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• Revision 37444 : Si les theora+vorbis plantent sur vorbis quand ils sont en mono, vorbis ...

  20 avril 2010, par kent1@… — Log

  Si les theora+vorbis plantent sur vorbis quand ils sont en mono, vorbis plantera aussi ... on rationalise tout cela

• Capture camera + mic and encode to h264/aac on macOS

  16 décembre 2018, par Flock Dawson

  I’m having trouble capturing and encoding audio+video on-the-fly on macOS.

  I tried two options :

  1. ffmpeg

     ffmpeg -threads 0 -f avfoundation -s 1920x1080 -framerate 25 -I 0:0 -async 441 -c:v libx264 -preset medium -pix_fmt yuv420p -crf 22 -c:a libfdk_aac -aq 95 -y

  2. gstreamer

     gst-launch-1.0 -ve avfvideosrc device-index=0 ! video/x-raw,width=1920,height=1080,framerate=25/1 ! vtenc_h264 ! queue ! mp4mux name=mux ! filesink location=out.mp4  osxaudiosrc device=0 ! audio/x-raw ! faac midside=false ! queue ! mux.

  The ffmpeg option works, but only for lower resolutions. With higher resolutions, the Mac mini (2018 gen) can’t do the heavy lifting. I think because I installed ffmpeg with brew, so it wasn’t compiled on my machine, meaning it doesn’t use the h264 hardware encoder in the Mac ?

  The gstreamer option works as well, but there’s a slight audio/video sync issue (audio is 100ms ahead of the video). I can’t seem to add delay to the GStreamer queue (it ignores it) :

  queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 min-threshold-time=100000000

  Anyone who has any experience with this ? Thanks !

• Using libav, how to force RTSP stream over TCP ?

  15 août 2023, par hamidi

  By using ffmpeg or ffplay you may specify "-rtsp_transport tcp" switch to force the rtsp stream be opened via tcp. In programming, I'm using libav to open the rtsp stream with the following code :

  


  AVDictionary *options = NULL;
av_dict_set(&options, "rtsp_transport", "tcp", 0);
int status = avformat_open_input(&mFormatContext, path.c_str(), inputFormat, &options);


  


  and I expect the same behavior. But the url 'path' is opened via udp. I may figure this out this way :

  


  I create an SDP file with libav and write the received packets to it. Then I use ffplay to process and play the sdp file. I get errors like :

  


  


  RTP : missed 1 packets

  


  


  


  error while decoding MB 75 38, bytestream -48

  


  


  If the rtsp stream has been opened via TCP, I should not miss any packets, and the second type of error, I think, is caused by losing packets too. So, it seems that the rtsp_transport option is not working.

  


  Any idea ?

  


  Update

  


  I found that av_dict_set works as expected, but the SDP file is made incorrectly (maybe). Its content is like :

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 226.1.1.15/0
t=0 0
a=tool:libavformat 56.40.101
m=video 43908 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1; sprop-parameter-sets=Z00AKpY1QPAET8s3AQEBAgA=,aO48gAA=; profile-level-id=4D002A


  


  I researched about it and found that the line beginning with 'm' should contain TCP instead of RTP/AVP, although all of this is my guess. The ffplay is reading the sdp file in the same system, so no socket or network connection exists between ffplay and the sdp file. So my guess may be incorrect. What do you think about the sdp file ? May it be incorrect ? I make it by av_sdp_create function and don't change it. I write it directly to the file.