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c7642cc628
By centralizing AudioSession management onto the AudioService, we can avoid enabling the RTCAudioSession while we're mid-ring. Also allows us to centralize and remove redundant audio session logic. // FREEBIE
46 lines
1.5 KiB
Swift
46 lines
1.5 KiB
Swift
//
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// Copyright (c) 2017 Open Whisper Systems. All rights reserved.
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//
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import Foundation
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import WebRTC
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/**
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* By default WebRTC starts the audio session (PlayAndRecord) immediately upon creating the peer connection
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* but we want to create the peer connection and set up all the signaling channels before we prompt the user
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* for an incoming call. Without manually handling the session, this would result in the user seeing a recording
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* permission requested (and recording banner) before they even know they have an incoming call.
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*
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* By using the `useManualAudio` and `isAudioEnabled` attributes of the RTCAudioSession we can delay recording until
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* it makes sense.
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*/
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class CallAudioSession {
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let TAG = "[CallAudioSession]"
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/**
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* The private class that manages AVAudioSession for WebRTC
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*/
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private let rtcAudioSession = RTCAudioSession.sharedInstance()
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/**
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* This must be called before any audio tracks are added to the peerConnection, else we'll start recording before all
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* our signaling is set up.
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*/
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func configure() {
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Logger.info("\(TAG) in \(#function)")
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rtcAudioSession.useManualAudio = true
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}
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/**
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* Because we useManualAudio with our RTCAudioSession, we have to start/stop the recording audio session ourselves.
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* Else, we start recording before the next call is ringing.
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*/
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var isRTCAudioEnabled: Bool {
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get {
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return rtcAudioSession.isAudioEnabled
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}
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set {
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rtcAudioSession.isAudioEnabled = newValue
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}
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}
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}
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