Answers.Body: <p>Actually, the filter only supports 16-bit samples, so anything else will be converted. You can open a bug report of type enhancement at <a href="http://trac.ffmpeg.org" rel="nofollow noreferrer">trac.ffmpeg.org</a></p>
  
Answers.ContentLicense: CC BY-SA 4.0
Answers.CreationDate: 2022-01-25T06:17:42.590Z
Answers.Text: Thank you! -- I have done so now: https://trac.ffmpeg.org/ticket/9613
Answers.UserDisplayName: user272901
Answers.ContentLicense: CC BY-SA 4.0
Answers.CreationDate: 2022-01-24T04:18:27.903Z
Answers.LastActivityDate: 2022-01-24T04:18:27.903Z
Answers.UserDisplayName: Gyan
Body: <p><a href="https://lacinato.com/pub/reaper/test3.wav" rel="nofollow noreferrer">Here is a test file</a> with peaks at +200 dBFS (it's an extreme example, but the same holds for a file with peaks a +1 dBFS).</p>
  <p>When running the &quot;volumedetect&quot; filter on this file, ffmpeg appears to change to 16bit int to run the volume detection, causing everything above 0 dBFS to be clipped (see &quot;Stream mapping&quot; in the transcript below). The result is that it detects the peak at 0 dBFS instead of 200 dBFS (see &quot;max volume&quot; below.)</p>
  <p>Am I missing something? Any way to work around this? The documentation says nothing about 32bit float being unsupported, so it seems like it should &quot;just work&quot;?</p>
  <p>Example run:</p>
  <pre><code>    $ ffmpeg -i test3.wav -filter:a volumedetect -f null /dev/null
      ffmpeg version 4.3.2-0+deb11u1ubuntu1 Copyright (c) 2000-2021 the FFmpeg developers
        built with gcc 10 (Ubuntu 10.2.1-20ubuntu1)
        configuration: --prefix=/usr --extra-version=0+deb11u1ubuntu1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
        libavutil      56. 51.100 / 56. 51.100
        libavcodec     58. 91.100 / 58. 91.100
        libavformat    58. 45.100 / 58. 45.100
        libavdevice    58. 10.100 / 58. 10.100
        libavfilter     7. 85.100 /  7. 85.100
        libavresample   4.  0.  0 /  4.  0.  0
        libswscale      5.  7.100 /  5.  7.100
        libswresample   3.  7.100 /  3.  7.100
        libpostproc    55.  7.100 / 55.  7.100
      Guessed Channel Layout for Input Stream #0.0 : stereo
      Input #0, wav, from 'test3.wav':
        Metadata:
          encoded_by      : REAPER
          date            : 2022-01-22
          creation_time   : 17-06-55
          time_reference  : 0
        Duration: 00:00:03.94, bitrate: 2823 kb/s
          Stream #0:0: Audio: pcm_f32le ([3][0][0][0] / 0x0003), 44100 Hz, stereo, flt, 2822 kb/s
      Stream mapping:
        Stream #0:0 -&gt; #0:0 (pcm_f32le (native) -&gt; pcm_s16le (native))
      Press [q] to stop, [?] for help
      Output #0, null, to '/dev/null':
        Metadata:
          encoded_by      : REAPER
          date            : 2022-01-22
          time_reference  : 0
          encoder         : Lavf58.45.100
          Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
          Metadata:
            encoder         : Lavc58.91.100 pcm_s16le
      size=N/A time=00:00:03.93 bitrate=N/A speed= 911x
      video:0kB audio:678kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
      [Parsed_volumedetect_0 @ 0x55617ae5e140] n_samples: 347156
      [Parsed_volumedetect_0 @ 0x55617ae5e140] mean_volume: -0.0 dB
      [Parsed_volumedetect_0 @ 0x55617ae5e140] max_volume: 0.0 dB
      [Parsed_volumedetect_0 @ 0x55617ae5e140] histogram_0db: 347155
  </code></pre>
  
Tags: <ffmpeg><filter><loudness>
Title: ffmpeg volumedetect filter doesn't work on 32bit float audio?
UserDisplayName: user272901
ThreadText: # ffmpeg volumedetect filter doesn't work on 32bit float audio?
  
  (Asked by: username_0 on 2022-01-23)
  
  [Here is a test file](https://lacinato.com/pub/reaper/test3.wav) with peaks at +200 dBFS (it's an extreme example, but the same holds for a file with peaks a +1 dBFS).
  
  When running the "volumedetect" filter on this file, ffmpeg appears to change to 16bit int to run the volume detection, causing everything above 0 dBFS to be clipped (see "Stream mapping" in the transcript below). The result is that it detects the peak at 0 dBFS instead of 200 dBFS (see "max volume" below.)
  
  Am I missing something? Any way to work around this? The documentation says nothing about 32bit float being unsupported, so it seems like it should "just work"?
  
  Example run:
  
  ```
      $ ffmpeg -i test3.wav -filter:a volumedetect -f null /dev/null
      ffmpeg version 4.3.2-0+deb11u1ubuntu1 Copyright (c) 2000-2021 the FFmpeg developers
        built with gcc 10 (Ubuntu 10.2.1-20ubuntu1)
        configuration: --prefix=/usr --extra-version=0+deb11u1ubuntu1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
        libavutil      56. 51.100 / 56. 51.100
        libavcodec     58. 91.100 / 58. 91.100
        libavformat    58. 45.100 / 58. 45.100
        libavdevice    58. 10.100 / 58. 10.100
        libavfilter     7. 85.100 /  7. 85.100
        libavresample   4.  0.  0 /  4.  0.  0
        libswscale      5.  7.100 /  5.  7.100
        libswresample   3.  7.100 /  3.  7.100
        libpostproc    55.  7.100 / 55.  7.100
      Guessed Channel Layout for Input Stream #0.0 : stereo
      Input #0, wav, from 'test3.wav':
        Metadata:
          encoded_by      : REAPER
          date            : 2022-01-22
          creation_time   : 17-06-55
          time_reference  : 0
        Duration: 00:00:03.94, bitrate: 2823 kb/s
          Stream #0:0: Audio: pcm_f32le ([3][0][0][0] / 0x0003), 44100 Hz, stereo, flt, 2822 kb/s
      Stream mapping:
        Stream #0:0 -> #0:0 (pcm_f32le (native) -> pcm_s16le (native))
      Press [q] to stop, [?] for help
      Output #0, null, to '/dev/null':
        Metadata:
          encoded_by      : REAPER
          date            : 2022-01-22
          time_reference  : 0
          encoder         : Lavf58.45.100
          Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
          Metadata:
            encoder         : Lavc58.91.100 pcm_s16le
      size=N/A time=00:00:03.93 bitrate=N/A speed= 911x
      video:0kB audio:678kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
      [Parsed_volumedetect_0 @ 0x55617ae5e140] n_samples: 347156
      [Parsed_volumedetect_0 @ 0x55617ae5e140] mean_volume: -0.0 dB
      [Parsed_volumedetect_0 @ 0x55617ae5e140] max_volume: 0.0 dB
      [Parsed_volumedetect_0 @ 0x55617ae5e140] histogram_0db: 347155
  
  ```
  
  **Tags:** ffmpeg, filter, loudness
  
  
  ---
  
  ## 1 answers
  
  ### Answer by username_1 on 2022-01-24. Score: 0 | Accepted answer
  
  Actually, the filter only supports 16-bit samples, so anything else will be converted. You can open a bug report of type enhancement at [trac.ffmpeg.org](http://trac.ffmpeg.org)
  
  > **Comments:**
  > **username_0**: Thank you! -- I have done so now: https://trac.ffmpeg.org/ticket/9613  
  
  ---
  
  