Recherche avancée

Sur d’autres sites (4589)

• 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 ?

  


• Interval option when downloading HLS format videos published on Internet sites using ffmpeg

  28 septembre 2021, par atsu8492

  I'm trying to download HLS format video as mp4 using ffmpeg.

  


  I don't want to bother the server side due to excessive requests, so I want to create an interval for ts file acquisition.

  


  When I actually run ffmpeg, I sometimes get an HTTP 429 error as shown below.

  


  In addition, it looked like there are no good options in ffmpeg.

  


  [https @ 000001bfef570100] Opening 'https://～/ts/9.ts' for reading
[https @ 000001bfeef31140] HTTP error 429 Too Many Requests
[hls @ 000001bfee6be200] keepalive request failed for 'https://～/ts/9.ts' with error: 'Server returned 4XX Client Error, but not one of 40{0,1,3,4}' when opening url, retrying with new connection
[https @ 000001bfef570100] Opening 'https://～/ts/9.ts' for reading
[https @ 000001bfef570100] Opening 'https://～/ts/10.ts' for reading
[https @ 000001bfef570100] Opening 'https://～/ts/11.ts' for reading
[https @ 000001bfef570100] Opening 'https://～/ts/12.ts' for reading
[https @ 000001bfef570100] Opening 'https://～/ts/13.ts' for reading
[https @ 000001bfeef31140] HTTP error 429 Too Many Requests
[hls @ 000001bfee6be200] keepalive request failed for 'https://～/ts/13.ts' with error: 'Server returned 4XX Client Error, but not one of 40{0,1,3,4}' when opening url, retrying with new connection
[https @ 000001bfef570100] Opening 'https://～/ts/13.ts' for reading
…


  


  Can you come up with any good way ?

  


  I'm not good at English, so I may not be able to respond well. sorry.
This post was written by Google Translate.

  


• FFMPEG how to record aac codec audio same time with video ?

  13 décembre 2017, par Gomi Odabaşıoğlu

  I am trying to use FFMPEG console tools effectively. Now I can record video and audio same time but my issue audio is delaying. I use following code :

  C:\Users\gomid_000\Desktop\ffmpeg.exe -y -f dshow -i audio="Stereo Mix (Realtek High Definition Audio)" -f gdigrab -show_region 1 -framerate 60 -i desktop -map 0 -map 1 -c:v libx264 -preset ultrafast -pix_fmt yuv420p C:\Users\gomid_000\Desktop\out.mp4 -threads:1 4

  I want to use h264 - Intel quick sync codec for video and for the audio
  AAC codec My machine accepts it.

  I tried to use Intel QSV with following code, is that correct ? (scroll "-c:v h264_qsv" part of the code)

  C:\Users\gomid_000\Desktop\ffmpeg.exe -y -f dshow -i audio="Stereo Mix (Realtek High Definition Audio)" -f gdigrab -show_region 1 -framerate 60 -i desktop -map 0 -map 1 -c:v h264_qsv -pix_fmt yuv420p C:\Users\gomid_000\Desktop\out.mp4

  Still do not know How to use AAC to record audio and keeep video and audio matching times automaticly. It is not installed maybe (?) (I do not know how to tell this in English will try :"Person speaks but audio comes very near but delays a little bit when I try bandicam they just match each other without any delay this is not hardware related.")