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Concatenating videos Audio/Video out of sync : Non-monotonous TDS output
6 août 2020, par inviderI am trying to add two videos using ffmpeg and the output video/audio is out of sync (and fast forwarded). Goal is to put
intro.mp4
before the original fileclip.flv



My approach is to



- 

- Change the format of
clip.flv
toclip.mp4





ffmpeg -i clip.flv -q 0 -c copy clip.mp4




- 

- Concat
intro.mp4
withclip.mp4





ffmpeg -f concat -safe 0 -i filesToJoin.txt -c copy combinedvideo.mp4




I see this in output log for command#2



[mp4 @ 0x3ebcd60] Non-monotonous DTS in output stream 0:0; previous: 392311, current: 391925; changing to 392312. This may result in incorrect timestamps in the output file.
frame= 1566 fps=0.0 q=-1.0 Lsize= 8711kB time=00:00:48.86 bitrate=1460.2kbits/s speed= 272x 
video:7363kB audio:1294kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.619341%






Files Metadata



Original source file
clip.flv
metadata


Metadata:
 encoder : Lavf57.83.100
 Duration: 00:00:40.09, start: 0.010000, bitrate: 1632 kb/s
 Stream #0:0: Video: h264 (High), yuv420p(progressive), 1280x720, 1500 kb/s, 30 fps, 30 tbr, 1k tbn, 60 tbc
 Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 160 kb/s




Intermediary file
intro.mp4
metadata


Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: mp42mp41
 creation_time : 2020-06-02T10:36:51.000000Z
 Duration: 00:00:13.21, start: 0.000000, bitrate: 484 kb/s
 Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 130 kb/s, 30 fps, 30 tbr, 30k tbn, 60k tbc (default)
 Metadata:
 creation_time : 2020-06-02T10:36:51.000000Z
 handler_name : Alias Data Handler
 encoder : AVC Coding
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 317 kb/s (default)
 Metadata:
 creation_time : 2020-06-02T10:36:52.000000Z
 handler_name : Alias Data Handler




File
clip.mp4
metadata


Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf57.83.100
 Duration: 00:00:40.01, start: 0.000000, bitrate: 1635 kb/s
 Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720, 1500 kb/s, 30 fps, 30 tbr, 16k tbn, 60 tbc (default)
 Metadata:
 handler_name : VideoHandler
 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 160 kb/s (default)
 Metadata:
 handler_name : SoundHandler






Things I have tried



- 

- As you'll see that the both videos have same frame rate but different timescale, I have tried changing the timescale of one of the video to match the other before combining the video but with no luck. For timescale change I used this command





ffmpeg -i clip.mp4 -video_track_timescale 30000 clip_ts30000.mp4



- 

- I have seen the similar questions on SO, no luck




- Change the format of
-
ffmpeg mkv to mp4 container not changed
26 août 2020, par Rioconverting from mkv to mp4 is not working for me ffmpeg is running fine but the container is still mkv this is cmd i am running


for %%i in (*.mkv) do ffmpeg -i "%%i" -c copy -c:a aac -b:a 320k -movflags +faststart "%%i.mp4"



output after ffmpeg is done


General
Complete name : C:\laragon\www\run\730428.mp4
Format : Matroska
Format version : Version 4
Overall bit rate mode : Variable
Encoded date : UTC 2020-08-21 07:01:37
Writing application : mkvmerge v48.0.0 ('Fortress Around Your Heart') 64-bit
Writing library : libebml v1.4.0 + libmatroska v1.6.0
FileExtension_Invalid : mkv mk3d mka mks



-
Is ffmpeg broken for h.264 RTP Output ?
13 mars 2023, par GroovyDotComI used wireshark to capture the RTP stream sent with :


ffmpeg -f lavfi -i "testsrc=duration=5:size=cif:rate=25" -pix_fmt yuv420p -g 25 -bf 2 -an -c:v libx264 -f rtp rtp://127.0.0.1:1234 > play.sdp


ffmpeg -version
ffmpeg version git-2020-03-15-c467328 Copyright (c) 2000-2020 the FFmpeg developers


As can be seen in bold, RTP timestamps go forward and backward. I expect them to be the same for every packet in the frame and then only go forward by 40ms (+3600 at 90khz clock) as per the H.264/RTP spec.


Also, according to that spec, the last packet in a frame should have its marker-bit set but here almost all the packets have this bit set.


Am I doing something wrong ? Not understanding something ? Or is ffmpeg support for writing H.264 RTP simply broken ?


SSRC=0xA49C3DC9, Seq=3595, Time=3153114809

SSRC=0xA49C3DC9, Seq=3596, Time=3153114809

SSRC=0xA49C3DC9, Seq=3597, Time=3153114809

SSRC=0xA49C3DC9, Seq=3598, Time=3153114809, Mark

SSRC=0xA49C3DC9, Seq=3599, Time=3153125609, Mark

SSRC=0xA49C3DC9, Seq=3600, Time=3153118409, Mark

SSRC=0xA49C3DC9, Seq=3601, Time=3153122009, Mark

SSRC=0xA49C3DC9, Seq=3602, Time=3153136409, Mark

SSRC=0xA49C3DC9, Seq=3603, Time=3153129209, Mark

SSRC=0xA49C3DC9, Seq=3604, Time=3153132809, Mark

SSRC=0xA49C3DC9, Seq=3605, Time=3153147209, Mark

SSRC=0xA49C3DC9, Seq=3606, Time=3153140009, Mark

SSRC=0xA49C3DC9, Seq=3607, Time=3153143609, Mark

SSRC=0xA49C3DC9, Seq=3608, Time=3153158009, Mark

SSRC=0xA49C3DC9, Seq=3609, Time=3153150809, Mark

SSRC=0xA49C3DC9, Seq=3610, Time=3153154409, Mark

SSRC=0xA49C3DC9, Seq=3611, Time=3153168809, Mark

SSRC=0xA49C3DC9, Seq=3612, Time=3153161609, Mark

SSRC=0xA49C3DC9, Seq=3613, Time=3153165209, Mark

SSRC=0xA49C3DC9, Seq=3614, Time=3153179609, Mark

SSRC=0xA49C3DC9, Seq=3615, Time=3153172409, Mark

SSRC=0xA49C3DC9, Seq=3616, Time=3153176009, Mark

SSRC=0xA49C3DC9, Seq=3617, Time=3153190409, Mark

SSRC=0xA49C3DC9, Seq=3618, Time=3153183209, Mark

The RTP specification, defined in RFC 3550, states that "the timestamp reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations" (Section 5.1).