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Don't require recording permissions until call is ringing.
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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)

// FREEBIE
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michaelkirk committed Jan 18, 2017
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7 changes: 7 additions & 0 deletions Libraries/WebRTC/README.md
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The RTCAudioSession.h header isn't included in the standard build of
WebRTC, so we've vendored it here. Otherwise we're using the vanilla
framework.

We use the RTCAudioSession header to manually manage the RTC audio
session, so as to not start recording until the call is connected.

224 changes: 224 additions & 0 deletions Libraries/WebRTC/RTCAudioSession.h
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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/

#import <AVFoundation/AVFoundation.h>
#import <Foundation/Foundation.h>

#import "WebRTC/RTCMacros.h"

NS_ASSUME_NONNULL_BEGIN

extern NSString * const kRTCAudioSessionErrorDomain;
/** Method that requires lock was called without lock. */
extern NSInteger const kRTCAudioSessionErrorLockRequired;
/** Unknown configuration error occurred. */
extern NSInteger const kRTCAudioSessionErrorConfiguration;

@class RTCAudioSession;
@class RTCAudioSessionConfiguration;

// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
// from AVAudioSession and handle them before calling these delegate methods,
// at which point applications can perform additional processing if required.
RTC_EXPORT
@protocol RTCAudioSessionDelegate <NSObject>

@optional
/** Called on a system notification thread when AVAudioSession starts an
* interruption event.
*/
- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;

/** Called on a system notification thread when AVAudioSession ends an
* interruption event.
*/
- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
shouldResumeSession:(BOOL)shouldResumeSession;

/** Called on a system notification thread when AVAudioSession changes the
* route.
*/
- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
reason:(AVAudioSessionRouteChangeReason)reason
previousRoute:(AVAudioSessionRouteDescription *)previousRoute;

/** Called on a system notification thread when AVAudioSession media server
* terminates.
*/
- (void)audioSessionMediaServicesWereLost:(RTCAudioSession *)session;

/** Called on a system notification thread when AVAudioSession media server
* restarts.
*/
- (void)audioSessionMediaServicesWereReset:(RTCAudioSession *)session;

// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.

- (void)audioSession:(RTCAudioSession *)session
didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;

/** Called on a WebRTC thread when the audio device is notified to begin
* playback or recording.
*/
- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;

/** Called on a WebRTC thread when the audio device is notified to stop
* playback or recording.
*/
- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;

@end

/** Proxy class for AVAudioSession that adds a locking mechanism similar to
* AVCaptureDevice. This is used to that interleaving configurations between
* WebRTC and the application layer are avoided.
*
* RTCAudioSession also coordinates activation so that the audio session is
* activated only once. See |setActive:error:|.
*/
RTC_EXPORT
@interface RTCAudioSession : NSObject

/** Convenience property to access the AVAudioSession singleton. Callers should
* not call setters on AVAudioSession directly, but other method invocations
* are fine.
*/
@property(nonatomic, readonly) AVAudioSession *session;

/** Our best guess at whether the session is active based on results of calls to
* AVAudioSession.
*/
@property(nonatomic, readonly) BOOL isActive;
/** Whether RTCAudioSession is currently locked for configuration. */
@property(nonatomic, readonly) BOOL isLocked;

/** If YES, WebRTC will not initialize the audio unit automatically when an
* audio track is ready for playout or recording. Instead, applications should
* call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
* as soon as an audio track is ready for playout or recording.
*/
@property(nonatomic, assign) BOOL useManualAudio;

/** This property is only effective if useManualAudio is YES.
* Represents permission for WebRTC to initialize the VoIP audio unit.
* When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
* stopped and uninitialized. This will stop incoming and outgoing audio.
* When set to YES, WebRTC will initialize and start the audio unit when it is
* needed (e.g. due to establishing an audio connection).
* This property was introduced to work around an issue where if an AVPlayer is
* playing audio while the VoIP audio unit is initialized, its audio would be
* either cut off completely or played at a reduced volume. By preventing
* the audio unit from being initialized until after the audio has completed,
* we are able to prevent the abrupt cutoff.
*/
@property(nonatomic, assign) BOOL isAudioEnabled;

// Proxy properties.
@property(readonly) NSString *category;
@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
@property(readonly) NSString *mode;
@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
@property(readonly) AVAudioSessionRouteDescription *currentRoute;
@property(readonly) NSInteger maximumInputNumberOfChannels;
@property(readonly) NSInteger maximumOutputNumberOfChannels;
@property(readonly) float inputGain;
@property(readonly) BOOL inputGainSettable;
@property(readonly) BOOL inputAvailable;
@property(readonly, nullable)
NSArray<AVAudioSessionDataSourceDescription *> * inputDataSources;
@property(readonly, nullable)
AVAudioSessionDataSourceDescription *inputDataSource;
@property(readonly, nullable)
NSArray<AVAudioSessionDataSourceDescription *> * outputDataSources;
@property(readonly, nullable)
AVAudioSessionDataSourceDescription *outputDataSource;
@property(readonly) double sampleRate;
@property(readonly) double preferredSampleRate;
@property(readonly) NSInteger inputNumberOfChannels;
@property(readonly) NSInteger outputNumberOfChannels;
@property(readonly) float outputVolume;
@property(readonly) NSTimeInterval inputLatency;
@property(readonly) NSTimeInterval outputLatency;
@property(readonly) NSTimeInterval IOBufferDuration;
@property(readonly) NSTimeInterval preferredIOBufferDuration;

/** Default constructor. */
+ (instancetype)sharedInstance;
- (instancetype)init NS_UNAVAILABLE;

/** Adds a delegate, which is held weakly. */
- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
/** Removes an added delegate. */
- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;

/** Request exclusive access to the audio session for configuration. This call
* will block if the lock is held by another object.
*/
- (void)lockForConfiguration;
/** Relinquishes exclusive access to the audio session. */
- (void)unlockForConfiguration;

/** If |active|, activates the audio session if it isn't already active.
* Successful calls must be balanced with a setActive:NO when activation is no
* longer required. If not |active|, deactivates the audio session if one is
* active and this is the last balanced call. When deactivating, the
* AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
* AVAudioSession.
*/
- (BOOL)setActive:(BOOL)active
error:(NSError **)outError;

// The following methods are proxies for the associated methods on
// AVAudioSession. |lockForConfiguration| must be called before using them
// otherwise they will fail with kRTCAudioSessionErrorLockRequired.

- (BOOL)setCategory:(NSString *)category
withOptions:(AVAudioSessionCategoryOptions)options
error:(NSError **)outError;
- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
error:(NSError **)outError;
- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
error:(NSError **)outError;
- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
error:(NSError **)outError;
- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
error:(NSError **)outError;
- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
error:(NSError **)outError;
- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError;
- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
error:(NSError **)outError;

@end

@interface RTCAudioSession (Configuration)

/** Applies the configuration to the current session. Attempts to set all
* properties even if previous ones fail. Only the last error will be
* returned.
* |lockForConfiguration| must be called first.
*/
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
error:(NSError **)outError;

/** Convenience method that calls both setConfiguration and setActive.
* |lockForConfiguration| must be called first.
*/
- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
active:(BOOL)active
error:(NSError **)outError;

@end

NS_ASSUME_NONNULL_END
34 changes: 17 additions & 17 deletions MAINTAINING.md
Expand Up @@ -20,23 +20,23 @@ https://webrtc.org/native-code/ios/
Once you have your build environment set up and the WebRTC source downloaded:

# The specific set of commands that worked for me were somewhat different.
# 1. Install depot tools
cd <somewhere>
git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git
cd depot_tools
export PATH=<somewhere>/depot_tools:"$PATH"
# 2. Fetch webrtc source
cd <somewhere else>
mkdir webrtc
cd webrtc
fetch --nohooks webrtc_ios
gclient sync
# 3. Build webrtc
# NOTE: build_ios_libs.sh only worked for me from inside "src"
cd src
webrtc/build/ios/build_ios_libs.sh
# NOTE: It's Carthage/Build/iOS, not Carthage/Builds
mv out_ios_libs/WebRTC.framework ../../Signal-iOS/Carthage/Build/iOS/
# 1. Install depot tools
cd <somewhere>
git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git
cd depot_tools
export PATH=<somewhere>/depot_tools:"$PATH"
# 2. Fetch webrtc source
cd <somewhere else>
mkdir webrtc
cd webrtc
fetch --nohooks webrtc_ios
gclient sync
# 3. Build webrtc
# NOTE: build_ios_libs.sh only worked for me from inside "src"
cd src
webrtc/build/ios/build_ios_libs.sh
# NOTE: It's Carthage/Build/iOS, not Carthage/Builds
mv out_ios_libs/WebRTC.framework ../../Signal-iOS/Carthage/Build/iOS/

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