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• aac-eld decode linux

  28 février 2013, par bernard

  I'm was in Vietnam and tried to learn Vietnames.
  The difficult thing is to learn the pronunciation and I made an Android App to record voices. Vietnamese people are very nice and I collected more than 500 sound files.

  Now I try to make a language learning app. To do that I try to reduce noise and normalize volume.

  I have chosen the following settings :

  this.recorder.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4);
  
  this.recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AAC_ELD);
  
  this.recorder.setAudioEncodingBitRate(96000);
  
  this.recorder.setAudioSamplingRate(44100);

  (cordova AudioPlayer.java - default settings are crap)

  Now I have a problem. I'm not able to open the files with audacity or ffmpeg.

  ./ffmpeg -i /tmp/speaker__ualp__x%E1%BA%A5u.mp4 /tmp/uu.wav
  
  ffmpeg version 1.1.2 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Feb 20 2013 10:50:54 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  
    configuration: --enable-libfdk-aac
  
    libavutil      52. 13.100 / 52. 13.100
  
    libavcodec     54. 86.100 / 54. 86.100
  
    libavformat    54. 59.106 / 54. 59.106
  
    libavdevice    54.  3.102 / 54.  3.102
  
    libavfilter     3. 32.100 /  3. 32.100
  
    libswscale      2.  1.103 /  2.  1.103
  
    libswresample   0. 17.102 /  0. 17.102
  
  [aac @ 0xaada980] Audio object type 39 is not supported.
  
      Last message repeated 1 times
  
  Guessed Channel Layout for  Input Stream #0.0 : mono
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/tmp/speaker__ualp__x%E1%BA%A5u.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 0
  
      compatible_brands: isom3gp4
  
      creation_time   : 2013-01-29 15:53:11
  
    Duration: 00:00:01.21, start: 0.000000, bitrate: 116 kb/s
  
      Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 96 kb/s
  
      Metadata:
  
        creation_time   : 2013-01-29 15:53:11
  
        handler_name    : SoundHandle
  
  [aac @ 0xaada980] Audio object type 39 is not supported.
  
  Output #0, wav, to '/tmp/uu.wav':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 0
  
      compatible_brands: isom3gp4
  
      Stream #0:0(eng): Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
  
      Metadata:
  
        creation_time   : 2013-01-29 15:53:11
  
        handler_name    : SoundHandle
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (aac -> pcm_s16le)
  
  Error while opening decoder for input stream #0:0

  What can I do ?

• 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.

• Anomalie #3386 : Spip derrière Varnish : port non-standard dans l’URL ?

  17 février 2015, par jluc -

  quote : "Parfois l’hébergeur te met un varnish en front et ne te permet pas de le configurer. C’est pour ça que je ne suis pas pour le fait d’exiger un réglage spécifique côté Varnish." Tout à fait. C’est le cas notamment de Gandi Simple Hosting pour lequel l’hébergé ne peut aucunement paramétrer varnish.