
  
  
    std::snprintf(filename, sizeof(filename), "%s", "out.mkv");
  
  
  
    // Create encoder
  
    codec = avcodec_find_encoder_by_name("libx264");
  
    if (codec == NULL) {
  
      std::cerr &lt;&lt; "codec not found" &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
    encoder = avcodec_alloc_context3(codec);
  
    if (encoder == nullptr) {
  
      std::cerr &lt;&lt; "failed to allocate encoder" &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
    // Configure encoder
  
    encoder->color_range = AVCOL_RANGE_JPEG;
  
    encoder->height = 1024;
  
    encoder->width = 1280;
  
    encoder->sample_aspect_ratio = {1, 1};
  
    encoder->pix_fmt = AV_PIX_FMT_YUVJ420P;
  
    encoder->time_base = {1, 10};
  
    encoder->framerate = {10, 1};
  
  
  
    err = av_opt_set_double(encoder->priv_data, "crf", 0.0, 0);
  
    if (err &lt; 0) {
  
      av_make_error_string(error_msg, sizeof(error_msg), err);
  
      std::cerr &lt;&lt; "failed to initialize encoder: " &lt;&lt; error_msg &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
    err = avcodec_open2(encoder, codec, nullptr);
  
    if (err &lt; 0) {
  
      av_make_error_string(error_msg, sizeof(error_msg), err);
  
      std::cerr &lt;&lt; "failed to initialize encoder: " &lt;&lt; error_msg &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
    // Create container
  
    avformat_alloc_output_context2(&amp;container, NULL, NULL, filename);
  
    if (container == NULL) {
  
      std::cerr &lt;&lt; "failed to allocate container" &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
    // Add stream
  
    stream = avformat_new_stream(container, nullptr);
  
    if (stream == NULL) {
  
      std::cerr &lt;&lt; "failed to create new stream" &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
    stream->time_base = encoder->time_base;
  
    err = avcodec_parameters_from_context(stream->codecpar, encoder);
  
    if (err &lt; 0) {
  
      av_make_error_string(error_msg, sizeof(error_msg), err);
  
      std::cerr &lt;&lt; "failed to configure stream: " &lt;&lt; error_msg &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
    // encoder->extradata_size == 0 at this point!
  
  
  
    // Open file and write file header
  
    err = avio_open(&amp;container->pb, filename, AVIO_FLAG_WRITE);
  
    if (err &lt; 0) {
  
      av_make_error_string(error_msg, sizeof(error_msg), err);
  
      std::cerr &lt;&lt; "failed to open output file: " &lt;&lt; error_msg &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
    err = avformat_write_header(container, nullptr);
  
    // fails in ff_isom_write_avcc because stream->codecpar->extradata is null.
  
    if (err &lt; 0) {
  
      av_make_error_string(error_msg, sizeof(error_msg), err);
  
      std::cerr &lt;&lt; "failed to set-up container: " &lt;&lt; error_msg &lt;&lt; std::endl;
  
      goto cleanup;
  
    }
  
  
  
  cleanup:
  
    avcodec_close(encoder);
  
    if (container != nullptr) {
  
      avio_closep(&amp;container->pb);
  
    }
  
    avcodec_free_context(&amp;encoder);
  
    sws_freeContext(color_converter);
  
    avformat_free_context(container);
  
    av_free(filename);
  
  
  
    return 0;
  
  }
  
  </iostream></cstring></cstdio>

  This fails with EINVAL (Invalid data found when processing input) at the avformat_write_header line (more precisely in ff_isom_write_avcc inside the ffmpeg library) because stream->codecpar->extradata is null. The same code works for an mp4 container and I have no clue how to initialize stream->codecpar->extradata (or encoder->extradata) manually.

  I have checked similar questions about that problem but could not find a conclusive answer :

  • https://stackoverflow.com/a/22231863/5786475 says extradata can be null when h264 is in annex b format
  • https://stackoverflow.com/a/27471912/5786475 says it shouldn’t be null.

• FFMpeg returns error : Unrecognized option 'master_pl_name'

  14 février 2021, par jonatasos

  I used this command (described at the processing message bellow) on my application several times before and always worked well. Now returns that error.

  


  '/usr/bin/ffmpeg' '-y' '-threads' '12' '-i' 
'/home/vagrant/code/myapp/storage/videos/videos_disk/a3.mp4' '-map' '0' '-vcodec' 'libx264' 
'-b:v' '500k' '-g' '48' '-hls_playlist_type' 'vod' '-hls_time' '10' '-hls_segment_filename' 
'/home/vagrant/code/myapp/storage/app/public/test_0_500_%05d.ts' '-master_pl_name' 
'master_playlist_guide_0.m3u8' '-acodec' 'aac' '-b:a' '128k' 
'/home/vagrant/code/myapp/storage/app/public/test_0_500.m3u8' '-map' '0' '-vcodec' 
'libx264' '-b:v' '1500k' '-g' '48' '-hls_playlist_type' 'vod' '-hls_time' '10' '-hls_segment_filename' 
'/home/vagrant/code/myapp/storage/app/public/test_1_1500_%05d.ts' '-master_pl_name' 
'master_playlist_guide_1.m3u8' '-acodec' 'aac' '-b:a' '128k' 
'/home/vagrant/code/myapp/storage/app/public/test_1_1500.m3u8' '-map' '0' '-vcodec' 
'libx264' '-b:v' '3000k' '-g' '48' '-hls_playlist_type' 'vod' '-hls_time' '10' '-hls_segment_filename' 
'/home/vagrant/code/myapp/storage/app/public/test_2_3000_%05d.ts' '-master_pl_name' 
'master_playlist_guide_2.m3u8' '-acodec' 'aac' '-b:a' '128k' 
'/home/vagrant/code/myapp/storage/app/public/test_2_3000.m3u8'


  


  (single line:) '/usr/bin/ffmpeg' '-y' '-threads' '12' '-i' '/home/vagrant/code/myapp/storage/videos/videos_disk/a3.mp4' '-map' '0' '-vcodec' 'libx264' '-b:v' '500k' '-g' '48' '-hls_playlist_type' 'vod' '-hls_time' '10' '-hls_segment_filename' '/home/vagrant/code/myapp/storage/app/public/test_0_500_%05d.ts' '-master_pl_name' 'master_playlist_guide_0.m3u8' '-acodec' 'aac' '-b:a' '128k' '/home/vagrant/code/myapp/storage/app/public/test_0_500.m3u8' '-map' '0' '-vcodec' 'libx264' '-b:v' '1500k' '-g' '48' '-hls_playlist_type' 'vod' '-hls_time' '10' '-hls_segment_filename' '/home/vagrant/code/myapp/storage/app/public/test_1_1500_%05d.ts' '-master_pl_name' 'master_playlist_guide_1.m3u8' '-acodec' 'aac' '-b:a' '128k' '/home/vagrant/code/myapp/storage/app/public/test_1_1500.m3u8' '-map' '0' '-vcodec' 'libx264' '-b:v' '3000k' '-g' '48' '-hls_playlist_type' 'vod' '-hls_time' '10' '-hls_segment_filename' '/home/vagrant/code/myapp/storage/app/public/test_2_3000_%05d.ts' '-master_pl_name' 'master_playlist_guide_2.m3u8' '-acodec' 'aac' '-b:a' '128k' '/home/vagrant/code/myapp/storage/app/public/test_2_3000.m3u8':



  


  This is the error returned :

  


   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
Unrecognized option 'master_pl_name'.
Error splitting the argument list: Option not found


  


  I`ve reinstalled the ffmpeg, also tested in a virtual machine and the same error happens.

  


  Live server : CentOS 7 / Apache2 / PHP-fpm 7.3 ;
Virtual machine : Ubuntu 20.04 / nginx / PHP-fpm 7.4.5

  


  I really have no clue about what to do about it now.