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diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.m b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.m
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+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.m
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+/*
+ *  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 "RTCAudioSessionConfiguration.h"
+#import "RTCAudioSession.h"
+
+#import "helpers/RTCDispatcher.h"
+#import "helpers/UIDevice+RTCDevice.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.
+const int kRTCAudioSessionPreferredNumberOfChannels = 1;
+
+// Preferred hardware sample rate (unit is in Hertz). The client sample rate
+// will be set to this value as well to avoid resampling the the audio unit's
+// format converter. Note that, some devices, e.g. BT headsets, only supports
+// 8000Hz as native sample rate.
+const double kRTCAudioSessionHighPerformanceSampleRate = 48000.0;
+
+// A lower sample rate will be used for devices with only one core
+// (e.g. iPhone 4). The goal is to reduce the CPU load of the application.
+const double kRTCAudioSessionLowComplexitySampleRate = 16000.0;
+
+// Use a hardware I/O buffer size (unit is in seconds) that matches the 10ms
+// size used by WebRTC. The exact actual size will differ between devices.
+// Example: using 48kHz on iPhone 6 results in a native buffer size of
+// ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
+// take care of any buffering required to convert between native buffers and
+// buffers used by WebRTC. It is beneficial for the performance if the native
+// size is as an even multiple of 10ms as possible since it results in "clean"
+// callback sequence without bursts of callbacks back to back.
+const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.02;
+
+// 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 RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *gWebRTCConfiguration = nil;
+
+@implementation RTC_OBJC_TYPE (RTCAudioSessionConfiguration)
+
+@synthesize category = _category;
+@synthesize categoryOptions = _categoryOptions;
+@synthesize mode = _mode;
+@synthesize sampleRate = _sampleRate;
+@synthesize ioBufferDuration = _ioBufferDuration;
+@synthesize inputNumberOfChannels = _inputNumberOfChannels;
+@synthesize outputNumberOfChannels = _outputNumberOfChannels;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    // Use a category which supports simultaneous recording and playback.
+    // By default, using this category implies that our app’s audio is
+    // nonmixable, hence activating the session will interrupt any other
+    // audio sessions which are also nonmixable.
+    _category = AVAudioSessionCategoryPlayAndRecord;
+    _categoryOptions = AVAudioSessionCategoryOptionAllowBluetooth;
+
+    // Specify mode for two-way voice communication (e.g. VoIP).
+    _mode = AVAudioSessionModeVoiceChat;
+
+    // Set the session's sample rate or the hardware sample rate.
+    // It is essential that we use the same sample rate as stream format
+    // to ensure that the I/O unit does not have to do sample rate conversion.
+    // Set the preferred audio I/O buffer duration, in seconds.
+    NSUInteger processorCount = [NSProcessInfo processInfo].processorCount;
+    // Use best sample rate and buffer duration if the CPU has more than one
+    // core.
+    if (processorCount > 1 && [UIDevice deviceType] != RTCDeviceTypeIPhone4S) {
+      _sampleRate = kRTCAudioSessionHighPerformanceSampleRate;
+      _ioBufferDuration = kRTCAudioSessionHighPerformanceIOBufferDuration;
+    } else {
+      _sampleRate = kRTCAudioSessionLowComplexitySampleRate;
+      _ioBufferDuration = kRTCAudioSessionLowComplexityIOBufferDuration;
+    }
+
+    // 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 {
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *config =
+      [[RTC_OBJC_TYPE(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 {
+  @synchronized(self) {
+    return (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)gWebRTCConfiguration;
+  }
+}
+
++ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration {
+  @synchronized(self) {
+    gWebRTCConfiguration = configuration;
+  }
+}
+
+@end