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#7 Ambience
16 octobre 2011, par
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16 octobre 2011, par
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Extracted .wav file from .mp4 video with ffmpeg only has half of the duration
16 mars 2021, par TheSprinterI've been using ffmpeg to extract .wav files from .mp4 videos to then generate waveforms and spectrograms for it, but I stumbled upon a file for which ffmpeg does not generate the full audio file. The video duration is 25:25 (25 minutes and 25 seconds), but the .wav file is 12:28 (12 minutes and 28 seconds).


This is what I've been doing :


C:\Users\<user>> ffmpeg -i video.mp4 -ab 160k -ac 1 -ar 44100 -vn audio.wav
</user>


I tried with
-ac 2
only to see if I get something different, but the result is the same. In fact, I can't generate the plots I need with a stereo audio file.

This is what I get in the terminal when I run the command :


C:\Users\<user>>ffmpeg -i video.mp4 -ab 160k -ac 1 -ar 44100 -vn audio.wav
ffmpeg version 4.3 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200621
 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libgsm --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 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
 libswscale 5. 7.100 / 5. 7.100
 libswresample 3. 7.100 / 3. 7.100
 libpostproc 55. 7.100 / 55. 7.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'audio.mp4':
 Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: mp42mp41
 creation_time : 2021-03-09T05:00:00.000000Z
 Duration: 00:25:25.29, start: 0.000000, bitrate: 2308 kb/s
 Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720, 4410 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
 Metadata:
 creation_time : 2021-03-09T05:00:00.000000Z
 handler_name : ?Mainconcept Video Media Handler
 encoder : AVC Coding
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 253 kb/s (default)
 Metadata:
 creation_time : 2021-03-09T05:00:00.000000Z
 handler_name : #Mainconcept MP4 Sound Media Handler
Stream mapping:
 Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'audio.wav':
 Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: mp42mp41
 ISFT : Lavf58.45.100
 Stream #0:0(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s (default)
 Metadata:
 creation_time : 2021-03-09T05:00:00.000000Z
 handler_name : #Mainconcept MP4 Sound Media Handler
 encoder : Lavc58.91.100 pcm_s16le
size= 64463kB time=00:12:28.41 bitrate= 705.6kbits/s speed= 156x
video:0kB audio:64463kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000118%
</user>


If this information on the video is not enough to troubleshoot, the video says the following (when it starts, for the first few seconds) :



EP TRT (23.976): 01:24:53:18
EP TRT (29.976): 01:24:55:06

AUDIO LAYOUT

Channel 1: Left Front (L)
Channel 2: Right Front (R)
Channel 3: Center (C)
Channel 4: Low Frequency/Effects (LFE)
Channel 5: Left Surround (Ls)
Channel 6: Right Surround (Rs)
Channel 7: Mono Mix (L)
Channel 8: Mono Mix (R)



This is for TV, that's why you see that longer duration. I don't know if all this is relevant, actually.


Some additional info


Something curious is that I'm doing this for manual tests, but I'm actually creating an app with python and, for that, I'm using
python-ffmpeg
module, and I with that I'm getting the correct .wav file (with the correct duration). The python lines for this are the following :

(
 ffmpeg
 .input(
 
 )
 .output(
 ,
 audio_bitrate=316000,
 ac=1,
 ar=48000
 )
 .run()
)