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Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
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record mediasoup RTP stream using FFmpeg for Firefox
30 juillet 2024, par Hadi AghandehI 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 setandCould not write header for output filewhen 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.rtpCompatibilitiesto the client. clients get thertpCompatibilitiesand 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;
 }



-
Merged Video Contains Inverted Clips After First Video Ends
3 février, par Nikunj AgrawalI am working on a Flutter application that merges multiple videos using
ffmpeg_kit_flutter. However, after merging, I notice that the second video (and any subsequent ones) appear inverted or rotated in the final output.

Issue Details :


- 

- The first video appears normal.
- The videos can be recorded using both front and back cameras.
- The second (and later) videos are flipped or rotated upside down.
- This happens after merging using
ffmpeg_kit_flutter.










Question :
How can I correctly merge multiple videos in Flutter without rotation issues ? Is there a way to normalize video orientation before merging using
ffmpeg_kit_flutter?

Any help would be appreciated ! 🚀


Code :


import 'dart:io';
import 'dart:math';

import 'package:camera/camera.dart';
import 'package:ffmpeg_kit_flutter/ffmpeg_kit.dart';
import 'package:ffmpeg_kit_flutter/return_code.dart';
import 'package:flutter/material.dart';
import 'package:path_provider/path_provider.dart';
import 'package:permission_handler/permission_handler.dart';
import 'package:record/record.dart';
import 'package:videotest/video_player.dart';

class MergeVideoRecording extends StatefulWidget {
 const MergeVideoRecording({super.key});

 @override
 State<mergevideorecording> createState() => _MergeVideoRecordingState();
}

class _MergeVideoRecordingState extends State<mergevideorecording> {
 CameraController? _cameraController;
 final AudioRecorder _audioRecorder = AudioRecorder();

 bool _isRecording = false;
 String? _videoPath;
 String? _audioPath;
 List<cameradescription> _cameras = [];
 int _currentCameraIndex = 0;
 final List<string> _recordedVideos = [];

 @override
 Widget build(BuildContext context) {
 return Scaffold(
 body: Column(
 mainAxisAlignment: MainAxisAlignment.center,
 children: [
 _cameraController != null && _cameraController!.value.isInitialized
 ? SizedBox(
 width: MediaQuery.of(context).size.width * 0.4,
 height: MediaQuery.of(context).size.height * 0.3,
 child: Stack(
 children: [
 ClipRRect(
 borderRadius: BorderRadius.circular(16),
 child: SizedBox(
 width: MediaQuery.of(context).size.width * 0.4,
 height: MediaQuery.of(context).size.height * 0.3,
 child: Transform(
 alignment: Alignment.center,
 transform:
 _cameras[_currentCameraIndex].lensDirection ==
 CameraLensDirection.front
 ? Matrix4.rotationY(pi)
 : Matrix4.identity(),
 child: CameraPreview(_cameraController!),
 ),
 ),
 ),
 Align(
 alignment: Alignment.topRight,
 child: InkWell(
 onTap: _switchCamera,
 child: const Padding(
 padding: EdgeInsets.all(8.0),
 child: CircleAvatar(
 radius: 18,
 backgroundColor: Colors.white,
 child: Icon(
 Icons.flip_camera_android,
 color: Colors.black,
 ),
 ),
 ),
 ),
 ),
 ],
 ),
 )
 : const CircularProgressIndicator(),
 const SizedBox(height: 16),
 Row(
 mainAxisAlignment: MainAxisAlignment.center,
 children: [
 FloatingActionButton(
 heroTag: 'record_button',
 onPressed: _toggleRecording,
 child: Icon(
 _isRecording ? Icons.stop : Icons.video_camera_back,
 ),
 ),
 const SizedBox(
 width: 50,
 ),
 FloatingActionButton(
 heroTag: 'merge_button',
 onPressed: _mergeVideos,
 child: const Icon(
 Icons.merge,
 ),
 ),
 ],
 ),
 if (!_isRecording)
 ListView.builder(
 shrinkWrap: true,
 itemCount: _recordedVideos.length,
 itemBuilder: (context, index) => InkWell(
 onTap: () {
 Navigator.push(
 context,
 MaterialPageRoute(
 builder: (context) => VideoPlayerScreen(
 videoPath: _recordedVideos[index],
 ),
 ),
 );
 },
 child: ListTile(
 title: Text('Video ${index + 1}'),
 subtitle: Text('Path ${_recordedVideos[index]}'),
 trailing: const Icon(Icons.play_arrow),
 ),
 ),
 ),
 ],
 ),
 );
 }

 @override
 void dispose() {
 _cameraController?.dispose();
 _audioRecorder.dispose();
 super.dispose();
 }

 @override
 void initState() {
 super.initState();
 _initializeDevices();
 }

 Future<void> _initializeCameraController(CameraDescription camera) async {
 _cameraController = CameraController(
 camera,
 ResolutionPreset.high,
 enableAudio: true,
 imageFormatGroup: ImageFormatGroup.yuv420, // Add this line
 );

 await _cameraController!.initialize();
 await _cameraController!.setExposureMode(ExposureMode.auto);
 await _cameraController!.setFocusMode(FocusMode.auto);
 setState(() {});
 }

 Future<void> _initializeDevices() async {
 final cameraStatus = await Permission.camera.request();
 final micStatus = await Permission.microphone.request();

 if (!cameraStatus.isGranted || !micStatus.isGranted) {
 _showError('Camera and microphone permissions required');
 return;
 }

 _cameras = await availableCameras();
 if (_cameras.isNotEmpty) {
 final frontCameraIndex = _cameras.indexWhere(
 (camera) => camera.lensDirection == CameraLensDirection.front);
 _currentCameraIndex = frontCameraIndex != -1 ? frontCameraIndex : 0;
 await _initializeCameraController(_cameras[_currentCameraIndex]);
 }
 }

 // Merge video
 Future<void> _mergeVideos() async {
 if (_recordedVideos.isEmpty) {
 _showError('No videos to merge');
 return;
 }

 try {
 // Debug logging
 print('Starting merge process');
 print('Number of videos to merge: ${_recordedVideos.length}');
 for (var i = 0; i < _recordedVideos.length; i++) {
 final file = File(_recordedVideos[i]);
 final exists = await file.exists();
 final size = exists ? await file.length() : 0;
 print('Video $i: ${_recordedVideos[i]}');
 print('Exists: $exists, Size: $size bytes');
 }

 final Directory appDir = await getApplicationDocumentsDirectory();
 final String outputPath =
 '${appDir.path}/merged_${DateTime.now().millisecondsSinceEpoch}.mp4';
 final String listFilePath = '${appDir.path}/list.txt';

 print('Output path: $outputPath');
 print('List file path: $listFilePath');

 // Create and verify list file
 final listFile = File(listFilePath);
 final fileContent = _recordedVideos
 .map((path) => "file '${path.replaceAll("'", "'\\''")}'")
 .join('\n');
 await listFile.writeAsString(fileContent);

 print('List file content:');
 print(await listFile.readAsString());

 // Simpler FFmpeg command for testing
 final command = '''
 -f concat
 -safe 0
 -i "$listFilePath"
 -c copy
 -y
 "$outputPath"
 '''
 .trim()
 .replaceAll('\n', ' ');

 print('Executing FFmpeg command: $command');

 final session = await FFmpegKit.execute(command);
 final returnCode = await session.getReturnCode();
 final logs = await session.getAllLogsAsString();
 final failStackTrace = await session.getFailStackTrace();

 print('FFmpeg return code: ${returnCode?.getValue() ?? "null"}');
 print('FFmpeg logs: $logs');
 if (failStackTrace != null) {
 print('FFmpeg fail stack trace: $failStackTrace');
 }

 if (ReturnCode.isSuccess(returnCode)) {
 final outputFile = File(outputPath);
 final outputExists = await outputFile.exists();
 final outputSize = outputExists ? await outputFile.length() : 0;

 print('Output file exists: $outputExists');
 print('Output file size: $outputSize bytes');

 if (outputExists && outputSize > 0) {
 setState(() => _recordedVideos.add(outputPath));
 _showSuccess('Videos merged successfully');
 } else {
 _showError('Merged file is empty or not created');
 }
 } else {
 _showError('Failed to merge videos. Check logs for details.');
 }

 // Clean up
 try {
 await listFile.delete();
 print('List file cleaned up successfully');
 } catch (e) {
 print('Failed to delete list file: $e');
 }
 } catch (e, s) {
 print('Error during merge: $e');
 print('Stack trace: $s');
 _showError('Error merging videos: ${e.toString()}');
 }
 }

 void _showError(String message) {
 ScaffoldMessenger.of(context).showSnackBar(
 SnackBar(content: Text(message), backgroundColor: Colors.red),
 );
 }

 void _showSuccess(String message) {
 ScaffoldMessenger.of(context).showSnackBar(
 SnackBar(content: Text(message), backgroundColor: Colors.green),
 );
 }

 Future<void> _startAudioRecording() async {
 try {
 final Directory tempDir = await getTemporaryDirectory();
 final audioPath = '${tempDir.path}/recording.wav';
 await _audioRecorder.start(const RecordConfig(), path: audioPath);
 setState(() => _isRecording = true);
 } catch (e) {
 _showError('Recording start error: $e');
 }
 }

 Future<void> _startVideoRecording() async {
 try {
 await _cameraController!.startVideoRecording();
 setState(() => _isRecording = true);
 } catch (e) {
 _showError('Recording start error: $e');
 }
 }

 Future<void> _stopAndSaveAudioRecording() async {
 _audioPath = await _audioRecorder.stop();
 if (_audioPath != null) {
 final Directory appDir = await getApplicationDocumentsDirectory();
 final timestamp = DateTime.now().millisecondsSinceEpoch;
 final String audioFileName = 'audio_$timestamp.wav';
 await File(_audioPath!).copy('${appDir.path}/$audioFileName');
 _showSuccess('Saved: $audioFileName');
 }
 }

 Future<void> _stopAndSaveVideoRecording() async {
 try {
 final video = await _cameraController!.stopVideoRecording();
 _videoPath = video.path;

 if (_videoPath != null) {
 final Directory appDir = await getApplicationDocumentsDirectory();
 final timestamp = DateTime.now().millisecondsSinceEpoch;
 final String videoFileName = 'video_$timestamp.mp4';
 final savedVideoPath = '${appDir.path}/$videoFileName';
 await File(_videoPath!).copy(savedVideoPath);

 setState(() {
 _recordedVideos.add(savedVideoPath);
 _isRecording = false;
 });

 _showSuccess('Saved: $videoFileName');
 }
 } catch (e) {
 _showError('Recording stop error: $e');
 }
 }

 Future<void> _switchCamera() async {
 if (_cameras.length <= 1) return;

 if (_isRecording) {
 await _stopAndSaveVideoRecording();
 _currentCameraIndex = (_currentCameraIndex + 1) % _cameras.length;
 await _initializeCameraController(_cameras[_currentCameraIndex]);
 await _startVideoRecording();
 } else {
 _currentCameraIndex = (_currentCameraIndex + 1) % _cameras.length;
 await _initializeCameraController(_cameras[_currentCameraIndex]);
 }
 }

 Future<void> _toggleRecording() async {
 if (_cameraController == null) return;

 if (_isRecording) {
 await _stopAndSaveVideoRecording();
 await _stopAndSaveAudioRecording();
 } else {
 _startVideoRecording();
 _startAudioRecording();
 setState(() => _recordedVideos.clear());
 }
 }
}
</void></void></void></void></void></void></void></void></void></string></cameradescription></mergevideorecording></mergevideorecording>

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Revision 33365 : ajout d’un modèle pour afficher les tags d’un auteur (necessite le plugin ...
28 novembre 2009, par brunobergot@… — Logajout d’un modèle pour afficher les tags d’un auteur (necessite le plugin nuage)
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