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• fftools/ffmpeg : rework -shortest implementation

  10 juin 2022, par Anton Khirnov

  fftools/ffmpeg : rework -shortest implementation
  The -shortest option (which finishes the output file at the time the
  
  shortest stream ends) is currently implemented by faking the -t option
  
  when an output stream ends. This approach is fragile, since it depends
  
  on the frames/packets being processed in a specific order. E.g. there
  
  are currently some situations in which the output file length will
  
  depend unpredictably on unrelated factors like encoder delay. More
  
  importantly, the present work aiming at splitting various ffmpeg
  
  components into different threads will make this approach completely
  
  unworkable, since the frames/packets will arrive in effectively random
  
  order.
  This commit introduces a "sync queue", which is essentially a collection
  
  of FIFOs, one per stream. Frames/packets are submitted to these FIFOs
  
  and are then released for further processing (encoding or muxing) when
  
  it is ensured that the frame in question will not cause its stream to
  
  get ahead of the other streams (the logic is similar to libavformat's
  
  interleaving queue).
  These sync queues are then used for encoding and/or muxing when the
  
   shortest option is specified.
  A new option – -shortest_buf_duration – controls the maximum number of
  
  queued packets, to avoid runaway memory usage.
  This commit changes the results of the following tests :
  
   copy-shortest[12] : the last audio frame is now gone. This is
  
    correct, since it actually outlasts the last video frame.
  
   shortest-sub : the video packets following the last subtitle packet are
  
    now gone. This is also correct.
  

  • [DH] Changelog
  • [DH] doc/ffmpeg.texi
  • [DH] fftools/Makefile
  • [DH] fftools/ffmpeg.c
  • [DH] fftools/ffmpeg.h
  • [DH] fftools/ffmpeg_mux.c
  • [DH] fftools/ffmpeg_opt.c
  • [DH] fftools/sync_queue.c
  • [DH] fftools/sync_queue.h
  • [DH] tests/fate/ffmpeg.mak
  • [DH] tests/ref/fate/copy-shortest1
  • [DH] tests/ref/fate/copy-shortest2
  • [DH] tests/ref/fate/shortest-sub

• 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 ?

  


• 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 ?