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• ffmpeg conversion for apple tv

  24 janvier 2013, par Sam

  I have one particular movie file that is giving me grief while I am trying to convert my movie library to be able to be viewed on my Apple TV. I have been using iFlicks to convert all of my files and have only had one issue. The original of this particular movie file plays fine but after it has been converted the video freezes after a few minutes but the audio keeps playing. I tried using ffmpeg to convert the file but now the file is very choppy. The first time it is played the video will be choppy, the next time the audio will be choppy... but it plays fine in VLC for some reason. So I was thinking that maybe I have chosen the wrong codecs to suit Quicktime/Apple TV. Below is the command I used for ffmpeg. Have I chosen the right codecs and actually written the command correctly ? (I haven't really used ffmpeg before...)

  ffmpeg -i input.avi -vcodec libx264 -acodec libfaac output.m4v

• Convert .m4a to PCM using libavcodec

  17 décembre 2013, par gmcc051

  I'm trying to convert a .m4a file to raw PCM file so that I can play it back in Audacity.

  According to the AVCodecContext it is a 44100 Hz track using the sample format AV_SAMPLE_FMT_FLTP which, to my understanding, when decodeded using avcodec_decode_audio4, I should get two arrays of floating point values (one for each channel).

  I'm unsure of the significance of the AVCodecContext's bits_per_coded_sample = 16

  Unfortunately Audacity plays the result back as if I have the original track is mixed in with some white noise.

  Here is some sample code of what I've been done. Note that I've also added a case for a track that uses signed 16bit non-interleaved data (sample_format = AC_SAMPLE_FMT_S16P), which Audacity plays back fine.

  int AudioDecoder::decode(std::string path)
  
  {
  
    const char* input_filename=path.c_str();
  
  
  
    av_register_all();
  
  
  
    AVFormatContext* container=avformat_alloc_context();
  
    if(avformat_open_input(&container,input_filename,NULL,NULL)<0){
  
      printf("Could not open file");
  
    }
  
  
  
    if(avformat_find_stream_info(container, NULL)<0){
  
        printf("Could not find file info");
  
    }
  
    av_dump_format(container,0,input_filename,false);
  
  
  
    int stream_id=-1;
  
    int i;
  
    for(i=0;inb_streams;i++){
  
      if(container->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO){
  
          stream_id=i;
  
          break;
  
      }
  
    }
  
    if(stream_id==-1){
  
      printf("Could not find Audio Stream");
  
    }
  
  
  
    AVDictionary *metadata=container->metadata;
  
    AVCodecContext *ctx=container->streams[stream_id]->codec;
  
    AVCodec *codec=avcodec_find_decoder(ctx->codec_id);
  
  
  
    if(codec==NULL){
  
      printf("cannot find codec!");
  
    }
  
  
  
    if(avcodec_open2(ctx,codec,NULL)<0){
  
       printf("Codec cannot be found");
  
    }
  
  
  
    AVSampleFormat sfmt = ctx->sample_fmt;
  
  
  
    AVPacket packet;
  
    av_init_packet(&packet);
  
    AVFrame *frame = avcodec_alloc_frame();
  
  
  
    int buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE+ FF_INPUT_BUFFER_PADDING_SIZE;;
  
    uint8_t buffer[buffer_size];
  
    packet.data=buffer;
  
    packet.size =buffer_size;
  
  
  
    FILE *outfile = fopen("test.raw", "wb");
  
  
  
    int len;
  
    int frameFinished=0;
  
  
  
    while(av_read_frame(container,&packet) >= 0)
  
    {
  
        if(packet.stream_index==stream_id)
  
        {
  
          //printf("Audio Frame read \n");
  
          int len=avcodec_decode_audio4(ctx, frame, &frameFinished, &packet);
  
  
  
          if(frameFinished)
  
          {       
  
            if (sfmt==AV_SAMPLE_FMT_S16P)
  
            { // Audacity: 16bit PCM little endian stereo
  
              int16_t* ptr_l = (int16_t*)frame->extended_data[0];
  
              int16_t* ptr_r = (int16_t*)frame->extended_data[1];
  
              for (int i=0; inb_samples; i++)
  
              {
  
                fwrite(ptr_l++, sizeof(int16_t), 1, outfile);
  
                fwrite(ptr_r++, sizeof(int16_t), 1, outfile);
  
              }
  
            }
  
            else if (sfmt==AV_SAMPLE_FMT_FLTP)
  
            { //Audacity: big endian 32bit stereo start offset 7 (but has noise)
  
              float* ptr_l = (float*)frame->extended_data[0];
  
              float* ptr_r = (float*)frame->extended_data[1];
  
              for (int i=0; inb_samples; i++)
  
              {
  
                  fwrite(ptr_l++, sizeof(float), 1, outfile);
  
                  fwrite(ptr_r++, sizeof(float), 1, outfile);
  
               }
  
             }            
  
          }
  
      }
  
  }
  
  fclose(outfile);
  
  av_close_input_file(container);
  
  return 0;   

  }

  I'm hoping I've just done a naive conversion (most/less significant bit issues), but at present I've been unable to figure it out. Note that Audacity can only import RAW float data if its 32bit or 64 bit float (big or little endian).

  Thanks for any insight.

• How to add more codec support in FFmpeg library used with Android ?

  2 janvier 2013, par AB1209

  I want to use ffmpeg library in my Android app to play .ts video file format.I am trying to run sample app given at GitHub Appunite.

  So far I have managed to compile ffmpeg library & build .so files.I am working on Ubuntu 10.10 & using NDK r8.

  Sample app works fine & plays file formats like .mp4 & .3gp but it doesn't play .FLV & .ts file (which is my requirement).

  Whenever I try to play .ts or .FLV files it says "Could not open stream"

  In Logcat it shows

  E/player.c(28005): player_set_data_source Could not open video file: file:///mnt/sdcard/football.ts (-1094995529: Invalid data found when processing input)

  I am trying this for weeks.So far I have not got any way. Please help

  Thanks