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• record mediasoup RTP stream using FFmpeg for Firefox

  30 juillet 2024, par Hadi Aghandeh

  I am trying to record WebRTC stream using mediasoup. I could record successfully on chrome and safari 13/14/15. However on Firefox the does not work.

  


  Client side code is a vue js component which gets rtp-compabilities using socket.io and create producers after the server creates the transports. This works good on chrome and safari.

  


  const { connect , createLocalTracks } = require('twilio-video');
const SocketClient = require("socket.io-client");
const SocketPromise = require("socket.io-promise").default;
const MediasoupClient = require("mediasoup-client");

export default {
    data() {
        return {
            errors: [],
            isReady: false,
            isRecording: false,
            loading: false,
            sapio: {
                token: null,
                connectionId: 0
            },
            server: {
                host: 'https://rtc.test',
                ws: '/server',
                socket: null,
            },
            peer: {},
        }
    },
    mounted() {
        this.init();
    },
    methods: {
        async init() {
            await this.startCamera();

            if (this.takeId) {
                await this.recordBySapioServer();
            }
        },
        startCamera() {
            return new Promise( (resolve, reject) => {
                if (window.videoMediaStreamObject) {
                    this.setVideoElementStream(window.videoMediaStreamObject);
                    resolve();
                } else {
                    // Get user media as required
                    try {
                        this.localeStream = navigator.mediaDevices.getUserMedia({
                            audio: true,
                            video: true,
                        }).then((stream) => {
                            this.setVideoElementStream(stream);
                            resolve();
                        })
                    } catch (err) {
                        console.error(err);
                        reject();
                    }
                }
            })
        },
        setVideoElementStream(stream) {
            this.localStream = stream;
            this.$refs.video.srcObject = stream;
            this.$refs.video.muted = true;
            this.$refs.video.play().then((video) => {
                this.isStreaming = true;
                this.height = this.$refs.video.videoHeight;
                this.width = this.$refs.video.videoWidth;
            });
        },
        // first thing we need is connecting to websocket
        connectToSocket() {
            const serverUrl = this.server.host;
            console.log("Connect with sapio rtc server:", serverUrl);

            const socket = SocketClient(serverUrl, {
                path:  this.server.ws,
                transports: ["websocket"],
            });
            this.socket = socket;

            socket.on("connect", () => {
                console.log("WebSocket connected");
                // we ask for rtp-capabilities from server to send to us
                socket.emit('send-rtp-capabilities');
            });

            socket.on("error", (err) => {
                this.loading = true;
                console.error("WebSocket error:", err);
            });

            socket.on("router-rtp-capabilities", async (msg) => {
                const { routerRtpCapabilities, sessionId, externalId } = msg;
                console.log('[rtpCapabilities:%o]', routerRtpCapabilities);
                this.routerRtpCapabilities = routerRtpCapabilities;

                try {
                    const device = new MediasoupClient.Device();
                    // Load the mediasoup device with the router rtp capabilities gotten from the server
                    await device.load({ routerRtpCapabilities });

                    this.peer.sessionId = sessionId;
                    this.peer.externalId = externalId;
                    this.peer.device = device;

                    this.createTransport();
                } catch (error) {
                    console.error('failed to init device [error:%o]', error);
                    socket.disconnect();
                }
            });

            socket.on("create-transport", async (msg) => {
                console.log('handleCreateTransportRequest() [data:%o]', msg);

                try {
                    // Create the local mediasoup send transport
                    this.peer.sendTransport = await this.peer.device.createSendTransport(msg);
                    console.log('send transport created [id:%s]', this.peer.sendTransport.id);

                    // Set the transport listeners and get the users media stream
                    this.handleSendTransportListeners();
                    this.setTracks();
                    this.loading = false;
                } catch (error) {
                    console.error('failed to create transport [error:%o]', error);
                    socket.disconnect();
                }
            });

            socket.on("connect-transport", async (msg) => {
                console.log('handleTransportConnectRequest()');
                try {
                    const action = this.connectTransport;

                    if (!action) {
                        throw new Error('transport-connect action was not found');
                    }

                    await action(msg);
                } catch (error) {
                    console.error('ailed [error:%o]', error);
                }
            });

            socket.on("produce", async (msg) => {
                console.log('handleProduceRequest()');
                try {
                    if (!this.produce) {
                        throw new Error('produce action was not found');
                    }
                    await this.produce(msg);
                } catch (error) {
                    console.error('failed [error:%o]', error);
                }
            });

            socket.on("recording", async (msg) => {
                this.isRecording = true;
            });

            socket.on("recording-error", async (msg) => {
                this.isRecording = false;
                console.error(msg);
            });

            socket.on("recording-closed", async (msg) => {
                this.isRecording = false;
                console.warn(msg)
            });

        },
        createTransport() {
            console.log('createTransport()');

            if (!this.peer || !this.peer.device.loaded) {
                throw new Error('Peer or device is not initialized');
            }

            // First we must create the mediasoup transport on the server side
            this.socket.emit('create-transport',{
                sessionId: this.peer.sessionId
            });
        },
        handleSendTransportListeners() {
            this.peer.sendTransport.on('connect', this.handleTransportConnectEvent);
            this.peer.sendTransport.on('produce', this.handleTransportProduceEvent);
            this.peer.sendTransport.on('connectionstatechange', connectionState => {
                console.log('send transport connection state change [state:%s]', connectionState);
            });
        },
        handleTransportConnectEvent({ dtlsParameters }, callback, errback) {
            console.log('handleTransportConnectEvent()');
            try {
                this.connectTransport = (msg) => {
                    console.log('connect-transport action');
                    callback();
                    this.connectTransport = null;
                };

                this.socket.emit('connect-transport',{
                    sessionId: this.peer.sessionId,
                    transportId: this.peer.sendTransport.id,
                    dtlsParameters
                });

            } catch (error) {
                console.error('handleTransportConnectEvent() failed [error:%o]', error);
                errback(error);
            }
        },
        handleTransportProduceEvent({ kind, rtpParameters }, callback, errback)  {
            console.log('handleTransportProduceEvent()');
            try {
                this.produce = jsonMessage => {
                    console.log('handleTransportProduceEvent callback [data:%o]', jsonMessage);
                    callback({ id: jsonMessage.id });
                    this.produce = null;
                };

                this.socket.emit('produce', {
                    sessionId: this.peer.sessionId,
                    transportId: this.peer.sendTransport.id,
                    kind,
                    rtpParameters
                });
            } catch (error) {
                console.error('handleTransportProduceEvent() failed [error:%o]', error);
                errback(error);
            }
        },
        async recordBySapioServer() {
            this.loading = true;
            this.connectToSocket();
        },
        async setTracks() {
            // Start mediasoup-client's WebRTC producers
            const audioTrack = this.localStream.getAudioTracks()[0];
            this.peer.audioProducer = await this.peer.sendTransport.produce({
                track: audioTrack,
                codecOptions :
                    {
                        opusStereo : 1,
                        opusDtx    : 1
                    }
            });


            let encodings;
            let codec;
            const codecOptions = {videoGoogleStartBitrate : 1000};

            codec = this.peer.device.rtpCapabilities.codecs.find((c) => c.kind.toLowerCase() === 'video');
            if (codec.mimeType.toLowerCase() === 'video/vp9') {
                encodings = { scalabilityMode: 'S3T3_KEY' };
            } else {
                encodings = [
                    { scaleResolutionDownBy: 4, maxBitrate: 500000 },
                    { scaleResolutionDownBy: 2, maxBitrate: 1000000 },
                    { scaleResolutionDownBy: 1, maxBitrate: 5000000 }
                ];
            }
            const videoTrack = this.localStream.getVideoTracks()[0];
            this.peer.videoProducer =await this.peer.sendTransport.produce({
                track: videoTrack,
                encodings,
                codecOptions,
                codec
            });

        },
        startRecording() {
            this.Q.answer.recordingId = this.peer.externalId;
            this.socket.emit("start-record", {
                sessionId: this.peer.sessionId
            });
        },
        stopRecording() {
            this.socket.emit("stop-record" , {
                sessionId: this.peer.sessionId
            });
        },
    },

}





  


  console.log of my ffmpeg process :

  


  // sdp string
[sdpString:v=0
  o=- 0 0 IN IP4 127.0.0.1
  s=FFmpeg
  c=IN IP4 127.0.0.1
  t=0 0
  m=video 25549 RTP/AVP 101 
  a=rtpmap:101 VP8/90000
  a=sendonly
  m=audio 26934 RTP/AVP 100 
  a=rtpmap:100 opus/48000/2
  a=sendonly
  ]

// ffmpeg args
commandArgs:[
  '-loglevel',
  'debug',
  '-protocol_whitelist',
  'pipe,udp,rtp',
  '-fflags',
  '+genpts',
  '-f',
  'sdp',
  '-i',
  'pipe:0',
  '-map',
  '0:v:0',
  '-c:v',
  'copy',
  '-map',
  '0:a:0',
  '-strict',
  '-2',
  '-c:a',
  'copy',
  '-f',
  'webm',
  '-flags',
  '+global_header',
  '-y',
  'storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm',
  [length]: 26
]
// ffmpeg log
ffmpeg::process::data [data:'ffmpeg version n4.4']
ffmpeg::process::data [data:' Copyright (c) 2000-2021 the FFmpeg developers']
ffmpeg::process::data [data:'\n']
ffmpeg::process::data [data:'  built with gcc 11.1.0 (GCC)\n']
ffmpeg::process::data [data:'  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-shared --enable-version3\n']
ffmpeg::process::data [data:'  libavutil      56. 70.100 / 56. 70.100\n' +
  '  libavcodec     58.134.100 / 58.134.100\n' +
  '  libavformat    58. 76.100 / 58. 76.100\n' +
  '  libavdevice    58. 13.100 / 58. 13.100\n' +
  '  libavfilter     7.110.100 /  7.110.100\n' +
  '  libswscale      5.  9.100 /  5.  9.100\n' +
  '  libswresample   3.  9.100 /  3.  9.100\n' +
  '  libpostproc    55.  9.100 / 55.  9.100\n' +
  'Splitting the commandline.\n' +
  "Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.\n" +
  "Reading option '-protocol_whitelist' ..."]
ffmpeg::process::data [data:" matched as AVOption 'protocol_whitelist' with argument 'pipe,udp,rtp'.\n" +
  "Reading option '-fflags' ..."]
ffmpeg::process::data [data:" matched as AVOption 'fflags' with argument '+genpts'.\n" +
  "Reading option '-f' ... matched as option 'f' (force format) with argument 'sdp'.\n" +
  "Reading option '-i' ... matched as input url with argument 'pipe:0'.\n" +
  "Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:v:0'.\n" +
  "Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.\n" +
  "Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:a:0'.\n" +
  "Reading option '-strict' ...Routing option strict to both codec and muxer layer\n" +
  " matched as AVOption 'strict' with argument '-2'.\n" +
  "Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'.\n" +
  "Reading option '-f' ... matched as option 'f' (force format) with argument 'webm'.\n" +
  "Reading option '-flags' ... matched as AVOption 'flags' with argument '+global_header'.\n" +
  "Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.\n" +
  "Reading option 'storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm' ... matched as output url.\n" +
  'Finished splitting the commandline.\n' +
  'Parsing a group of options: global .\n' +
  'Applying option loglevel (set logging level) with argument debug.\n' +
  'Applying option y (overwrite output files) with argument 1.\n' +
  'Successfully parsed a group of options.\n' +
  'Parsing a group of options: input url pipe:0.\n' +
  'Applying option f (force format) with argument sdp.\n' +
  'Successfully parsed a group of options.\n' +
  'Opening an input file: pipe:0.\n' +
  "[sdp @ 0x55604dc58400] Opening 'pipe:0' for reading\n" +
  '[sdp @ 0x55604dc58400] video codec set to: vp8\n' +
  '[sdp @ 0x55604dc58400] audio codec set to: opus\n' +
  '[sdp @ 0x55604dc58400] audio samplerate set to: 48000\n' +
  '[sdp @ 0x55604dc58400] audio channels set to: 2\n' +
  '[udp @ 0x55604dc6c500] end receive buffer size reported is 425984\n' +
  '[udp @ 0x55604dc6c7c0] end receive buffer size reported is 425984\n' +
  '[sdp @ 0x55604dc58400] setting jitter buffer size to 500\n' +
  '[udp @ 0x55604dc6d900] end receive buffer size reported is 425984\n' +
  '[udp @ 0x55604dc6d2c0] end receive buffer size reported is 425984\n' +
  '[sdp @ 0x55604dc58400] setting jitter buffer size to 500\n']
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] Before avformat_find_stream_info() pos: 210 bytes read:210 seeks:0 nb_streams:2\n']
  **mediasoup:Consumer resume() +1s**
  **mediasoup:Channel request() [method:consumer.resume, id:12] +1s**
  **mediasoup:Channel request succeeded [method:consumer.resume, id:12] +0ms**
  **mediasoup:Consumer resume() +1ms**
  **mediasoup:Channel request() [method:consumer.resume, id:13] +0ms**
  **mediasoup:Channel request succeeded [method:consumer.resume, id:13] +0ms**
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] Could not find codec parameters for stream 0 (Video: vp8, 1 reference frame, yuv420p): unspecified size\n' +
  "Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options\n"]
ffmpeg::process::data [data:'[sdp @ 0x55604dc58400] After avformat_find_stream_info() pos: 210 bytes read:210 seeks:0 frames:0\n' +
  "Input #0, sdp, from 'pipe:0':\n" +
  '  Metadata:\n' +
  '    title           : FFmpeg\n' +
  '  Duration: N/A, bitrate: N/A\n' +
  '  Stream #0:0, 0, 1/90000: Video: vp8, 1 reference frame, yuv420p, 90k tbr, 90k tbn, 90k tbc\n' +
  '  Stream #0:1, 0, 1/48000: Audio: opus, 48000 Hz, stereo, fltp\n' +
  'Successfully opened the file.\n' +
  'Parsing a group of options: output url storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm.\n' +
  'Applying option map (set input stream mapping) with argument 0:v:0.\n' +
  'Applying option c:v (codec name) with argument copy.\n' +
  'Applying option map (set input stream mapping) with argument 0:a:0.\n' +
  'Applying option c:a (codec name) with argument copy.\n' +
  'Applying option f (force format) with argument webm.\n' +
  'Successfully parsed a group of options.\n' +
  'Opening an output file: storage/recordings/26e63cb3-4f81-499e-941a-c0bb7f7f52ce.webm.\n' +
  "[file @ 0x55604dce5bc0] Setting default whitelist 'file,crypto,data'\n"]
ffmpeg::process::data [data:'Successfully opened the file.\n' +
  '[webm @ 0x55604dce0fc0] dimensions not set\n' +
  'Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument\n' +
  'Error initializing output stream 0:1 -- \n' +
  'Stream mapping:\n' +
  '  Stream #0:0 -> #0:0 (copy)\n' +
  '  Stream #0:1 -> #0:1 (copy)\n' +
  '    Last message repeated 1 times\n' +
  '[AVIOContext @ 0x55604dc6dcc0] Statistics: 0 seeks, 0 writeouts\n' +
  '[AVIOContext @ 0x55604dc69380] Statistics: 210 bytes read, 0 seeks\n']
ffmpeg::process::close



  


  FFmpeg says dimensions not  set and Could not write header for output file when I use Firefox. This might be enough for understanding the problem, but if you need more information you can read how server side is performing.
Server-Side in summary can be something like this :
lets say we initialized worker and router at run time using following functions.

  


      // Start the mediasoup workers
module.exports.initializeWorkers = async () => {
  const { logLevel, logTags, rtcMinPort, rtcMaxPort } = config.worker;

  console.log('initializeWorkers() creating %d mediasoup workers', config.numWorkers);

  for (let i = 0; i < config.numWorkers; ++i) {
    const worker = await mediasoup.createWorker({
      logLevel, logTags, rtcMinPort, rtcMaxPort
    });

    worker.once('died', () => {
      console.error('worker::died worker has died exiting in 2 seconds... [pid:%d]', worker.pid);
      setTimeout(() => process.exit(1), 2000);
    });

    workers.push(worker);
  }
};


  


  module.exports.createRouter = async () => {
  const worker = getNextWorker();

  console.log('createRouter() creating new router [worker.pid:%d]', worker.pid);

  console.log(`config.router.mediaCodecs:${JSON.stringify(config.router.mediaCodecs)}`)

  return await worker.createRouter({ mediaCodecs: config.router.mediaCodecs });
};


  


  We pass router.rtpCompatibilities to the client. clients get the rtpCompatibilities and create a device and loads it. after that a transport must be created at server side.

  


      const handleCreateTransportRequest = async (jsonMessage) => {

  const transport = await createTransport('webRtc', router);

  var peer;
  try {peer = peers.get(jsonMessage.sessionId);}
  catch{console.log('peer not found')}
  
  peer.addTransport(transport);

  peer.socket.emit('create-transport',{
    id: transport.id,
    iceParameters: transport.iceParameters,
    iceCandidates: transport.iceCandidates,
    dtlsParameters: transport.dtlsParameters
  });
};


  


  Then after the client side also created the transport we listen to connect event an at the time of event, we request the server to create connection.

  


  const handleTransportConnectRequest = async (jsonMessage) => {
  var peer;
  try {peer = peers.get(jsonMessage.sessionId);}
  catch{console.log('peer not found')}

  if (!peer) {
    throw new Error(`Peer with id ${jsonMessage.sessionId} was not found`);
  }

  const transport = peer.getTransport(jsonMessage.transportId);

  if (!transport) {
    throw new Error(`Transport with id ${jsonMessage.transportId} was not found`);
  }

  await transport.connect({ dtlsParameters: jsonMessage.dtlsParameters });
  console.log('handleTransportConnectRequest() transport connected');
  peer.socket.emit('connect-transport');
};


  


  Similar thing happen on produce event.

  


  const handleProduceRequest = async (jsonMessage) => {
  console.log('handleProduceRequest [data:%o]', jsonMessage);

  var peer;
  try {peer = peers.get(jsonMessage.sessionId);}
  catch{console.log('peer not found')}

  if (!peer) {
    throw new Error(`Peer with id ${jsonMessage.sessionId} was not found`);
  }

  const transport = peer.getTransport(jsonMessage.transportId);

  if (!transport) {
    throw new Error(`Transport with id ${jsonMessage.transportId} was not found`);
  }

  const producer = await transport.produce({
    kind: jsonMessage.kind,
    rtpParameters: jsonMessage.rtpParameters
  });

  peer.addProducer(producer);

  console.log('handleProducerRequest() new producer added [id:%s, kind:%s]', producer.id, producer.kind);

  peer.socket.emit('produce',{
    id: producer.id,
    kind: producer.kind
  });
};


  


  For Recording, first I create plain transports for audio and video producers.

  


  const rtpTransport = router.createPlainTransport(config.plainRtpTransport);


  


  then rtp transport must be connected to ports :

  


    await rtpTransport.connect({
    ip: '127.0.0.1',
    port: remoteRtpPort,
    rtcpPort: remoteRtcpPort
  });


  


  Then the consumer must also be created.

  


    const rtpConsumer = await rtpTransport.consume({
    producerId: producer.id,
    rtpCapabilities,
    paused: true
  });


  


  After that we can start recording using following code :

  


   this._rtpParameters = args;
    this._process = undefined;
    this._observer = new EventEmitter();
    this._peer = args.peer;

    this._sdpString = createSdpText(this._rtpParameters);
    this._sdpStream = convertStringToStream(this._sdpString);
    // create dir
    const dir = process.env.REOCRDING_PATH ?? 'storage/recordings';
    if (!fs.existsSync(dir)) shelljs.mkdir('-p', dir);
  
    this._extension = 'webm';
    // create file path
    this._path = `${dir}/${args.peer.sessionId}.${this._extension}`
    let loop = 0;
    while(fs.existsSync(this._path)) {
      this._path = `${dir}/${args.peer.sessionId}-${++loop}.${this._extension}`
    }

this._recordingnModel = await Recording.findOne({sessionIds: { $in: [this._peer.sessionId] }})
    this._recordingnModel.files.push(this._path);
    this._recordingnModel.save();

let proc  = ffmpeg(this._sdpStream)
    .inputOptions([
      '-protocol_whitelist','pipe,udp,rtp',
      '-f','sdp',
    ])
    .format(this._extension)
    .output(this._path)
    .size('720x?')
    .on('start', ()=>{
      this._peer.socket.emit('recording');
    })
    .on('end', ()=>{
      let path = this._path.replace('storage/recordings/', '');
      this._peer.socket.emit('recording-closed', {
        url: `${process.env.APP_URL}/recording/file/${path}`
      });
    });

    proc.run();
    this._process =  proc;
  }



  


• Non-RSA TLS1.2 Packet decryption

  7 février 2017, par Joseph Wahba

  I am trying to decrypt a pcap file. This pcap file contains a capture of an HLS encrypted video stream. The pcap contains TLSv1.2 packets.

  Below are some information from the pcap file

  Server Hello message Cipher Suite :

  TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384.

  EC Diffie-Hellman server Params : pubkey (1)

  The Certificate Status message :

  Signature Hash Algorithm Hash : SHA256

  Signature Hash Algorithm Signature : ECDSA

  Client Key Exchange Message

  EC Diffie-Hellman server Params : pubkey (2)

  I tried to follow this Wireshark SSL decryption tutorial. But it seems that it works only for RSA encryptions.
  I have been researching for a while and found this discussion. I am quoting an extract from this discussion :

  There is an important parameter to mind : decryption of a passively
  recorded session (with a copy of the server private key) works only if
  the key exchange was of type RSA or static DH ; with "DHE" and "ECDHE"
  cipher suites, you won’t be able to decrypt such a session, even with
  knowledge of the server private key. In that case, you will need
  either the negotiated "master secret", or to use the server private
  key to actively intercept the connection

  It’s note worthy that I have the client private key. In my case, the client is FFmpeg video streamer (FFplay). I had a look also on the TLS v1.2 RFC.

  My question :

  Is it possible to do a decryption in this scenario ? If yes, what do I need to have to do so ?

  Is the decryption done using the client’s private key or using the pre_shared_master (i.e. Diffie-Hellman) ?

• avformat : add vapoursynth wrapper

  27 avril 2018, par wm4

  avformat : add vapoursynth wrapper
  This can "demux" .vpy files. Autodetection of .vpy scripts is
  
  intentionally not done, because it would be a major security issue. You
  
  need to force the format, for example with "-f vapoursynth" for the
  
  FFmpeg CLI tools.
  Some minor code copied from other LGPL parts of FFmpeg.
  I did not find a good way to test a few of the more obscure VS features,
  
  like VFR nodes, compat pixel formats, or nodes with dynamic size/format
  
  changes. These can be easily implemented on demand.
  

  • [DH] configure
  • [DH] libavformat/Makefile
  • [DH] libavformat/allformats.c
  • [DH] libavformat/vapoursynth.c
  • [DH] libavformat/version.h