Different audio rates on input DV files

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Different audio rates on input DV files

Bill Moseley
I'm running ffmpeg version 4.1 (-version output below) on Mojave. I'm
pretty new to ffmpeg, so please let me know what more I could provide.

Some years ago I ripped a large number of Mini-DV tapes into clips. My
input files are named:
...
clip-1999-07-29 11;27;40.dv
clip-1999-07-29 11;33;26.dv
clip-1999-07-29 11;33;35.dv
clip-1999-07-29 11;33;47.dv
clip-1999-07-29 11;54;03.dv
clip-1999-07-29 11;54;03 1.dv
clip-1999-07-29 11;57;32.dv
...

I'm now trying to create a single mp4 file for all the clips on a given
day. I run this where "input.txt" is sorted list of the files for a given
day:

$ ffmpeg -f concat -safe 0 -i input.txt -vf yadif out.mp4

Apparently audio settings on the camera changed during that day.

$ ffmpeg -i 'clip-1999-07-29 13;25;19.dv' 2>&1 | grep Audio
    Stream #0:1: Audio: pcm_s16le, 32000 Hz, stereo, s16, 1024 kb/s
    Stream #0:2: Audio: pcm_s16le, 32000 Hz, stereo, s16, 1024 kb/s

$ ffmpeg -i 'clip-1999-07-29 13;26;20.dv' 2>&1 | grep Audio
    Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s


When ffmpeg hits this change I get these messages:

[aac @ 0x7f9d33011e00] Queue input is backward in time
[mp4 @ 0x7f9d3300f400] Non-monotonous DTS in output stream 0:1; previous:
29973504, current: 19983150; changing to 29973505. This may result in
incorrect timestamps in the output file.
[and many more until the end]

And then during playback VLC stops playing audio, and mplayer gets
confused, but video continues to play.

I tried specifying output audio like this, but no luck:

$ ffmpeg -f concat -safe 0 -i input.txt -vf yadif -c:a aac -b:a 384k
-profile:a aac_low out.mp4

If ffmpeg not detecting the change in input rate?

Thanks,


$ ffmpeg -version
ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers
built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1_1 --enable-shared
--enable-pthreads --enable-version3 --enable-hardcoded-tables
--enable-avresample --cc=clang --host-cflags= --host-ldflags=
--enable-ffplay --enable-gpl --enable-libmp3lame --enable-libopus
--enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx
--enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma
--enable-opencl --enable-videotoolbox
libavutil      56. 22.100 / 56. 22.100
libavcodec     58. 35.100 / 58. 35.100
libavformat    58. 20.100 / 58. 20.100
libavdevice    58.  5.100 / 58.  5.100
libavfilter     7. 40.101 /  7. 40.101
libavresample   4.  0.  0 /  4.  0.  0
libswscale      5.  3.100 /  5.  3.100
libswresample   3.  3.100 /  3.  3.100
libpostproc    55.  3.100 / 55.  3.100

--
Bill Moseley
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Re: Different audio rates on input DV files

Bill Moseley
On Tue, Jan 15, 2019 at 6:27 PM Bill Moseley <[hidden email]> wrote:


> $ ffmpeg -f concat -safe 0 -i input.txt -vf yadif out.mp4
>
> Apparently audio settings on the camera changed during that day.
>
> $ ffmpeg -i 'clip-1999-07-29 13;25;19.dv' 2>&1 | grep Audio
>     Stream #0:1: Audio: pcm_s16le, 32000 Hz, stereo, s16, 1024 kb/s
>     Stream #0:2: Audio: pcm_s16le, 32000 Hz, stereo, s16, 1024 kb/s
>
> $ ffmpeg -i 'clip-1999-07-29 13;26;20.dv' 2>&1 | grep Audio
>     Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
>
>
I guess the answer is in the documentation:
https://ffmpeg.org/ffmpeg-formats.html#concat

"All files must have the same streams (same codecs, same time base, etc.)."


Looks like I need to normalized the dv files first, so any that are not
48000 Hz:

$ ffmpeg -i input.dv -ar 48000 out.dv


--
Bill Moseley
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Re: Different audio rates on input DV files

Carl Eugen Hoyos-2
2019-01-16 21:10 GMT+01:00, Bill Moseley <[hidden email]>:

> On Tue, Jan 15, 2019 at 6:27 PM Bill Moseley <[hidden email]> wrote:
>
>
>> $ ffmpeg -f concat -safe 0 -i input.txt -vf yadif out.mp4
>>
>> Apparently audio settings on the camera changed during that day.
>>
>> $ ffmpeg -i 'clip-1999-07-29 13;25;19.dv' 2>&1 | grep Audio
>>     Stream #0:1: Audio: pcm_s16le, 32000 Hz, stereo, s16, 1024 kb/s
>>     Stream #0:2: Audio: pcm_s16le, 32000 Hz, stereo, s16, 1024 kb/s
>>
>> $ ffmpeg -i 'clip-1999-07-29 13;26;20.dv' 2>&1 | grep Audio
>>     Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
>>
>>
> I guess the answer is in the documentation:
> https://ffmpeg.org/ffmpeg-formats.html#concat
>
> "All files must have the same streams (same codecs, same time base, etc.)."
>
>
> Looks like I need to normalized the dv files first, so any that are not
> 48000 Hz:
>
> $ ffmpeg -i input.dv -ar 48000 out.dv

$ ffmpeg -i input.dv -ar 48k -vcodec copy out.dv

The alternative is to use the concat filter that does not
have the format limitations.

Carl Eugen
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