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swift - 在 swift 中使用 WebRTC 进行 iOS 屏幕共享(使用 ReplayKit)

转载 作者:行者123 更新时间:2023-12-05 01:35:46 29 4
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我已经成功实现了 ReplayKit。

样本处理器.swift

class SampleHandler: RPBroadcastSampleHandler {

override func broadcastStarted(withSetupInfo setupInfo: [String : NSObject]?) {

}

override func processSampleBuffer(_ sampleBuffer: CMSampleBuffer, with sampleBufferType: RPSampleBufferType) {
switch sampleBufferType {

case RPSampleBufferType.video:
break

case RPSampleBufferType.audioApp:
break

case RPSampleBufferType.audioMic:
break
@unknown default:
return
}
}
}

问题:

  1. 如何使用 WebRTC 显示屏幕共享视频和接收器?
  2. 如果我想将屏幕共享视频保存在我的文档目录文件夹或画廊中,而不是我需要做什么?

我正在快速使用 WebRTC SDK for mobile。我的 WebRTCClient 文件。

    final class WebRTCClient: NSObject {

// The `RTCPeerConnectionFactory` is in charge of creating new RTCPeerConnection instances.
// A new RTCPeerConnection should be created every new call, but the factory is shared.
static let factory: RTCPeerConnectionFactory = {
RTCInitializeSSL()
let videoEncoderFactory = RTCDefaultVideoEncoderFactory()
let videoDecoderFactory = RTCDefaultVideoDecoderFactory()
return RTCPeerConnectionFactory(encoderFactory: videoEncoderFactory, decoderFactory: videoDecoderFactory)
}()

weak var delegate: WebRTCClientDelegate?
let peerConnection: RTCPeerConnection
private let rtcAudioSession = RTCAudioSession.sharedInstance()
private let audioQueue = DispatchQueue(label: "audio")
private let mediaConstrains = [kRTCMediaConstraintsOfferToReceiveAudio: kRTCMediaConstraintsValueTrue,
kRTCMediaConstraintsOfferToReceiveVideo: kRTCMediaConstraintsValueTrue]
private var videoCapturer: RTCVideoCapturer?
private var localVideoTrack: RTCVideoTrack?
private var remoteVideoTrack: RTCVideoTrack?
private var localDataChannel: RTCDataChannel?
private var remoteDataChannel: RTCDataChannel?

@available(*, unavailable)
override init() {
fatalError("WebRTCClient:init is unavailable")
}

required init(iceServers: [String]) {

let config = RTCConfiguration()
// config.iceServers = [RTCIceServer(urlStrings: iceServers)]

config.iceServers = [RTCIceServer(urlStrings:["//Turn server URL"],
username:"username",
credential:"password")]


// Unified plan is more superior than planB
config.sdpSemantics = .unifiedPlan

// gatherContinually will let WebRTC to listen to any network changes and send any new candidates to the other client
config.continualGatheringPolicy = .gatherContinually

let constraints = RTCMediaConstraints(mandatoryConstraints: nil,
optionalConstraints: ["DtlsSrtpKeyAgreement":kRTCMediaConstraintsValueTrue])
self.peerConnection = WebRTCClient.factory.peerConnection(with: config, constraints: constraints, delegate: nil)

super.init()
self.createMediaSenders()
self.configureAudioSession()
self.peerConnection.delegate = self
}

// MARK: Signaling
func offer(completion: @escaping (_ sdp: RTCSessionDescription) -> Void) {
let constrains = RTCMediaConstraints(mandatoryConstraints: self.mediaConstrains,
optionalConstraints: nil)
self.peerConnection.offer(for: constrains) { (sdp, error) in
guard let sdp = sdp else {
return
}

self.peerConnection.setLocalDescription(sdp, completionHandler: { (error) in
completion(sdp)
})
}
}

func answer(completion: @escaping (_ sdp: RTCSessionDescription) -> Void) {
let constrains = RTCMediaConstraints(mandatoryConstraints: self.mediaConstrains,
optionalConstraints: nil)
self.peerConnection.answer(for: constrains) { (sdp, error) in
guard let sdp = sdp else {
return
}

self.peerConnection.setLocalDescription(sdp, completionHandler: { (error) in
completion(sdp)
})
}
}

func set(remoteSdp: RTCSessionDescription, completion: @escaping (Error?) -> ()) {
self.peerConnection.setRemoteDescription(remoteSdp, completionHandler: completion)
}

func set(remoteCandidate: RTCIceCandidate) {
self.peerConnection.add(remoteCandidate)
}

// MARK: Media
func startCaptureLocalVideo(renderer: RTCVideoRenderer, position : AVCaptureDevice.Position) {
guard let capturer = self.videoCapturer as? RTCCameraVideoCapturer else {
return
}

guard
let frontCamera = (RTCCameraVideoCapturer.captureDevices().first { $0.position == position }),
// choose highest res
let format = (RTCCameraVideoCapturer.supportedFormats(for: frontCamera).sorted { (f1, f2) -> Bool in
let width1 = CMVideoFormatDescriptionGetDimensions(f1.formatDescription).width
let width2 = CMVideoFormatDescriptionGetDimensions(f2.formatDescription).width
return width1 < width2
}).last,

// choose highest fps
let fps = (format.videoSupportedFrameRateRanges.sorted { return $0.maxFrameRate < $1.maxFrameRate }.last) else {
return
}

capturer.startCapture(with: frontCamera,
format: format,
fps: Int(fps.maxFrameRate))

self.localVideoTrack?.add(renderer)
}

func startCaptureLocalTestVideo(renderer: RTCVideoRenderer, position : AVCaptureDevice.Position) {
guard (self.videoCapturer as? RTCCameraVideoCapturer) != nil else {
return
}

guard
let frontCamera = (RTCCameraVideoCapturer.captureDevices().first { $0.position == position }),
// choose highest res
let format = (RTCCameraVideoCapturer.supportedFormats(for: frontCamera).sorted { (f1, f2) -> Bool in
let width1 = CMVideoFormatDescriptionGetDimensions(f1.formatDescription).width
let width2 = CMVideoFormatDescriptionGetDimensions(f2.formatDescription).width
return width1 < width2
}).last,

// choose highest fps
let _ = (format.videoSupportedFrameRateRanges.sorted { return $0.maxFrameRate < $1.maxFrameRate }.last) else {
return
}

capturer.startCapture(with: frontCamera,
format: format,
fps: Int(fps.maxFrameRate))

self.localVideoTrack?.add(renderer)
}

func renderRemoteVideo(to renderer: RTCVideoRenderer) {
self.remoteVideoTrack?.add(renderer)
}

private func configureAudioSession() {
self.rtcAudioSession.lockForConfiguration()
do {
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
try self.rtcAudioSession.setMode(AVAudioSession.Mode.voiceChat.rawValue)
} catch let error {
debugPrint("Error changeing AVAudioSession category: \(error)")
}
self.rtcAudioSession.unlockForConfiguration()
}

private func createMediaSenders() {
let streamId = "stream"

// Audio
let audioTrack = self.createAudioTrack()
self.peerConnection.add(audioTrack, streamIds: [streamId])

// Video
let videoTrack = self.createVideoTrack()
self.localVideoTrack = videoTrack
self.peerConnection.add(videoTrack, streamIds: [streamId])
self.remoteVideoTrack = self.peerConnection.transceivers.first { $0.mediaType == .video }?.receiver.track as? RTCVideoTrack

// Data
if let dataChannel = createDataChannel() {
dataChannel.delegate = self
self.localDataChannel = dataChannel
}
}

private func createAudioTrack() -> RTCAudioTrack {
let audioConstrains = RTCMediaConstraints(mandatoryConstraints: nil, optionalConstraints: nil)
let audioSource = WebRTCClient.factory.audioSource(with: audioConstrains)
let audioTrack = WebRTCClient.factory.audioTrack(with: audioSource, trackId: "audio0")
return audioTrack
}

private func createVideoTrack() -> RTCVideoTrack {
let videoSource = WebRTCClient.factory.videoSource()

#if TARGET_OS_SIMULATOR
self.videoCapturer = RTCFileVideoCapturer(delegate: videoSource)
#else
self.videoCapturer = RTCCameraVideoCapturer(delegate: videoSource)
#endif

let videoTrack = WebRTCClient.factory.videoTrack(with: videoSource, trackId: "video0")
return videoTrack
}

// MARK: Data Channels
private func createDataChannel() -> RTCDataChannel? {
let config = RTCDataChannelConfiguration()
guard let dataChannel = self.peerConnection.dataChannel(forLabel: "WebRTCData", configuration: config) else {
debugPrint("Warning: Couldn't create data channel.")
return nil
}
return dataChannel
}

func sendData(_ data: Data) {
let buffer = RTCDataBuffer(data: data, isBinary: true)
self.remoteDataChannel?.sendData(buffer)
}
}

extension WebRTCClient: RTCPeerConnectionDelegate {

func peerConnection(_ peerConnection: RTCPeerConnection, didChange stateChanged: RTCSignalingState) {
debugPrint("peerConnection new signaling state: \(stateChanged)")
}

func peerConnection(_ peerConnection: RTCPeerConnection, didAdd stream: RTCMediaStream) {
debugPrint("peerConnection did add stream")
}

func peerConnection(_ peerConnection: RTCPeerConnection, didRemove stream: RTCMediaStream) {
debugPrint("peerConnection did remote stream")
}

func peerConnectionShouldNegotiate(_ peerConnection: RTCPeerConnection) {
debugPrint("peerConnection should negotiate")
}

func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceConnectionState) {
debugPrint("peerConnection new connection state: \(newState)")
self.delegate?.webRTCClient(self, didChangeConnectionState: newState)
}

func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceGatheringState) {
debugPrint("peerConnection new gathering state: \(newState)")
}

func peerConnection(_ peerConnection: RTCPeerConnection, didGenerate candidate: RTCIceCandidate) {
self.delegate?.webRTCClient(self, didDiscoverLocalCandidate: candidate)
}

func peerConnection(_ peerConnection: RTCPeerConnection, didRemove candidates: [RTCIceCandidate]) {
debugPrint("peerConnection did remove candidate(s)")
}

func peerConnection(_ peerConnection: RTCPeerConnection, didOpen dataChannel: RTCDataChannel) {
debugPrint("peerConnection did open data channel")
self.remoteDataChannel = dataChannel
}
}

// MARK:- Audio control
extension WebRTCClient {
func muteAudio() {
// self.setAudioEnabled(false)
swapCameraToFront()
}

func unmuteAudio() {
self.setAudioEnabled(true)
}

func muteVideo() {
self.setVideoEnabled(false)
}

func unmuteVideo() {
self.setVideoEnabled(true)
}


// Fallback to the default playing device: headphones/bluetooth/ear speaker
func speakerOff() {
self.audioQueue.async { [weak self] in
guard let self = self else {
return
}

self.rtcAudioSession.lockForConfiguration()
do {
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
try self.rtcAudioSession.overrideOutputAudioPort(.none)
} catch let error {
debugPrint("Error setting AVAudioSession category: \(error)")
}
self.rtcAudioSession.unlockForConfiguration()
}
}

// Force speaker
func speakerOn() {
self.audioQueue.async { [weak self] in
guard let self = self else {
return
}

self.rtcAudioSession.lockForConfiguration()
do {
try self.rtcAudioSession.setCategory(AVAudioSession.Category.playAndRecord.rawValue)
try self.rtcAudioSession.overrideOutputAudioPort(.speaker)
try self.rtcAudioSession.setActive(true)
} catch let error {
debugPrint("Couldn't force audio to speaker: \(error)")
}
self.rtcAudioSession.unlockForConfiguration()
}
}

private func setAudioEnabled(_ isEnabled: Bool) {

let audioTracks = self.peerConnection.transceivers.compactMap { return $0.sender.track as? RTCAudioTrack }
audioTracks.forEach { $0.isEnabled = isEnabled }
}

private func setVideoEnabled(_ isEnabled: Bool) {

let videoTracks = self.peerConnection.transceivers.compactMap { return $0.sender.track as? RTCVideoTrack }
videoTracks.forEach { $0.isEnabled = isEnabled }

}

func swapCameraToFront() {

let localStream: RTCMediaStream? = peerConnection.localStreams.first
localStream?.removeVideoTrack((localStream?.videoTracks.first)!)

let localVideoTrack: RTCVideoTrack? = self.createVideoTrack()
if localVideoTrack != nil {
localStream?.addVideoTrack(localVideoTrack!)
// delegate?.appClient(self, didReceiveLocalVideoTrack: localVideoTrack!)
}
peerConnection.remove(localStream!)
peerConnection.add(localStream!)
}

func swapCameraToBack() {
let localStream: RTCMediaStream? = peerConnection.localStreams.first
localStream?.removeVideoTrack((localStream?.videoTracks.first)!)
let localVideoTrack: RTCVideoTrack? = self.createVideoTrack()
if localVideoTrack != nil {
localStream?.addVideoTrack(localVideoTrack!)
}
peerConnection.remove(localStream!)
peerConnection.add(localStream!)
}

}

extension WebRTCClient: RTCDataChannelDelegate {
func dataChannelDidChangeState(_ dataChannel: RTCDataChannel) {
debugPrint("dataChannel did change state: \(dataChannel.readyState)")
}

func dataChannel(_ dataChannel: RTCDataChannel, didReceiveMessageWith buffer: RTCDataBuffer) {
self.delegate?.webRTCClient(self, didReceiveData: buffer.data)
}
}

ViewController.swift文件

override func viewDidLoad() {
super.viewDidLoad()
let localRenderer = RTCMTLVideoView(frame: self.localVideoView?.frame ?? CGRect.zero
let remoteRenderer = RTCMTLVideoView(frame: self.view.frame)

self.view.insertSubview(localRenderer, at: 0)
self.view.insertSubview(remoteRenderer, at: -1)

self.webRTCClient.startCaptureLocalVideo(renderer: localRenderer, position: .front)
self.webRTCClient.renderRemoteVideo(to: remoteRenderer)

我正在使用下面的 WEBRTC 项目 https://github.com/stasel/WebRTC-iOS

用于演示 WebRTC 的更详细的应用程序:

最佳答案

您应该以RTCVideoFrame 的相关格式发送每个字节的samplebuffer 数据,然后使用推送视频帧发送到WebRTC。

videoSource.capturer(videoCapturer, didCapture: videoFrame)

像这样。

执行步骤:

  1. 将样本缓冲区转换为图像
    var cgImage:CGImage?
guard let sourceImageBuffer = CMSampleBufferGetImageBuffer(sampleBuffer) else{return}
VTCreateCGImageFromCVPixelBuffer(sourceImageBuffer, options: nil, imageOut: &cgImage)
let image = UIImage(cgImage: cgImage!)
  1. 由于样本缓冲区的速度非常快,约为 15fps,因此我们将其降低了 6fps。
    let kDesiredFrameRate = 6.0
let currentTimestamp = CMSampleBufferGetPresentationTimeStamp(sampleBuffer)
let delta = CMTimeSubtract(currentTimestamp, lastVideoTimestamp).seconds
let threshold = Double(1.0/kDesiredFrameRate)
guard delta > threshold else {return}
lastVideoTimestamp = currentTimestamp
  1. 屏幕方向
    enum VideoRotation: Int {
case _0 = 0
case _90 = 90
case _180 = 180
case _270 = 270
}
var videoOrientation = VideoRotation._0
let orientationAttachment = CMGetAttachment(sampleBuffer, key: RPVideoSampleOrientationKey as CFString, attachmentModeOut: nil) as? NSNumber
let orientation: CGImagePropertyOrientation = CGImagePropertyOrientation(rawValue: orientationAttachment.uint32Value) ?? .up
switch (orientation) {
case .up, .upMirrored, .down, .downMirrored:
videoOrientation = ._0
break
case .leftMirrored:
videoOrientation = ._90
case .left:
videoOrientation = ._90
case .rightMirrored:
videoOrientation = ._270
case .right:
videoOrientation = ._270
@unknown default:
break
}

现在接下来的 2 个步骤将在主应用程序中处理:

  1. 转换 CVPixelBuffer 帧中的图像:

    guard let videoImage = UIImage(data: imageData!) else {return}
    guard let cgImage = videoImage.cgImage else {return}
    guard let imageBuffer = videoImage.pixelBuffer(forImage: cgImage) else {return}

辅助方法

```
func pixelBuffer(forImage image:CGImage) -> CVPixelBuffer? {
let frameSize = CGSize(width: image.width, height: image.height)

var pixelBuffer:CVPixelBuffer? = nil

let status = CVPixelBufferCreate(kCFAllocatorDefault, Int(frameSize.width), Int(frameSize.height), kCVPixelFormatType_32BGRA , nil, &pixelBuffer)

guaed status == kCVReturnSuccess else {return nil}

CVPixelBufferLockBaseAddress(pixelBuffer!, CVPixelBufferLockFlags.init(rawValue: 0))
let data = CVPixelBufferGetBaseAddress(pixelBuffer!)
let rgbColorSpace = CGColorSpaceCreateDeviceRGB()
let bitmapInfo = CGBitmapInfo(rawValue: CGBitmapInfo.byteOrder32Little.rawValue | CGImageAlphaInfo.premultipliedFirst.rawValue)
let context = CGContext(data: data, width: Int(frameSize.width), height: Int(frameSize.height), bitsPerComponent: 8, bytesPerRow: CVPixelBufferGetBytesPerRow(pixelBuffer!), space: rgbColorSpace, bitmapInfo: bitmapInfo.rawValue)

context?.draw(image, in: CGRect(x: 0, y: 0, width: image.width, height: image.height))

CVPixelBufferUnlockBaseAddress(pixelBuffer!, CVPixelBufferLockFlags(rawValue: 0))

return pixelBuffer
}
```
  1. 最后一步 - CVPixelBuffer 到 RTVideoFrame 的对话

    let rotation = RTCVideoRotation(rawValue: videoRotation)
    let rtcPixlBuffer = RTCCVPixelBuffer(pixelBuffer: imageBuffer)
    let rtcVideoFrame = RTCVideoFrame(buffer: rtcPixlBuffer,
    rotation: rotation ?? ._0,
    timeStampNs: Int64(timeStampNs))

关于swift - 在 swift 中使用 WebRTC 进行 iOS 屏幕共享(使用 ReplayKit),我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/62664114/

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