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    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
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  • bash loop through list of file paths - characters get removed when sending to ffmpeg [duplicate]

    9 octobre 2018, par Brad Johnson

    This question already has an answer here :

    I have a text file that contains a list of paths to flac files I want to convert to wav. Here is a small section of it :

    /mnt/nfs/Music/Rob D/1995 - Clubbed To Death/Rob D - 02 - Clubbed To Death _Kurayamino Variation_.flac
    
/mnt/nfs/Music/Blonde Redhead/2000 - Melody of Certain Damaged Lemons/11 - For the Damaged Coda.flac
    
/mnt/nfs/Music/I Monster/2001 - Daydream In Blue/01 - Daydream In Blue.flac
    
/mnt/nfs/Music/Moby/2002 - Extreme Ways/01 - Extreme Ways.flac
    
/mnt/nfs/Music/RJD2/2002 - Deadringer/01 - The Horror.flac
    
/mnt/nfs/Music/RJD2/2002 - Deadringer/03 - Smoke & Mirrors.flac
    
/mnt/nfs/Music/RJD2/2002 - Deadringer/06 - Ghostwriter.flac
    
/mnt/nfs/Music/RJD2/2002 - Deadringer/10 - Chicken-Bone Circuit.flac
    
/mnt/nfs/Music/FC Kahuna/2003 - Hayling/01 - Hayling _Original_.flac
    
/mnt/nfs/Music/Lamb/2003 - Between Darkness and Wonder/04 - Angelica.flac

    I’m trying to loop through it like so :

    while read -r line; do
    
 wavfile=$(basename "$line")
    
 wavfile="${wavfile%.*}"
    
 ffmpeg -i "$line" "$2/$wavfile.wav"
    
done <$1

    ...where $1 is where I would pass the name of the text file and $2 is the destination directory.

    Here is the output with the irrelevant ffmpeg junk pruned out :

    Input #0, flac, from '/mnt/nfs/Music/Rob D/1995 - Clubbed To Death/Rob D - 02 - Clubbed To Death _Kurayamino Variation_.flac':
    
...
    
Output #0, wav, to '/mnt/gray/Clubbed To Death/Rob D - 02 - Clubbed To Death _Kurayamino Variation_.wav':
    
...
    
/nfs/Music/Blonde Redhead/2000 - Melody of Certain Damaged Lemons/11 - For the Damaged Coda.flac: No such file or directory
    
...
    
Input #0, flac, from '/mnt/nfs/Music/I Monster/2001 - Daydream In Blue/01 - Daydream In Blue.flac':
    
...
    
Output #0, wav, to '/mnt/gray/Clubbed To Death/01 - Daydream In Blue.wav':
    
...
    
nt/nfs/Music/Moby/2002 - Extreme Ways/01 - Extreme Ways.flac: No such file or directory
    
...
    
Input #0, flac, from '/mnt/nfs/Music/RJD2/2002 - Deadringer/01 - The Horror.flac':
    
...
    
Output #0, wav, to '/mnt/gray/Clubbed To Death/01 - The Horror.wav':
    
...
    
nt/nfs/Music/RJD2/2002 - Deadringer/03 - Smoke & Mirrors.flac: No such file or directory
    
...
    
Input #0, flac, from '/mnt/nfs/Music/RJD2/2002 - Deadringer/06 - Ghostwriter.flac':
    
...
    
Output #0, wav, to '/mnt/gray/Clubbed To Death/06 - Ghostwriter.wav':
    
...
    
nt/nfs/Music/RJD2/2002 - Deadringer/10 - Chicken-Bone Circuit.flac: No such file or directory
    
...
    
Input #0, flac, from '/mnt/nfs/Music/FC Kahuna/2003 - Hayling/01 - Hayling _Original_.flac':
    
...
    
Output #0, wav, to '/mnt/gray/Clubbed To Death/01 - Hayling _Original_.wav':
    
...
    
/nfs/Music/Lamb/2003 - Between Darkness and Wonder/04 - Angelica.flac: No such file or directory

    If you pay attention to the paths that ffmpeg reports don’t exist, you’ll see that a seemingly random number of characters has been removed from the beginning. This appears to happen on even numbered lines, but odd numbered lines work. I can only reproduce this behavior when using ffmpeg. If I replace the ffmpeg line with a simple echo statement, every file path is shown to be correct. How can this be ?

    Other suggestions of accomplishing this are also welcome, however I do need the files processed in the order by which I have them listed in the file.

  • Script that will use ffmpeg to add AAC 2.0 audio streams to videos

    9 juillet 2018, par jsr

    I need to cast some videos to my Chromecast that only allows AAC audio. I have some video files and I am continuously adding more. They all have AC3 audio. The ideal would be to have a script that uses ffmpeg/ffprobe to add (not replace) an AAC stream converting the AC3 stream. The script should be looking in a path (and all the subfolders) every night for movies without AAC 2.0 streams.

    The container should me mkv. Can anyone help me getting started ? I’m not very into python yet so I don’t know where to start, but I do understand the coding. Help is much appreciated !

  • Transcode HLS Segments individually using FFMPEG

    27 mai 2013, par rayh

    I am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).

    Here is an example ffmpeg command line :

    ffmpeg -threads 1 -nostdin -loglevel verbose \
    
   -nostdin -y -i input.ts -c:a libfdk_aac \
    
   -ac 2 -b:a 64k -y -metadata -vn output.ts

    Inspecting an example sound file shows that there is a gap at the end of the audio :

    End

    And the start of the file looks suspiciously attenuated (although this may not be an issue) :

    Start

    My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.

    Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?

    ** UPDATE 1 **

    Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)

    Original Start
    Original End

    ** UPDATED 2 **

    So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :

    Side-by-side start
    Side-by-side end

    I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).

    ** UPDATE 3 **

    According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.

    For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.

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