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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.

  


• command for Php ffmpeg exec

  24 juin 2015, par shilpa shri

  Can anyone please help me with ffmpeg exec command for php , i need converting video file , so that output file should be 640*480 resolution, 128Kbps sound ,output file should be mp4 container video codec H.264 Audio codec AAC

  Currently i am using code
  $a = exec(’"/usr/local/bin/ffmpeg" -y -i "test.mp4" -ab 128k -s 640x480 -vcodec mpeg4 -acodec:a copy "output.mp4" 2>&1 ’, $output, $error) ;

  where output.mp4 file is generated but the video is not clear (very blurred video).
  and the result which displayed on browser shows

  [79] =>     Last message repeated 391 times
  
  [80] => frame= 5159 fps=723 q=24.8 size=   12857kB time=00:03:26.91 bitrate= 509.0kbits/s dup=4591 drop=0    
  
  Error while decoding stream #0:0: Invalid data found when processing input
  
  [81] =>     Last message repeated 359 times
  
  [82] => frame= 5627 fps=739 q=31.0 Lsize=   14071kB time=00:03:45.16 bitrate= 511.9kbits/s dup=5009 drop=0
  
  [83] => video:8650kB audio:5277kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.030902%
  
  [84] => Conversion failed!

  Input file was 82.21mb
  output file was 13.74mb

• ffmpeg -ss then apply filter then concat producing timestamp errors

  6 août 2020, par Bob Ramsey

  Using ffmpeg, I have split a file into multiple parts using -ss. Then I apply a filter to some of the files, then concat the files back together. When I do that, I get : Non-monotonous DTS in output stream 0:0 ; previous : 341334, current : 340526 ; changing to 341335. This may result in incorrect timestamps in the output file. The output file plays, but there are noticeable skips where the files are joined.

  


  Here's how I am splitting the file :

  


  ffmpeg -i full_source.mp4 -ss 0 -to 14.264250 -c copy  01-plain.mp4
ffmpeg -i full_source.mp4 -ss 14.264250 -to 18.435083 -c copy  01-filtered.mp4

ffmpeg -i full_source.mp4 -ss 18.435083 -to 29.988292 -c copy  02-plain.mp4
ffmpeg -i full_source.mp4 -ss 29.988292 -to 31.865167 -c copy  02-filtered.mp4
...
ffmpeg -i full_source.mp4 -ss 0 -to 14.264250 -c copy  10-plain.mp4
ffmpeg -i full_source.mp4 -ss 234.484203 -to 300.000 -c copy  10-filtered.mp4


  


  Then I apply a different drawtext filter on each of the 10 filtered files and save them with a new name, like :

  


  ffmpeg -hide_banner -loglevel warning -y -i 01-filtered.mp4 -filter_complex "drawtext=fontfile=calibri.ttf:fontsize=24:fontcolor=white:x=300:y=500:text='hello world'" -crf 15 01-filtered-complete.mp4


  


  Finally, I join all of the plain and complete files back together like this :

  


  ffmpeg  -f concat -safe 0 -i mylist.txt -c:a copy -c:v copy  outfile.mp4


  


  And that's where the timing error comes in. I've tried adding -vsync drop in the concat command, but that didn't really work either. Same version of ffmpeg does the split, the filter, and the concat. I've tried different versions, everything from 20170519 to one from May 2020 with the same result. Always making sure that all three steps are done by the same version of ffmpeg.

  


  The only thing I can see is that ffprobe shows a duration of 14.27 for 01-plain.mp4 when it should be 14.264250. All of the other files show a similar rounding difference. The files are 23.98 fps. If I do all of my filters in really long command without splitting the file, I can use the more precise numbers with no problem. It just takes 10 times as long. This is all scripted, it happens a couple of hundred times a day and time is money, so I can't take 10 times as long to do each file.

  


  Any ideas ? Thanks in advance !