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• avformat/rtsp : Send mode=record instead of mode=receive in Transport header

  15 janvier 2024, par Paul Orlyk

  avformat/rtsp : Send mode=record instead of mode=receive in Transport header
  Fixes server compatibility issues with rtspclientsink GStreamer plugin.
  >From specification :
  
  RFC 7826 "Real-Time Streaming Protocol Version 2.0" (https://datatracker.ietf.org/doc/html/rfc7826), section 18.54 :
  
     mode : The mode parameter indicates the methods to be supported for
  
           this session.  The currently defined valid value is "PLAY".  If
  
           not provided, the default is "PLAY".  The "RECORD" value was
  
           defined in RFC 2326 ; in this specification, it is unspecified
  
           but reserved.  RECORD and other values may be specified in the
  
           future.
  
  RFC 2326 "Real Time Streaming Protocol (RTSP)" (https://datatracker.ietf.org/doc/html/rfc2326), section 12.39 :
  
     mode :
  
            The mode parameter indicates the methods to be supported for
  
            this session. Valid values are PLAY and RECORD. If not
  
            provided, the default is PLAY.
  mode=receive was always like this, from the initial commit 'a8ad6ffa rtsp : Add listen mode'.
  For comparison, Wowza was used to push RTSP stream to. Both GStreamer and FFmpeg had no issues.
  
  Here is the capture of Wowza responding to SETUP request :
  
  200 OK
  
  CSeq : 3
  
  Server : Wowza Streaming Engine 4.8.26+4 build20231212155517
  
  Cache-Control : no-cache
  
  Expires : Mon, 15 Jan 2024 19:40:31 GMT
  
  Transport : RTP/AVP/UDP ;unicast ;client_port=11640-11641 ;mode=record ;source=172.17.0.2 ;server_port=6976-6977
  
  Date : Mon, 15 Jan 2024 19:40:31 GMT
  
  Session : 1401457689 ;timeout=60
  Test setup :
  
      Server : ffmpeg -loglevel trace -y -rtsp_flags listen -i rtsp ://0.0.0.0:30800/live.stream t.mp4
  
      FFmpeg client : ffmpeg -re -i "Big Buck Bunny - FULL HD 30FPS.mp4" -c:v libx264 -f rtsp rtsp ://127.0.0.1:30800/live.stream
  
      GStreamer client : gst-launch-1.0 videotestsrc is-live=true pattern=smpte ! queue ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=60/1 ! timeoverlay font-desc="Sans, 84" halignment=center valignment=center ! queue ! videoconvert ! tee name=t t. ! x264enc bitrate=9000 pass=cbr speed-preset=ultrafast byte-stream=false key-int-max=15 threads=1 ! video/x-h264,profile=baseline ! queue ! rsink. audiotestsrc ! voaacenc ! queue ! rsink. t. ! queue ! autovideosink rtspclientsink name=rsink location=rtsp ://localhost:30800/live.stream
  Test results :
  
  modified FFmpeg client -> stock server : ok
  
  stock FFmpeg client    -> modified server : ok
  
  modified FFmpeg client -> modified server : ok
  
  GStreamer client       -> modified server : ok
  Signed-off-by : Paul Orlyk <paul.orlyk@gmail.com>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/rtspdec.c

• ffmpeg mp4 to mp3 conversion generates 1 second audio files only [closed]

  24 janvier 2024, par doesnotcompile

  I am having the following problem : I am having trouble converting an mp4 to an mp3.

  


  I have mp4 audio recordings that I need to convert in to mp3 audio files. The mp4 files have been recorded on iOS.

  


  I am currently doing this by using ffmpeg in python like so :

  


  subprocess.run(
            [
                "ffmpeg",
                "-loglevel", "error",
                "-i", ,
                "-q:a", "0",
                "-map", "a",
                
            ]
        )


  


  This runs with no error outputs.

  


  However, the resulting mp3 is only 1 second long, even though the input mp4 is around 5 seconds long.

  


  The mp4 is playable in Quicktime and Apple Music.

  


  ffprobe output :

  


  > ffprobe -v error -show_format input.mp4
[FORMAT]
filename=input.mp4
nb_streams=1
nb_programs=0
format_name=mov,mp4,m4a,3gp,3g2,mj2
format_long_name=QuickTime / MOV
start_time=0.000000
duration=5.120000
size=119636
bit_rate=186931
probe_score=100
TAG:major_brand=iso5
TAG:minor_version=1
TAG:compatible_brands=isomiso5hlsf
TAG:creation_time=2024-01-24T12:47:24.000000Z
[/FORMAT]


  


  No errors shown, and clearly 5 seconds long audio.

  


  Has anyone else experienced similar issues before ?

  


  Happy about any pointers !

  


• libavformat : add RCWT closed caption muxex

  14 janvier 2024, par Marth64

  libavformat : add RCWT closed caption muxex
  Signed-off-by : Marth64 <marth64@proxyid.net>
  Raw Captions With Time (RCWT) is a format native to ccextractor, a commonly
  
  used open source tool for processing 608/708 closed caption (CC) sources.
  
  It can be used to archive the original, raw CC bitstream and to produce
  
  a source file file for later CC processing or conversion. As a result,
  
  it also allows for interopability with ccextractor for processing CC data
  
  extracted via ffmpeg. The format is simple to parse and can be used
  
  to retain all lines and variants of CC.
  A free specification of RCWT can be found here :
  
  https://github.com/CCExtractor/ccextractor/blob/master/docs/BINARY_FILE_FORMAT.TXT
  
  This muxer implements the specification as of 01/05/2024, which has
  
  been stable and unchanged for 10 years as of this writing.
  This muxer will have some nuances from the way that ccextractor muxes RCWT.
  
  No compatibility issues when processing the output with ccextractor
  
  have been observed as a result of this so far, but mileage may vary
  
  and outputs will not be a bit-exact match.
  Specifically, the differences are :
  
  (1)  This muxer will identify as "FF" as the writing program identifier, so
  
  as to be honest about the output's origin.
  (2)  ffmpeg's MPEG-1/2, H264, HEVC, etc. decoders extract closed captioning
  
  data differently than ccextractor from embedded SEI/user data.
  
  For example, DVD captioning bytes will be translated to ATSC A53 format.
  
  This allows ffmpeg to handle 608/708 in a consistant way downstream.
  
  This is a lossless conversion and the meaningful data is retained.
  (3)  This muxer will not alter the extracted data except to remove invalid
  
  packets in between valid CC blocks. On the other hand, ccextractor
  
  will by default remove mid-stream padding, and add padding at the end
  
  of the stream (in order to convey the end time of the source video).
  

  • [DH] Changelog
  • [DH] doc/muxers.texi
  • [DH] libavformat/Makefile
  • [DH] libavformat/allformats.c
  • [DH] libavformat/rcwtenc.c
  • [DH] tests/fate/subtitles.mak