Hi,
When I run ffmpeg command on a wav file, the received output file is of
large size as compared to the original file. The command and the output are
below,
*FFMPEG command*
% ~/Downloads/audio/ffmpeg -i call-redacted.wav output.wav
ffmpeg version 4.4.1-tessus https://evermeet.cx/ffmpeg/ Copyright (c)
2000-2021 the FFmpeg developers
built with Apple clang version 11.0.0 (clang-1100.0.33.17)
configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
--enable-libfreetype --enable-libgsm --enable-libmodplug
--enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
--enable-libopus --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvmaf
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
--pkg-config-flags=--static --disable-ffplay
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
Input #0, mp3, from 'call-redacted.wav':
Metadata:
encoder : Lavf58.45.100
Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s
Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'output.wav':
Metadata:
ISFT : Lavf58.76.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, stereo,
s16, 256 kb/s
Metadata:
encoder : Lavc58.134.100 pcm_s16le
size= 24569kB time=00:13:06.17 bitrate= 256.0kbits/s speed=2.97e+03x
video:0kB audio:24569kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.000310
*Input and output file sizes*
du -sh call-redacted.wav
6.5M call-redacted.wav
du -sh output.wav
24M output.wav
Alternatively, when I run the same command but change the output
file's extension to mp3, the correct sized file is returned. But the time
taken (6 seconds) in this case is quite higher than the previous case(less
than 1 second). The command and output is below,
~/Downloads/audio/ffmpeg -i call-redacted.wav output.mp3
ffmpeg version 4.4.1-tessus https://evermeet.cx/ffmpeg/ Copyright (c)
2000-2021 the FFmpeg developers
built with Apple clang version 11.0.0 (clang-1100.0.33.17)
configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
--enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
--enable-libfreetype --enable-libgsm --enable-libmodplug
--enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
--enable-libopus --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvmaf
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
--pkg-config-flags=--static --disable-ffplay
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
Input #0, mp3, from 'call-redacted.wav':
Metadata:
encoder : Lavf58.45.100
Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s
Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (mp3float) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'output.mp3':
Metadata:
TSSE : Lavf58.76.100
Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp
Metadata:
encoder : Lavc58.134.100 libmp3lame
size= 2304kB time=00:13:06.17 bitrate= 24.0kbits/s speed= 130x
video:0kB audio:2304kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.011063
du -sh output.mp3
3.3M output.mp3
Please help me out in identifying the right command options for optimum
time and filesize.
Regards,
Shubham
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The fact that the file extension is .wav does not mean it also is an
uncompressed pcm wav file. In this case as correctly identified by
ffmpeg it is an mp3 file (lossy compressed audio) that will always get
bigger when converted to uncompressed pcm. Which is what your first
command does. It decodes the mp3 and stores the uncompressed stream in
pcm format.
Your second command actually encodes the already encoded file again,
reducing the quality of the audio even further as 64 kb is already very
low quality for an mp3 file. This is also why it takes longer.
So what are you trying to achieve?
Greetings Ferdi
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