Provide isAudioEnabled flag to control audio unit.

webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m

 /*
  *  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 "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
 
+#import "WebRTC/RTCDispatcher.h"
+
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 
 // Try to use mono to save resources. Also avoids channel format conversion
 // in the I/O audio unit. Initial tests have shown that it is possible to use
 // mono natively for built-in microphones and for BT headsets but not for
 // wired headsets. Wired headsets only support stereo as native channel format
 // but it is a low cost operation to do a format conversion to mono in the
 // audio unit. Hence, we will not hit a RTC_CHECK in
 // VerifyAudioParametersForActiveAudioSession() for a mismatch between the
 // preferred number of channels and the actual number of channels.
@@ skipping 19 matching lines @@
 // sequence without bursts of callbacks back to back.
 const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.01;
 
 // Use a larger buffer size on devices with only one core (e.g. iPhone 4).
 // It will result in a lower CPU consumption at the cost of a larger latency.
 // The size of 60ms is based on instrumentation that shows a significant
 // reduction in CPU load compared with 10ms on low-end devices.
 // TODO(henrika): monitor this size and determine if it should be modified.
 const double kRTCAudioSessionLowComplexityIOBufferDuration = 0.06;
 
+static RTCAudioSessionConfiguration *gWebRTCConfiguration = nil;
+
 @implementation RTCAudioSessionConfiguration
 
 @synthesize category = _category;
 @synthesize categoryOptions = _categoryOptions;
 @synthesize mode = _mode;
 @synthesize sampleRate = _sampleRate;
 @synthesize ioBufferDuration = _ioBufferDuration;
 @synthesize inputNumberOfChannels = _inputNumberOfChannels;
 @synthesize outputNumberOfChannels = _outputNumberOfChannels;
 
@@ skipping 27 matching lines @@
     // We try to use mono in both directions to save resources and format
     // conversions in the audio unit. Some devices does only support stereo;
     // e.g. wired headset on iPhone 6.
     // TODO(henrika): add support for stereo if needed.
     _inputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
     _outputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
   }
   return self;
 }
 
++ (void)initialize {
+  gWebRTCConfiguration = [[self alloc] init];
+}
+
 + (instancetype)currentConfiguration {
   RTCAudioSession *session = [RTCAudioSession sharedInstance];
   RTCAudioSessionConfiguration *config =
       [[RTCAudioSessionConfiguration alloc] init];
   config.category = session.category;
   config.categoryOptions = session.categoryOptions;
   config.mode = session.mode;
   config.sampleRate = session.sampleRate;
   config.ioBufferDuration = session.IOBufferDuration;
   config.inputNumberOfChannels = session.inputNumberOfChannels;
   config.outputNumberOfChannels = session.outputNumberOfChannels;
   return config;
 }
 
 + (instancetype)webRTCConfiguration {
-  return [[self alloc] init];
+  @synchronized(self) {
+    return (RTCAudioSessionConfiguration *)gWebRTCConfiguration;
+  }
+}
+
++ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration {
+  @synchronized(self) {
+    gWebRTCConfiguration = configuration;
+  }
 }
 
 @end