mirror of https://github.com/oxen-io/session-ios
Don't require recording permissions until call is ringing.
We do this by manually managing the RTCAudioSession. Unfortunately to do this we have to include a couple of RTC headers not exported by the default build of WebRTC.framework (see: Libraries/WebRTC) // FREEBIEpull/1/head
Michael Kirk
8 years ago
parent
ca218ebb64
commit
dbb29d7d7e
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The RTCAudioSession.h header isn't included in the standard build of
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WebRTC, so we've vendored it here. Otherwise we're using the vanilla
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framework.
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We use the RTCAudioSession header to manually manage the RTC audio
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session, so as to not start recording until the call is connected.
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/*
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* Copyright 2016 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import <AVFoundation/AVFoundation.h>
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#import <Foundation/Foundation.h>
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#import "WebRTC/RTCMacros.h"
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NS_ASSUME_NONNULL_BEGIN
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extern NSString * const kRTCAudioSessionErrorDomain;
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/** Method that requires lock was called without lock. */
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extern NSInteger const kRTCAudioSessionErrorLockRequired;
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/** Unknown configuration error occurred. */
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extern NSInteger const kRTCAudioSessionErrorConfiguration;
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@class RTCAudioSession;
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@class RTCAudioSessionConfiguration;
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// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
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// from AVAudioSession and handle them before calling these delegate methods,
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// at which point applications can perform additional processing if required.
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RTC_EXPORT
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@protocol RTCAudioSessionDelegate <NSObject>
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@optional
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/** Called on a system notification thread when AVAudioSession starts an
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* interruption event.
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*/
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- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
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/** Called on a system notification thread when AVAudioSession ends an
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* interruption event.
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*/
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- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
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shouldResumeSession:(BOOL)shouldResumeSession;
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/** Called on a system notification thread when AVAudioSession changes the
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* route.
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*/
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- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
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reason:(AVAudioSessionRouteChangeReason)reason
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previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
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/** Called on a system notification thread when AVAudioSession media server
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* terminates.
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*/
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- (void)audioSessionMediaServicesWereLost:(RTCAudioSession *)session;
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/** Called on a system notification thread when AVAudioSession media server
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* restarts.
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*/
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- (void)audioSessionMediaServicesWereReset:(RTCAudioSession *)session;
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// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
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- (void)audioSession:(RTCAudioSession *)session
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didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
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/** Called on a WebRTC thread when the audio device is notified to begin
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* playback or recording.
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*/
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- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;
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/** Called on a WebRTC thread when the audio device is notified to stop
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* playback or recording.
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*/
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- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
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@end
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/** Proxy class for AVAudioSession that adds a locking mechanism similar to
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* AVCaptureDevice. This is used to that interleaving configurations between
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* WebRTC and the application layer are avoided.
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*
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* RTCAudioSession also coordinates activation so that the audio session is
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* activated only once. See |setActive:error:|.
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*/
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RTC_EXPORT
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@interface RTCAudioSession : NSObject
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/** Convenience property to access the AVAudioSession singleton. Callers should
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* not call setters on AVAudioSession directly, but other method invocations
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* are fine.
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*/
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@property(nonatomic, readonly) AVAudioSession *session;
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/** Our best guess at whether the session is active based on results of calls to
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* AVAudioSession.
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*/
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@property(nonatomic, readonly) BOOL isActive;
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/** Whether RTCAudioSession is currently locked for configuration. */
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@property(nonatomic, readonly) BOOL isLocked;
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/** If YES, WebRTC will not initialize the audio unit automatically when an
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* audio track is ready for playout or recording. Instead, applications should
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* call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
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* as soon as an audio track is ready for playout or recording.
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*/
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@property(nonatomic, assign) BOOL useManualAudio;
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/** This property is only effective if useManualAudio is YES.
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* Represents permission for WebRTC to initialize the VoIP audio unit.
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* When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
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* stopped and uninitialized. This will stop incoming and outgoing audio.
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* When set to YES, WebRTC will initialize and start the audio unit when it is
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* needed (e.g. due to establishing an audio connection).
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* This property was introduced to work around an issue where if an AVPlayer is
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* playing audio while the VoIP audio unit is initialized, its audio would be
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* either cut off completely or played at a reduced volume. By preventing
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* the audio unit from being initialized until after the audio has completed,
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* we are able to prevent the abrupt cutoff.
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*/
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@property(nonatomic, assign) BOOL isAudioEnabled;
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// Proxy properties.
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@property(readonly) NSString *category;
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@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
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@property(readonly) NSString *mode;
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@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
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@property(readonly) AVAudioSessionRouteDescription *currentRoute;
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@property(readonly) NSInteger maximumInputNumberOfChannels;
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@property(readonly) NSInteger maximumOutputNumberOfChannels;
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@property(readonly) float inputGain;
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@property(readonly) BOOL inputGainSettable;
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@property(readonly) BOOL inputAvailable;
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@property(readonly, nullable)
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NSArray<AVAudioSessionDataSourceDescription *> * inputDataSources;
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@property(readonly, nullable)
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AVAudioSessionDataSourceDescription *inputDataSource;
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@property(readonly, nullable)
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NSArray<AVAudioSessionDataSourceDescription *> * outputDataSources;
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@property(readonly, nullable)
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AVAudioSessionDataSourceDescription *outputDataSource;
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@property(readonly) double sampleRate;
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@property(readonly) double preferredSampleRate;
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@property(readonly) NSInteger inputNumberOfChannels;
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@property(readonly) NSInteger outputNumberOfChannels;
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@property(readonly) float outputVolume;
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@property(readonly) NSTimeInterval inputLatency;
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@property(readonly) NSTimeInterval outputLatency;
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@property(readonly) NSTimeInterval IOBufferDuration;
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@property(readonly) NSTimeInterval preferredIOBufferDuration;
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/** Default constructor. */
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+ (instancetype)sharedInstance;
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- (instancetype)init NS_UNAVAILABLE;
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/** Adds a delegate, which is held weakly. */
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- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
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/** Removes an added delegate. */
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- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;
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/** Request exclusive access to the audio session for configuration. This call
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* will block if the lock is held by another object.
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*/
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- (void)lockForConfiguration;
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/** Relinquishes exclusive access to the audio session. */
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- (void)unlockForConfiguration;
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/** If |active|, activates the audio session if it isn't already active.
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* Successful calls must be balanced with a setActive:NO when activation is no
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* longer required. If not |active|, deactivates the audio session if one is
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* active and this is the last balanced call. When deactivating, the
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* AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
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* AVAudioSession.
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*/
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- (BOOL)setActive:(BOOL)active
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error:(NSError **)outError;
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// The following methods are proxies for the associated methods on
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// AVAudioSession. |lockForConfiguration| must be called before using them
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// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
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- (BOOL)setCategory:(NSString *)category
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withOptions:(AVAudioSessionCategoryOptions)options
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error:(NSError **)outError;
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- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
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- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
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- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
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- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
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error:(NSError **)outError;
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- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
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error:(NSError **)outError;
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- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
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error:(NSError **)outError;
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- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
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error:(NSError **)outError;
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- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
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error:(NSError **)outError;
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- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
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error:(NSError **)outError;
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- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
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error:(NSError **)outError;
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@end
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@interface RTCAudioSession (Configuration)
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/** Applies the configuration to the current session. Attempts to set all
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* properties even if previous ones fail. Only the last error will be
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* returned.
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* |lockForConfiguration| must be called first.
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*/
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- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
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error:(NSError **)outError;
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/** Convenience method that calls both setConfiguration and setActive.
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* |lockForConfiguration| must be called first.
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*/
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- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
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active:(BOOL)active
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error:(NSError **)outError;
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@end
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NS_ASSUME_NONNULL_END
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@ -0,0 +1,51 @@
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// Copyright © 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. However, the headers for RTCAudioSession are not exported by default, so we've vendored the header
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* into our project. See "Libraries/WebRTC"
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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 the recording audio session ourselves.
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*/
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func start() {
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Logger.info("\(TAG) in \(#function)")
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rtcAudioSession.isAudioEnabled = true
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}
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/**
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* Because we useManualAudio with our RTCAudioSession, we have to 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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func stop() {
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Logger.info("\(TAG) in \(#function)")
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rtcAudioSession.isAudioEnabled = false
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}
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}
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@ -0,0 +1,35 @@
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// Copyright © 2017 Open Whisper Systems. All rights reserved.
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//
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import XCTest
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import AVKit
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import WebRTC
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/**
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* These tests are obtuse - they just assert the exact implementation of the methods. Normally I wouldn't include them,
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* but these methods make use of a header not included in the standard distribution of the WebRTC.framework. We've
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* included the header in our local project, and test the methods here to make sure that they are still available when
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* we upgrade the framework.
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*
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* If they are failing, it's possible the RTCAudioSession header, and our usage of it, need to be updated.
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*/
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class CallAudioSessionTest: XCTestCase {
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func testAudioSession() {
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let rtcAudioSession = RTCAudioSession.sharedInstance()
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// Sanity Check
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XCTAssertFalse(rtcAudioSession.useManualAudio)
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CallAudioSession().configure()
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XCTAssertTrue(rtcAudioSession.useManualAudio)
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XCTAssertFalse(rtcAudioSession.isAudioEnabled)
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CallAudioSession().start()
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XCTAssertTrue(rtcAudioSession.useManualAudio)
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XCTAssertTrue(rtcAudioSession.isAudioEnabled)
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CallAudioSession().stop()
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XCTAssertTrue(rtcAudioSession.useManualAudio)
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XCTAssertFalse(rtcAudioSession.isAudioEnabled)
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}
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}
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