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• Files dissapearing with ffmpeg recursive conversion

  13 août 2014, par CaRoXo

  I started in askubuntu, asking for a way to convert recursively more than 14K of wma to mp3 extracting the wma files path from a txt file.
  There was an answer that could cover my needs, but a bug appears. The first run with some hundreds worked ok. The second, some wma albums got converted, others entirely deleted. There were some modifications. And last time completely, deleted all wma without converting.

  this was the original script

  #!/usr/bin/env bash
  
  
  
  readarray -t files < wma-files.txt
  
  
  
  for file in "${files[@]}"; do
  
      out=`echo $file | sed "s:wma:mp3:"`
  
      probe=`avprobe -show_streams "$file" 2>/dev/null`
  
      rate=`echo "$probe" | grep "bit_rate" | sed "s:.*=\(.*\)[0-9][0-9][0-9][.].*:\1:"`
  
      avconv -i "$file" -ab "$rate"k "$out"
  
      rm "$file"
  
  done

  Then the adaptation with ffmpeg

  #!/usr/bin/env bash
  
  
  
  readarray -t files < wma-files.txt
  
  
  
  for file in "${files[@]}"; do
  
      out=`echo $file | sed "s:wma:mp3:"`
  
      probe=`avprobe -show_streams "$file" 2>/dev/null`
  
      rate=`echo "$probe" | grep "bit_rate" | sed "s:.*=\(.*\)[0-9][0-9][0-9][.].*:\1:"`
  
      ffmpeg -i "$file" -ab "$rate"k "$out" && rm "$file"
  
  done

  With the first one I converted many files. Other just get deleted. The ones deleted were always the same release (so, all tracks from a release). I can listen, and even convert them with Soundkonverter.

  Both of them produces "no such file of directory" and when this happens, everything get deleted.

  The partition where files are stored is a usb HDD ntfs, but also happens in my ext4 internal HD.
  Im under Xubuntu 14.04

  Here the script running with avconv (wich what i managed to convert some, but other get dissapeared) http://pastebin.com/w5weqEws and with ffmpeg (that didn’t convert any) http://pastebin.com/3QkaPzvW

  I can’t find differences between successfully and deleted original wma’s. But for example, while other progs like beets read and write the tags, puddletag and mp3tag (under wine) don’t, until I converted them with soundkonverter.

  As the person trying to help me there redirect me here on the original post http://askubuntu.com/questions/508278/how-to-use-ffmpeg-to-convert-wma-to-mp3-recursively-importing-from-txt-file/508304#508304
  Im here asking for any help to make run an script like this. Or any to use ffmpeg to convert recursively the audio files. My capacity of understanding is short for being able to make something working just reading the docs.

  So I ask a help to run this. If I miss any relevant information, just tell me.

  NOTE : I want to add that doing the conversion with

  for file in "${files[@]}"; do
  
      out=`echo "$file" | sed s:wma:mp3:`
  
      avconv -i "$file" -ab 192k "$out"
  
      rm "$file"
  
  done

  It works in the same files (the ones that are deleted with the other). Only that it makes everything to 192k, so not good if Im converting lower bitrate ones. And get this error "Application provided invalid, non monotonically increasing dts to muxer in stream 0" that seems something typical from avconv in 14.04. With ffmpeg I cant try becouse I don’t find the way how to use it, even out of the script. I really don’t understand the docs seems
  .

  NOTE2 : This is a mediainfo exit from :

  1- A typical wma that get dissapeared always with the script

  Audio
  
  ID                                       : 1
  
  Format                                   : WMA
  
  Format version                           : Version 2
  
  Codec ID                                 : 161
  
  Codec ID/Info                            : Windows Media Audio
  
  Description of the codec                 : Windows Media Audio 9 - 128 kbps, 44 kHz, stereo 1-pass CBR
  
  Duration                                 : 2mn 25s
  
  Bit rate mode                            : Constant
  
  Bit rate                                 : 128 Kbps
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 44.1 KHz
  
  Bit depth                                : 16 bits
  
  Stream size                              : 2.21 MiB (99%)
  
  Language                                 : English (US)

  2- A Wma that got succesfully converted (yes Im using copies now, I cant risk specially some rares audios that I got on the road)

  Audio
  
  ID                                       : 1
  
  Format                                   : WMA
  
  Format version                           : Version 2
  
  Codec ID                                 : 161
  
  Codec ID/Info                            : Windows Media Audio
  
  Description of the codec                 : Windows Media Audio 9 - 128 kbps, 44 kHz, stereo 1-pass CBR
  
  Duration                                 : 4mn 35s
  
  Bit rate mode                            : Constant
  
  Bit rate                                 : 128 Kbps
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 44.1 KHz
  
  Bit depth                                : 16 bits
  
  Stream size                              : 4.21 MiB (99%)
  
  Language                                 : English (US)

  So, as I don’t see difference, but maybe, I’m losing any data to look into ?

• Files dissapearing with ffmpeg recursive conversion

  22 mai 2021, par CaRoXo

  I started in askubuntu, asking for a way to convert recursively more than 14K of wma to mp3 extracting the wma files path from a txt file.
There was an answer that could cover my needs, but a bug appears. The first run with some hundreds worked ok. The second, some wma albums got converted, others entirely deleted. There were some modifications. And last time completely, deleted all wma without converting.

  


  this was the original script

  


  #!/usr/bin/env bash

readarray -t files < wma-files.txt

for file in "${files[@]}"; do
    out=`echo $file | sed "s:wma:mp3:"`
    probe=`avprobe -show_streams "$file" 2>/dev/null`
    rate=`echo "$probe" | grep "bit_rate" | sed "s:.*=\(.*\)[0-9][0-9][0-9][.].*:\1:"`
    avconv -i "$file" -ab "$rate"k "$out"
    rm "$file"
done


  


  Then the adaptation with ffmpeg

  


  #!/usr/bin/env bash

readarray -t files < wma-files.txt

for file in "${files[@]}"; do
    out=`echo $file | sed "s:wma:mp3:"`
    probe=`avprobe -show_streams "$file" 2>/dev/null`
    rate=`echo "$probe" | grep "bit_rate" | sed "s:.*=\(.*\)[0-9][0-9][0-9][.].*:\1:"`
    ffmpeg -i "$file" -ab "$rate"k "$out" && rm "$file"
done


  


  With the first one I converted many files. Other just get deleted. The ones deleted were always the same release (so, all tracks from a release). I can listen, and even convert them with Soundkonverter.

  


  Both of them produces "no such file of directory" and when this happens, everything get deleted.

  


  The partition where files are stored is a usb HDD ntfs, but also happens in my ext4 internal HD.
I'm under Xubuntu 14.04

  


  Here the script running with avconv (which what I managed to convert some, but other get disappeared) http://pastebin.com/w5weqEws and with ffmpeg (that didn't convert any) http://pastebin.com/3QkaPzvW

  


  I can't find differences between successfully and deleted original wma's. But for example, while other progs like beets read and write the tags, puddletag and mp3tag (under wine) don't, until I converted them with soundkonverter.

  


  As the person trying to help me there redirect me here on the original post https://askubuntu.com/questions/508278/how-to-use-ffmpeg-to-convert-wma-to-mp3-recursively-importing-from-txt-file/508304#508304
I'm here asking for any help to make run an script like this. Or any to use ffmpeg to convert recursively the audio files. My capacity of understanding is short for being able to make something working just reading the docs.

  


  So I ask a help to run this. If I miss any relevant information, just tell me.

  


  NOTE : I want to add that doing the conversion with

  


  for file in "${files[@]}"; do
    out=`echo "$file" | sed s:wma:mp3:`
    avconv -i "$file" -ab 192k "$out"
    rm "$file"
done


  


  It works in the same files (the ones that are deleted with the other). Only that it makes everything to 192k, so not good if I'm converting lower bitrate ones. And get this error "Application provided invalid, non monotonically increasing dts to muxer in stream 0" that seems something typical from avconv in 14.04. With ffmpeg I cant try because I don't find the way how to use it, even out of the script. I really don't understand the docs seems
.

  


  NOTE2 : This is a mediainfo exit from :

  


  1- A typical wma that get disappeared always with the script

  


  Audio
ID                                       : 1
Format                                   : WMA
Format version                           : Version 2
Codec ID                                 : 161
Codec ID/Info                            : Windows Media Audio
Description of the codec                 : Windows Media Audio 9 - 128 kbps, 44 kHz, stereo 1-pass CBR
Duration                                 : 2mn 25s
Bit rate mode                            : Constant
Bit rate                                 : 128 Kbps
Channel(s)                               : 2 channels
Sampling rate                            : 44.1 KHz
Bit depth                                : 16 bits
Stream size                              : 2.21 MiB (99%)
Language                                 : English (US)


  


  2- A Wma that got successfully converted (yes I'm using copies now, I can't risk specially some rare audios that I got on the road)

  


  Audio
ID                                       : 1
Format                                   : WMA
Format version                           : Version 2
Codec ID                                 : 161
Codec ID/Info                            : Windows Media Audio
Description of the codec                 : Windows Media Audio 9 - 128 kbps, 44 kHz, stereo 1-pass CBR
Duration                                 : 4mn 35s
Bit rate mode                            : Constant
Bit rate                                 : 128 Kbps
Channel(s)                               : 2 channels
Sampling rate                            : 44.1 KHz
Bit depth                                : 16 bits
Stream size                              : 4.21 MiB (99%)
Language                                 : English (US)


  


  So, as I don't see difference, but maybe, I'm losing any data to look into ?

  


• ffmpeg produced .wav reads only zeros with scipy.io.wavfile

  8 janvier 2015, par question_mark

  Hi everyone and thanks for reading.

  I wanted to do some analysis on a song using Python’s scipy.io.wavfile. Since I only have the song as .mp3 I converted the file to .wav using ffmpeg the following way :

  ffmpeg -i test.mp3 test.wav

  The .wav file plays perfectly well with vlc player, but wavfile shows only zeroes when reading it :

  from scipy.io import wavfile as wf
  
  
  
  data = wf.read("test.wav")
  
  C:\Program Files\Anaconda\lib\site-packages\scipy\io\wavfile.py:42: WavFileWarning: Unknown wave file format
  
    warnings.warn("Unknown wave file format", WavFileWarning)
  
  
  
  data
  
  (44100, array([[0, 0],
  
          [0, 0],
  
          [0, 0],
  
          ..., 
  
          [0, 0],
  
          [0, 0],
  
          [0, 0]], dtype=int16))

  I tried getting the data with Python’s built-in wave module before to the same effect (only zeros).
  I am using the 64bit version of ffmpeg (ffmpeg-20140218-git-61d5970-win64-static).

  Any help is appreciated :-)

  Edit : Included .wav header and tried forcing ffmpeg output format

  I guess the header information of the .wav file is included here :

  ffmpeg -i .\test.wav
  
  Guessed Channel Layout for  Input Stream #0.0 : stereo
  
  Input #0, wav, from '.\test.wav':
  
    Metadata:
  
      artist          : Joe Cocker
  
      copyright       : (C) 1987 Capitol Records, Inc.
  
      date            : 1987
  
      genre           : Pop
  
      title           : Unchain My Heart
  
      album           : Unchain My Heart
  
      track           : 1/10
  
      encoder         : Lavf55.33.100
  
    Duration: 00:05:04.33, bitrate: 1411 kb/s
  
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s

  If I try to specify the ffmpeg output format explicitly for the .mp3 conversion :

  ffmpeg -i .\test.mp3 -f s16le -ar 44100 -ac 2 test.wav
  
  Input #0, mp3, from '.\test.mp3':
  
    Metadata:
  
      title           : Unchain My Heart
  
      artist          : Joe Cocker
  
      album           : Unchain My Heart
  
      genre           : Pop
  
      composer        : Bobby Sharp
  
      track           : 1/10
  
      disc            : 1/1
  
      album_artist    : Joe Cocker
  
      copyright       : (C) 1987 Capitol Records, Inc.
  
      date            : 1987
  
    Duration: 00:05:04.35, start: 0.025056, bitrate: 240 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 235 kb/s
  
      Stream #0:1: Video: mjpeg, yuvj420p(pc), 600x600 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        title           :
  
        comment         : Cover (front)
  
  Output #0, s16le, to 'test.wav':
  
    Metadata:
  
      title           : Unchain My Heart
  
      artist          : Joe Cocker
  
      album           : Unchain My Heart
  
      genre           : Pop
  
      composer        : Bobby Sharp
  
      track           : 1/10
  
      disc            : 1/1
  
      album_artist    : Joe Cocker
  
      copyright       : (C) 1987 Capitol Records, Inc.
  
      date            : 1987
  
      encoder         : Lavf55.33.100
  
      Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 -> pcm_s16le)
  
  Press [q] to stop, [?] for help
  
  video:0kB audio:52425kB subtitle:0 data:0 global headers:0kB muxing overhead 0.000000%
  
  size=   52425kB time=00:05:04.32 bitrate=1411.2kbits/s

  But in this case (forced format), both ffmpeg and wavfile are not able to read the file :

  ffmpeg -i .\test.wav
  
  .\test.wav: Invalid data found when processing input

  and

  data = wf.read("test2.wav")
  
  ---------------------------------------------------------------------------
  
  ValueError                                Traceback (most recent call last)
  
   in <module>()
  
  ----> 1 data = wf.read("test2.wav")
  
  
  
  C:\Program Files\Anaconda\lib\site-packages\scipy\io\wavfile.pyc in read(filename, mmap)
  
      152 
  
      153     try:
  
  --> 154         fsize = _read_riff_chunk(fid)
  
      155         noc = 1
  
      156         bits = 8
  
  
  
  C:\Program Files\Anaconda\lib\site-packages\scipy\io\wavfile.pyc in _read_riff_chunk(fid)
  
       98         _big_endian = True
  
       99     elif str1 != b'RIFF':
  
  --> 100         raise ValueError("Not a WAV file.")
  
      101     if _big_endian:
  
      102         fmt = '>I'
  
  
  
  ValueError: Not a WAV file.
  
  </module>