Add observer for AVAudioSession.outputVolume

webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm

 /*
  *  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 <Foundation/Foundation.h>
 #import <OCMock/OCMock.h>
 
 #include "webrtc/test/gtest.h"
 
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
 
 @interface RTCAudioSessionTestDelegate : NSObject <RTCAudioSessionDelegate>
+
+@property (nonatomic, readonly) float outputVolume;
+
 @end
 
 @implementation RTCAudioSessionTestDelegate
 
+@synthesize outputVolume = _outputVolume;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    _outputVolume = -1;
+  }
+  return self;
+}
+
 - (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session {
 }
 
 - (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
                    shouldResumeSession:(BOOL)shouldResumeSession {
 }
 
 - (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
            reason:(AVAudioSessionRouteChangeReason)reason
     previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
 }
 
 - (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session {
 }
 
 - (void)audioSessionMediaServerReset:(RTCAudioSession *)session {
 }
 
 - (void)audioSessionShouldConfigure:(RTCAudioSession *)session {
 }
 
 - (void)audioSessionShouldUnconfigure:(RTCAudioSession *)session {
 }
 
+- (void)audioSession:(RTCAudioSession *)audioSession
+    didChangeOutputVolume:(float)outputVolume {
+  _outputVolume = outputVolume;
+}
+
 @end
 
 // A delegate that adds itself to the audio session on init and removes itself
 // in its dealloc.
 @interface RTCTestRemoveOnDeallocDelegate : RTCAudioSessionTestDelegate
 @end
 
 @implementation RTCTestRemoveOnDeallocDelegate
 
 - (instancetype)init {
@@ skipping 180 matching lines @@
   [audioSession unlockForConfiguration];
 
   OCMVerify([mockAudioSession session]);
   OCMVerify([[mockAVAudioSession ignoringNonObjectArgs] setActive:YES withOptions:0 error:&error]);
   OCMVerify([[mockAVAudioSession ignoringNonObjectArgs] setActive:NO withOptions:0 error:&error]);
 
   [mockAVAudioSession stopMocking];
   [mockAudioSession stopMocking];
 }
 
+- (void)testAudioVolumeDidNotify {
+  RTCAudioSession *session = [RTCAudioSession sharedInstance];
+  RTCAudioSessionTestDelegate *delegate =
+      [[RTCAudioSessionTestDelegate alloc] init];
+  [session addDelegate:delegate];
+
+  [session observeValueForKeyPath:@"outputVolume"
+                         ofObject:[AVAudioSession sharedInstance]
+                           change:
+        @{NSKeyValueChangeNewKey :
+            @([AVAudioSession sharedInstance].outputVolume) }
+                          context:nil];
+
+  EXPECT_NE(delegate.outputVolume, -1);
+  EXPECT_EQ([AVAudioSession sharedInstance].outputVolume, delegate.outputVolume);
+}
+
 @end
 
 namespace webrtc {
 
 class AudioSessionTest : public ::testing::Test {
  protected:
   void TearDown() {
     RTCAudioSession *session = [RTCAudioSession sharedInstance];
     for (id<RTCAudioSessionDelegate> delegate : session.delegates) {
       [session removeDelegate:delegate];
@@ skipping 29 matching lines @@
 TEST_F(AudioSessionTest, AudioSessionActivation) {
   RTCAudioSessionTest *test = [[RTCAudioSessionTest alloc] init];
   [test testAudioSessionActivation];
 }
 
 TEST_F(AudioSessionTest, ConfigureWebRTCSession) {
   RTCAudioSessionTest *test = [[RTCAudioSessionTest alloc] init];
   [test testConfigureWebRTCSession];
 }
 
+TEST_F(AudioSessionTest, AudioVolumeDidNotify) {
+  RTCAudioSessionTest *test = [[RTCAudioSessionTest alloc] init];
+  [test testAudioVolumeDidNotify];
+}
+
 }  // namespace webrtc