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• Decoding AAC to PCM with ffmpeg results in noise

  18 octobre 2022, par userDtrm

  I have a .mp4 file generated with ffmpeg as follows.

  


  


  ffmpeg -y -i video_extended.mp4 -itsoffset 00:00:04.00 -i output5-1.wav -map 0:0 -map 1:0 -c:v copy -c:a aac -ac 6 -ar 48000 -b:a 128k -async 1 mixed.mp4

  


  


  Playing mixed.mp4 file with ffplay is fine and there is no impact to the sound quality. Below is the output I get from ffplay when using the command ffplay -i  mixed.mp4

  


  > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
> 'mixed_h264_aac_512k_async_qp0_all_I.mp4':   Metadata:
>     major_brand     : isom
>     minor_version   : 512
>     compatible_brands: isomiso2avc1mp41
>     encoder         : Lavf58.76.100   Duration: 00:00:16.02, start: 0.000000, bitrate: 49136 kb/s   Stream #0:0[0x1](und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv422p10le(progressive),
> 1920x1080, 65409 kb/s, 59.94 fps, 59.94 tbr, 11988 tbn (default)
>     Metadata:
>       handler_name    : VideoHandler
>       vendor_id       : [0][0][0][0]   Stream #0:1[0x2](und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 71 kb/s (default)
>     Metadata:
>       handler_name    : SoundHandler
>       vendor_id       : [0][0][0][0] Switch subtitle stream from #-1 to #-1 vq= 1606KB sq=    0B f=0/0


  


  Then, I decode the mixed.mp4 file back to raw PCM using the following command.

  


  


  ffmpeg -i mixed.mp4 -vn -acodec pcm_s16le -f s16le -ar 48000 -ac 6 raw_audio.pcm

  


  


  However, this raw_audio.pcm contains a lot of noise and ffplay output shows the following output

  


  [s16le @ 0x7f7490000c80] Estimating duration from bitrate, this may be inaccurate
Input #0, s16le, from 'separated_audio_s16.pcm':
  Duration: 00:00:16.02, bitrate: 4607 kb/s
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, 6 channels, s16, 4608 kb/s
[pcm_s16le @ 0x7f749002b940] Multiple frames in a packet.
[pcm_s16le @ 0x7f749002b940] Invalid PCM packet, data has size 8 but at least a size of 12 was expected
    Last message repeated 32 times
[pcm_s16le @ 0x7f749002b940] Invalid PCM packet, data has size 8 but at least a size of 12 was expected
    Last message repeated 11 times
Switch subtitle stream from #-1 to #-1 vq=    0KB sq=    0B f=0/0   
[pcm_s16le @ 0x7f749002b940] Invalid PCM packet, data has size 8 but at least a size of 12 was expected
    Last message repeated 11 times
[pcm_s16le @ 0x7f749002b940] Invalid PCM packet, data has size 8 but at least a size of 12 was expected
    Last message repeated 11 times
[pcm_s16le @ 0x7f749002b940] Invalid PCM packet, data has size 8 but at least a size of 12 was expected


  


  Can someone please explain the issue here ? Note that the ffplay command that works correctly for mixed.mp4 shows fltp as the audio format, whereas when playing the raw_audio.pcm file, it is seen as s16.

  


  Is this a resampling issue in ffmpeg, and how can I rectify this ?

  


  I’m using ffmpeg and ffplay versions 5.0.1 in a Fedora 36 system.

  


  Thank you.

  


• lavfi/vf_fieldmatch : keep fields as-is if not matched properly

  3 novembre 2022, par mail@nodoa.me

  lavfi/vf_fieldmatch : keep fields as-is if not matched properly
  Makes it possible to use deinterlacers which output one frame for each field as fallback if field
  
  matching fails (combmatch=full).
  Currently, the documented example with fallback on a post-deinterlacer will only work in case the
  
  deinterlacer outputs one frame per first field (as yadif=mode=0). The reason for that is that
  
  fieldmatch will attempt to match the second field regardless of whether it recognizes the end
  
  result is still interlaced. This produces garbled output with for example mixed telecined 24fps and
  
  60i content combined with a field-based deinterlaced such as yadif=mode=1.
  
  This patch orders fieldmatch to revert to using the second field of the current frame in case the
  
  end result is still interlaced and a post-deinterlacer is assumed to be used.
  Signed-off-by : lovesyk <lovesyk@users.noreply.github.com>
  

  • [DH] libavfilter/vf_fieldmatch.c

• ffmpeg video filters on each input video from txt input-file using stream_loop

  16 octobre 2022, par thereceptionist

  I'm using the below command to loop a number of video clips over the duration of an audio track :

  


  ffmpeg -stream_loop -1 -f concat -i input.txt -i audio.mp3 -map 0:v -map 1:a -shortest output.mp4

  


  Where input.txt looks like this :

  


  file '1.mp4'
file '2.mp4'
file '3.mp4'
file '4.mp4'
file '5.mp4'
... 
file 'n.mp4'


  


  I would like to add a fade in and fade out effect every time a new "clip" is mixed in. I have seen a few examples where the input files must be explicitly referenced in the effects chain. But since the number of videos in input.txt can vary it would be great to be able to add the same effect over and over again.