| /* |
| * 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; |
| |
| - (BOOL)checkLock:(NSError **)outError; |
| |
| /** 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; |
| |
| // 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; |
| - (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches; |
| |
| @end |
| |
| NS_ASSUME_NONNULL_END |