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#3 The Safest Place
16 octobre 2011, par
Mis à jour : Février 2013
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ED-ME-5 1-DVD
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Revolution of Open-source and film making towards open film making
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Autres articles (48)
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Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
L’utiliser, en parler, le critiquer
10 avril 2011La première attitude à adopter est d’en parler, soit directement avec les personnes impliquées dans son développement, soit autour de vous pour convaincre de nouvelles personnes à l’utiliser.
Plus la communauté sera nombreuse et plus les évolutions seront rapides ...
Une liste de discussion est disponible pour tout échange entre utilisateurs. -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...)
Sur d’autres sites (10394)
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HTML5 video player doesn't work on iPad/iPhone
16 avril 2015, par Avrohom YisroelNOTE This turned out to be a simulator problem, not a video encoding problem, see my edit lower down...
I’m creating a web site for a local college, and they want to be able to add short videos that people can view online. I’ve spent quite a bit of time trying to work out how to get the videos to play on iDevices, but have failed.
I’m using Video.js (http://www.videojs.com), and have HTML that looks like this...
<video class="video-js vjs-default-skin" controls="controls" preload="auto" width="640" height="352" poster="/Content/Images/logobg.png" data-setup="{}">
<source src="/Content/Videos/video.m4v" type="video/mp4">
<source src="/Content/Videos/video.webm" type="video/webm">
<p class="vjs-no-js">
To view this video please enable JavaScript, and consider upgrading to a web browser that <a href="http://videojs.com/html5-video-support/" target="_blank">supports HTML5 video</a>
</p>
</source></source></video>This works fine on desktop browsers, where it uses the m4v file. However, if I load the page on an iDevice, the video player says "no compatible source was found for this video", which sounds like it doesn’t like either the m4v or the webm file.
I created the webm file using instructions found at http://daniemon.com/blog/how-to-convert-videos-to-webm-with-ffmpeg/. I tried creating a .mov file using the accepted answer at iPad Doesn’t Render H.264 Video with HTML5, but this gave the same error.
Anyone any ideas how I can support iDevices ? Please don’t blind me with science, I’m a real newbie with all this video stuff, and need simple instructions !
Edit The problem I was having was when trying to view the site on a mobile simulator. When I uploaded the site to a real server, and tried it on an iPad, it worked fine. So, if anyone is having a similar problem, first use something like Handbrake to encode the videos, as it seems to do it fine, then make sure you’re testing on a real mobile device, not a simulator !
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How to simultaneously capture mic, stream it to RTSP server and play it on iPhone's speaker ?
24 août 2021, par Norbert TowiańskiI want to capture sound from mic, stream it to RTSP server and play it simultaneously on iPhone's speaker after getting samples from RTSP server. I mean such kind of loop. I use FFMPEGKit and I want to use MobileVLCKit, but unfortunately microphone is off when I start play stream.
I think I've done first step (capturing from microphone and send OutputStream to RTSP server) :


@IBAction func transmitBtnPressed(_ sender: Any) {
 ffmpeg_transmit()
}

@IBAction func recordBtnPressed(_ sender: Any) {
 switch recordingState {
 case .idle:
 recordingState = .start
 startRecording()
 recordBtn.setTitle("Started", for: .normal)
 let urlToFile = URL(fileURLWithPath: outPipePath!)
 outputStream = OutputStream(url: urlToFile, append: false)
 outputStream!.open()
 case .capturing:
 recordingState = .end
 stopRecording()
 recordBtn.setTitle("End", for: .normal)
 default:
 break
 }
}

override func viewDidLoad() {
 super.viewDidLoad()
 outPipePath = FFmpegKitConfig.registerNewFFmpegPipe()
 self.setup()
}

override func viewDidAppear(_ animated: Bool) {
 super.viewDidAppear(animated)
 setUpAuthStatus()
}

func setUpAuthStatus() {
 if AVCaptureDevice.authorizationStatus(for: AVMediaType.audio) != .authorized {
 AVCaptureDevice.requestAccess(for: AVMediaType.audio, completionHandler: { (authorized) in
 DispatchQueue.main.async {
 if authorized {
 self.setup()
 }
 }
 })
 }
}

func setup() {
 self.session.sessionPreset = AVCaptureSession.Preset.high
 
 self.recordingURL = URL(fileURLWithPath: "\(NSTemporaryDirectory() as String)/file.m4a")
 if self.fileManager.isDeletableFile(atPath: self.recordingURL!.path) {
 _ = try? self.fileManager.removeItem(atPath: self.recordingURL!.path)
 }
 
 self.assetWriter = try? AVAssetWriter(outputURL: self.recordingURL!,
 fileType: AVFileType.m4a)
 self.assetWriter!.movieFragmentInterval = CMTime.invalid
 self.assetWriter!.shouldOptimizeForNetworkUse = true
 
 let audioSettings = [
 AVFormatIDKey: kAudioFormatLinearPCM,
 AVSampleRateKey: 48000.0,
 AVNumberOfChannelsKey: 1,
 AVLinearPCMIsFloatKey: false,
 AVLinearPCMBitDepthKey: 16,
 AVLinearPCMIsBigEndianKey: false,
 AVLinearPCMIsNonInterleaved: false,
 
 ] as [String : Any]
 
 
 self.audioInput = AVAssetWriterInput(mediaType: AVMediaType.audio,
 outputSettings: audioSettings)
 
 self.audioInput?.expectsMediaDataInRealTime = true
 
 if self.assetWriter!.canAdd(self.audioInput!) {
 self.assetWriter?.add(self.audioInput!)
 }
 
 self.session.startRunning()
 
 DispatchQueue.main.async {
 self.session.beginConfiguration()
 
 self.session.commitConfiguration()
 
 let audioDevice = AVCaptureDevice.default(for: AVMediaType.audio)
 let audioIn = try? AVCaptureDeviceInput(device: audioDevice!)
 
 if self.session.canAddInput(audioIn!) {
 self.session.addInput(audioIn!)
 }
 
 if self.session.canAddOutput(self.audioOutput) {
 self.session.addOutput(self.audioOutput)
 }
 
 self.audioConnection = self.audioOutput.connection(with: AVMediaType.audio)
 }
}

func startRecording() {
 if self.assetWriter?.startWriting() != true {
 print("error: \(self.assetWriter?.error.debugDescription ?? "")")
 }
 
 self.audioOutput.setSampleBufferDelegate(self, queue: self.recordingQueue)
}

func stopRecording() {
 self.audioOutput.setSampleBufferDelegate(nil, queue: nil)
 
 self.assetWriter?.finishWriting {
 print("Saved in folder \(self.recordingURL!)")
 }
}
func captureOutput(_ captureOutput: AVCaptureOutput, didOutput
 sampleBuffer: CMSampleBuffer, from connection: AVCaptureConnection) {
 
 if !self.isRecordingSessionStarted {
 let presentationTime = CMSampleBufferGetPresentationTimeStamp(sampleBuffer)
 self.assetWriter?.startSession(atSourceTime: presentationTime)
 self.isRecordingSessionStarted = true
 recordingState = .capturing
 }
 
 var blockBuffer: CMBlockBuffer?
 var audioBufferList: AudioBufferList = AudioBufferList.init()
 
 CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, bufferListSizeNeededOut: nil, bufferListOut: &audioBufferList, bufferListSize: MemoryLayout<audiobufferlist>.size, blockBufferAllocator: nil, blockBufferMemoryAllocator: nil, flags: kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, blockBufferOut: &blockBuffer)
 let buffers = UnsafeMutableAudioBufferListPointer(&audioBufferList)
 
 for buffer in buffers {
 let u8ptr = buffer.mData!.assumingMemoryBound(to: UInt8.self)
 let output = outputStream!.write(u8ptr, maxLength: Int(buffer.mDataByteSize))
 
 if (output == -1) {
 let error = outputStream?.streamError
 print("\(#file) > \(#function) > Error on outputStream: \(error!.localizedDescription)")
 }
 else {
 print("\(#file) > \(#function) > Data sent")
 }
 }
}

func ffmpeg_transmit() {
 
 let cmd1: String = "-f s16le -ar 48000 -ac 1 -i "
 let cmd2: String = " -probesize 32 -analyzeduration 0 -c:a libopus -application lowdelay -ac 1 -ar 48000 -f rtsp -rtsp_transport udp rtsp://localhost:18556/mystream"
 let cmd = cmd1 + outPipePath! + cmd2
 
 print(cmd)
 
 ffmpegSession = FFmpegKit.executeAsync(cmd, withExecuteCallback: { ffmpegSession in
 
 let state = ffmpegSession?.getState()
 let returnCode = ffmpegSession?.getReturnCode()
 if let returnCode = returnCode, let get = ffmpegSession?.getFailStackTrace() {
 print("FFmpeg process exited with state \(String(describing: FFmpegKitConfig.sessionState(toString: state!))) and rc \(returnCode).\(get)")
 }
 }, withLogCallback: { log in
 
 }, withStatisticsCallback: { statistics in
 
 })
}
</audiobufferlist>


I want to use MobileVLCKit in that way :


func startStream(){
 guard let url = URL(string: "rtsp://localhost:18556/mystream") else {return}
 audioPlayer!.media = VLCMedia(url: url)

 audioPlayer!.media.addOption( "-vv")
 audioPlayer!.media.addOption( "--network-caching=10000")

 audioPlayer!.delegate = self
 audioPlayer!.audio.volume = 100

 audioPlayer!.play()

}



Could you give me some hints how to implement that ?


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passing additional values to s3 event notification for lambda consumption
8 septembre 2017, par user1790300I have to write code in react-native that allows a user to upload videos to amazon s3 to be transcoded for consumption by various devices. For the processing after the upload occurs ; I am reviewing two approaches :
1) I can use Lambda with ffmpeg to handle the transcoding immediately after the uploading occurs (my fear here would be the amount of time required to transcode the videos and the effect on pricing if it takes a considerable amount of time).
2) I can have s3 pass an sns message to a rest api after the created event occurs and the rest api generate a rabbitmq message that will be processed by worker that will perform the transcoding using ffmpeg.
Option 1) seems to be the preferable option based on a completion time perspective. How concerned should I be with using 1) considering how long video transcoding might take as opposed to option 2) ?
Also, regardless, I need a way to pass additional parameters to lambda or along the sns messaging that would allow me to somehow associate the user who uploaded the video with their account. Is there a way to pass additional text-based values to s3 to pass along to lambda or along sns when the upload completes, as a caveat I plan to upload the video directly to s3 using the rest layer(found this answer here : http://docs.aws.amazon.com/AmazonS3/latest/API/RESTObjectPUT.html#RESTObjectPUT-responses-examples) ?