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• Ffmpeg inaccurate cutting with ts and m3u8 files despite resamping audio filter

  31 juillet 2020, par Lemon Sky

  I need to accurately seek and cut a video. Some online sources say put -ss in front or after the source. The result is the same for me. In the examples below, the start time is accurate but the duration is not accurate.

  


  ffmpeg -y -ss 00:00:05 -t 00:00:05 -i output.ts 5s.wav
ffprobe 5s.wav

Duration: 00:00:04.74

ffmpeg -y -i output.ts -ss 00:00:05 -t 00:00:05 5s.wav
ffprobe 5s.wav

Duration: 00:00:04.74


  


  Sometimes the starting point is not accurate but the duration is. This is clearly audible by cutting the ts file and cutting an uncompressed wav file, and listening to both.

  


  ffmpeg -y -i output.ts -ss 00:00:15 -t 00:00:05 5s.wav

ffmpeg -y -i output.wav -ss 00:00:15 -t 00:00:05 5s-reference.wav


  


  What fixes the starting time is if I use an m3u8 file that contains the byte offset for every keyframe AND I put the -ss option in front of the source file (if I put it after the source, the start time is inaccurate but the duration is accurate) :

  


  ffmpeg -y -ss 00:00:15 -t 00:00:05 -i output.m3u8 5s.wav


  


  This fixes the start time but the duration is at the location that I would get had I used no m3u8 file (duration is just 4.47s).

  


  It seems different timestamps are involved, and sometimes one or the other gets used.

  


  The ts file was generated by capturing a UDP stream and storing it with ffmpeg and -codec:v copy.

  


  Is ffmpeg broken, or the ts file ? How do I work around this issue or fix the ts file ? What I realize is that the video starts later than the audio, probably because the video does not start with a keyframe. Can I get ffmpeg to start the -codec:v copy after the first keyframe ? What I also notice is that using ffprobe reports "start : 1.400000". Can I force it to start at 0 ?

  


  Any hints would be appreciated.

  


  I tried both ffmpeg 4.3.1 and ffmpeg git-2020-07-24-21442a8.

  


  The output.* files were generated as follows. The statement "af aresample=async=1" should fill missing audio according to Duration of source video and subtracted audio are different. Adding this statement makes no difference in terms of accuracy or duration. The question is not a duplicate.

  


  ffmpeg -i udp://example:port ^
-af aresample=async=1 ^
-codec:v copy ^
-codec:a aac -ac 2 -ar 44100 -b:a 160k ^
-hls_time 4 -hls_flags single_file -hls_list_size 0 -hls_segment_filename output.ts -hls_segment_type mpegts output.m3u8 ^
-codec:a pcm_s16le -bitexact -ar 11025 -ac 1 output.wav


  


• Ffmpeg inaccurate cut duration with ts [duplicate]

  30 juillet 2020, par Lemon Sky

  I need to accurately seek and cut a video. Some online sources say put -ss in front or after the source. The result is the same for me. In the examples below, the start time is accurate but the duration is not accurate.

  


  ffmpeg -y -ss 00:00:05 -t 00:00:05 -i http://tyberis.com/output.ts 5s.wav
ffprobe 5s.wav

Duration: 00:00:04.74

ffmpeg -y -i http://tyberis.com/output.ts -ss 00:00:05 -t 00:00:05 5s.wav
ffprobe 5s.wav

Duration: 00:00:04.74


  


  Sometimes the starting point is not accurate but the duration is. This is clearly audible by cutting the ts file and cutting an uncompressed wav file, and listening to both.

  


  ffmpeg -y -i http://tyberis.com/output.ts -ss 00:00:15 -t 00:00:05 5s.wav

ffmpeg -y -i http://tyberis.com/output.wav -ss 00:00:15 -t 00:00:05 5s-reference.wav


  


  What fixes the starting time is if I use an m3u8 file that contains the byte offset for every keyframe AND I put the -ss option in front of the source file (if I put it after the source, the start time is inaccurate but the duration is accurate) :

  


  ffmpeg -y -ss 00:00:15 -t 00:00:05 -i http://tyberis.com/output.m3u8 5s.wav


  


  This fixes the start time but the duration is at the location that I would get had I used no m3u8 file (duration is just 4.47s).

  


  It seems different timestamps are involved, and sometimes one or the other gets used.

  


  The ts file was generated by capturing a UDP stream and storing it with ffmpeg and -codec:v copy.

  


  Is ffmpeg broken, or the ts file ? How do I work around this issue or fix the ts file ? What I realize is that the video starts later than the audio, probably because the video does not start with a keyframe. Can I get ffmpeg to start the -codec:v copy after the first keyframe ? What I also notice is that using ffprobe reports "start : 1.400000". Can I force it to start at 0 ?

  


  Any hints would be appreciated.

  


  I tried both ffmpeg 4.3.1 and ffmpeg git-2020-07-24-21442a8.

  


• unable to steam rtsp from mp4(h264) file using ffmpeg on os x : Connection refused Could not write header for output file

  28 janvier 2023, par TalG

  im ussing the following command on my macbook os high sierra to stream rtsp from mp4 file using ffmpeg :

  


  sudo ffmpeg -re -i ./Big_Buck_Bunny_1080_10s_1MB.mp4 -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f rtsp -rtsp_transport udp rtsp://127.0.0.1:8888/live


  


  but get the following error :

  


  [tcp @ 0x7fb979e22ec0] Connection to tcp://127.0.0.1:1935?timeout=0 failed: Connection refused
Could not write header for output file #0 (incorrect codec parameters ?): Connection refused
Error initializing output stream 0:0 -- 
Conversion failed!


  


  here is the whole output of the command :

  


  ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  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
  libavresample   4.  0.  0 /  4.  0.  0
  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 './Big_Buck_Bunny_1080_10s_1MB.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    title           : Big Buck Bunny, Sunflower version
    artist          : Blender Foundation 2008, Janus Bager Kristensen 2013
    composer        : Sacha Goedegebure
    encoder         : Lavf57.63.100
    comment         : Creative Commons Attribution 3.0 - http://bbb3d.renderfarming.net
    genre           : Animation
  Duration: 00:00:10.00, start: 0.000000, bitrate: 815 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 812 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
    Metadata:
      handler_name    : VideoHandler
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
Press [q] to stop, [?] for help
[libx264 @ 0x7fb97a00de00] using SAR=1/1
[libx264 @ 0x7fb97a00de00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0x7fb97a00de00] profile High, level 4.0, 4:2:0, 8-bit
[libx264 @ 0x7fb97a00de00] 264 - core 160 r3011 cde9a93 - H.264/MPEG-4 AVC codec - Copyleft 2003-2020 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x3 me=dia subme=1 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 lookahead_threads=12 sliced_threads=1 slices=12 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
[tcp @ 0x7fb979e22ec0] Connection to tcp://127.0.0.1:1935?timeout=0 failed: Connection refused
Could not write header for output file #0 (incorrect codec parameters ?): Connection refused
Error initializing output stream 0:0 -- 
Conversion failed!


  


  tried with and without sudo, tried changing rtsp ://... to http://
also tried udp but get same output..

  


  chacked that the port is not in use(8888) and different ports (1935...) but still the same.

  


  i installed ffmpeg via brew install...

  


  when i run some test server on my localhost i never have issues ussing an unused port

  


  really stuck here and any help would be amazing...thank you

  


  EDIT :
Problem was in the command i used : "rtsp ://127.0.0.1:8888/live" - but i did not have a running server capable of accepting the data from ffmpeg and redestributing it - so i had to first run such server and only after that to run ffmpeg :

  


  Servers which can receive from FFmpeg (to restream to multiple clients) include ffserver (linux only, though with cygwin it might work on windows), or ​Wowza Media Server, or ​Flash Media Server, Red5, or ​various others. Even ​VLC can pick up the stream from ffmpeg, then redistribute it, acting as a server.

  


  i used the VLC option. You can read about it here : http://trac.ffmpeg.org/wiki/StreamingGuide