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• Why every audio part is louder in FFmpeg when I join them in one audio ?

  14 mai 2024, par Volodymyr Bilovus

  I trying to make dubbing for audio. I have original audio track and I want to put translated audio parts on top of the original.

  


  translated audio 100% vol : —p1--- ---p2— -----p3--- —p4—

  


  original audio 5% vol : -----------------------------------------

  


  Here is my FFmpeg command with filter_complex

  


  ffmpeg -i video_wpmXlZF4XiE.opus -i 989-audio.mp3 -i 989-audio.mp3 -i 989-audio.mp3 -i 989-audio.mp3 \
-filter_complex "\
[0:a]loudnorm=I=-14:TP=-2:LRA=7, volume=0.05[original]; \
[1:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=5000|5000, volume=1.0[sent1]; \
[2:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=10000|10000, volume=1.0[sent2]; \
[3:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=20000|20000, volume=1.0[sent3]; \
[4:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=30000|30000, volume=1.0[sent4]; \
[original][sent1][sent2][sent3][sent4]amix=inputs=5:duration=longest[out]" \
-map "[out]" output.mp3


  


  Audios I put on top of the original audio track is the same -i 989-audio.mp3 I made it by purpose to show the problem
And here is the audio levels on final generated track.


  


  As you can see, first and second only slightly different but third
and fourth have totally different(higher) volume level (Notice, audio is the same).
Why it's happened ? And how can I workaround this odd behaviour ?

  


• Using ffmpeg with libfdk-aac, encoding HE-AAC v1 and mono actually, but in decoding ffmpeg show HE-AAC v2 and stereo [closed]

  13 novembre 2024, par olojzyg

  ffmpeg encode command :

  


  


  ffmpeg -i aac_128000_f32le_22050_1.wav -c:a libfdk_aac -profile:a
aac_he -b:a 64k -channels 1 test.aac

  


  


  output :

  


  [aist#0:0/pcm_s16le @ 000001d38ecfe340] Guessed Channel Layout: mono
  Input #0, wav, from 'aac_128000_f32le_22050_1.wav':   Duration: 00:00:06.48, bitrate: 359 kb/s
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s
  Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> aac (libfdk_aac)) 
  Press [q] to stop, [?] for help Output #0, adts, to 'test.aac':
  Metadata:
        encoder         : Lavf61.7.100   
        Stream #0:0: Audio: aac (HE-AAC), 22050 Hz, mono, s16, 64 kb/s
          Metadata:
            encoder         : Lavc61.19.100 libfdk_aac 
[out#0/adts @ 000001d38ecf8680] video:0KiB audio:53KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 0.000000%
 size=      53KiB time=00:00:06.47 bitrate=  67.0kbits/s speed= 296x


  


  but decode this test.aac, I think HE-AAC and mono is corrent, but it show HE-AAC v2 and stereo, why ?
decode command :

  


  


  ffmpeg -i test.aac -f null -

  


  


  decode output :

  


  [aac @ 0000020ffcab3d40] Estimating duration from bitrate, this may be inaccurate
 Input #0, aac, from 'test.aac':   Duration: 00:00:06.78, bitrate: 63 kb/s
 Stream #0:0: Audio: aac (HE-AACv2), 22050 Hz, stereo, fltp, 63 kb/s
 Stream mapping:
 Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
 Press [q] to stop, [?] for help Output
#0, null, to 'pipe:':
   Metadata:
    encoder         : Lavf61.7.100
   Stream #0:0: Audio: pcm_s16le, 22050 Hz, stereo, s16, 705 kb/s
      Metadata:
        encoder         : Lavc61.19.100 pcm_s16le
[out#0/null @ 0000020ffcac8bc0] video:0KiB audio:584KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: unknown
size=N/A time=00:00:06.78 bitrate=N/A speed=1.6e+03x


  


  what happened ? test.aac is also displayed as 2 channels in Audition. But 1 channel is displayed in MediaInfo, and ADTS header is corrent :

  


  


  channel_configuration : 1 (0x1) - (3 bits)

  


  


• avcodec/prores{dec,dsp} : Remove always-false checks

  24 février, par Andreas Rheinhardt

  avcodec/proresdec,dsp : Remove always-false checks
  avctx->bits_per_raw_sample is always 10 or 12 here ;
  
  the checks have been added in preparation for making
  
  bits_per_raw_sample user-settable via an AVOption,
  
  but this never happened.
  While just at it, also set unpack_alpha earlier
  
  (where bits_per_raw_sample is set).
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
  

  • [DH] libavcodec/proresdec.c
  • [DH] libavcodec/proresdsp.c
  • [DH] libavcodec/proresdsp.h