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• Add a side data type for audio service type.

  14 novembre 2014, par Anton Khirnov

  Add a side data type for audio service type.
  Currently, audio service type is a field in AVCodecContext. However,
  
  side data is more appropriate for this kind of information.
  

  • [DBH] doc/APIchanges
  • [DBH] libavcodec/avcodec.h
  • [DBH] libavcodec/utils.c
  • [DBH] libavcodec/version.h
  • [DBH] libavfilter/af_ashowinfo.c
  • [DBH] libavutil/frame.h
  • [DBH] libavutil/version.h

• Is it possible to use something other than a circular buffer when using ffmpeg to pass RTSP video to a WebRTC client ?

  3 novembre 2024, par Doctico

  I have a Node application that uses ffmpeg to receive the RTSP stream of an ip camera (h264), and then pass send it using node-webrtc to a remote client. However, in my current implementation, ffmpeg only outputs 8192 byte chunks, which must be buffered in order to create the full frame that my current node-webrtc flow expects. This circular buffer results in a 1-2 second delay when the client is viewing the video. Is there any way to pass the stream through node-webrtc as the chunks come in, or at least extract complete frames so that circular buffering is not necessary ?

  


  So far, I have tried this, which works but has a 1-2 second delay, and even higher delay with higher resolution cameras :

  


      async startStream() {

        const rtspUrl = 'rtsp://my-rtsp-url';

        const videoSource = new wrtc.nonstandard.RTCVideoSource();
        const videoTrack = videoSource.createTrack();

        const width = 640;
        const height = 480;
        const frameSize = width * height * 1.5; // YUV420p format

        //circular buffer:
        let frameBuffer = Buffer.alloc(0);
        const frameStream = new Writable({
            write: (chunk, encoding, callback) => {
                frameBuffer = Buffer.concat([frameBuffer, chunk]);

                while (frameBuffer.length >= frameSize) {
                    const frame = frameBuffer.slice(0, frameSize);
                    frameBuffer = frameBuffer.slice(frameSize);

                    videoSource.onFrame({
                        width: width,
                        height: height,
                        data: new Uint8ClampedArray(frame)
                    });
                }
                callback();
            }
        });

        const ffmpegProcess = ffmpeg(rtspUrl)
            .inputOptions([
                `-fflags nobuffer`,
                `-flags low_delay`,
                `-rtsp_transport tcp`,
                `-strict experimental`,
                `-analyzeduration 0`,
                `-threads 0`,
                `-hwaccel auto`
            ])
            .outputOptions([
                `-f rawvideo`,
                `-c:v rawvideo`,
                '-b:v', streamId === 1 ? '2000k' : '1000k',
                '-bf', '0',
                `-s ${width}x${height}`,
                `-pix_fmt yuv420p`,
                `-tune zerolatency`
            ])
            .on('start', (cmd) => console.log('FFmpeg started:', cmd))
            .on('error', error => {
                console.error('FFmpeg error:', error);
                this.stopStream();
            })
            .on('end', () => {
                console.log('FFmpeg stream ended');
                this.stopStream();
            })

        ffmpegProcess
            .pipe(frameStream);

        return videoTrack;
    }


  


• How to use ffmpeg.wasm client side

  9 avril 2022, par Yordan Radev

  I am looking to use ffmpeg.wasm in a serverless Vue app, but have no idea how to integrate it. Ideally I want to use it to stitch together Canvas HTMLElement into a video output that the user can download. If anyone can make a JSFiddle, Codepen or whatever other medium demo, that would be great. Working off of this example I am not sure how to add a Canvas element as a frame to a file and then createURL for the resulting file.