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• hw_base_encode : make recon_frames_ref optional

  30 août 2024, par Lynne

  hw_base_encode : make recon_frames_ref optional
  Vulkan supports some stupidly odd hardware, that unfortunately,
  
  most modern GPUs happen to be.
  
  The DPB images for encoders may be required to be preallocated
  
  all at once, and rather than be individual frames, be layers of
  
  a single frame.
  As the hw_base_encode code is written with the thought that either
  
  the driver or the device itself supports sane image allocation,
  
  Vulkan does not leave us with this option.
  So, in the case that the hardware does not support individual
  
  frames to be used as DPBs, make the DBP frames context optional,
  
  and let the subsystem manage this.
  

  • [DH] libavcodec/hw_base_encode.c

• ffmpeg : concatinating files creates audio artefacts

  28 octobre 2022, par LML

  I'm currently trying to create a video out of multiple short video files. However, the final video always has audio artefacts, where it sounds like a short high pitch or echo at certain times during the audio. All the audio is a text-to-speech generated voice. No music. The artefacts appear sometimes more, sometimes less. But I would obviously prefer to have 0 of it.

  


  My starting point is a long audio file (mono with audio codec "mp3" according to ffprobe). Within that file are a bunch of short pauses of 4-5 seconds. I detect the silences and create individual audio files from there. Afterwards I create an mp4 file with this audio and a still image. Up to this point, the audio is perfectly fine and sounds the exact same as in the original file.

  


  After this I want to create the final video : each of the individual parts added into one long video. There is a transition between each file to mark the changing of image and audio. But even when skipping the transition and simply adding all of these clips that were generated the same way together, the artefacts are still present.

  


  The commands I use to create the different files.

  


  Create individual audio files :
.\ffmpeg.exe -y -hide_banner -i TTSAudio.mp3 -ss 359.944 -to 372.02479 -c copy partXY.mp3

  


  Create individual video files by using a .png file as the video stream and the partXY.mp3 as the audio stream :
.\ffmpeg.exe -y -hide_banner -framerate 30 -loop 1 -i XY_full.png -i partXY.mp3 -c:v libx265 -c:a copy -shortest partXY.mp4

  


  For concatenating the files :
.\ffmpeg.exe -y -hide_banner -i part000.mp4 -i part001.mp4 -i part002.mp4 -filter_complex "[0:v] [0:a] [1:v] [1:a] [2:v] [2:a] concat=n=3:v=1:a=1 [v] [a]" -map "[v]" -map "[a]" -c:v copy -c:a copy final_video.mp4

  


  I've tried a lot of different things and codecs for the audio, without any luck. I use h265, as using h264 was causing weird video artefacts after uploading the file to YouTube.
I have tried reencoding, instead of copying (-c:a copy) at various stages, especially the final video. All without any luck.
I've used the different concatenation where you provide a list of files, which created a whole different set of problems.

  


  I've managed to filter the artefacts out by using -af "lowpass=f=2800", but that changes the voice a lot. I was also not able to notice the pitch visually when opening the audio in audacity, for example.

  


  Example :
https://soundcloud.com/thelml/sets/ffmpeg-audio-artefacts/s-LNr6UaMPgz9?si=f7b30e1e64bf4333ad055fa1fe21e9ec
Due to the files being so short, I seem to have to sometimes have to replay the bugged file to hear the artefact.

  


  So my question is : how do I fix this, without using a lowpass that basically changes the whole voice ?

  


• How to export wave slices to the same bits per sample as the original file

  23 avril 2019, par tzujan

  I am looping through a large wave file, via a dictionary of new file names, lengths and versions. The loop exports the individual slices as files :

  mix.export(key, format='wav')

  However, it converts the original 24-bit file to 32-bit slices. I have been doing a round trip to pro tools to get the files back to 24, as I can’t figure out either ffmpeg settings, or getting the slice into a subprocess.

  I have tried several variations of this theme :

  mix.export(key, format='wav', codec='pcm_s24le')

  As well as this one :

  mix.export(k, format='wav', parameters=['ffmpeg', '-i', '-acodec', 'pcm_s24le', '-ar', '48000'])

  I can’t seem to get the individual slices to work in the following subprocess call. key is the file name from the key-value pair. This works well in the 32-bit export, but not sure how to make it work when a slice’s temp file, such as /var/folders/vc/q7jggn7900l099w45463lgx40000gn/T/tmpw_6mxyg8 needs to be called.

  subprocess.call(['ffmpeg', '-i', key, 
  
                   '-acodec', 'pcm_s24le', '-ar', '48000', 'output.wav'])

  Hoping for slices of the exact same format as the original input :

  mix_file = AudioSegment.from_wav(file_name)