Recherche avancée

Sur d’autres sites (10382)

• Audioqueue try to read ADPCM sound

  4 septembre 2013, par Kamax

  I try to read a .asf stream with FFMPEG and the audio part is ADPCM IMA WAV codec.

  The sound i hear when i hear something is only noise. I suppose my AudioStreamBasicDescription are wrong. How can i get the correct values ?

  From ffprobe i have this :

  [STREAM]
  
  index=1
  
  codec_name=adpcm_ima_wav
  
  codec_long_name=ADPCM IMA WAV
  
  codec_type=audio
  
  codec_time_base=1/8000
  
  codec_tag_string=[17][0][0][0]
  
  codec_tag=0x0011
  
  sample_rate=8000.000000 
  
  channels=1
  
  bits_per_sample=4
  
  r_frame_rate=0/0
  
  avg_frame_rate=250/41
  
  time_base=1/1000
  
  start_time=0.000000 
  
  duration=N/A
  
  [/STREAM]

  and

  Stream #0.1: Audio: adpcm_ima_wav, 8000 Hz, 1 channels, s16, 32 kb/s

  my code for the moment is :

  asbd.mSampleRate = 8000;
  
  asbd.mFormatID = kAudioFormatAppleIMA4;
  
  asbd.mBytesPerPacket = 34;
  
  asbd.mFramesPerPacket = 64;
  
  asdb.mbitsPerChannel = 1;

  All the rest is to 0 (flags, byte per frame, bits per channel, reserved).

  Edit :

  I just find that the codec code 17 is maybe for kAudioFormatDVIIntelIMA and not kAudioFormatAppleIMA4. Can someone confirm this ?

  This is from ffprobe :

  [PACKET]
  
  codec_type=audio
  
  stream_index=1
  
  pts=11200
  
  pts_time=11.200000 
  
  dts=11200
  
  dts_time=11.200000 
  
  duration=164
  
  duration_time=0.164000 
  
  size=656.000000 
  
  pos=1171105
  
  flags=K
  
  [/PACKET]

  Is the size 656 for mBytesperpacket ?

  With this value and the half 328 for mFramesPerPacket i can hear something but it's not continuous and has a lot of noise.

  Please help !

• which format to choose in SDL_AudioSpec

  8 mars 2013, par user1896048

  I'm following the ffmpeg tutorial by Dranger (http://dranger.com/ffmpeg) which also uses SDL. For opening the audio device desired parameters are specified in an SDL_AudioSpec. There is a format field which is set to AUDIO_S16SYS. I'm trying to play an mp4 video with aac audio (Audio : aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 151 kb/s) and it just gives noise. Should I use a different format ? If yes, how to decide ?

  This is the code used to set the specs :

  wanted_spec.freq = aCodecCtx->sample_rate;
  
  wanted_spec.format = AUDIO_S16SYS;
  
  wanted_spec.channels = aCodecCtx->channels;
  
  wanted_spec.silence = 0;
  
  wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
  
  wanted_spec.callback = audio_callback;
  
  wanted_spec.userdata = aCodecCtx;

• aac-eld decoding in Linux

  7 mars 2013, par bernard

  I was in Vietnam and tried to learn Vietnamese.

  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'm trying 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 ?