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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.

  


• TypeError : parameter 'url' is missing a type annotation in callback 'play' + ffmpeg

  10 décembre 2022, par morichh

  I can't understand why Python interpreter doesn't like the fact that I use the url in the last line

  


  @client.tree.command()
async def play(interaction: Interaction, url = str):
    global voice
    channel = interaction.user.voice.channel
    voice = get(client.voice_clients, Guild=interaction.guild)
    if voice and voice.is_connected():
        await voice.move_to(channel)
        await interaction.response.send_message(f"*1*")
    else:
        voice = await channel.connect()
        await interaction.response.send_message(f"*2*")
    voice.play(discord.FFmpegPCMAudio(executable="C:/ffmpeg/bin/ffmpeg.exe", source = url))    



  


  By the way, ffmpeg also does not work.