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• ffmpeg transcode

  12 novembre 2015, par user2004388

  I want to do a audio trancode using ffmpeg library. Now i have out file but I can listen only noise .
  The steps of my program are :
  1) Open input file and decode in raw format using avcodec_decode_audio4
  2) encode and save the raw format .
  I don’t Know where I wrong. This is my code.

  /* 
  
   * File:   newmain.c
  
   * Author: antonello
  
   *
  
   * Created on 23 gennaio 2013, 11.24
  
   */
  
  
  
  #include 
  
  #include 
  
  
  
  
  
  #include <libavutil></libavutil>samplefmt.h>
  
  #include <libavutil></libavutil>timestamp.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libavcodec></libavcodec>old_codec_ids.h>
  
  
  
  static AVCodecContext *get_encoder(int sampleRate, int channels, int audioBitrate)
  
  { 
  
      AVCodecContext  *audioCodec;
  
      AVCodec *codec;
  
  
  
  
  
  
  
      //Set up audio encoder
  
      codec = avcodec_find_encoder(CODEC_ID_AAC);
  
      if (codec == NULL) 
  
      {
  
          printf("avcodec_find_encoder: ERROR\n");
  
          return NULL;
  
      }
  
      audioCodec = avcodec_alloc_context();
  
      audioCodec->bit_rate = audioBitrate;
  
      audioCodec->sample_fmt = AV_SAMPLE_FMT_S16P;
  
      audioCodec->sample_rate = sampleRate;
  
      audioCodec->channels = channels;
  
      audioCodec->profile = FF_PROFILE_AAC_MAIN;
  
      audioCodec->channel_layout=AV_CH_LAYOUT_MONO;
  
      //audioCodec->time_base = (AVRational){1, sampleRate};
  
      audioCodec->time_base.num  = 1;
  
      audioCodec->time_base.den  = sampleRate;
  
  
  
      audioCodec->codec_type = AVMEDIA_TYPE_AUDIO;
  
      if (avcodec_open(audioCodec, codec) &lt; 0) 
  
      {
  
          printf("avcodec_open: ERROR\n");
  
          return NULL;
  
      }
  
  
  
      return audioCodec;
  
  }
  
  
  
  
  
  int main(int argc, char** argv) {
  
    AVFormatContext *aFormatCtx_decoder = NULL;
  
    AVFormatContext *aFormatCtx_encoder = NULL;
  
    int             i, audioStream;
  
    AVPacket        packet_decoder;
  
    AVPacket        packet_encoder;
  
    int             got_frame=0;
  
    int             complete_decode=0;
  
    int             len;
  
    AVFrame         *decoded_frame = NULL;
  
    AVCodecContext  *aCodec_decoderCtx = NULL;
  
    AVCodec         *aCodec_decoder = NULL;
  
    FILE            *outfile;
  
    //reding input file
  
    avcodec_register_all();
  
  
  
     //register all codecs
  
      av_register_all();
  
  
  
   //open file
  
      if(avformat_open_input(&amp;aFormatCtx_decoder, "sample.aac", NULL, NULL)!=0){
  
          fprintf(stderr, "Could not open source file \n");
  
          return -1; // Couldn't open file
  
      }
  
  
  
    // Retrieve stream information
  
    if(avformat_find_stream_info(aFormatCtx_decoder, NULL)&lt;0){
  
        fprintf(stderr, "Couldn't find stream information \n");
  
        return -1; // Couldn't find stream information
  
    }
  
  
  
    // Dump information about file onto standard error
  
    //av_dump_format(aFormatCtx_decode, 0, argv[1], 0);
  
  
  
    // Find the first audio stream
  
    audioStream=-1;
  
  
  
    for(i=0; inb_streams; i++) {
  
      if(aFormatCtx_decoder->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO &amp;&amp;
  
         audioStream &lt; 0) {
  
        audioStream=i;
  
      }
  
    }
  
    if(audioStream==-1){
  
        fprintf(stderr, "File haven't sudio stream \n");
  
        return -1;
  
    }
  
  
  
    //get audio codec contex
  
    aCodec_decoderCtx=aFormatCtx_decoder->streams[audioStream]->codec;
  
    //get audio codec
  
    aCodec_decoder = avcodec_find_decoder(aCodec_decoderCtx->codec_id);
  
    aCodec_decoder->sample_fmts=AV_SAMPLE_FMT_S16P;
  
    if(!aCodec_decoder) {
  
      fprintf(stderr, "Unsupported codec!\n");
  
      return -1;//Unsupported codec!
  
    }
  
    //open codec
  
    // Open codec
  
    if(avcodec_open2(aCodec_decoderCtx, aCodec_decoder, NULL)&lt;0)
  
      return -1; // Could not open codec
  
    // allocate audio frame
  
    decoded_frame = avcodec_alloc_frame();
  
    if (!decoded_frame) {
  
      fprintf(stderr, "Could not allocate audio frame\n");
  
      return -1;//Could not allocate audio frame
  
      }
  
    aCodec_decoderCtx->bit_rate=12000;
  
    aFormatCtx_encoder=get_encoder(8000,1,12000);
  
    av_init_packet(&amp;packet_encoder);
  
  
  
    printf("param %d %d %d",aCodec_decoderCtx->sample_fmt,aCodec_decoderCtx->channels,aCodec_decoderCtx->bit_rate);
  
  
  
    outfile = fopen("out.aac", "wb");
  
      if (!outfile) {
  
          printf(stderr, "Could not open outfile \n");
  
          return -1;//Could not open outfile
  
      }
  
    while(av_read_frame(aFormatCtx_decoder, &amp;packet_decoder)>=0) {
  
       // decode frame
  
       len = avcodec_decode_audio4(aCodec_decoderCtx, decoded_frame, &amp;got_frame, &amp;packet_decoder);
  
          if (len &lt; 0) {
  
              fprintf(stderr, "Error while decoding\n");
  
              return -1;
  
              }
  
  
  
          if (got_frame){
  
            avcodec_encode_audio2(aFormatCtx_encoder,&amp;packet_encoder,decoded_frame,&amp;complete_decode);
  
            if(complete_decode){
  
            //    printf("complete decode frame");
  
                fwrite(packet_encoder.data, 1, packet_encoder.size, outfile);
  
                av_free_packet(&amp;packet_encoder);
  
            }
  
          }
  
  
  
  
  
  
  
      }
  
    fclose(outfile);
  
      return (EXIT_SUCCESS);
  
  }

• Trouble writing mp3 into flv container

  8 février 2013, par Sriram

  I am encoding a raw pcm stream with LAME and writing the mp3 encoded samples to an flv container using a c program. For debugging purposes, I am also writing the mp3 encoded samples to a file separately. The following is observed :

  1. The mp3 written to another file is correct. There are no clicks or any other artefacts observed.

  2. The flv file does not play anything. Examining with ffmpeg like so :

     $ ./ffmpeg.exe -i temp_local_flv.flv

  The above command gives the following :

  ffmpeg version N-49352-gc46943e Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Jan 26 2013 12:12:14 with gcc 4.7.2 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  
    libavutil      52. 17.100 / 52. 17.100
  
    libavcodec     54. 91.100 / 54. 91.100
  
    libavformat    54. 61.104 / 54. 61.104
  
    libavdevice    54.  3.102 / 54.  3.102
  
    libavfilter     3. 34.101 /  3. 34.101
  
    libswscale      2.  2.100 /  2.  2.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  2.100 / 52.  2.100
  
  [flv @ 0000000000307be0] Stream discovered after head already parsed
  
  [mp3 @ 000000000237d9c0] Header missing
  
  Input #0, flv, from 'temp_local_flv.flv':
  
    Duration: 00:00:00.07, start: 0.002000, bitrate: 643 kb/s
  
      Stream #0:0: Audio: mp3, 22050 Hz, mono, s16p, 32 kb/s
  
      Stream #0:1: Data: none
  
  At least one output file must be specified  

  My questions :
  1. What is this "Header missing" ? Is there a "special" mp3 header that needs to be written when writing encoded samples to an flv container ? If so, what does the header contain ? And given that the mp3 samples written to the file are decoded correctly by an audio player, what am I missing ?

• My FFMPEG converter is not allowing mp4 to webm conversions.

  2 novembre 2016, par Fred Garbutt

  heres the command :

  ffmpeg -i videos/test.mp4 -c:v libvpx -level 216 -profile 0 -qmax 42 -qmin 10 -c:a libvorbis -f webm out.webm

  i keep getting this error :

  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  here is the full response

  # ffmpeg -i videos/test.mp4 -c:v libvpx -level 216 -profile 0 -qmax 42 -qmin 10 -c:a 
  
  
  
  libvorbis -f webm out.webm
  
  ffmpeg version N-49225-g7e059c9 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Jan 24 2013 05:14:06 with gcc 4.1.2 (GCC) 20080704 (Red Hat 4.1.2-54)
  
    configuration: --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-gpl --enable-postproc --enable-nonfree
  
    libavutil      52. 15.101 / 52. 15.101
  
    libavcodec     54. 90.100 / 54. 90.100
  
    libavformat    54. 61.104 / 54. 61.104
  
    libavdevice    54.  3.102 / 54.  3.102
  
    libavfilter     3. 33.100 /  3. 33.100
  
    libswscale      2.  2.100 /  2.  2.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  2.100 / 52.  2.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'videos/test.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 1
  
      compatible_brands: mp42avc1
  
      creation_time   : 2010-08-12 15:42:21
  
    Duration: 00:00:34.20, start: 0.000000, bitrate: 358 kb/s
  
      Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 100 kb/s
  
      Metadata:
  
        creation_time   : 2010-08-12 15:42:21
  
        handler_name    : Apple Sound Media Handler
  
      Stream #0:1(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 640x320, 251 kb/s, 29.97 fps, 29.97 tbr, 2997 tbn, 5994 tbc
  
      Metadata:
  
        creation_time   : 2010-08-12 15:42:21
  
        handler_name    : Apple Video Media Handler
  
  Please use -profile:a or -profile:v, -profile is ambiguous
  
  File 'out.webm' already exists. Overwrite ? [y/N] y
  
  v0.9.6
  
  [libvpx @ 0x7e38d20] Failed to initialize encoder: ABI version mismatch
  
  Output #0, webm, to 'out.webm':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 1
  
      compatible_brands: mp42avc1
  
      Stream #0:0(eng): Video: vp8, yuv420p, 640x320, q=10-42, 200 kb/s, 90k tbn, 29.97 tbc
  
      Metadata:
  
        creation_time   : 2010-08-12 15:42:21
  
        handler_name    : Apple Video Media Handler
  
      Stream #0:1(eng): Audio: none, 44100 Hz, stereo, fltp
  
      Metadata:
  
        creation_time   : 2010-08-12 15:42:21
  
        handler_name    : Apple Sound Media Handler
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (h264 -> libvpx)
  
    Stream #0:0 -> #0:1 (aac -> libvorbis)
  
  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  does anybody have any idea whats going on ? I am completely lost. I have tried so many different commands. converting from mp4 to ogg works fine, and i have had mp4 to webm working before, but i reinstalled ffmpeg to get the mp4 to ogg working and now ive lost my mp4 to webm conversions. >.< this whole installation process is such a damn nightmare. does anyone havea good tutorial for centos installation of ffmpeg that will work for mp4 ogg and webm ? I have used the centos tut that came off of the ffmpeg site, but it failed me. If anyone has any ideas that could point me in the right direction i would much appreciate it !