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Sur d’autres sites (11577)
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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 ?


-
Revision 30079 : servait pour débuguer ... donc plus nécessaire
22 juillet 2009, par kent1@… — Logservait pour débuguer ... donc plus nécessaire
-
Ffmpeg - How can I create HLS multiple language streams, in multiple qualities ?
28 avril 2022, par Daniel EllisPreface


I'm working on converting videos from 4k to multiple qualities with multiple languages but am having issues with the multiple languages overlaying, sometimes losing quality and sometimes being out of sync. (this is less of a problem in the German audio, as this is voice over anyhow)


We as a team are complete noobs in terms of Video / Audio + HLS — I'm a front end developer who has no experience of this so apologies if my question is poorly phrased



Videos


I have the video in a 4k format and have removed the original sound as I have English and German audio files that need to be overlayed. I am then taking these files and throwing them together into a .ts file like this :


$ ffmpeg -i ep03-ns-4k.mp4 -i nkit-ep3-de-output.m4a -i nkit-ep3-en-output.m4a \
> -thread 0 -muxdelay 0 -y \
> -map 0:v -map 1 -map 2 -movflags +faststart -refs 1 \
> -vcodec libx264 -acodec aac -profile:v baseline -level 30 -ar 44100 -ab 64k -f mpegts out.ts 


This outputs a 4k
out.tsvideo, with both audio tracks playing.

The hard part


This is where I'm finding it tricky, I now need to convert this single file into multiple quality levels (480, 720, 1080, 1920) and I attempt this with the following command :


ffmpeg -hide_banner -y -i out.ts \
-crf 20 -sc_threshold 0 -g 48 -keyint_min 48 -ar 48000 \
-map 0:v:0 -map 0:v:0 -map 0:v:0 -map 0:v:0 \
-c:v:0 h264 -profile:v:0 main -filter:v:0 "scale=w=848:h=480:force_original_aspect_ratio=decrease" -b:v:0 1400k -maxrate:v:0 1498k -bufsize:v:0 2100k \
-c:v:1 h264 -profile:v:1 main -filter:v:1 "scale=w=1280:h=720:force_original_aspect_ratio=decrease" -b:v:1 2800k -maxrate:v:1 2996k -bufsize:v:1 4200k \
-c:v:2 h264 -profile:v:2 main -filter:v:2 "scale=w=1920:h=1080:force_original_aspect_ratio=decrease" -b:v:2 5600k -maxrate:v:2 5992k -bufsize:v:2 8400k \
-c:v:3 h264 -profile:v:3 main -filter:v:3 "scale=w=3840:h=1920:force_original_aspect_ratio=decrease" -b:v:3 11200k -maxrate:v:3 11984k -bufsize:v:3 16800k \
-var_stream_map "v:0 v:1 v:2 v:3" \
-master_pl_name master.m3u8 \
-f hls -hls_time 4 -hls_playlist_type vod -hls_list_size 0 \
-hls_segment_filename "%v/episode-%03d.ts" "%v/episode.m3u8"


This creates the required qualities, but I'm now at a loss of how this might work with the audio


Audio


For the audio I run this command :


ffmpeg -i out.ts -threads 0 -muxdelay 0 -y -map 0:a:0 -codec copy -f segment -segment_time 4 -segment_list_size 0 -segment_list audio-de/audio-de.m3u8 -segment_format mpegts audio-de/audio-de_%d.aac
ffmpeg -i out.ts -threads 0 -muxdelay 0 -y -map 0:a:1 -codec copy -f segment -segment_time 4 -segment_list_size 0 -segment_list audio-en/audio-en.m3u8 -segment_format mpegts audio-en/audio-en_%d.aac



This creates the required audio segments.


The question


I realise this is quite an ask, but is there anything wrong with our inputs ? Is there a way that this can be done a bit more streamlined ?


Any answers are greatly appreciated.


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