mirror of https://github.com/oxen-io/session-ios
Garther audio concerns, clean up session when done
- sync speakerphone state manipulated from system call screen
- Revert audio session after call failure, ensures media plays out of
speaker after placing a failing call.
- Replace notification with delegate pattern since we're already using
delegate pattern here.
- Fixes voiceover accessibility after voice memo
- Avoid audio blip after pressing hangup
- Rename CallAudioSession -> OWSAudioSession
Going to start using it for other non-call things since we want to
gather all our audio session concerns.
- Resume background audio when done playing video
- Extract OWSVideoPlayer which ensures audio is in proper state before
playback
- Move recording session logic to shared OWSAudioSession
- Deactivate audio session when complete
// FREEBIE
pull/1/head
Michael Kirk
8 years ago
parent
fa9ac5aa46
commit
c646f76335
@ -1,51 +0,0 @@
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//
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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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// Force singleton access
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static let shared = CallAudioSession()
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private init() {}
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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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* See header for details on manual audio.
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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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@ -0,0 +1,72 @@
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//
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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 AVFoundation
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@objc
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protocol OWSVideoPlayerDelegate: class {
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@available(iOSApplicationExtension 9.0, *)
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func videoPlayerDidPlayToCompletion(_ videoPlayer: OWSVideoPlayer)
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}
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@objc
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public class OWSVideoPlayer: NSObject {
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let avPlayer: AVPlayer
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weak var delegate: OWSVideoPlayerDelegate?
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@available(iOS 9.0, *)
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init(url: URL) {
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self.avPlayer = AVPlayer(url: url)
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super.init()
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NotificationCenter.default.addObserver(self,
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selector: #selector(playerItemDidPlayToCompletion(_:)),
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name: NSNotification.Name.AVPlayerItemDidPlayToEndTime,
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object: avPlayer.currentItem)
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}
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// MARK: Playback Controls
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@available(iOS 9.0, *)
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public func pause() {
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avPlayer.pause()
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OWSAudioSession.shared.endAudioActivity()
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}
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@available(iOS 9.0, *)
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public func play() {
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OWSAudioSession.shared.setPlaybackCategory()
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guard let item = avPlayer.currentItem else {
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owsFail("\(logTag) video player item was unexpectedly nil")
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return
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}
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if item.currentTime() == item.duration {
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// Rewind for repeated plays, but only if it previously played to end.
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avPlayer.seek(to: kCMTimeZero)
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}
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avPlayer.play()
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}
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@available(iOS 9.0, *)
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@objc(seekToTime:)
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public func seek(to time: CMTime) {
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avPlayer.seek(to: time)
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}
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// MARK: private
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@objc
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@available(iOS 9.0, *)
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private func playerItemDidPlayToCompletion(_ notification: Notification) {
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self.delegate?.videoPlayerDidPlayToCompletion(self)
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OWSAudioSession.shared.endAudioActivity()
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}
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}
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@ -0,0 +1,101 @@
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//
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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 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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// Ignores hardware mute switch, plays through external speaker
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public func setPlaybackCategory() {
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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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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() -> 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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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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public func endAudioActivity() {
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Logger.debug("\(logTag) in \(#function)")
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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 header for details on manual audio.
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*/
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public 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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Loading…
Reference in New Issue