Notify delegates about audio glitches in real time

Bug: webrtc:7819
Change-Id: I72ec77d216ce386dd45aef68eeac833b3a75b670
Reviewed-on: https://chromium-review.googlesource.com/543239
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18778}

webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m

@@ -22,7 +22,8 @@
 #import "WebRTC/RTCMediaConstraints.h"
 
 @interface ARDVideoCallViewController () <ARDAppClientDelegate,
-    ARDVideoCallViewDelegate>
+                                          ARDVideoCallViewDelegate,
+                                          RTCAudioSessionDelegate>
 @property(nonatomic, strong) RTCVideoTrack *remoteVideoTrack;
 @property(nonatomic, readonly) ARDVideoCallView *videoCallView;
 @end
@@ -57,6 +58,9 @@
   _videoCallView.statusLabel.text =
       [self statusTextForState:RTCIceConnectionStateNew];
   self.view = _videoCallView;
+
+  RTCAudioSession *session = [RTCAudioSession sharedInstance];
+  [session addDelegate:self];
 }
 
 #pragma mark - ARDAppClientDelegate
@@ -158,6 +162,13 @@
   _videoCallView.statsView.hidden = NO;
 }
 
+#pragma mark - RTCAudioSessionDelegate
+
+- (void)audioSession:(RTCAudioSession *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches {
+  RTCLog(@"Audio session detected glitch, total: %lld", totalNumberOfGlitches);
+}
+
 #pragma mark - Private
 
 - (void)setRemoteVideoTrack:(RTCVideoTrack *)remoteVideoTrack {

webrtc/modules/audio_device/ios/audio_device_ios.mm

@@ -672,6 +672,12 @@ void AudioDeviceIOS::HandlePlayoutGlitchDetected() {
   num_detected_playout_glitches_++;
   RTCLog(@"Number of detected playout glitches: %lld",
          num_detected_playout_glitches_);
+
+  int64_t glitch_count = num_detected_playout_glitches_;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    RTCAudioSession* session = [RTCAudioSession sharedInstance];
+    [session notifyDidDetectPlayoutGlitch:glitch_count];
+  });
 }
 
 void AudioDeviceIOS::HandleOutputVolumeChange() {

webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h

@@ -90,6 +90,7 @@ NS_ASSUME_NONNULL_BEGIN
 - (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
 - (void)notifyDidStartPlayOrRecord;
 - (void)notifyDidStopPlayOrRecord;
+- (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
 
 @end
 

webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm

@@ -914,4 +914,13 @@ NSString * const kRTCAudioSessionOutputVolumeSelector = @"outputVolume";
   }
 }
 
+- (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSession:didDetectPlayoutGlitch:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self didDetectPlayoutGlitch:totalNumberOfGlitches];
+    }
+  }
+}
+
 @end

webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h

@@ -78,6 +78,12 @@ RTC_EXPORT
 - (void)audioSession:(RTCAudioSession *)audioSession
     didChangeOutputVolume:(float)outputVolume;
 
+/** Called when the audio device detects a playout glitch. The argument is the
+ *  number of glitches detected so far in the current audio playout session.
+ */
+- (void)audioSession:(RTCAudioSession *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
 @end
 
 /** This is a protocol used to inform RTCAudioSession when the audio session