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157 lines
5.5 KiB
Swift
157 lines
5.5 KiB
Swift
//
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// Copyright (c) 2018 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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@objc
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public class AudioActivity: NSObject {
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let audioDescription: String
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override public var description: String {
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return "<\(self.logTag) audioDescription: \"\(audioDescription)\">"
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}
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public
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init(audioDescription: String) {
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self.audioDescription = audioDescription
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}
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deinit {
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OWSAudioSession.shared.ensureAudioSessionActivationState()
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}
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}
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@objc
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public class OWSAudioSession: NSObject {
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// Force singleton access
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public static let shared = OWSAudioSession()
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private override init() {}
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private let avAudioSession = AVAudioSession.sharedInstance()
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private var currentActivities: [Weak<AudioActivity>] = []
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// Ignores hardware mute switch, plays through external speaker
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public func setPlaybackCategory(audioActivity: AudioActivity) {
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Logger.debug("\(logTag) in \(#function)")
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// In general, we should have put the audio session back to it's default
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// category when we were done with whatever activity required it to be modified
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assert(avAudioSession.category == AVAudioSessionCategorySoloAmbient)
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startAudioActivity(audioActivity)
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do {
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try avAudioSession.setCategory(AVAudioSessionCategoryPlayback)
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} catch {
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owsFail("\(logTag) in \(#function) failed with error: \(error)")
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}
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}
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public func setRecordCategory(audioActivity: AudioActivity) -> Bool {
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Logger.debug("\(logTag) in \(#function)")
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// In general, we should have put the audio session back to it's default
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// category when we were done with whatever activity required it to be modified
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assert(avAudioSession.category == AVAudioSessionCategorySoloAmbient)
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assert(avAudioSession.recordPermission() == .granted)
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startAudioActivity(audioActivity)
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do {
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try avAudioSession.setCategory(AVAudioSessionCategoryRecord)
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return true
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} catch {
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owsFail("\(logTag) in \(#function) failed with error: \(error)")
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return false
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}
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}
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private func startAudioActivity(_ audioActivity: AudioActivity) {
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Logger.debug("\(logTag) in \(#function) with \(audioActivity)")
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self.currentActivities.append(Weak(value: audioActivity))
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}
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public func endAudioActivity(_ audioActivity: AudioActivity) {
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Logger.debug("\(logTag) in \(#function) with audioActivity: \(audioActivity)")
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currentActivities = currentActivities.filter { return $0.value != audioActivity }
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ensureAudioSessionActivationState()
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}
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fileprivate func ensureAudioSessionActivationState() {
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// Cull any stale activities
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currentActivities = currentActivities.flatMap { oldActivity in
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guard oldActivity.value != nil else {
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// Normally we should be explicitly stopping an audio activity, but this allows
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// for recovery if the owner of the AudioAcivity was GC'd without ending it's
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// audio activity
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Logger.warn("\(logTag) an old activity has been gc'd")
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return nil
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}
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// return any still-active activities
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return oldActivity
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}
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guard currentActivities.count == 0 else {
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Logger.debug("\(logTag) not deactivating due to currentActivities: \(currentActivities)")
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return
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}
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do {
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try avAudioSession.setCategory(AVAudioSessionCategorySoloAmbient)
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// When playing audio in Signal, other apps audio (e.g. Music) is paused.
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// By notifying when we deactivate, the other app can resume playback.
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try avAudioSession.setActive(false, with: [.notifyOthersOnDeactivation])
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} catch {
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owsFail("\(logTag) in \(#function) failed with error: \(error)")
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}
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}
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// MARK: - WebRTC Audio
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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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/**
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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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public func configureRTCAudio() {
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Logger.info("\(logTag) 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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* See "WebRTC Audio" comment for details on manual audio.
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*/
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public func enableRTCAudio(audioActivity: AudioActivity) {
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startAudioActivity(audioActivity)
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rtcAudioSession.isAudioEnabled = true
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}
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public func disableRTCAudio(audioActivity: AudioActivity) {
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rtcAudioSession.isAudioEnabled = false
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endAudioActivity(audioActivity)
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}
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}
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