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• ffprobe reports different format depending on file extension

  27 octobre 2015, par John Reichert

  I am using ffprobe on Windows to identify files. The command/parameters I use are :

  ffprobe -show_entries format=format_name -v quiet filename

  I noticed with certain image files ffprobe reports a different format if the file name has (or does not have) and extension. For example, for a BMP image file named XXX

  ffprobe -show_entries format=format_name -v quiet XXX

  reports

  [FORMAT]
format_name=mp3
[/FORMAT]

  But, if I rename the file to XXX.BMP and rerun the command

  ffprobe -show_entries format=format_name -v quiet XXX.BMP

  reports

  [FORMAT]
format_name=image2
[/FORMAT]

  Is this behavior expected ? Has anyone else encountered this issue ? I am using the most recent version of ffmpeg.

• Realtime http ogg stream with ffmpeg latency

  4 septembre 2013, par MastErAldo

  I'm using the Kinect as an infrared camera and through a named pipe i send each captured frame to FFMPEG which encodes them in an ogg/theora stream that is sent to an http server.
  The server than broadcasts the video to every connected user. The users use the Cortado applet to play the video.

  The video plays fine, but the latency keeps on increasing continuously. So if at the beginning the latency is about 100ms, in a minute it reaches up to 10 seconds.

  However the playback is smooth, the bandwidth is enough and the encoding machine and the server CPU is not overloaded, so I don't understand what can make the latency increase.

  This is the ffmpeg command line that we use :

  ffmpeg.exe -f segment -f rawvideo -pix_fmt rgb565 -s 640x480 -i \\.\pipe\CAH_STREAM_PIPE -fflags nobuffer -r 60 -b:v 128k -bufsize 1024k -s 240x160 -codec:v libtheora -qscale:v 1 -f ogg http://localhost/video_stream.ogg

  Am I missing some parameters or setting some of them to wrong values ? Or the problem is elsewhere ?

  BTW : I'm not sure about using nobuffer and bufsize at the same time

  Thanks

• Streaming audio from C++ [closed]

  27 août 2022, par Oliver Hawker

  I have a C++ audio application that generates audio, which I intend to stream over a network using RTSP and FFmpeg (or it's libs directly). Fundamentally, the audio is not coming from a file or device but is generated internally by the application.

  


  So far, I have looked at the following approaches :

  


  


  1. Spin up an FFmpeg process inside my application using FFmpeg-kit, and pipe data in to it using a named pipe
— Written in objective C, so I'd need a wrapper around it to call from C++.
— Feels like an awful lot of overhead for something so simple, I am dubious that it's necessary.

     


  


  2. Leverage libavformat etc. directly.
— I can imagine that the pipeline would look like a regular encode pipeline using raw frames, which I can sort-of see how it would work, but I'm struggling to mix this example with something that also transmits over RTSP.

     


  


  3. Make something myself that transmits raw samples over UDP
— Bandwidth-heavy and probably unnecessary

     


  


  


  Does anybody know of a tutorial, starting point, reference implementation or other project I could look at for some guidance ? Any solution I have come up with seems ridiculously heavyweight for something so simple.

  


  Thanks all.