在Swift中将麦克风和InApp音频CMSampleBuffer发送到WebRTC,可以通过以下步骤实现:
下面是一个示例代码,展示了如何在Swift中实现上述步骤:
import WebRTC
import AVFoundation
class WebRTCManager: NSObject, AVCaptureAudioDataOutputSampleBufferDelegate {
var audioSource: RTCAudioSource?
var peerConnection: RTCPeerConnection?
func startWebRTC() {
// 创建音频捕获会话
let captureSession = AVCaptureSession()
guard let audioDevice = AVCaptureDevice.default(for: .audio),
let audioInput = try? AVCaptureDeviceInput(device: audioDevice),
captureSession.canAddInput(audioInput) else {
return
}
captureSession.addInput(audioInput)
let audioOutput = AVCaptureAudioDataOutput()
let audioQueue = DispatchQueue(label: "audioQueue")
audioOutput.setSampleBufferDelegate(self, queue: audioQueue)
captureSession.addOutput(audioOutput)
// 创建音频源
audioSource = RTCAudioSource()
// 创建PeerConnection
let rtcConfig = RTCConfiguration()
let rtcConstraints = RTCMediaConstraints(mandatoryConstraints: nil, optionalConstraints: nil)
peerConnection = factory.peerConnection(with: rtcConfig, constraints: rtcConstraints, delegate: nil)
// 设置音频源
if let audioTrack = factory.audioTrack(with: audioSource!, trackId: "audioTrack") {
let rtcMediaStream = factory.mediaStream(withStreamId: "mediaStream")
rtcMediaStream.addAudioTrack(audioTrack)
peerConnection?.add(rtcMediaStream)
}
// 连接到WebRTC服务器
let rtcIceServer = RTCIceServer(urlStrings: ["stun:stun.l.google.com:19302"])
peerConnection?.setConfiguration(RTCConfiguration())
peerConnection?.add(rtcIceServer)
}
// AVCaptureAudioDataOutputSampleBufferDelegate方法,获取音频数据
func captureOutput(_ output: AVCaptureOutput, didOutput sampleBuffer: CMSampleBuffer, from connection: AVCaptureConnection) {
// 将音频数据添加到音频源中
audioSource?.audioQueue?.push(sampleBuffer)
}
}
这是一个简单的示例,展示了如何在Swift中将麦克风和InApp音频CMSampleBuffer发送到WebRTC。请注意,这只是一个基本的实现,实际应用中可能需要更多的配置和处理。
关于WebRTC的更多信息和使用方法,你可以参考腾讯云的实时音视频解决方案,链接地址:腾讯云实时音视频解决方案。
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