Non-monotonous DTS in output stream

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Non-monotonous DTS in output stream

Pieter Venter
Hello,

I'm trying to process some AVI files that originally came from a Sony
Handycam to mp4 and managed to get that to work.
However, a couple of files are giving me trouble and I've spent days trying
to figure it out.

I'm new here but have done my homework as best I could:
* I downloaded the latest ffmpeg build I could find.
* Searched the forums for solutions/switches (-af
aresample=async=1, -fflags +igndts, -fflags +sortdts).
* Tried other tools to extract the audio.
* Past full command and output for your reference.
* Will try to not "top post". That seems to be a thing here.

If I run the following command, audio is processed correctly up to about
the 43s mark, then becomes slower than expected (i.e. voices are deep,
audio in "slow mo" but video plays normal).

The original file plays correctly with VLC and Video on Linux. It does not
play correctly with mplayer (same slowed down audio issue past 43 seconds).

fmpeg -i input.avi -c:v libx264 -preset fast -crf 21 output.mp4
ffmpeg version N-55863-g9f38fac053-static https://johnvansickle.com/ffmpeg/
 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 8 (Debian 8.3.0-6)
  configuration: --enable-gpl --enable-version3 --enable-static
--disable-debug --disable-ffplay --disable-indev=sndio
--disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom
--enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-librubberband --enable-libsoxr
--enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
--enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi
--enable-libzimg
  libavutil      56. 64.100 / 56. 64.100
  libavcodec     58.119.100 / 58.119.100
  libavformat    58. 65.101 / 58. 65.101
  libavdevice    58. 11.103 / 58. 11.103
  libavfilter     7.100.100 /  7.100.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
[avi @ 0x620c340] Switching to NI mode, due to poor interleaving
Input #0, avi, from 'input.avi':
  Duration: 00:07:00.00, start: 0.000000, bitrate: 28806 kb/s
    Stream #0:0: Video: dvvideo, yuv420p, 720x576 [SAR 16:15 DAR 4:3],
25000 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
    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
Stream mapping:
  Stream #0:0 -> #0:0 (dvvideo (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
[libx264 @ 0x6233580] using SAR=16/15
[libx264 @ 0x6233580] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
AVX FMA3 BMI2 AVX2
[libx264 @ 0x6233580] profile High, level 3.0, 4:2:0, 8-bit
[libx264 @ 0x6233580] 264 - core 161 r3040 35417dc - H.264/MPEG-4 AVC codec
- Copyleft 2003-2021 - http://www.videolan.org/x264.html - options: cabac=1
ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=6 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11
fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2
sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1
weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=25 scenecut=40
intra_refresh=0 rc_lookahead=30 rc=crf mbtree=1 crf=21.0 qcomp=0.60 qpmin=0
qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'output.mp4':
  Metadata:
    encoder         : Lavf58.65.101
    Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p(bottom coded
first (swapped)), 720x576 [SAR 16:15 DAR 4:3], q=2-31, 25 fps, 12800 tbn
    Metadata:
      encoder         : Lavc58.119.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
    Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 32000 Hz, stereo,
fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.119.100 aac
frame=10500 fps=116 q=-1.0 Lsize=  168876kB time=00:07:00.02
bitrate=3293.7kbits/s speed=4.63x
video:158898kB audio:9545kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead: 0.257381%
[libx264 @ 0x6233580] frame I:93    Avg QP:20.65  size: 54098
[libx264 @ 0x6233580] frame P:3607  Avg QP:23.10  size: 22302
[libx264 @ 0x6233580] frame B:6800  Avg QP:24.65  size: 11358
[libx264 @ 0x6233580] consecutive B-frames: 13.4%  0.6%  0.7% 85.3%
[libx264 @ 0x6233580] mb I  I16..4:  1.2% 98.2%  0.6%
[libx264 @ 0x6233580] mb P  I16..4:  0.5% 32.9%  0.4%  P16..4: 35.4% 17.7%
 9.8%  0.0%  0.0%    skip: 3.2%
[libx264 @ 0x6233580] mb B  I16..4:  2.2% 25.7%  0.2%  B16..8: 22.8%  8.7%
 0.7%  direct:28.9%  skip:10.8%  L0:40.2% L1:32.6% BI:27.2%
[libx264 @ 0x6233580] 8x8 transform intra:93.9% inter:82.3%
[libx264 @ 0x6233580] coded y,uvDC,uvAC intra: 82.4% 88.7% 30.9% inter:
40.1% 65.9% 1.2%
[libx264 @ 0x6233580] i16 v,h,dc,p: 21% 19% 34% 26%
[libx264 @ 0x6233580] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 15% 14% 40%  5%  5%
 5%  5%  6%  6%
[libx264 @ 0x6233580] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu:  9% 54% 15%  4%  4%
 3%  4%  3%  4%
[libx264 @ 0x6233580] i8c dc,h,v,p: 59% 19% 19%  3%
[libx264 @ 0x6233580] Weighted P-Frames: Y:8.2% UV:1.4%
[libx264 @ 0x6233580] ref P L0: 56.9% 43.1%
[libx264 @ 0x6233580] ref B L0: 74.0% 26.0%
[libx264 @ 0x6233580] ref B L1: 95.3%  4.7%
[libx264 @ 0x6233580] kb/s:3099.25
[aac @ 0x6234fc0] Qavg: 189.772

Trying to narrow down the problem area, I did the following - just encode
up to the 43s mark and dump the audio.
If I run it for the whole file, there are hundreds, if not thousands of
entries like " Non-monotonous DTS..."

 ffmpeg -t 00:00:43 -i  input.avi -map 0:a:0 -c:a aac output.avi
ffmpeg version N-55863-g9f38fac053-static https://johnvansickle.com/ffmpeg/
 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 8 (Debian 8.3.0-6)
  configuration: --enable-gpl --enable-version3 --enable-static
--disable-debug --disable-ffplay --disable-indev=sndio
--disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom
--enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-librubberband --enable-libsoxr
--enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
--enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi
--enable-libzimg
  libavutil      56. 64.100 / 56. 64.100
  libavcodec     58.119.100 / 58.119.100
  libavformat    58. 65.101 / 58. 65.101
  libavdevice    58. 11.103 / 58. 11.103
  libavfilter     7.100.100 /  7.100.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
[avi @ 0x5b922c0] Switching to NI mode, due to poor interleaving
Input #0, avi, from 'input.avi':
  Duration: 00:07:00.00, start: 0.000000, bitrate: 28806 kb/s
    Stream #0:0: Video: dvvideo, yuv420p, 720x576 [SAR 16:15 DAR 4:3],
25000 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
    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
File 'output.avi' already exists. Overwrite? [y/N] y
Stream mapping:
  Stream #0:1 -> #0:0 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, avi, to 'output.avi':
  Metadata:
    ISFT            : Lavf58.65.101
    Stream #0:0: Audio: aac (LC) ([255][0][0][0] / 0x00FF), 32000 Hz,
stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.119.100 aac
[avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1341,
current: 1339; changing to 1342. This may result in incorrect timestamps in
the output file.
[avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1342,
current: 1340; changing to 1343. This may result in incorrect timestamps in
the output file.
[avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1343,
current: 1340; changing to 1344. This may result in incorrect timestamps in
the output file.
[avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1344,
current: 1341; changing to 1345. This may result in incorrect timestamps in
the output file.
[avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1345,
current: 1342; changing to 1346. This may result in incorrect timestamps in
the output file.
[avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1346,
current: 1342; changing to 1347. This may result in incorrect timestamps in
the output file.
[avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1347,
current: 1343; changing to 1348. This may result in incorrect timestamps in
the output file.
[avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1348,
current: 1343; changing to 1349. This may result in incorrect timestamps in
the output file.
size=     714kB time=00:00:43.16 bitrate= 135.6kbits/s speed=96.2x
video:0kB audio:676kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 5.592729%
[aac @ 0x5bb9980] Qavg: 225.541

The only solution I could find so far is to
a. extract all the audio using ffmpeg (total audio file lengh is 10m09s)
b. use audacity to carefully select the audio from 0m43s to the end and
"shrink" it down to a total of 7m00s (the original file length)
c. create a new video file from the processed audio stream

Is there a way to troubleshoot, ignore, correct the Non-monotonous DTS
error?
Thanks for your help.
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Re: Non-monotonous DTS in output stream

Pieter Venter
I've found out some new info, but worried about this top post etiquette.
Please help if I'm doing this wrong.

On Tue, Feb 9, 2021 at 7:19 PM Pieter Venter <[hidden email]> wrote:

> Hello,
>
> I'm trying to process some AVI files that originally came from a Sony
> Handycam to mp4 and managed to get that to work.
> However, a couple of files are giving me trouble and I've spent days
> trying to figure it out.
>
> I'm new here but have done my homework as best I could:
> * I downloaded the latest ffmpeg build I could find.
> * Searched the forums for solutions/switches (-af
> aresample=async=1, -fflags +igndts, -fflags +sortdts).
> * Tried other tools to extract the audio.
> * Past full command and output for your reference.
> * Will try to not "top post". That seems to be a thing here.
>
> If I run the following command, audio is processed correctly up to about
> the 43s mark, then becomes slower than expected (i.e. voices are deep,
> audio in "slow mo" but video plays normal).
>
> The original file plays correctly with VLC and Video on Linux. It does not
> play correctly with mplayer (same slowed down audio issue past 43 seconds).
>
> fmpeg -i input.avi -c:v libx264 -preset fast -crf 21 output.mp4
> ffmpeg version N-55863-g9f38fac053-static
> https://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2021 the FFmpeg
> developers
>   built with gcc 8 (Debian 8.3.0-6)
>   configuration: --enable-gpl --enable-version3 --enable-static
> --disable-debug --disable-ffplay --disable-indev=sndio
> --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r
> --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom
> --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype
> --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libopenjpeg --enable-librubberband --enable-libsoxr
> --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
> --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
> --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi
> --enable-libzimg
>   libavutil      56. 64.100 / 56. 64.100
>   libavcodec     58.119.100 / 58.119.100
>   libavformat    58. 65.101 / 58. 65.101
>   libavdevice    58. 11.103 / 58. 11.103
>   libavfilter     7.100.100 /  7.100.100
>   libswscale      5.  8.100 /  5.  8.100
>   libswresample   3.  8.100 /  3.  8.100
>   libpostproc    55.  8.100 / 55.  8.100
> [avi @ 0x620c340] Switching to NI mode, due to poor interleaving
> Input #0, avi, from 'input.avi':
>   Duration: 00:07:00.00, start: 0.000000, bitrate: 28806 kb/s
>     Stream #0:0: Video: dvvideo, yuv420p, 720x576 [SAR 16:15 DAR 4:3],
> 25000 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
>     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
> Stream mapping:
>   Stream #0:0 -> #0:0 (dvvideo (native) -> h264 (libx264))
>   Stream #0:1 -> #0:1 (pcm_s16le (native) -> aac (native))
> Press [q] to stop, [?] for help
> [libx264 @ 0x6233580] using SAR=16/15
> [libx264 @ 0x6233580] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> AVX FMA3 BMI2 AVX2
> [libx264 @ 0x6233580] profile High, level 3.0, 4:2:0, 8-bit
> [libx264 @ 0x6233580] 264 - core 161 r3040 35417dc - H.264/MPEG-4 AVC
> codec - Copyleft 2003-2021 - http://www.videolan.org/x264.html - options:
> cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=6 psy=1
> psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1
> cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12
> lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0
> bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1
> b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=25
> scenecut=40 intra_refresh=0 rc_lookahead=30 rc=crf mbtree=1 crf=21.0
> qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> Output #0, mp4, to 'output.mp4':
>   Metadata:
>     encoder         : Lavf58.65.101
>     Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p(bottom coded
> first (swapped)), 720x576 [SAR 16:15 DAR 4:3], q=2-31, 25 fps, 12800 tbn
>     Metadata:
>       encoder         : Lavc58.119.100 libx264
>     Side data:
>       cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
>     Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 32000 Hz, stereo,
> fltp, 128 kb/s
>     Metadata:
>       encoder         : Lavc58.119.100 aac
> frame=10500 fps=116 q=-1.0 Lsize=  168876kB time=00:07:00.02
> bitrate=3293.7kbits/s speed=4.63x
> video:158898kB audio:9545kB subtitle:0kB other streams:0kB global
> headers:0kB muxing overhead: 0.257381%
> [libx264 @ 0x6233580] frame I:93    Avg QP:20.65  size: 54098
> [libx264 @ 0x6233580] frame P:3607  Avg QP:23.10  size: 22302
> [libx264 @ 0x6233580] frame B:6800  Avg QP:24.65  size: 11358
> [libx264 @ 0x6233580] consecutive B-frames: 13.4%  0.6%  0.7% 85.3%
> [libx264 @ 0x6233580] mb I  I16..4:  1.2% 98.2%  0.6%
> [libx264 @ 0x6233580] mb P  I16..4:  0.5% 32.9%  0.4%  P16..4: 35.4% 17.7%
>  9.8%  0.0%  0.0%    skip: 3.2%
> [libx264 @ 0x6233580] mb B  I16..4:  2.2% 25.7%  0.2%  B16..8: 22.8%  8.7%
>  0.7%  direct:28.9%  skip:10.8%  L0:40.2% L1:32.6% BI:27.2%
> [libx264 @ 0x6233580] 8x8 transform intra:93.9% inter:82.3%
> [libx264 @ 0x6233580] coded y,uvDC,uvAC intra: 82.4% 88.7% 30.9% inter:
> 40.1% 65.9% 1.2%
> [libx264 @ 0x6233580] i16 v,h,dc,p: 21% 19% 34% 26%
> [libx264 @ 0x6233580] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 15% 14% 40%  5%  5%
>  5%  5%  6%  6%
> [libx264 @ 0x6233580] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu:  9% 54% 15%  4%  4%
>  3%  4%  3%  4%
> [libx264 @ 0x6233580] i8c dc,h,v,p: 59% 19% 19%  3%
> [libx264 @ 0x6233580] Weighted P-Frames: Y:8.2% UV:1.4%
> [libx264 @ 0x6233580] ref P L0: 56.9% 43.1%
> [libx264 @ 0x6233580] ref B L0: 74.0% 26.0%
> [libx264 @ 0x6233580] ref B L1: 95.3%  4.7%
> [libx264 @ 0x6233580] kb/s:3099.25
> [aac @ 0x6234fc0] Qavg: 189.772
>
> Trying to narrow down the problem area, I did the following - just encode
> up to the 43s mark and dump the audio.
> If I run it for the whole file, there are hundreds, if not thousands of
> entries like " Non-monotonous DTS..."
>
>  ffmpeg -t 00:00:43 -i  input.avi -map 0:a:0 -c:a aac output.avi
> ffmpeg version N-55863-g9f38fac053-static
> https://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2021 the FFmpeg
> developers
>   built with gcc 8 (Debian 8.3.0-6)
>   configuration: --enable-gpl --enable-version3 --enable-static
> --disable-debug --disable-ffplay --disable-indev=sndio
> --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r
> --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom
> --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype
> --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libopenjpeg --enable-librubberband --enable-libsoxr
> --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
> --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
> --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi
> --enable-libzimg
>   libavutil      56. 64.100 / 56. 64.100
>   libavcodec     58.119.100 / 58.119.100
>   libavformat    58. 65.101 / 58. 65.101
>   libavdevice    58. 11.103 / 58. 11.103
>   libavfilter     7.100.100 /  7.100.100
>   libswscale      5.  8.100 /  5.  8.100
>   libswresample   3.  8.100 /  3.  8.100
>   libpostproc    55.  8.100 / 55.  8.100
> [avi @ 0x5b922c0] Switching to NI mode, due to poor interleaving
> Input #0, avi, from 'input.avi':
>   Duration: 00:07:00.00, start: 0.000000, bitrate: 28806 kb/s
>     Stream #0:0: Video: dvvideo, yuv420p, 720x576 [SAR 16:15 DAR 4:3],
> 25000 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
>     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
> File 'output.avi' already exists. Overwrite? [y/N] y
> Stream mapping:
>   Stream #0:1 -> #0:0 (pcm_s16le (native) -> aac (native))
> Press [q] to stop, [?] for help
> Output #0, avi, to 'output.avi':
>   Metadata:
>     ISFT            : Lavf58.65.101
>     Stream #0:0: Audio: aac (LC) ([255][0][0][0] / 0x00FF), 32000 Hz,
> stereo, fltp, 128 kb/s
>     Metadata:
>       encoder         : Lavc58.119.100 aac
> [avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1341,
> current: 1339; changing to 1342. This may result in incorrect timestamps in
> the output file.
> [avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1342,
> current: 1340; changing to 1343. This may result in incorrect timestamps in
> the output file.
> [avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1343,
> current: 1340; changing to 1344. This may result in incorrect timestamps in
> the output file.
> [avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1344,
> current: 1341; changing to 1345. This may result in incorrect timestamps in
> the output file.
> [avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1345,
> current: 1342; changing to 1346. This may result in incorrect timestamps in
> the output file.
> [avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1346,
> current: 1342; changing to 1347. This may result in incorrect timestamps in
> the output file.
> [avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1347,
> current: 1343; changing to 1348. This may result in incorrect timestamps in
> the output file.
> [avi @ 0x5bb8040] Non-monotonous DTS in output stream 0:0; previous: 1348,
> current: 1343; changing to 1349. This may result in incorrect timestamps in
> the output file.
> size=     714kB time=00:00:43.16 bitrate= 135.6kbits/s speed=96.2x
> video:0kB audio:676kB subtitle:0kB other streams:0kB global headers:0kB
> muxing overhead: 5.592729%
> [aac @ 0x5bb9980] Qavg: 225.541
>
> The only solution I could find so far is to
> a. extract all the audio using ffmpeg (total audio file lengh is 10m09s)
> b. use audacity to carefully select the audio from 0m43s to the end and
> "shrink" it down to a total of 7m00s (the original file length)
> c. create a new video file from the processed audio stream
>
> Is there a way to troubleshoot, ignore, correct the Non-monotonous DTS
> error?
> Thanks for your help.
>
>
Using the -debug_ts switch, I can now see that the second audio channel
stops at 42.52

muxer <- type:audio pkt_pts:1060 pkt_pts_time:42.4 pkt_dts:1060
pkt_dts_time:42.4 size:5120
demuxer -> ist_index:1 type:audio next_dts:42400000 next_dts_time:42.4
next_pts:42400000 next_pts_time:42.4 pkt_pts:1061 pkt_pts_time:42.44
pkt_dts:1061 pkt_dts_time:42.44 off:0 off_time:0
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1061 pkt_pts_time:42.44
pkt_dts:1061 pkt_dts_time:42.44 off:0 off_time:0
muxer <- type:audio pkt_pts:1061 pkt_pts_time:42.44 pkt_dts:1061
pkt_dts_time:42.44 size:5120
demuxer -> ist_index:2 type:audio next_dts:42400000 next_dts_time:42.4
next_pts:42400000 next_pts_time:42.4 pkt_pts:1061 pkt_pts_time:42.44
pkt_dts:1061 pkt_dts_time:42.44 off:0 off_time:0
demuxer+ffmpeg -> ist_index:2 type:audio pkt_pts:1061 pkt_pts_time:42.44
pkt_dts:1061 pkt_dts_time:42.44 off:0 off_time:0
muxer <- type:audio pkt_pts:1061 pkt_pts_time:42.44 pkt_dts:1061
pkt_dts_time:42.44 size:5120
demuxer -> ist_index:1 type:audio next_dts:42440000 next_dts_time:42.44
next_pts:42440000 next_pts_time:42.44 pkt_pts:1062 pkt_pts_time:42.48
pkt_dts:1062 pkt_dts_time:42.48 off:0 off_time:0
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1062 pkt_pts_time:42.48
pkt_dts:1062 pkt_dts_time:42.48 off:0 off_time:0
muxer <- type:audio pkt_pts:1062 pkt_pts_time:42.48 pkt_dts:1062
pkt_dts_time:42.48 size:5120
demuxer -> ist_index:2 type:audio next_dts:42440000 next_dts_time:42.44
next_pts:42440000 next_pts_time:42.44 pkt_pts:1062 pkt_pts_time:42.48
pkt_dts:1062 pkt_dts_time:42.48 off:0 off_time:0
demuxer+ffmpeg -> ist_index:2 type:audio pkt_pts:1062 pkt_pts_time:42.48
pkt_dts:1062 pkt_dts_time:42.48 off:0 off_time:0
muxer <- type:audio pkt_pts:1062 pkt_pts_time:42.48 pkt_dts:1062
pkt_dts_time:42.48 size:5120
demuxer -> ist_index:1 type:audio next_dts:42480000 next_dts_time:42.48
next_pts:42480000 next_pts_time:42.48 pkt_pts:1063 pkt_pts_time:42.52
pkt_dts:1063 pkt_dts_time:42.52 off:0 off_time:0
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1063 pkt_pts_time:42.52
pkt_dts:1063 pkt_dts_time:42.52 off:0 off_time:0
muxer <- type:audio pkt_pts:1063 pkt_pts_time:42.52 pkt_dts:1063
pkt_dts_time:42.52 size:5120
demuxer -> ist_index:2 type:audio next_dts:42480000 next_dts_time:42.48
next_pts:42480000 next_pts_time:42.48 pkt_pts:1063 pkt_pts_time:42.52
pkt_dts:1063 pkt_dts_time:42.52 off:0 off_time:0
demuxer+ffmpeg -> ist_index:2 type:audio pkt_pts:1063 pkt_pts_time:42.52
pkt_dts:1063 pkt_dts_time:42.52 off:0 off_time:0

From this point forward, no more channel 2 for audio

muxer <- type:audio pkt_pts:1063 pkt_pts_time:42.52 pkt_dts:1063
pkt_dts_time:42.52 size:5120
demuxer -> ist_index:1 type:audio next_dts:42520000 next_dts_time:42.52
next_pts:42520000 next_pts_time:42.52 pkt_pts:1064 pkt_pts_time:42.56
pkt_dts:1064 pkt_dts_time:42.56 off:0 off_time:0
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1064 pkt_pts_time:42.56
pkt_dts:1064 pkt_dts_time:42.56 off:0 off_time:0
muxer <- type:audio pkt_pts:1064 pkt_pts_time:42.56 pkt_dts:1064
pkt_dts_time:42.56 size:7680
demuxer -> ist_index:1 type:audio next_dts:42560000 next_dts_time:42.56
next_pts:42560000 next_pts_time:42.56 pkt_pts:1065 pkt_pts_time:42.6
pkt_dts:1065 pkt_dts_time:42.6 off:0 off_time:0
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1065 pkt_pts_time:42.6
pkt_dts:1065 pkt_dts_time:42.6 off:0 off_time:0
muxer <- type:audio pkt_pts:1065 pkt_pts_time:42.6 pkt_dts:1065
pkt_dts_time:42.6 size:7680
demuxer -> ist_index:1 type:audio next_dts:42600000 next_dts_time:42.6
next_pts:42600000 next_pts_time:42.6 pkt_pts:1066 pkt_pts_time:42.64
pkt_dts:1066 pkt_dts_time:42.64 off:0 off_time:0
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1066 pkt_pts_time:42.64
pkt_dts:1066 pkt_dts_time:42.64 off:0 off_time:0
muxer <- type:audio pkt_pts:1066 pkt_pts_time:42.64 pkt_dts:1066
pkt_dts_time:42.64 size:7680
demuxer -> ist_index:1 type:audio next_dts:42640000 next_dts_time:42.64
next_pts:42640000 next_pts_time:42.64 pkt_pts:1067 pkt_pts_time:42.68
pkt_dts:1067 pkt_dts_time:42.68 off:0 off_time:0
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1067 pkt_pts_time:42.68
pkt_dts:1067 pkt_dts_time:42.68 off:0 off_time:0
muxer <- type:audio pkt_pts:1067 pkt_pts_time:42.68 pkt_dts:1067
pkt_dts_time:42.68 size:7680
demuxer -> ist_index:1 type:audio next_dts:42680000 next_dts_time:42.68
next_pts:42680000 next_pts_time:42.68 pkt_pts:1068 pkt_pts_time:42.72
pkt_dts:1068 pkt_dts_time:42.72 off:0 off_time:0
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1068 pkt_pts_time:42.72
pkt_dts:1068 pkt_dts_time:42.72 off:0 off_time:0

I've tried just excluding that audio stream, but the remaining audio is
still weird

 ffmpeg -i input.avi  -map 0 -map -0:a:1 -c copy output.avi
ffmpeg version N-55863-g9f38fac053-static https://johnvansickle.com/ffmpeg/
 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 8 (Debian 8.3.0-6)
  configuration: --enable-gpl --enable-version3 --enable-static
--disable-debug --disable-ffplay --disable-indev=sndio
--disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom
--enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-librubberband --enable-libsoxr
--enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
--enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi
--enable-libzimg
  libavutil      56. 64.100 / 56. 64.100
  libavcodec     58.119.100 / 58.119.100
  libavformat    58. 65.101 / 58. 65.101
  libavdevice    58. 11.103 / 58. 11.103
  libavfilter     7.100.100 /  7.100.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
[avi @ 0x599e300] Switching to NI mode, due to poor interleaving
Input #0, avi, from 'input.avi':
  Duration: 00:07:00.00, start: 0.000000, bitrate: 28806 kb/s
    Stream #0:0: Video: dvvideo, yuv420p, 720x576 [SAR 16:15 DAR 4:3],
25000 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
    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
File 'output.avi' already exists. Overwrite? [y/N] y
Output #0, avi, to 'output.avi':
  Metadata:
    ISFT            : Lavf58.65.101
    Stream #0:0: Video: dvvideo (dvsd / 0x64737664), yuv420p, 720x576 [SAR
16:15 DAR 4:3], q=2-31, 25000 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 32000 Hz,
stereo, s16, 1024 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=10500 fps=8868 q=-1.0 Lsize= 1553212kB time=00:07:00.00
bitrate=30295.0kbits/s speed= 355x
video:1476562kB audio:76090kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead: 0.036054%

Any advice would be welcome, thanks.
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