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• Extended client ownership of MediaCodec encoder output buffers for RTMP streaming

  13 février 2014, par dbro

  Background :

  I've connected Android's MediaCodec to FFmpeg for muxing a variety of formats not supported by MediaMuxer, including rtmp:// output via a .flv container. Such streaming muxers require longer, unpredictable ownership of MediaCodec's output buffers, as they may perform networking I/O on any packet processing step. For my video stream, I'm using MediaCodec configured for Surface input. To decouple muxing from encoding, I queue MediaCodec's ByteBuffer output buffers to my muxer via a Handler.

  All works splendidly if I mux the .flv output to file, rather than rtmp endpoint.

  Problem :

  When muxing to rtmp://... endpoint I notice my streaming application begins to block on calls to eglSwapBuffers(mEGLDisplay, mEncodingEGLSurface) at dequeueOutputBuffer() once I'm retaining even a few MediaCodec output buffers in my muxing queue as MediaCodec seems to be locked to only 4 output buffers.

  Any tricks to avoid copying all encoder output returned by MediaCodec#dequeueOutputBuffers and immediately calling releaseOutputBuffer(...) ?

  The full source of my project is available on Github. Specifically, see :

  • AndroidEncoder.java : Abstract Encoder class with shared behavior between Audio and Video encoders : mainly drainEncoder(). Writes data to a Muxer instance.
  • FFmpegMuxer.java : Implements Muxer
  • CameraEncoder.java. Sends camera frames to an AndroidEncoder subclass configured for Video encoding.

  Systrace

  

  Here's some systrace output streaming 720p @ 2Mbps video to Zencoder.

  Solved

  Copying then releasing the MediaCodec encoder output ByteBuffers as soon as they're available solves the issue without significantly affecting performance. I recycle the ByteBuffer copies in an ArrayDeque<bytebuffer></bytebuffer> for each muxer track, which limits the number of allocations.

• Stream microphone from client browser to remote server and pass audio in real time to ffmpeg to combine with a second video source

  4 mai 2021, par fakeguybrushthreepwood

  As a beginner at working with these kinds of real-time streaming services, I've spent hours trying to work out how this is possible, but can't seem to work out I'd precisely go about it.

  


  I'm prototyping a personal basic web app that does the following :

  


  


  1. In a web browser, the web application has a button that says 'Stream Microphone' - when pressed it streams the audio from the user's microphone (the user obviously has to consent to give permission to send their microphone audio) through to the server which I was presuming would be running node.js (no specific reason at this point, just thought this is how I'd go about doing it).

     


  


  2. The server receives the audio close enough to real-time somehow (not sure how I'd do this).

     


  


  3. I can then run ffmpeg on the command line and take the real-time audio coming in real-time and add it as the sound to a video file (let's just say I'm going to play testmovie.mp4) that I want to play.

     


  


  


  I've looked at various solutions - such as maybe using WebRTC, RTP/RTSP, Piping audio into ffmpeg, Gstreamer, Kurento, Flashphoner and/or Wowza - but somehow they look overly complicated and usually seem to focus on video along with audio. I just need to work with audio.

  


• How to compare/show the difference between 2 videos in ffmpeg ?

  25 janvier 2016, par polarka

  I am a newbie at encoding. I have read and tried x264 in lossless mode (-qp 0), however I’d like to make sure that in my new video, every single pixel contains the same information as the source file (which is in YUV 420 so the loss of color conversion is avoidable, as far as I know). I want to be able to check that, because I don’t believe in that if someone just says its lossless.

  I welcome answers suggesting other codecs for lossless encoding, my only requirements for codecs are having one of the best compression rate and let me to pick different calculation times (such as the range from placebo to veryfast in x264) in order to adjust the compression level and calc time to my needs. But keep in mind that the original question is about how can I calculate the differences frame by frame of two videos and export it to a 3rd file, so I can watch it myself. I think that knowledge (if its possible and doesnt have serious limitations) will be useful for me in the future too.