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Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
List of compatible distributions
26 avril 2011, parThe table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)
Sur d’autres sites (7742)
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Applescript to batch convert videos with ffmpeg
13 mai 2022, par Thad_SuperspermI'm trying to add padding to a large amount of videos in their folders. I created an app with AppleScriptEditor so I can drag and drop files and they're automatically converted. I found a script on the web, I edited it with the ffmpeg command I need, but it won't work because it wants to overwrite the source file.


on open argv
 set paths to ""
 repeat with f in argv
 set paths to paths & quoted form of POSIX path of f & " "
 end repeat
 tell application "Terminal"
 do script "for f in " & paths & "; do ffmpeg -i \"$f\" -vf pad=\"9/8*iw:ih:(ow-iw)/2:0:color=black\" \"$f\"; done"
 activate
 end tell
end open


Note that I want to keep the filename, filetype and put the new file next to the old one but just add an underscore at the end of the new file, before the extension ; e.g. : file.ext. > file_.ext


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Feed output of one filter to the input of one filter multiple times with ffmpeg
27 août 2021, par kentcdoddsI have the following ffmpeg commands to create a podcast episode :


# remove all silence at start and end of the audio files
ffmpeg -i call.mp3 -af silenceremove=1:0:-50dB call1.mp3
ffmpeg -i response.mp3 -af silenceremove=1:0:-50dB response1.mp3

# remove silence longer than 1 second anywhere within the audio files
ffmpeg -i call1.mp3 -af silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB call2.mp3
ffmpeg -i response1.mp3 -af silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB response2.mp3

# normalize audio files
ffmpeg -i call2.mp3 -af loudnorm=I=-16:LRA=11:TP=0.0 call3.mp3
ffmpeg -i response2.mp3 -af loudnorm=I=-16:LRA=11:TP=0.0 response3.mp3

# cross fade audio files with intro/interstitial/outro
ffmpeg -i intro.mp3 -i call3.mp3 -i interstitial.mp3 -i response3.mp3 -i outro.mp3
 -filter_complex "[0][1]acrossfade=d=1:c2=nofade[a01];
 [a01][2]acrossfade=d=1:c1=nofade[a02];
 [a02][3]acrossfade=d=1:c2=nofade[a03];
 [a03][4]acrossfade=d=1:c1=nofade"
 output.mp3


While this "works" fine, I can't help but feel like it would be more efficient to do this all in one ffmpeg command. Based on what I found online this should be possible, but I don't understand the syntax well enough to know how to make it work. Here's what I tried :


ffmpeg -i intro.mp3 -i call.mp3 -i interstitial.mp3 -i response.mp3 -i outro.mp3
 -af [1]silenceremove=1:0:-50dB[trimmedCall]
 -af [3]silenceremove=1:0:-50dB[trimmedResponse]
 -af [trimmedCall]silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB[noSilenceCall]
 -af [trimmedResponse]silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB[noSilenceResponse]
 -af [noSilenceCall]loudnorm=I=-16:LRA=11:TP=0.0[call]
 -af [noSilenceResponse]loudnorm=I=-16:LRA=11:TP=0.0[response]
 -filter_complex "[0][call]acrossfade=d=1:c2=nofade[a01];
 [a01][2]acrossfade=d=1:c1=nofade[a02];
 [a02][response]acrossfade=d=1:c2=nofade[a03];
 [a03][4]acrossfade=d=1:c1=nofade"
 output.mp3


But I have a feeling I have a fundamental misunderstanding of this because I got this error which I don't understand :


Stream specifier 'call' in filtergraph description 
[0][call]acrossfade=d=1:c2=nofade[a01];
[a01][2]acrossfade=d=1:c1=nofade[a02];
[a02][response]acrossfade=d=1:c2=nofade[a03];
[a03][4]acrossfade=d=1:c1=nofade
 matches no streams.


For added context, I'm running all these commands through @ffmpeg/ffmpeg so that last command actually looks like this (in JavaScript) :


await ffmpeg.run(
 '-i', 'intro.mp3',
 '-i', 'call.mp3',
 '-i', 'interstitial.mp3',
 '-i', 'response.mp3',
 '-i', 'outro.mp3',
 '-af', '[1]silenceremove=1:0:-50dB[trimmedCall]',
 '-af', '[3]silenceremove=1:0:-50dB[trimmedResponse]',
 '-af', '[trimmedCall]silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB[noSilenceCall]',
 '-af', '[trimmedResponse]silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB[noSilenceResponse]',
 '-af', '[noSilenceCall]loudnorm=I=-16:LRA=11:TP=0.0[call]',
 '-af', '[noSilenceResponse]loudnorm=I=-16:LRA=11:TP=0.0[response]',
 '-filter_complex', `
[0][call]acrossfade=d=1:c2=nofade[a01];
[a01][2]acrossfade=d=1:c1=nofade[a02];
[a02][response]acrossfade=d=1:c2=nofade[a03];
[a03][4]acrossfade=d=1:c1=nofade
 `,
 'output.mp3',
)


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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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