webrtc/sdk/objc/components/audio/RTCAudioSession+Private.h
Niels Möller be74b8058b Fix spelling of receiver and transceiver.
Bug: None
Change-Id: I439e217d67283b182833e48da15af9ae367ac14e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256015
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36257}
2022-03-18 14:54:10 +00:00

95 lines
3.7 KiB
Objective-C

/*
* 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 "RTCAudioSession.h"
NS_ASSUME_NONNULL_BEGIN
@class RTC_OBJC_TYPE(RTCAudioSessionConfiguration);
@interface RTC_OBJC_TYPE (RTCAudioSession)
()
/** Number of times setActive:YES has succeeded without a balanced call to
* setActive:NO.
*/
@property(nonatomic, readonly) int activationCount;
/** The number of times `beginWebRTCSession` was called without a balanced call
* to `endWebRTCSession`.
*/
@property(nonatomic, readonly) int webRTCSessionCount;
/** Convenience BOOL that checks useManualAudio and isAudioEnebled. */
@property(readonly) BOOL canPlayOrRecord;
/** Tracks whether we have been sent an interruption event that hasn't been matched by either an
* interrupted end event or a foreground event.
*/
@property(nonatomic, assign) BOOL isInterrupted;
/** Adds the delegate to the list of delegates, and places it at the front of
* the list. This delegate will be notified before other delegates of
* audio events.
*/
- (void)pushDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
/** Signals RTCAudioSession that a WebRTC session is about to begin and
* audio configuration is needed. Will configure the audio session for WebRTC
* if not already configured and if configuration is not delayed.
* Successful calls must be balanced by a call to endWebRTCSession.
*/
- (BOOL)beginWebRTCSession:(NSError **)outError;
/** Signals RTCAudioSession that a WebRTC session is about to end and audio
* unconfiguration is needed. Will unconfigure the audio session for WebRTC
* if this is the last unmatched call and if configuration is not delayed.
*/
- (BOOL)endWebRTCSession:(NSError **)outError;
/** Configure the audio session for WebRTC. This call will fail if the session
* is already configured. On other failures, we will attempt to restore the
* previously used audio session configuration.
* `lockForConfiguration` must be called first.
* Successful calls to configureWebRTCSession must be matched by calls to
* `unconfigureWebRTCSession`.
*/
- (BOOL)configureWebRTCSession:(NSError **)outError;
/** Unconfigures the session for WebRTC. This will attempt to restore the
* audio session to the settings used before `configureWebRTCSession` was
* called.
* `lockForConfiguration` must be called first.
*/
- (BOOL)unconfigureWebRTCSession:(NSError **)outError;
/** Returns a configuration error with the given description. */
- (NSError *)configurationErrorWithDescription:(NSString *)description;
/** Notifies the receiver that a playout glitch was detected. */
- (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
/** Notifies the receiver that there was an error when starting an audio unit. */
- (void)notifyAudioUnitStartFailedWithError:(OSStatus)error;
// Properties and methods for tests.
- (void)notifyDidBeginInterruption;
- (void)notifyDidEndInterruptionWithShouldResumeSession:(BOOL)shouldResumeSession;
- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
- (void)notifyMediaServicesWereLost;
- (void)notifyMediaServicesWereReset;
- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
- (void)notifyDidStartPlayOrRecord;
- (void)notifyDidStopPlayOrRecord;
@end
NS_ASSUME_NONNULL_END