Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

sdk/objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm

@@ -0,0 +1,172 @@
+ /*
+ *  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/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSessionConfiguration.h"
+
+#import "WebRTC/RTCLogging.h"
+#import "RTCAudioSession+Private.h"
+
+
+@implementation RTCAudioSession (Configuration)
+
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+                   error:(NSError **)outError {
+  return [self setConfiguration:configuration
+                         active:NO
+                shouldSetActive:NO
+                          error:outError];
+}
+
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError {
+  return [self setConfiguration:configuration
+                         active:active
+                shouldSetActive:YES
+                          error:outError];
+}
+
+#pragma mark - Private
+
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+                  active:(BOOL)active
+         shouldSetActive:(BOOL)shouldSetActive
+                   error:(NSError **)outError {
+  NSParameterAssert(configuration);
+  if (outError) {
+    *outError = nil;
+  }
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+
+  // Provide an error even if there isn't one so we can log it. We will not
+  // return immediately on error in this function and instead try to set
+  // everything we can.
+  NSError *error = nil;
+
+  if (self.category != configuration.category ||
+      self.categoryOptions != configuration.categoryOptions) {
+    NSError *categoryError = nil;
+    if (![self setCategory:configuration.category
+               withOptions:configuration.categoryOptions
+                     error:&categoryError]) {
+      RTCLogError(@"Failed to set category: %@",
+                  categoryError.localizedDescription);
+      error = categoryError;
+    } else {
+      RTCLog(@"Set category to: %@", configuration.category);
+    }
+  }
+
+  if (self.mode != configuration.mode) {
+    NSError *modeError = nil;
+    if (![self setMode:configuration.mode error:&modeError]) {
+      RTCLogError(@"Failed to set mode: %@",
+                  modeError.localizedDescription);
+      error = modeError;
+    } else {
+      RTCLog(@"Set mode to: %@", configuration.mode);
+    }
+  }
+
+  // Sometimes category options don't stick after setting mode.
+  if (self.categoryOptions != configuration.categoryOptions) {
+    NSError *categoryError = nil;
+    if (![self setCategory:configuration.category
+               withOptions:configuration.categoryOptions
+                     error:&categoryError]) {
+      RTCLogError(@"Failed to set category options: %@",
+                  categoryError.localizedDescription);
+      error = categoryError;
+    } else {
+      RTCLog(@"Set category options to: %ld",
+             (long)configuration.categoryOptions);
+    }
+  }
+
+  if (self.preferredSampleRate != configuration.sampleRate) {
+    NSError *sampleRateError = nil;
+    if (![self setPreferredSampleRate:configuration.sampleRate
+                                error:&sampleRateError]) {
+      RTCLogError(@"Failed to set preferred sample rate: %@",
+                  sampleRateError.localizedDescription);
+      error = sampleRateError;
+    } else {
+      RTCLog(@"Set preferred sample rate to: %.2f",
+             configuration.sampleRate);
+    }
+  }
+
+  if (self.preferredIOBufferDuration != configuration.ioBufferDuration) {
+    NSError *bufferDurationError = nil;
+    if (![self setPreferredIOBufferDuration:configuration.ioBufferDuration
+                                      error:&bufferDurationError]) {
+      RTCLogError(@"Failed to set preferred IO buffer duration: %@",
+                  bufferDurationError.localizedDescription);
+      error = bufferDurationError;
+    } else {
+      RTCLog(@"Set preferred IO buffer duration to: %f",
+             configuration.ioBufferDuration);
+    }
+  }
+
+  if (shouldSetActive) {
+    NSError *activeError = nil;
+    if (![self setActive:active error:&activeError]) {
+      RTCLogError(@"Failed to setActive to %d: %@",
+                  active, activeError.localizedDescription);
+      error = activeError;
+    }
+  }
+
+  if (self.isActive &&
+      // TODO(tkchin): Figure out which category/mode numChannels is valid for.
+      [self.mode isEqualToString:AVAudioSessionModeVoiceChat]) {
+    // Try to set the preferred number of hardware audio channels. These calls
+    // must be done after setting the audio session’s category and mode and
+    // activating the session.
+    NSInteger inputNumberOfChannels = configuration.inputNumberOfChannels;
+    if (self.inputNumberOfChannels != inputNumberOfChannels) {
+      NSError *inputChannelsError = nil;
+      if (![self setPreferredInputNumberOfChannels:inputNumberOfChannels
+                                             error:&inputChannelsError]) {
+       RTCLogError(@"Failed to set preferred input number of channels: %@",
+                   inputChannelsError.localizedDescription);
+       error = inputChannelsError;
+      } else {
+        RTCLog(@"Set input number of channels to: %ld",
+               (long)inputNumberOfChannels);
+      }
+    }
+    NSInteger outputNumberOfChannels = configuration.outputNumberOfChannels;
+    if (self.outputNumberOfChannels != outputNumberOfChannels) {
+      NSError *outputChannelsError = nil;
+      if (![self setPreferredOutputNumberOfChannels:outputNumberOfChannels
+                                              error:&outputChannelsError]) {
+        RTCLogError(@"Failed to set preferred output number of channels: %@",
+                    outputChannelsError.localizedDescription);
+        error = outputChannelsError;
+      } else {
+        RTCLog(@"Set output number of channels to: %ld",
+               (long)outputNumberOfChannels);
+      }
+    }
+  }
+
+  if (outError) {
+    *outError = error;
+  }
+
+  return error == nil;
+}
+
+@end

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

@@ -0,0 +1,97 @@
+/*
+ *  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/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
+
+#include <vector>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCAudioSessionConfiguration;
+
+@interface 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<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.
+@property(nonatomic, readonly)
+    std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
+
+- (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

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

@@ -0,0 +1,925 @@
+/*
+ *  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/RTCAudioSession.h"
+
+#import <UIKit/UIKit.h>
+
+#include "webrtc/rtc_base/atomicops.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/criticalsection.h"
+
+#import "WebRTC/RTCAudioSessionConfiguration.h"
+#import "WebRTC/RTCLogging.h"
+
+#import "RTCAudioSession+Private.h"
+
+
+NSString * const kRTCAudioSessionErrorDomain = @"org.webrtc.RTCAudioSession";
+NSInteger const kRTCAudioSessionErrorLockRequired = -1;
+NSInteger const kRTCAudioSessionErrorConfiguration = -2;
+NSString * const kRTCAudioSessionOutputVolumeSelector = @"outputVolume";
+
+// This class needs to be thread-safe because it is accessed from many threads.
+// TODO(tkchin): Consider more granular locking. We're not expecting a lot of
+// lock contention so coarse locks should be fine for now.
+@implementation RTCAudioSession {
+  rtc::CriticalSection _crit;
+  AVAudioSession *_session;
+  volatile int _activationCount;
+  volatile int _lockRecursionCount;
+  volatile int _webRTCSessionCount;
+  BOOL _isActive;
+  BOOL _useManualAudio;
+  BOOL _isAudioEnabled;
+  BOOL _canPlayOrRecord;
+  BOOL _isInterrupted;
+}
+
+@synthesize session = _session;
+@synthesize delegates = _delegates;
+
++ (instancetype)sharedInstance {
+  static dispatch_once_t onceToken;
+  static RTCAudioSession *sharedInstance = nil;
+  dispatch_once(&onceToken, ^{
+    sharedInstance = [[self alloc] init];
+  });
+  return sharedInstance;
+}
+
+- (instancetype)init {
+  if (self = [super init]) {
+    _session = [AVAudioSession sharedInstance];
+
+    NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+    [center addObserver:self
+               selector:@selector(handleInterruptionNotification:)
+                   name:AVAudioSessionInterruptionNotification
+                 object:nil];
+    [center addObserver:self
+               selector:@selector(handleRouteChangeNotification:)
+                   name:AVAudioSessionRouteChangeNotification
+                 object:nil];
+    [center addObserver:self
+               selector:@selector(handleMediaServicesWereLost:)
+                   name:AVAudioSessionMediaServicesWereLostNotification
+                 object:nil];
+    [center addObserver:self
+               selector:@selector(handleMediaServicesWereReset:)
+                   name:AVAudioSessionMediaServicesWereResetNotification
+                 object:nil];
+    // Posted on the main thread when the primary audio from other applications
+    // starts and stops. Foreground applications may use this notification as a
+    // hint to enable or disable audio that is secondary.
+    [center addObserver:self
+               selector:@selector(handleSilenceSecondaryAudioHintNotification:)
+                   name:AVAudioSessionSilenceSecondaryAudioHintNotification
+                 object:nil];
+    // Also track foreground event in order to deal with interruption ended situation.
+    [center addObserver:self
+               selector:@selector(handleApplicationDidBecomeActive:)
+                   name:UIApplicationDidBecomeActiveNotification
+                 object:nil];
+    [_session addObserver:self
+               forKeyPath:kRTCAudioSessionOutputVolumeSelector
+                  options:NSKeyValueObservingOptionNew | NSKeyValueObservingOptionOld
+                  context:nil];
+
+    RTCLog(@"RTCAudioSession (%p): init.", self);
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [[NSNotificationCenter defaultCenter] removeObserver:self];
+  [_session removeObserver:self forKeyPath:kRTCAudioSessionOutputVolumeSelector context:nil];
+  RTCLog(@"RTCAudioSession (%p): dealloc.", self);
+}
+
+- (NSString *)description {
+  NSString *format =
+      @"RTCAudioSession: {\n"
+       "  category: %@\n"
+       "  categoryOptions: %ld\n"
+       "  mode: %@\n"
+       "  isActive: %d\n"
+       "  sampleRate: %.2f\n"
+       "  IOBufferDuration: %f\n"
+       "  outputNumberOfChannels: %ld\n"
+       "  inputNumberOfChannels: %ld\n"
+       "  outputLatency: %f\n"
+       "  inputLatency: %f\n"
+       "  outputVolume: %f\n"
+       "}";
+  NSString *description = [NSString stringWithFormat:format,
+      self.category, (long)self.categoryOptions, self.mode,
+      self.isActive, self.sampleRate, self.IOBufferDuration,
+      self.outputNumberOfChannels, self.inputNumberOfChannels,
+      self.outputLatency, self.inputLatency, self.outputVolume];
+  return description;
+}
+
+- (void)setIsActive:(BOOL)isActive {
+  @synchronized(self) {
+    _isActive = isActive;
+  }
+}
+
+- (BOOL)isActive {
+  @synchronized(self) {
+    return _isActive;
+  }
+}
+
+- (BOOL)isLocked {
+  return _lockRecursionCount > 0;
+}
+
+- (void)setUseManualAudio:(BOOL)useManualAudio {
+  @synchronized(self) {
+    if (_useManualAudio == useManualAudio) {
+      return;
+    }
+    _useManualAudio = useManualAudio;
+  }
+  [self updateCanPlayOrRecord];
+}
+
+- (BOOL)useManualAudio {
+  @synchronized(self) {
+    return _useManualAudio;
+  }
+}
+
+- (void)setIsAudioEnabled:(BOOL)isAudioEnabled {
+  @synchronized(self) {
+    if (_isAudioEnabled == isAudioEnabled) {
+      return;
+    }
+    _isAudioEnabled = isAudioEnabled;
+  }
+  [self updateCanPlayOrRecord];
+}
+
+- (BOOL)isAudioEnabled {
+  @synchronized(self) {
+    return _isAudioEnabled;
+  }
+}
+
+// TODO(tkchin): Check for duplicates.
+- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate {
+  RTCLog(@"Adding delegate: (%p)", delegate);
+  if (!delegate) {
+    return;
+  }
+  @synchronized(self) {
+    _delegates.push_back(delegate);
+    [self removeZeroedDelegates];
+  }
+}
+
+- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate {
+  RTCLog(@"Removing delegate: (%p)", delegate);
+  if (!delegate) {
+    return;
+  }
+  @synchronized(self) {
+    _delegates.erase(std::remove(_delegates.begin(),
+                                 _delegates.end(),
+                                 delegate),
+                     _delegates.end());
+    [self removeZeroedDelegates];
+  }
+}
+
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wthread-safety-analysis"
+
+- (void)lockForConfiguration {
+  _crit.Enter();
+  rtc::AtomicOps::Increment(&_lockRecursionCount);
+}
+
+- (void)unlockForConfiguration {
+  // Don't let threads other than the one that called lockForConfiguration
+  // unlock.
+  if (_crit.TryEnter()) {
+    rtc::AtomicOps::Decrement(&_lockRecursionCount);
+    // One unlock for the tryLock, and another one to actually unlock. If this
+    // was called without anyone calling lock, we will hit an assertion.
+    _crit.Leave();
+    _crit.Leave();
+  }
+}
+
+#pragma clang diagnostic pop
+
+#pragma mark - AVAudioSession proxy methods
+
+- (NSString *)category {
+  return self.session.category;
+}
+
+- (AVAudioSessionCategoryOptions)categoryOptions {
+  return self.session.categoryOptions;
+}
+
+- (NSString *)mode {
+  return self.session.mode;
+}
+
+- (BOOL)secondaryAudioShouldBeSilencedHint {
+  return self.session.secondaryAudioShouldBeSilencedHint;
+}
+
+- (AVAudioSessionRouteDescription *)currentRoute {
+  return self.session.currentRoute;
+}
+
+- (NSInteger)maximumInputNumberOfChannels {
+  return self.session.maximumInputNumberOfChannels;
+}
+
+- (NSInteger)maximumOutputNumberOfChannels {
+  return self.session.maximumOutputNumberOfChannels;
+}
+
+- (float)inputGain {
+  return self.session.inputGain;
+}
+
+- (BOOL)inputGainSettable {
+  return self.session.inputGainSettable;
+}
+
+- (BOOL)inputAvailable {
+  return self.session.inputAvailable;
+}
+
+- (NSArray<AVAudioSessionDataSourceDescription *> *)inputDataSources {
+  return self.session.inputDataSources;
+}
+
+- (AVAudioSessionDataSourceDescription *)inputDataSource {
+  return self.session.inputDataSource;
+}
+
+- (NSArray<AVAudioSessionDataSourceDescription *> *)outputDataSources {
+  return self.session.outputDataSources;
+}
+
+- (AVAudioSessionDataSourceDescription *)outputDataSource {
+  return self.session.outputDataSource;
+}
+
+- (double)sampleRate {
+  return self.session.sampleRate;
+}
+
+- (double)preferredSampleRate {
+  return self.session.preferredSampleRate;
+}
+
+- (NSInteger)inputNumberOfChannels {
+  return self.session.inputNumberOfChannels;
+}
+
+- (NSInteger)outputNumberOfChannels {
+  return self.session.outputNumberOfChannels;
+}
+
+- (float)outputVolume {
+  return self.session.outputVolume;
+}
+
+- (NSTimeInterval)inputLatency {
+  return self.session.inputLatency;
+}
+
+- (NSTimeInterval)outputLatency {
+  return self.session.outputLatency;
+}
+
+- (NSTimeInterval)IOBufferDuration {
+  return self.session.IOBufferDuration;
+}
+
+- (NSTimeInterval)preferredIOBufferDuration {
+  return self.session.preferredIOBufferDuration;
+}
+
+// TODO(tkchin): Simplify the amount of locking happening here. Likely that we
+// can just do atomic increments / decrements.
+- (BOOL)setActive:(BOOL)active
+            error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  int activationCount = _activationCount;
+  if (!active && activationCount == 0) {
+    RTCLogWarning(@"Attempting to deactivate without prior activation.");
+  }
+  BOOL success = YES;
+  BOOL isActive = self.isActive;
+  // Keep a local error so we can log it.
+  NSError *error = nil;
+  BOOL shouldSetActive =
+      (active && !isActive) || (!active && isActive && activationCount == 1);
+  // Attempt to activate if we're not active.
+  // Attempt to deactivate if we're active and it's the last unbalanced call.
+  if (shouldSetActive) {
+    AVAudioSession *session = self.session;
+    // AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation is used to ensure
+    // that other audio sessions that were interrupted by our session can return
+    // to their active state. It is recommended for VoIP apps to use this
+    // option.
+    AVAudioSessionSetActiveOptions options =
+        active ? 0 : AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation;
+    success = [session setActive:active
+                     withOptions:options
+                           error:&error];
+    if (outError) {
+      *outError = error;
+    }
+  }
+  if (success) {
+    if (shouldSetActive) {
+      self.isActive = active;
+    }
+    if (active) {
+      [self incrementActivationCount];
+    }
+  } else {
+    RTCLogError(@"Failed to setActive:%d. Error: %@",
+                active, error.localizedDescription);
+  }
+  // Decrement activation count on deactivation whether or not it succeeded.
+  if (!active) {
+    [self decrementActivationCount];
+  }
+  RTCLog(@"Number of current activations: %d", _activationCount);
+  return success;
+}
+
+- (BOOL)setCategory:(NSString *)category
+        withOptions:(AVAudioSessionCategoryOptions)options
+              error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setCategory:category withOptions:options error:outError];
+}
+
+- (BOOL)setMode:(NSString *)mode error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setMode:mode error:outError];
+}
+
+- (BOOL)setInputGain:(float)gain error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setInputGain:gain error:outError];
+}
+
+- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredSampleRate:sampleRate error:outError];
+}
+
+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
+                               error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredIOBufferDuration:duration error:outError];
+}
+
+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
+                                    error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredInputNumberOfChannels:count error:outError];
+}
+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
+                                     error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredOutputNumberOfChannels:count error:outError];
+}
+
+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
+                          error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session overrideOutputAudioPort:portOverride error:outError];
+}
+
+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
+                    error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredInput:inPort error:outError];
+}
+
+- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                     error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setInputDataSource:dataSource error:outError];
+}
+
+- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                      error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setOutputDataSource:dataSource error:outError];
+}
+
+#pragma mark - Notifications
+
+- (void)handleInterruptionNotification:(NSNotification *)notification {
+  NSNumber* typeNumber =
+      notification.userInfo[AVAudioSessionInterruptionTypeKey];
+  AVAudioSessionInterruptionType type =
+      (AVAudioSessionInterruptionType)typeNumber.unsignedIntegerValue;
+  switch (type) {
+    case AVAudioSessionInterruptionTypeBegan:
+      RTCLog(@"Audio session interruption began.");
+      self.isActive = NO;
+      self.isInterrupted = YES;
+      [self notifyDidBeginInterruption];
+      break;
+    case AVAudioSessionInterruptionTypeEnded: {
+      RTCLog(@"Audio session interruption ended.");
+      self.isInterrupted = NO;
+      [self updateAudioSessionAfterEvent];
+      NSNumber *optionsNumber =
+          notification.userInfo[AVAudioSessionInterruptionOptionKey];
+      AVAudioSessionInterruptionOptions options =
+          optionsNumber.unsignedIntegerValue;
+      BOOL shouldResume =
+          options & AVAudioSessionInterruptionOptionShouldResume;
+      [self notifyDidEndInterruptionWithShouldResumeSession:shouldResume];
+      break;
+    }
+  }
+}
+
+- (void)handleRouteChangeNotification:(NSNotification *)notification {
+  // Get reason for current route change.
+  NSNumber* reasonNumber =
+      notification.userInfo[AVAudioSessionRouteChangeReasonKey];
+  AVAudioSessionRouteChangeReason reason =
+      (AVAudioSessionRouteChangeReason)reasonNumber.unsignedIntegerValue;
+  RTCLog(@"Audio route changed:");
+  switch (reason) {
+    case AVAudioSessionRouteChangeReasonUnknown:
+      RTCLog(@"Audio route changed: ReasonUnknown");
+      break;
+    case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
+      RTCLog(@"Audio route changed: NewDeviceAvailable");
+      break;
+    case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
+      RTCLog(@"Audio route changed: OldDeviceUnavailable");
+      break;
+    case AVAudioSessionRouteChangeReasonCategoryChange:
+      RTCLog(@"Audio route changed: CategoryChange to :%@",
+             self.session.category);
+      break;
+    case AVAudioSessionRouteChangeReasonOverride:
+      RTCLog(@"Audio route changed: Override");
+      break;
+    case AVAudioSessionRouteChangeReasonWakeFromSleep:
+      RTCLog(@"Audio route changed: WakeFromSleep");
+      break;
+    case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
+      RTCLog(@"Audio route changed: NoSuitableRouteForCategory");
+      break;
+    case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
+      RTCLog(@"Audio route changed: RouteConfigurationChange");
+      break;
+  }
+  AVAudioSessionRouteDescription* previousRoute =
+      notification.userInfo[AVAudioSessionRouteChangePreviousRouteKey];
+  // Log previous route configuration.
+  RTCLog(@"Previous route: %@\nCurrent route:%@",
+         previousRoute, self.session.currentRoute);
+  [self notifyDidChangeRouteWithReason:reason previousRoute:previousRoute];
+}
+
+- (void)handleMediaServicesWereLost:(NSNotification *)notification {
+  RTCLog(@"Media services were lost.");
+  [self updateAudioSessionAfterEvent];
+  [self notifyMediaServicesWereLost];
+}
+
+- (void)handleMediaServicesWereReset:(NSNotification *)notification {
+  RTCLog(@"Media services were reset.");
+  [self updateAudioSessionAfterEvent];
+  [self notifyMediaServicesWereReset];
+}
+
+- (void)handleSilenceSecondaryAudioHintNotification:(NSNotification *)notification {
+  // TODO(henrika): just adding logs here for now until we know if we are ever
+  // see this notification and might be affected by it or if further actions
+  // are required.
+  NSNumber *typeNumber =
+      notification.userInfo[AVAudioSessionSilenceSecondaryAudioHintTypeKey];
+  AVAudioSessionSilenceSecondaryAudioHintType type =
+      (AVAudioSessionSilenceSecondaryAudioHintType)typeNumber.unsignedIntegerValue;
+  switch (type) {
+    case AVAudioSessionSilenceSecondaryAudioHintTypeBegin:
+      RTCLog(@"Another application's primary audio has started.");
+      break;
+    case AVAudioSessionSilenceSecondaryAudioHintTypeEnd:
+      RTCLog(@"Another application's primary audio has stopped.");
+      break;
+  }
+}
+
+- (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
+  RTCLog(@"Application became active after an interruption. Treating as interruption "
+         " end. isInterrupted changed from %d to 0.", self.isInterrupted);
+  if (self.isInterrupted) {
+    self.isInterrupted = NO;
+    [self updateAudioSessionAfterEvent];
+  }
+  // Always treat application becoming active as an interruption end event.
+  [self notifyDidEndInterruptionWithShouldResumeSession:YES];
+}
+
+#pragma mark - Private
+
++ (NSError *)lockError {
+  NSDictionary *userInfo = @{
+    NSLocalizedDescriptionKey:
+        @"Must call lockForConfiguration before calling this method."
+  };
+  NSError *error =
+      [[NSError alloc] initWithDomain:kRTCAudioSessionErrorDomain
+                                 code:kRTCAudioSessionErrorLockRequired
+                             userInfo:userInfo];
+  return error;
+}
+
+- (std::vector<__weak id<RTCAudioSessionDelegate> >)delegates {
+  @synchronized(self) {
+    // Note: this returns a copy.
+    return _delegates;
+  }
+}
+
+// TODO(tkchin): check for duplicates.
+- (void)pushDelegate:(id<RTCAudioSessionDelegate>)delegate {
+  @synchronized(self) {
+    _delegates.insert(_delegates.begin(), delegate);
+  }
+}
+
+- (void)removeZeroedDelegates {
+  @synchronized(self) {
+    _delegates.erase(
+        std::remove_if(_delegates.begin(),
+                       _delegates.end(),
+                       [](id delegate) -> bool { return delegate == nil; }),
+        _delegates.end());
+  }
+}
+
+- (int)activationCount {
+  return _activationCount;
+}
+
+- (int)incrementActivationCount {
+  RTCLog(@"Incrementing activation count.");
+  return rtc::AtomicOps::Increment(&_activationCount);
+}
+
+- (NSInteger)decrementActivationCount {
+  RTCLog(@"Decrementing activation count.");
+  return rtc::AtomicOps::Decrement(&_activationCount);
+}
+
+- (int)webRTCSessionCount {
+  return _webRTCSessionCount;
+}
+
+- (BOOL)canPlayOrRecord {
+  return !self.useManualAudio || self.isAudioEnabled;
+}
+
+- (BOOL)isInterrupted {
+  @synchronized(self) {
+    return _isInterrupted;
+  }
+}
+
+- (void)setIsInterrupted:(BOOL)isInterrupted {
+  @synchronized(self) {
+    if (_isInterrupted == isInterrupted) {
+      return;
+   }
+   _isInterrupted = isInterrupted;
+  }
+}
+
+- (BOOL)checkLock:(NSError **)outError {
+  // Check ivar instead of trying to acquire lock so that we won't accidentally
+  // acquire lock if it hasn't already been called.
+  if (!self.isLocked) {
+    if (outError) {
+      *outError = [RTCAudioSession lockError];
+    }
+    return NO;
+  }
+  return YES;
+}
+
+- (BOOL)beginWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  rtc::AtomicOps::Increment(&_webRTCSessionCount);
+  [self notifyDidStartPlayOrRecord];
+  return YES;
+}
+
+- (BOOL)endWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  rtc::AtomicOps::Decrement(&_webRTCSessionCount);
+  [self notifyDidStopPlayOrRecord];
+  return YES;
+}
+
+- (BOOL)configureWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  RTCLog(@"Configuring audio session for WebRTC.");
+
+  // Configure the AVAudioSession and activate it.
+  // Provide an error even if there isn't one so we can log it.
+  NSError *error = nil;
+  RTCAudioSessionConfiguration *webRTCConfig =
+      [RTCAudioSessionConfiguration webRTCConfiguration];
+  if (![self setConfiguration:webRTCConfig active:YES error:&error]) {
+    RTCLogError(@"Failed to set WebRTC audio configuration: %@",
+                error.localizedDescription);
+    // Do not call setActive:NO if setActive:YES failed.
+    if (outError) {
+      *outError = error;
+    }
+    return NO;
+  }
+
+  // Ensure that the device currently supports audio input.
+  // TODO(tkchin): Figure out if this is really necessary.
+  if (!self.inputAvailable) {
+    RTCLogError(@"No audio input path is available!");
+    [self unconfigureWebRTCSession:nil];
+    if (outError) {
+      *outError = [self configurationErrorWithDescription:@"No input path."];
+    }
+    return NO;
+  }
+
+  // It can happen (e.g. in combination with BT devices) that the attempt to set
+  // the preferred sample rate for WebRTC (48kHz) fails. If so, make a new
+  // configuration attempt using the sample rate that worked using the active
+  // audio session. A typical case is that only 8 or 16kHz can be set, e.g. in
+  // combination with BT headsets. Using this "trick" seems to avoid a state
+  // where Core Audio asks for a different number of audio frames than what the
+  // session's I/O buffer duration corresponds to.
+  // TODO(henrika): this fix resolves bugs.webrtc.org/6004 but it has only been
+  // tested on a limited set of iOS devices and BT devices.
+  double sessionSampleRate = self.sampleRate;
+  double preferredSampleRate = webRTCConfig.sampleRate;
+  if (sessionSampleRate != preferredSampleRate) {
+    RTCLogWarning(
+        @"Current sample rate (%.2f) is not the preferred rate (%.2f)",
+        sessionSampleRate, preferredSampleRate);
+    if (![self setPreferredSampleRate:sessionSampleRate
+                                error:&error]) {
+      RTCLogError(@"Failed to set preferred sample rate: %@",
+                  error.localizedDescription);
+      if (outError) {
+        *outError = error;
+      }
+    }
+  }
+
+  return YES;
+}
+
+- (BOOL)unconfigureWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  RTCLog(@"Unconfiguring audio session for WebRTC.");
+  [self setActive:NO error:outError];
+
+  return YES;
+}
+
+- (NSError *)configurationErrorWithDescription:(NSString *)description {
+  NSDictionary* userInfo = @{
+    NSLocalizedDescriptionKey: description,
+  };
+  return [[NSError alloc] initWithDomain:kRTCAudioSessionErrorDomain
+                                    code:kRTCAudioSessionErrorConfiguration
+                                userInfo:userInfo];
+}
+
+- (void)updateAudioSessionAfterEvent {
+  BOOL shouldActivate = self.activationCount > 0;
+  AVAudioSessionSetActiveOptions options = shouldActivate ?
+      0 : AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation;
+  NSError *error = nil;
+  if ([self.session setActive:shouldActivate
+                  withOptions:options
+                        error:&error]) {
+    self.isActive = shouldActivate;
+  } else {
+    RTCLogError(@"Failed to set session active to %d. Error:%@",
+                shouldActivate, error.localizedDescription);
+  }
+}
+
+- (void)updateCanPlayOrRecord {
+  BOOL canPlayOrRecord = NO;
+  BOOL shouldNotify = NO;
+  @synchronized(self) {
+    canPlayOrRecord = !self.useManualAudio || self.isAudioEnabled;
+    if (_canPlayOrRecord == canPlayOrRecord) {
+      return;
+    }
+    _canPlayOrRecord = canPlayOrRecord;
+    shouldNotify = YES;
+  }
+  if (shouldNotify) {
+    [self notifyDidChangeCanPlayOrRecord:canPlayOrRecord];
+  }
+}
+
+- (void)audioSessionDidActivate:(AVAudioSession *)session {
+  if (_session != session) {
+    RTCLogError(@"audioSessionDidActivate called on different AVAudioSession");
+  }
+  [self incrementActivationCount];
+  self.isActive = YES;
+}
+
+- (void)audioSessionDidDeactivate:(AVAudioSession *)session {
+  if (_session != session) {
+    RTCLogError(@"audioSessionDidDeactivate called on different AVAudioSession");
+  }
+  self.isActive = NO;
+  [self decrementActivationCount];
+}
+
+- (void)observeValueForKeyPath:(NSString *)keyPath
+                      ofObject:(id)object
+                        change:(NSDictionary *)change
+                       context:(void *)context {
+  if (object == _session) {
+    NSNumber *newVolume = change[NSKeyValueChangeNewKey];
+    RTCLog(@"OutputVolumeDidChange to %f", newVolume.floatValue);
+    [self notifyDidChangeOutputVolume:newVolume.floatValue];
+  } else {
+    [super observeValueForKeyPath:keyPath
+                         ofObject:object
+                           change:change
+                          context:context];
+  }
+}
+
+- (void)notifyDidBeginInterruption {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidBeginInterruption:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidBeginInterruption:self];
+    }
+  }
+}
+
+- (void)notifyDidEndInterruptionWithShouldResumeSession:
+    (BOOL)shouldResumeSession {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidEndInterruption:shouldResumeSession:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidEndInterruption:self
+                           shouldResumeSession:shouldResumeSession];
+    }
+  }
+}
+
+- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
+    previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidChangeRoute:reason:previousRoute:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidChangeRoute:self
+                                    reason:reason
+                             previousRoute:previousRoute];
+    }
+  }
+}
+
+- (void)notifyMediaServicesWereLost {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionMediaServerTerminated:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionMediaServerTerminated:self];
+    }
+  }
+}
+
+- (void)notifyMediaServicesWereReset {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionMediaServerReset:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionMediaServerReset:self];
+    }
+  }
+}
+
+- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSession:didChangeCanPlayOrRecord:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self didChangeCanPlayOrRecord:canPlayOrRecord];
+    }
+  }
+}
+
+- (void)notifyDidStartPlayOrRecord {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidStartPlayOrRecord:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidStartPlayOrRecord:self];
+    }
+  }
+}
+
+- (void)notifyDidStopPlayOrRecord {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidStopPlayOrRecord:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidStopPlayOrRecord:self];
+    }
+  }
+}
+
+- (void)notifyDidChangeOutputVolume:(float)volume {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSession:didChangeOutputVolume:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self didChangeOutputVolume:volume];
+    }
+  }
+}
+
+- (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

sdk/objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m

@@ -0,0 +1,134 @@
+/*
+ *  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/RTCAudioSession.h"
+#import "WebRTC/RTCAudioSessionConfiguration.h"
+
+#import "WebRTC/RTCDispatcher.h"
+#import "WebRTC/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 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;
+
+- (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 {
+  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 {
+  @synchronized(self) {
+    return (RTCAudioSessionConfiguration *)gWebRTCConfiguration;
+  }
+}
+
++ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration {
+  @synchronized(self) {
+    gWebRTCConfiguration = configuration;
+  }
+}
+
+@end

sdk/objc/Framework/Classes/Common/NSString+StdString.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2015 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>
+
+#include <string>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface NSString (StdString)
+
+@property(nonatomic, readonly) std::string stdString;
+
++ (std::string)stdStringForString:(NSString *)nsString;
++ (NSString *)stringForStdString:(const std::string&)stdString;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/Common/NSString+StdString.mm

@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2015 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 "NSString+StdString.h"
+
+@implementation NSString (StdString)
+
+- (std::string)stdString {
+  return [NSString stdStringForString:self];
+}
+
++ (std::string)stdStringForString:(NSString *)nsString {
+  NSData *charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
+  return std::string(reinterpret_cast<const char *>(charData.bytes),
+                     charData.length);
+}
+
++ (NSString *)stringForStdString:(const std::string&)stdString {
+  // std::string may contain null termination character so we construct
+  // using length.
+  return [[NSString alloc] initWithBytes:stdString.data()
+                                  length:stdString.length()
+                                encoding:NSUTF8StringEncoding];
+}
+
+@end

sdk/objc/Framework/Classes/Common/RTCDispatcher+Private.h

@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2015 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/RTCDispatcher.h"
+
+@interface RTCDispatcher ()
+
++ (dispatch_queue_t)dispatchQueueForType:(RTCDispatcherQueueType)dispatchType;
+
+@end

sdk/objc/Framework/Classes/Common/RTCDispatcher.m

@@ -0,0 +1,61 @@
+/*
+ *  Copyright 2015 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 "RTCDispatcher+Private.h"
+
+static dispatch_queue_t kAudioSessionQueue = nil;
+static dispatch_queue_t kCaptureSessionQueue = nil;
+
+@implementation RTCDispatcher
+
++ (void)initialize {
+  static dispatch_once_t onceToken;
+  dispatch_once(&onceToken, ^{
+    kAudioSessionQueue = dispatch_queue_create(
+        "org.webrtc.RTCDispatcherAudioSession",
+        DISPATCH_QUEUE_SERIAL);
+    kCaptureSessionQueue = dispatch_queue_create(
+        "org.webrtc.RTCDispatcherCaptureSession",
+        DISPATCH_QUEUE_SERIAL);
+  });
+}
+
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType
+                      block:(dispatch_block_t)block {
+  dispatch_queue_t queue = [self dispatchQueueForType:dispatchType];
+  dispatch_async(queue, block);
+}
+
++ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType {
+  dispatch_queue_t targetQueue = [self dispatchQueueForType:dispatchType];
+  const char* targetLabel = dispatch_queue_get_label(targetQueue);
+  const char* currentLabel = dispatch_queue_get_label(DISPATCH_CURRENT_QUEUE_LABEL);
+
+  NSAssert(strlen(targetLabel) > 0, @"Label is required for the target queue.");
+  NSAssert(strlen(currentLabel) > 0, @"Label is required for the current queue.");
+
+  return strcmp(targetLabel, currentLabel) == 0;
+}
+
+#pragma mark - Private
+
++ (dispatch_queue_t)dispatchQueueForType:(RTCDispatcherQueueType)dispatchType {
+  switch (dispatchType) {
+    case RTCDispatcherTypeMain:
+      return dispatch_get_main_queue();
+    case RTCDispatcherTypeCaptureSession:
+      return kCaptureSessionQueue;
+    case RTCDispatcherTypeAudioSession:
+      return kAudioSessionQueue;
+  }
+}
+
+@end
+

sdk/objc/Framework/Classes/Common/RTCFieldTrials.mm

@@ -0,0 +1,69 @@
+/*
+ *  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/RTCFieldTrials.h"
+
+#include <memory>
+
+#import "WebRTC/RTCLogging.h"
+
+// Adding 'nogncheck' to disable the gn include headers check.
+// We don't want to depend on 'system_wrappers:field_trial_default' because
+// clients should be able to provide their own implementation.
+#include "webrtc/system_wrappers/include/field_trial_default.h"  // nogncheck
+
+NSString * const kRTCFieldTrialAudioSendSideBweKey = @"WebRTC-Audio-SendSideBwe";
+NSString * const kRTCFieldTrialSendSideBweWithOverheadKey = @"WebRTC-SendSideBwe-WithOverhead";
+NSString * const kRTCFieldTrialFlexFec03AdvertisedKey = @"WebRTC-FlexFEC-03-Advertised";
+NSString * const kRTCFieldTrialFlexFec03Key = @"WebRTC-FlexFEC-03";
+NSString * const kRTCFieldTrialImprovedBitrateEstimateKey = @"WebRTC-ImprovedBitrateEstimate";
+NSString * const kRTCFieldTrialMedianSlopeFilterKey = @"WebRTC-BweMedianSlopeFilter";
+NSString * const kRTCFieldTrialTrendlineFilterKey = @"WebRTC-BweTrendlineFilter";
+NSString * const kRTCFieldTrialH264HighProfileKey = @"WebRTC-H264HighProfile";
+NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey =
+    @"WebRTC-Audio-MinimizeResamplingOnMobile";
+NSString * const kRTCFieldTrialEnabledValue = @"Enabled";
+
+static std::unique_ptr<char[]> gFieldTrialInitString;
+
+NSString *RTCFieldTrialMedianSlopeFilterValue(
+    size_t windowSize, double thresholdGain) {
+  NSString *format = @"Enabled-%zu,%lf";
+  return [NSString stringWithFormat:format, windowSize, thresholdGain];
+}
+
+NSString *RTCFieldTrialTrendlineFilterValue(
+    size_t windowSize, double smoothingCoeff, double thresholdGain) {
+  NSString *format = @"Enabled-%zu,%lf,%lf";
+  return [NSString stringWithFormat:format, windowSize, smoothingCoeff, thresholdGain];
+}
+
+void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials) {
+  if (!fieldTrials) {
+    RTCLogWarning(@"No fieldTrials provided.");
+    return;
+  }
+  // Assemble the keys and values into the field trial string.
+  // We don't perform any extra format checking. That should be done by the underlying WebRTC calls.
+  NSMutableString *fieldTrialInitString = [NSMutableString string];
+  for (NSString *key in fieldTrials) {
+    NSString *fieldTrialEntry = [NSString stringWithFormat:@"%@/%@/", key, fieldTrials[key]];
+    [fieldTrialInitString appendString:fieldTrialEntry];
+  }
+  size_t len = fieldTrialInitString.length + 1;
+  gFieldTrialInitString.reset(new char[len]);
+  if (![fieldTrialInitString getCString:gFieldTrialInitString.get()
+                              maxLength:len
+                               encoding:NSUTF8StringEncoding]) {
+    RTCLogError(@"Failed to convert field trial string.");
+    return;
+  }
+  webrtc::field_trial::InitFieldTrialsFromString(gFieldTrialInitString.get());
+}

sdk/objc/Framework/Classes/Common/RTCFileLogger.mm

@@ -0,0 +1,176 @@
+/*
+ *  Copyright 2015 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/RTCFileLogger.h"
+
+#include <memory>
+
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/filerotatingstream.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/logsinks.h"
+
+NSString *const kDefaultLogDirName = @"webrtc_logs";
+NSUInteger const kDefaultMaxFileSize = 10 * 1024 * 1024; // 10MB.
+const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
+
+@implementation RTCFileLogger {
+  BOOL _hasStarted;
+  NSString *_dirPath;
+  NSUInteger _maxFileSize;
+  std::unique_ptr<rtc::FileRotatingLogSink> _logSink;
+}
+
+@synthesize severity = _severity;
+@synthesize rotationType = _rotationType;
+@synthesize shouldDisableBuffering = _shouldDisableBuffering;
+
+- (instancetype)init {
+  NSArray *paths = NSSearchPathForDirectoriesInDomains(
+      NSDocumentDirectory, NSUserDomainMask, YES);
+  NSString *documentsDirPath = [paths firstObject];
+  NSString *defaultDirPath =
+      [documentsDirPath stringByAppendingPathComponent:kDefaultLogDirName];
+  return [self initWithDirPath:defaultDirPath
+                   maxFileSize:kDefaultMaxFileSize];
+}
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize {
+  return [self initWithDirPath:dirPath
+                   maxFileSize:maxFileSize
+                  rotationType:RTCFileLoggerTypeCall];
+}
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize
+                   rotationType:(RTCFileLoggerRotationType)rotationType {
+  NSParameterAssert(dirPath.length);
+  NSParameterAssert(maxFileSize);
+  if (self = [super init]) {
+    BOOL isDir = NO;
+    NSFileManager *fileManager = [NSFileManager defaultManager];
+    if ([fileManager fileExistsAtPath:dirPath isDirectory:&isDir]) {
+      if (!isDir) {
+        // Bail if something already exists there.
+        return nil;
+      }
+    } else {
+      if (![fileManager createDirectoryAtPath:dirPath
+                  withIntermediateDirectories:NO
+                                   attributes:nil
+                                        error:nil]) {
+        // Bail if we failed to create a directory.
+        return nil;
+      }
+    }
+    _dirPath = dirPath;
+    _maxFileSize = maxFileSize;
+    _severity = RTCFileLoggerSeverityInfo;
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self stop];
+}
+
+- (void)start {
+  if (_hasStarted) {
+    return;
+  }
+  switch (_rotationType) {
+    case RTCFileLoggerTypeApp:
+      _logSink.reset(
+          new rtc::FileRotatingLogSink(_dirPath.UTF8String,
+                                       kRTCFileLoggerRotatingLogPrefix,
+                                       _maxFileSize,
+                                       _maxFileSize / 10));
+      break;
+    case RTCFileLoggerTypeCall:
+      _logSink.reset(
+          new rtc::CallSessionFileRotatingLogSink(_dirPath.UTF8String,
+                                                  _maxFileSize));
+      break;
+  }
+  if (!_logSink->Init()) {
+    LOG(LS_ERROR) << "Failed to open log files at path: "
+                  << _dirPath.UTF8String;
+    _logSink.reset();
+    return;
+  }
+  if (_shouldDisableBuffering) {
+    _logSink->DisableBuffering();
+  }
+  rtc::LogMessage::LogThreads(true);
+  rtc::LogMessage::LogTimestamps(true);
+  rtc::LogMessage::AddLogToStream(_logSink.get(), [self rtcSeverity]);
+  _hasStarted = YES;
+}
+
+- (void)stop {
+  if (!_hasStarted) {
+    return;
+  }
+  RTC_DCHECK(_logSink);
+  rtc::LogMessage::RemoveLogToStream(_logSink.get());
+  _hasStarted = NO;
+  _logSink.reset();
+}
+
+- (NSData *)logData {
+  if (_hasStarted) {
+    return nil;
+  }
+  NSMutableData* logData = [NSMutableData data];
+  std::unique_ptr<rtc::FileRotatingStream> stream;
+  switch(_rotationType) {
+    case RTCFileLoggerTypeApp:
+      stream.reset(
+          new rtc::FileRotatingStream(_dirPath.UTF8String,
+                                      kRTCFileLoggerRotatingLogPrefix));
+      break;
+    case RTCFileLoggerTypeCall:
+      stream.reset(new rtc::CallSessionFileRotatingStream(_dirPath.UTF8String));
+      break;
+  }
+  if (!stream->Open()) {
+    return logData;
+  }
+  size_t bufferSize = 0;
+  if (!stream->GetSize(&bufferSize) || bufferSize == 0) {
+    return logData;
+  }
+  size_t read = 0;
+  // Allocate memory using malloc so we can pass it direcly to NSData without
+  // copying.
+  std::unique_ptr<uint8_t[]> buffer(static_cast<uint8_t*>(malloc(bufferSize)));
+  stream->ReadAll(buffer.get(), bufferSize, &read, nullptr);
+  logData = [[NSMutableData alloc] initWithBytesNoCopy:buffer.release()
+                                                length:read];
+  return logData;
+}
+
+#pragma mark - Private
+
+- (rtc::LoggingSeverity)rtcSeverity {
+  switch (_severity) {
+    case RTCFileLoggerSeverityVerbose:
+      return rtc::LS_VERBOSE;
+    case RTCFileLoggerSeverityInfo:
+      return rtc::LS_INFO;
+    case RTCFileLoggerSeverityWarning:
+      return rtc::LS_WARNING;
+    case RTCFileLoggerSeverityError:
+      return rtc::LS_ERROR;
+  }
+}
+
+@end

sdk/objc/Framework/Classes/Common/RTCLogging.mm

@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2015 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/RTCLogging.h"
+
+#include "webrtc/rtc_base/logging.h"
+
+rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
+  switch (severity) {
+    case RTCLoggingSeverityVerbose:
+      return rtc::LS_VERBOSE;
+    case RTCLoggingSeverityInfo:
+      return rtc::LS_INFO;
+    case RTCLoggingSeverityWarning:
+      return rtc::LS_WARNING;
+    case RTCLoggingSeverityError:
+      return rtc::LS_ERROR;
+  }
+}
+
+void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string) {
+  if (log_string.length) {
+    const char* utf8_string = log_string.UTF8String;
+    LOG_V(RTCGetNativeLoggingSeverity(severity)) << utf8_string;
+  }
+}
+
+void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity) {
+  rtc::LogMessage::LogToDebug(RTCGetNativeLoggingSeverity(severity));
+}
+
+NSString* RTCFileName(const char* file_path) {
+  NSString* ns_file_path =
+      [[NSString alloc] initWithBytesNoCopy:const_cast<char*>(file_path)
+                                     length:strlen(file_path)
+                                   encoding:NSUTF8StringEncoding
+                               freeWhenDone:NO];
+  return ns_file_path.lastPathComponent;
+}
+

sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h

@@ -0,0 +1,24 @@
+/*
+ *  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.
+ */
+
+#if defined(WEBRTC_IOS)
+
+#import <Foundation/Foundation.h>
+
+@interface RTCUIApplicationStatusObserver : NSObject
+
++ (instancetype)sharedInstance;
++ (void)prepareForUse;
+
+- (BOOL)isApplicationActive;
+
+@end
+
+#endif  // WEBRTC_IOS

sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.m

@@ -0,0 +1,114 @@
+/*
+ *  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.
+ */
+
+#include "RTCUIApplicationStatusObserver.h"
+
+#if defined(WEBRTC_IOS)
+
+#import <UIKit/UIKit.h>
+
+#include "webrtc/rtc_base/checks.h"
+
+@interface RTCUIApplicationStatusObserver ()
+
+@property(nonatomic, assign) BOOL initialized;
+@property(nonatomic, assign) UIApplicationState state;
+
+@end
+
+@implementation RTCUIApplicationStatusObserver {
+  BOOL _initialized;
+  dispatch_block_t _initializeBlock;
+  dispatch_semaphore_t _waitForInitializeSemaphore;
+  UIApplicationState _state;
+
+  id<NSObject> _activeObserver;
+  id<NSObject> _backgroundObserver;
+}
+
+@synthesize initialized = _initialized;
+@synthesize state = _state;
+
++ (instancetype)sharedInstance {
+  static id sharedInstance;
+  static dispatch_once_t onceToken;
+  dispatch_once(&onceToken, ^{
+    sharedInstance = [[self alloc] init];
+  });
+
+  return sharedInstance;
+}
+
+// Method to make sure observers are added and the initialization block is
+// scheduled to run on the main queue.
++ (void)prepareForUse {
+  __unused RTCUIApplicationStatusObserver *observer = [self sharedInstance];
+}
+
+- (id)init {
+  if (self = [super init]) {
+    NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+    __weak RTCUIApplicationStatusObserver *weakSelf = self;
+    _activeObserver = [center addObserverForName:UIApplicationDidBecomeActiveNotification
+                                          object:nil
+                                           queue:[NSOperationQueue mainQueue]
+                                      usingBlock:^(NSNotification *note) {
+                                        weakSelf.state =
+                                            [UIApplication sharedApplication].applicationState;
+                                      }];
+
+    _backgroundObserver = [center addObserverForName:UIApplicationDidEnterBackgroundNotification
+                                              object:nil
+                                               queue:[NSOperationQueue mainQueue]
+                                          usingBlock:^(NSNotification *note) {
+                                            weakSelf.state =
+                                                [UIApplication sharedApplication].applicationState;
+                                          }];
+
+    _waitForInitializeSemaphore = dispatch_semaphore_create(1);
+    _initialized = NO;
+    _initializeBlock = dispatch_block_create(DISPATCH_BLOCK_INHERIT_QOS_CLASS, ^{
+      weakSelf.state = [UIApplication sharedApplication].applicationState;
+      weakSelf.initialized = YES;
+    });
+
+    dispatch_async(dispatch_get_main_queue(), _initializeBlock);
+  }
+
+  return self;
+}
+
+- (void)dealloc {
+  NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+  [center removeObserver:_activeObserver];
+  [center removeObserver:_backgroundObserver];
+}
+
+- (BOOL)isApplicationActive {
+  // NOTE: The function `dispatch_block_wait` can only legally be called once.
+  // Because of this, if several threads call the `isApplicationActive` method before
+  // the `_initializeBlock` has been executed, instead of multiple threads calling
+  // `dispatch_block_wait`, the other threads need to wait for the first waiting thread
+  // instead.
+  if (!_initialized) {
+    dispatch_semaphore_wait(_waitForInitializeSemaphore, DISPATCH_TIME_FOREVER);
+    if (!_initialized) {
+      long ret = dispatch_block_wait(_initializeBlock,
+                                     dispatch_time(DISPATCH_TIME_NOW, 10.0 * NSEC_PER_SEC));
+      RTC_DCHECK_EQ(ret, 0);
+    }
+    dispatch_semaphore_signal(_waitForInitializeSemaphore);
+  }
+  return _state == UIApplicationStateActive;
+}
+
+@end
+
+#endif  // WEBRTC_IOS

sdk/objc/Framework/Classes/Common/UIDevice+RTCDevice.mm

@@ -0,0 +1,176 @@
+/*
+ *  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/UIDevice+RTCDevice.h"
+
+#include <memory>
+#include <sys/sysctl.h>
+
+@implementation UIDevice (RTCDevice)
+
++ (RTCDeviceType)deviceType {
+  NSDictionary *machineNameToType = @{
+    @"iPhone1,1": @(RTCDeviceTypeIPhone1G),
+    @"iPhone1,2": @(RTCDeviceTypeIPhone3G),
+    @"iPhone2,1": @(RTCDeviceTypeIPhone3GS),
+    @"iPhone3,1": @(RTCDeviceTypeIPhone4),
+    @"iPhone3,3": @(RTCDeviceTypeIPhone4Verizon),
+    @"iPhone4,1": @(RTCDeviceTypeIPhone4S),
+    @"iPhone5,1": @(RTCDeviceTypeIPhone5GSM),
+    @"iPhone5,2": @(RTCDeviceTypeIPhone5GSM_CDMA),
+    @"iPhone5,3": @(RTCDeviceTypeIPhone5CGSM),
+    @"iPhone5,4": @(RTCDeviceTypeIPhone5CGSM_CDMA),
+    @"iPhone6,1": @(RTCDeviceTypeIPhone5SGSM),
+    @"iPhone6,2": @(RTCDeviceTypeIPhone5SGSM_CDMA),
+    @"iPhone7,1": @(RTCDeviceTypeIPhone6Plus),
+    @"iPhone7,2": @(RTCDeviceTypeIPhone6),
+    @"iPhone8,1": @(RTCDeviceTypeIPhone6S),
+    @"iPhone8,2": @(RTCDeviceTypeIPhone6SPlus),
+    @"iPod1,1": @(RTCDeviceTypeIPodTouch1G),
+    @"iPod2,1": @(RTCDeviceTypeIPodTouch2G),
+    @"iPod3,1": @(RTCDeviceTypeIPodTouch3G),
+    @"iPod4,1": @(RTCDeviceTypeIPodTouch4G),
+    @"iPod5,1": @(RTCDeviceTypeIPodTouch5G),
+    @"iPad1,1": @(RTCDeviceTypeIPad),
+    @"iPad2,1": @(RTCDeviceTypeIPad2Wifi),
+    @"iPad2,2": @(RTCDeviceTypeIPad2GSM),
+    @"iPad2,3": @(RTCDeviceTypeIPad2CDMA),
+    @"iPad2,4": @(RTCDeviceTypeIPad2Wifi2),
+    @"iPad2,5": @(RTCDeviceTypeIPadMiniWifi),
+    @"iPad2,6": @(RTCDeviceTypeIPadMiniGSM),
+    @"iPad2,7": @(RTCDeviceTypeIPadMiniGSM_CDMA),
+    @"iPad3,1": @(RTCDeviceTypeIPad3Wifi),
+    @"iPad3,2": @(RTCDeviceTypeIPad3GSM_CDMA),
+    @"iPad3,3": @(RTCDeviceTypeIPad3GSM),
+    @"iPad3,4": @(RTCDeviceTypeIPad4Wifi),
+    @"iPad3,5": @(RTCDeviceTypeIPad4GSM),
+    @"iPad3,6": @(RTCDeviceTypeIPad4GSM_CDMA),
+    @"iPad4,1": @(RTCDeviceTypeIPadAirWifi),
+    @"iPad4,2": @(RTCDeviceTypeIPadAirCellular),
+    @"iPad4,4": @(RTCDeviceTypeIPadMini2GWifi),
+    @"iPad4,5": @(RTCDeviceTypeIPadMini2GCellular),
+    @"i386": @(RTCDeviceTypeSimulatori386),
+    @"x86_64": @(RTCDeviceTypeSimulatorx86_64),
+  };
+
+  size_t size = 0;
+  sysctlbyname("hw.machine", NULL, &size, NULL, 0);
+  std::unique_ptr<char[]> machine;
+  machine.reset(new char[size]);
+  sysctlbyname("hw.machine", machine.get(), &size, NULL, 0);
+  NSString *machineName = [[NSString alloc] initWithCString:machine.get()
+                                                   encoding:NSUTF8StringEncoding];
+  RTCDeviceType deviceType = RTCDeviceTypeUnknown;
+  NSNumber *typeNumber = machineNameToType[machineName];
+  if (typeNumber) {
+    deviceType = static_cast<RTCDeviceType>(typeNumber.integerValue);
+  }
+  return deviceType;
+}
+
++ (NSString *)stringForDeviceType:(RTCDeviceType)deviceType {
+  switch (deviceType) {
+    case RTCDeviceTypeUnknown:
+      return @"Unknown";
+    case RTCDeviceTypeIPhone1G:
+      return @"iPhone 1G";
+    case RTCDeviceTypeIPhone3G:
+      return @"iPhone 3G";
+    case RTCDeviceTypeIPhone3GS:
+      return @"iPhone 3GS";
+    case RTCDeviceTypeIPhone4:
+      return @"iPhone 4";
+    case RTCDeviceTypeIPhone4Verizon:
+      return @"iPhone 4 Verizon";
+    case RTCDeviceTypeIPhone4S:
+      return @"iPhone 4S";
+    case RTCDeviceTypeIPhone5GSM:
+      return @"iPhone 5 (GSM)";
+    case RTCDeviceTypeIPhone5GSM_CDMA:
+      return @"iPhone 5 (GSM+CDMA)";
+    case RTCDeviceTypeIPhone5CGSM:
+      return @"iPhone 5C (GSM)";
+    case RTCDeviceTypeIPhone5CGSM_CDMA:
+      return @"iPhone 5C (GSM+CDMA)";
+    case RTCDeviceTypeIPhone5SGSM:
+      return @"iPhone 5S (GSM)";
+    case RTCDeviceTypeIPhone5SGSM_CDMA:
+      return @"iPhone 5S (GSM+CDMA)";
+    case RTCDeviceTypeIPhone6Plus:
+      return @"iPhone 6 Plus";
+    case RTCDeviceTypeIPhone6:
+      return @"iPhone 6";
+    case RTCDeviceTypeIPhone6S:
+      return @"iPhone 6S";
+    case RTCDeviceTypeIPhone6SPlus:
+      return @"iPhone 6S Plus";
+    case RTCDeviceTypeIPodTouch1G:
+      return @"iPod Touch 1G";
+    case RTCDeviceTypeIPodTouch2G:
+      return @"iPod Touch 2G";
+    case RTCDeviceTypeIPodTouch3G:
+      return @"iPod Touch 3G";
+    case RTCDeviceTypeIPodTouch4G:
+      return @"iPod Touch 4G";
+    case RTCDeviceTypeIPodTouch5G:
+      return @"iPod Touch 5G";
+    case RTCDeviceTypeIPad:
+      return @"iPad";
+    case RTCDeviceTypeIPad2Wifi:
+      return @"iPad 2 (WiFi)";
+    case RTCDeviceTypeIPad2GSM:
+      return @"iPad 2 (GSM)";
+    case RTCDeviceTypeIPad2CDMA:
+      return @"iPad 2 (CDMA)";
+    case RTCDeviceTypeIPad2Wifi2:
+      return @"iPad 2 (WiFi) 2";
+    case RTCDeviceTypeIPadMiniWifi:
+      return @"iPad Mini (WiFi)";
+    case RTCDeviceTypeIPadMiniGSM:
+      return @"iPad Mini (GSM)";
+    case RTCDeviceTypeIPadMiniGSM_CDMA:
+      return @"iPad Mini (GSM+CDMA)";
+    case RTCDeviceTypeIPad3Wifi:
+      return @"iPad 3 (WiFi)";
+    case RTCDeviceTypeIPad3GSM_CDMA:
+      return @"iPad 3 (GSM+CDMA)";
+    case RTCDeviceTypeIPad3GSM:
+      return @"iPad 3 (GSM)";
+    case RTCDeviceTypeIPad4Wifi:
+      return @"iPad 4 (WiFi)";
+    case RTCDeviceTypeIPad4GSM:
+      return @"iPad 4 (GSM)";
+    case RTCDeviceTypeIPad4GSM_CDMA:
+      return @"iPad 4 (GSM+CDMA)";
+    case RTCDeviceTypeIPadAirWifi:
+      return @"iPad Air (WiFi)";
+    case RTCDeviceTypeIPadAirCellular:
+      return @"iPad Air (Cellular)";
+    case RTCDeviceTypeIPadMini2GWifi:
+      return @"iPad Mini 2G (Wifi)";
+    case RTCDeviceTypeIPadMini2GCellular:
+      return @"iPad Mini 2G (Cellular)";
+    case RTCDeviceTypeSimulatori386:
+      return @"i386 Simulator";
+    case RTCDeviceTypeSimulatorx86_64:
+      return @"x86_64 Simulator";
+  }
+  return @"Unknown";
+}
+
++ (double)currentDeviceSystemVersion {
+  return [self currentDevice].systemVersion.doubleValue;
+}
+
++ (BOOL)isIOS9OrLater {
+  return [self currentDeviceSystemVersion] >= 9.0;
+}
+
+@end

sdk/objc/Framework/Classes/Common/helpers.h

@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#ifndef WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_HELPERS_H_
+#define WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_HELPERS_H_
+
+#include <string>
+
+namespace webrtc {
+namespace ios {
+
+bool CheckAndLogError(BOOL success, NSError* error);
+
+NSString *NSStringFromStdString(const std::string& stdString);
+std::string StdStringFromNSString(NSString* nsString);
+
+// Return thread ID as a string.
+std::string GetThreadId();
+
+// Return thread ID as string suitable for debug logging.
+std::string GetThreadInfo();
+
+// Returns [NSThread currentThread] description as string.
+// Example: <NSThread: 0x170066d80>{number = 1, name = main}
+std::string GetCurrentThreadDescription();
+
+#if defined(WEBRTC_IOS)
+// Returns the current name of the operating system.
+std::string GetSystemName();
+
+// Returns the current version of the operating system as a string.
+std::string GetSystemVersionAsString();
+
+// Returns the version of the operating system in double representation.
+// Uses a cached value of the system version.
+double GetSystemVersion();
+
+// Returns the device type.
+// Examples: ”iPhone” and ”iPod touch”.
+std::string GetDeviceType();
+#endif  // defined(WEBRTC_IOS)
+
+// Returns a more detailed device name.
+// Examples: "iPhone 5s (GSM)" and "iPhone 6 Plus".
+std::string GetDeviceName();
+
+// Returns the name of the process. Does not uniquely identify the process.
+std::string GetProcessName();
+
+// Returns the identifier of the process (often called process ID).
+int GetProcessID();
+
+// Returns a string containing the version of the operating system on which the
+// process is executing. The string is string is human readable, localized, and
+// is appropriate for displaying to the user.
+std::string GetOSVersionString();
+
+// Returns the number of processing cores available on the device.
+int GetProcessorCount();
+
+#if defined(WEBRTC_IOS)
+// Indicates whether Low Power Mode is enabled on the iOS device.
+bool GetLowPowerModeEnabled();
+#endif
+
+}  // namespace ios
+}  // namespace webrtc
+
+#endif  // WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_HELPERS_H_

sdk/objc/Framework/Classes/Common/helpers.mm

@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2015 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 <sys/sysctl.h>
+#if defined(WEBRTC_IOS)
+#import <UIKit/UIKit.h>
+#endif
+
+#include <memory>
+
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h"
+
+namespace webrtc {
+namespace ios {
+
+#if defined(WEBRTC_IOS)
+bool isOperatingSystemAtLeastVersion(double version) {
+  return GetSystemVersion() >= version;
+}
+#endif
+
+NSString* NSStringFromStdString(const std::string& stdString) {
+  // std::string may contain null termination character so we construct
+  // using length.
+  return [[NSString alloc] initWithBytes:stdString.data()
+                                  length:stdString.length()
+                                encoding:NSUTF8StringEncoding];
+}
+
+std::string StdStringFromNSString(NSString* nsString) {
+  NSData* charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
+  return std::string(reinterpret_cast<const char*>([charData bytes]),
+                     [charData length]);
+}
+
+bool CheckAndLogError(BOOL success, NSError* error) {
+  if (!success) {
+    NSString* msg =
+        [NSString stringWithFormat:@"Error: %ld, %@, %@", (long)error.code,
+                                   error.localizedDescription,
+                                   error.localizedFailureReason];
+    LOG(LS_ERROR) << StdStringFromNSString(msg);
+    return false;
+  }
+  return true;
+}
+
+// TODO(henrika): see if it is possible to move to GetThreadName in
+// platform_thread.h and base it on pthread methods instead.
+std::string GetCurrentThreadDescription() {
+  NSString* name = [NSString stringWithFormat:@"%@", [NSThread currentThread]];
+  return StdStringFromNSString(name);
+}
+
+#if defined(WEBRTC_IOS)
+std::string GetSystemName() {
+  NSString* osName = [[UIDevice currentDevice] systemName];
+  return StdStringFromNSString(osName);
+}
+
+std::string GetSystemVersionAsString() {
+  NSString* osVersion = [[UIDevice currentDevice] systemVersion];
+  return StdStringFromNSString(osVersion);
+}
+
+double GetSystemVersion() {
+  static dispatch_once_t once_token;
+  static double system_version;
+  dispatch_once(&once_token, ^{
+    system_version = [UIDevice currentDevice].systemVersion.doubleValue;
+  });
+  return system_version;
+}
+
+std::string GetDeviceType() {
+  NSString* deviceModel = [[UIDevice currentDevice] model];
+  return StdStringFromNSString(deviceModel);
+}
+#endif
+
+std::string GetDeviceName() {
+  size_t size;
+  sysctlbyname("hw.machine", NULL, &size, NULL, 0);
+  std::unique_ptr<char[]> machine;
+  machine.reset(new char[size]);
+  sysctlbyname("hw.machine", machine.get(), &size, NULL, 0);
+  return std::string(machine.get());
+}
+
+std::string GetProcessName() {
+  NSString* processName = [NSProcessInfo processInfo].processName;
+  return StdStringFromNSString(processName);
+}
+
+int GetProcessID() {
+  return [NSProcessInfo processInfo].processIdentifier;
+}
+
+std::string GetOSVersionString() {
+  NSString* osVersion =
+      [NSProcessInfo processInfo].operatingSystemVersionString;
+  return StdStringFromNSString(osVersion);
+}
+
+int GetProcessorCount() {
+  return [NSProcessInfo processInfo].processorCount;
+}
+
+#if defined(__IPHONE_9_0) && defined(__IPHONE_OS_VERSION_MAX_ALLOWED) \
+    && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_9_0
+bool GetLowPowerModeEnabled() {
+  if (isOperatingSystemAtLeastVersion(9.0)) {
+    // lowPoweredModeEnabled is only available on iOS9+.
+    return [NSProcessInfo processInfo].lowPowerModeEnabled;
+  }
+  LOG(LS_WARNING) << "webrtc::ios::GetLowPowerModeEnabled() is not "
+                     "supported. Requires at least iOS 9.0";
+  return false;
+}
+#endif
+
+}  // namespace ios
+}  // namespace webrtc
+

sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h

@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2017 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 "RTCMTLRenderer.h"
+
+NS_AVAILABLE(10_11, 9_0)
+@interface RTCMTLI420Renderer : RTCMTLRenderer
+@end

sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.mm

@@ -0,0 +1,152 @@
+/*
+ *  Copyright 2017 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 "RTCMTLI420Renderer.h"
+#import "WebRTC/RTCVideoFrameBuffer.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoFrame.h"
+
+#import "RTCMTLRenderer+Private.h"
+
+#define MTL_STRINGIFY(s) @ #s
+
+static NSString *const shaderSource = MTL_STRINGIFY(
+    using namespace metal; typedef struct {
+      packed_float2 position;
+      packed_float2 texcoord;
+    } Vertex;
+
+    typedef struct {
+      float4 position[[position]];
+      float2 texcoord;
+    } Varyings;
+
+    vertex Varyings vertexPassthrough(device Vertex * verticies[[buffer(0)]],
+                                      unsigned int vid[[vertex_id]]) {
+      Varyings out;
+      device Vertex &v = verticies[vid];
+      out.position = float4(float2(v.position), 0.0, 1.0);
+      out.texcoord = v.texcoord;
+
+      return out;
+    }
+
+    fragment half4 fragmentColorConversion(
+        Varyings in[[stage_in]], texture2d<float, access::sample> textureY[[texture(0)]],
+        texture2d<float, access::sample> textureU[[texture(1)]],
+        texture2d<float, access::sample> textureV[[texture(2)]]) {
+      constexpr sampler s(address::clamp_to_edge, filter::linear);
+      float y;
+      float u;
+      float v;
+      float r;
+      float g;
+      float b;
+      // Conversion for YUV to rgb from http://www.fourcc.org/fccyvrgb.php
+      y = textureY.sample(s, in.texcoord).r;
+      u = textureU.sample(s, in.texcoord).r;
+      v = textureV.sample(s, in.texcoord).r;
+      u = u - 0.5;
+      v = v - 0.5;
+      r = y + 1.403 * v;
+      g = y - 0.344 * u - 0.714 * v;
+      b = y + 1.770 * u;
+
+      float4 out = float4(r, g, b, 1.0);
+
+      return half4(out);
+    });
+
+@implementation RTCMTLI420Renderer {
+  // Textures.
+  id<MTLTexture> _yTexture;
+  id<MTLTexture> _uTexture;
+  id<MTLTexture> _vTexture;
+
+  MTLTextureDescriptor *_descriptor;
+  MTLTextureDescriptor *_chromaDescriptor;
+
+  int _width;
+  int _height;
+  int _chromaWidth;
+  int _chromaHeight;
+}
+
+#pragma mark - Virtual
+
+- (NSString *)shaderSource {
+  return shaderSource;
+}
+
+- (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame {
+  [super setupTexturesForFrame:frame];
+
+  id<MTLDevice> device = [self currentMetalDevice];
+  if (!device) {
+    return NO;
+  }
+
+  id<RTCI420Buffer> buffer = [frame.buffer toI420];
+
+  // Luma (y) texture.
+  if (!_descriptor || (_width != frame.width && _height != frame.height)) {
+    _width = frame.width;
+    _height = frame.height;
+    _descriptor = [MTLTextureDescriptor texture2DDescriptorWithPixelFormat:MTLPixelFormatR8Unorm
+                                                                     width:_width
+                                                                    height:_height
+                                                                 mipmapped:NO];
+    _descriptor.usage = MTLTextureUsageShaderRead;
+    _yTexture = [device newTextureWithDescriptor:_descriptor];
+  }
+
+  // Chroma (u,v) textures
+  [_yTexture replaceRegion:MTLRegionMake2D(0, 0, _width, _height)
+               mipmapLevel:0
+                 withBytes:buffer.dataY
+               bytesPerRow:buffer.strideY];
+
+  if (!_chromaDescriptor ||
+      (_chromaWidth != frame.width / 2 && _chromaHeight != frame.height / 2)) {
+    _chromaWidth = frame.width / 2;
+    _chromaHeight = frame.height / 2;
+    _chromaDescriptor =
+        [MTLTextureDescriptor texture2DDescriptorWithPixelFormat:MTLPixelFormatR8Unorm
+                                                           width:_chromaWidth
+                                                          height:_chromaHeight
+                                                       mipmapped:NO];
+    _chromaDescriptor.usage = MTLTextureUsageShaderRead;
+    _uTexture = [device newTextureWithDescriptor:_chromaDescriptor];
+    _vTexture = [device newTextureWithDescriptor:_chromaDescriptor];
+  }
+
+  [_uTexture replaceRegion:MTLRegionMake2D(0, 0, _chromaWidth, _chromaHeight)
+               mipmapLevel:0
+                 withBytes:buffer.dataU
+               bytesPerRow:buffer.strideU];
+  [_vTexture replaceRegion:MTLRegionMake2D(0, 0, _chromaWidth, _chromaHeight)
+               mipmapLevel:0
+                 withBytes:buffer.dataV
+               bytesPerRow:buffer.strideV];
+
+  return (_uTexture != nil) && (_yTexture != nil) && (_vTexture != nil);
+}
+
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder {
+  [renderEncoder setFragmentTexture:_yTexture atIndex:0];
+  [renderEncoder setFragmentTexture:_uTexture atIndex:1];
+  [renderEncoder setFragmentTexture:_vTexture atIndex:2];
+}
+
+@end

sdk/objc/Framework/Classes/Metal/RTCMTLNSVideoView.m

@@ -0,0 +1,118 @@
+/*
+ *  Copyright 2017 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/RTCMTLNSVideoView.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "WebRTC/RTCVideoFrame.h"
+
+#import "RTCMTLI420Renderer.h"
+
+@interface RTCMTLNSVideoView ()<MTKViewDelegate>
+@property(nonatomic) id<RTCMTLRenderer> renderer;
+@property(nonatomic, strong) MTKView *metalView;
+@property(atomic, strong) RTCVideoFrame *videoFrame;
+@end
+
+@implementation RTCMTLNSVideoView {
+  id<RTCMTLRenderer> _renderer;
+}
+
+@synthesize renderer = _renderer;
+@synthesize metalView = _metalView;
+@synthesize videoFrame = _videoFrame;
+
+- (instancetype)initWithFrame:(CGRect)frameRect {
+  self = [super initWithFrame:frameRect];
+  if (self) {
+    [self configure];
+  }
+  return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aCoder {
+  self = [super initWithCoder:aCoder];
+  if (self) {
+    [self configure];
+  }
+  return self;
+}
+
+#pragma mark - Private
+
++ (BOOL)isMetalAvailable {
+  return [MTLCopyAllDevices() count] > 0;
+}
+
+- (void)configure {
+  if ([[self class] isMetalAvailable]) {
+    _metalView = [[MTKView alloc] initWithFrame:self.bounds];
+    [self addSubview:_metalView];
+    _metalView.layerContentsPlacement = NSViewLayerContentsPlacementScaleProportionallyToFit;
+    _metalView.translatesAutoresizingMaskIntoConstraints = NO;
+    _metalView.framebufferOnly = YES;
+    _metalView.delegate = self;
+
+    _renderer = [[RTCMTLI420Renderer alloc] init];
+    if (![(RTCMTLI420Renderer *)_renderer addRenderingDestination:_metalView]) {
+      _renderer = nil;
+    };
+  }
+}
+
+- (void)updateConstraints {
+  NSDictionary *views = NSDictionaryOfVariableBindings(_metalView);
+
+  NSArray *constraintsHorizontal =
+      [NSLayoutConstraint constraintsWithVisualFormat:@"H:|-0-[_metalView]-0-|"
+                                              options:0
+                                              metrics:nil
+                                                views:views];
+  [self addConstraints:constraintsHorizontal];
+
+  NSArray *constraintsVertical =
+      [NSLayoutConstraint constraintsWithVisualFormat:@"V:|-0-[_metalView]-0-|"
+                                              options:0
+                                              metrics:nil
+                                                views:views];
+  [self addConstraints:constraintsVertical];
+  [super updateConstraints];
+}
+
+#pragma mark - MTKViewDelegate methods
+- (void)drawInMTKView:(nonnull MTKView *)view {
+  if (self.videoFrame == nil) {
+    return;
+  }
+  if (view == self.metalView) {
+    [_renderer drawFrame:self.videoFrame];
+  }
+}
+
+- (void)mtkView:(MTKView *)view drawableSizeWillChange:(CGSize)size {
+}
+
+#pragma mark - RTCVideoRenderer
+
+- (void)setSize:(CGSize)size {
+  _metalView.drawableSize = size;
+  [_metalView draw];
+}
+
+- (void)renderFrame:(nullable RTCVideoFrame *)frame {
+  if (frame == nil) {
+    return;
+  }
+  self.videoFrame = [frame newI420VideoFrame];
+}
+
+@end

sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h

@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 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 "RTCMTLRenderer.h"
+
+NS_AVAILABLE(10_11, 9_0)
+@interface RTCMTLNV12Renderer : RTCMTLRenderer
+
+@end

sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.mm

@@ -0,0 +1,135 @@
+/*
+ *  Copyright 2017 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 "RTCMTLNV12Renderer.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoFrame.h"
+#import "WebRTC/RTCVideoFrameBuffer.h"
+
+#import "RTCMTLRenderer+Private.h"
+
+#define MTL_STRINGIFY(s) @ #s
+
+static NSString *const shaderSource = MTL_STRINGIFY(
+    using namespace metal; typedef struct {
+      packed_float2 position;
+      packed_float2 texcoord;
+    } Vertex;
+
+    typedef struct {
+      float4 position[[position]];
+      float2 texcoord;
+    } Varyings;
+
+    vertex Varyings vertexPassthrough(device Vertex * verticies[[buffer(0)]],
+                                      unsigned int vid[[vertex_id]]) {
+      Varyings out;
+      device Vertex &v = verticies[vid];
+      out.position = float4(float2(v.position), 0.0, 1.0);
+      out.texcoord = v.texcoord;
+      return out;
+    }
+
+    // Receiving YCrCb textures.
+    fragment half4 fragmentColorConversion(
+        Varyings in[[stage_in]], texture2d<float, access::sample> textureY[[texture(0)]],
+        texture2d<float, access::sample> textureCbCr[[texture(1)]]) {
+      constexpr sampler s(address::clamp_to_edge, filter::linear);
+      float y;
+      float2 uv;
+      y = textureY.sample(s, in.texcoord).r;
+      uv = textureCbCr.sample(s, in.texcoord).rg - float2(0.5, 0.5);
+
+      // Conversion for YUV to rgb from http://www.fourcc.org/fccyvrgb.php
+      float4 out = float4(y + 1.403 * uv.y, y - 0.344 * uv.x - 0.714 * uv.y, y + 1.770 * uv.x, 1.0);
+
+      return half4(out);
+    });
+
+@implementation RTCMTLNV12Renderer {
+  // Textures.
+  CVMetalTextureCacheRef _textureCache;
+  id<MTLTexture> _yTexture;
+  id<MTLTexture> _CrCbTexture;
+}
+
+- (BOOL)addRenderingDestination:(__kindof MTKView *)view {
+  if ([super addRenderingDestination:view]) {
+    [self initializeTextureCache];
+    return YES;
+  }
+  return NO;
+}
+
+- (void)initializeTextureCache {
+  CVReturn status = CVMetalTextureCacheCreate(kCFAllocatorDefault, nil, [self currentMetalDevice],
+                                              nil, &_textureCache);
+  if (status != kCVReturnSuccess) {
+    RTCLogError(@"Metal: Failed to initialize metal texture cache. Return status is %d", status);
+  }
+}
+
+- (NSString *)shaderSource {
+  return shaderSource;
+}
+
+- (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame {
+  [super setupTexturesForFrame:frame];
+  CVPixelBufferRef pixelBuffer = ((RTCCVPixelBuffer *)frame.buffer).pixelBuffer;
+
+  id<MTLTexture> lumaTexture = nil;
+  id<MTLTexture> chromaTexture = nil;
+  CVMetalTextureRef outTexture = nullptr;
+
+  // Luma (y) texture.
+  int lumaWidth = CVPixelBufferGetWidthOfPlane(pixelBuffer, 0);
+  int lumaHeight = CVPixelBufferGetHeightOfPlane(pixelBuffer, 0);
+
+  int indexPlane = 0;
+  CVReturn result = CVMetalTextureCacheCreateTextureFromImage(
+      kCFAllocatorDefault, _textureCache, pixelBuffer, nil, MTLPixelFormatR8Unorm, lumaWidth,
+      lumaHeight, indexPlane, &outTexture);
+
+  if (result == kCVReturnSuccess) {
+    lumaTexture = CVMetalTextureGetTexture(outTexture);
+  }
+
+  // Same as CFRelease except it can be passed NULL without crashing.
+  CVBufferRelease(outTexture);
+  outTexture = nullptr;
+
+  // Chroma (CrCb) texture.
+  indexPlane = 1;
+  result = CVMetalTextureCacheCreateTextureFromImage(
+      kCFAllocatorDefault, _textureCache, pixelBuffer, nil, MTLPixelFormatRG8Unorm, lumaWidth / 2,
+      lumaHeight / 2, indexPlane, &outTexture);
+  if (result == kCVReturnSuccess) {
+    chromaTexture = CVMetalTextureGetTexture(outTexture);
+  }
+  CVBufferRelease(outTexture);
+
+  if (lumaTexture != nil && chromaTexture != nil) {
+    _yTexture = lumaTexture;
+    _CrCbTexture = chromaTexture;
+    return YES;
+  }
+  return NO;
+}
+
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder {
+  [renderEncoder setFragmentTexture:_yTexture atIndex:0];
+  [renderEncoder setFragmentTexture:_CrCbTexture atIndex:1];
+}
+
+@end

sdk/objc/Framework/Classes/Metal/RTCMTLRenderer+Private.h

@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2017 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 <Metal/Metal.h>
+#import "RTCMTLRenderer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+@interface RTCMTLRenderer (Private)
+- (nullable id<MTLDevice>)currentMetalDevice;
+- (NSString *)shaderSource;
+- (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame;
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder;
+@end
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h

@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2017 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>
+#if TARGET_OS_IPHONE
+#import <UIKit/UIKit.h>
+#else
+#import <AppKit/AppKit.h>
+#endif
+
+#import "WebRTC/RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+/**
+ * Protocol defining ability to render RTCVideoFrame in Metal enabled views.
+ */
+@protocol RTCMTLRenderer<NSObject>
+
+/**
+ * Method to be implemented to perform actual rendering of the provided frame.
+ *
+ * @param frame The frame to be rendered.
+ */
+- (void)drawFrame:(RTCVideoFrame *)frame;
+
+/**
+ * Sets the provided view as rendering destination if possible.
+ *
+ * If not possible method returns NO and callers of the method are responisble for performing
+ * cleanups.
+ */
+
+#if TARGET_OS_IOS
+- (BOOL)addRenderingDestination:(__kindof UIView *)view;
+#else
+- (BOOL)addRenderingDestination:(__kindof NSView *)view;
+#endif
+
+@end
+
+/**
+ * Implementation of RTCMTLRenderer protocol for rendering native nv12 video frames.
+ */
+NS_AVAILABLE(10_11, 9_0)
+@interface RTCMTLRenderer : NSObject<RTCMTLRenderer>
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.mm

@@ -0,0 +1,244 @@
+/*
+ *  Copyright 2017 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 "RTCMTLRenderer+Private.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoFrame.h"
+
+#include "webrtc/api/video/video_rotation.h"
+#include "webrtc/rtc_base/checks.h"
+
+// As defined in shaderSource.
+static NSString *const vertexFunctionName = @"vertexPassthrough";
+static NSString *const fragmentFunctionName = @"fragmentColorConversion";
+
+static NSString *const pipelineDescriptorLabel = @"RTCPipeline";
+static NSString *const commandBufferLabel = @"RTCCommandBuffer";
+static NSString *const renderEncoderLabel = @"RTCEncoder";
+static NSString *const renderEncoderDebugGroup = @"RTCDrawFrame";
+
+static const float cubeVertexData[64] = {
+    -1.0, -1.0, 0.0, 1.0, 1.0, -1.0, 1.0, 1.0, -1.0, 1.0, 0.0, 0.0, 1.0, 1.0, 1.0, 0.0,
+
+    // rotation = 90, offset = 16.
+    -1.0, -1.0, 1.0, 1.0, 1.0, -1.0, 1.0, 0.0, -1.0, 1.0, 0.0, 1.0, 1.0, 1.0, 0.0, 0.0,
+
+    // rotation = 180, offset = 32.
+    -1.0, -1.0, 1.0, 0.0, 1.0, -1.0, 0.0, 0.0, -1.0, 1.0, 1.0, 1.0, 1.0, 1.0, 0.0, 1.0,
+
+    // rotation = 270, offset = 48.
+    -1.0, -1.0, 0.0, 0.0, 1.0, -1.0, 0.0, 1.0, -1.0, 1.0, 1.0, 0.0, 1.0, 1.0, 1.0, 1.0,
+};
+
+static inline int offsetForRotation(RTCVideoRotation rotation) {
+  switch (rotation) {
+    case RTCVideoRotation_0:
+      return 0;
+    case RTCVideoRotation_90:
+      return 16;
+    case RTCVideoRotation_180:
+      return 32;
+    case RTCVideoRotation_270:
+      return 48;
+  }
+  return 0;
+}
+
+// The max number of command buffers in flight (submitted to GPU).
+// For now setting it up to 1.
+// In future we might use triple buffering method if it improves performance.
+static const NSInteger kMaxInflightBuffers = 1;
+
+@implementation RTCMTLRenderer {
+  __kindof MTKView *_view;
+
+  // Controller.
+  dispatch_semaphore_t _inflight_semaphore;
+
+  // Renderer.
+  id<MTLDevice> _device;
+  id<MTLCommandQueue> _commandQueue;
+  id<MTLLibrary> _defaultLibrary;
+  id<MTLRenderPipelineState> _pipelineState;
+
+  // Buffers.
+  id<MTLBuffer> _vertexBuffer;
+
+  // RTC Frame parameters.
+  int _offset;
+}
+
+- (instancetype)init {
+  if (self = [super init]) {
+    // _offset of 0 is equal to rotation of 0.
+    _offset = 0;
+    _inflight_semaphore = dispatch_semaphore_create(kMaxInflightBuffers);
+  }
+
+  return self;
+}
+
+- (BOOL)addRenderingDestination:(__kindof MTKView *)view {
+  return [self setupWithView:view];
+}
+
+#pragma mark - Private
+
+- (BOOL)setupWithView:(__kindof MTKView *)view {
+  BOOL success = NO;
+  if ([self setupMetal]) {
+    [self setupView:view];
+    [self loadAssets];
+    [self setupBuffers];
+    success = YES;
+  }
+  return success;
+}
+#pragma mark - Inheritance
+
+- (id<MTLDevice>)currentMetalDevice {
+  return _device;
+}
+
+- (NSString *)shaderSource {
+  RTC_NOTREACHED() << "Virtual method not implemented in subclass.";
+  return nil;
+}
+
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder {
+  RTC_NOTREACHED() << "Virtual method not implemented in subclass.";
+}
+
+- (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame {
+  _offset = offsetForRotation(frame.rotation);
+  return YES;
+}
+
+#pragma mark - GPU methods
+
+- (BOOL)setupMetal {
+  // Set the view to use the default device.
+  _device = MTLCreateSystemDefaultDevice();
+  if (!_device) {
+    return NO;
+  }
+
+  // Create a new command queue.
+  _commandQueue = [_device newCommandQueue];
+
+  // Load metal library from source.
+  NSError *libraryError = nil;
+
+  id<MTLLibrary> sourceLibrary =
+      [_device newLibraryWithSource:[self shaderSource] options:NULL error:&libraryError];
+
+  if (libraryError) {
+    RTCLogError(@"Metal: Library with source failed\n%@", libraryError);
+    return NO;
+  }
+
+  if (!sourceLibrary) {
+    RTCLogError(@"Metal: Failed to load library. %@", libraryError);
+    return NO;
+  }
+  _defaultLibrary = sourceLibrary;
+
+  return YES;
+}
+
+- (void)setupView:(__kindof MTKView *)view {
+  view.device = _device;
+
+  view.preferredFramesPerSecond = 30;
+  view.autoResizeDrawable = NO;
+
+  // We need to keep reference to the view as it's needed down the rendering pipeline.
+  _view = view;
+}
+
+- (void)loadAssets {
+  id<MTLFunction> vertexFunction = [_defaultLibrary newFunctionWithName:vertexFunctionName];
+  id<MTLFunction> fragmentFunction = [_defaultLibrary newFunctionWithName:fragmentFunctionName];
+
+  MTLRenderPipelineDescriptor *pipelineDescriptor = [[MTLRenderPipelineDescriptor alloc] init];
+  pipelineDescriptor.label = pipelineDescriptorLabel;
+  pipelineDescriptor.vertexFunction = vertexFunction;
+  pipelineDescriptor.fragmentFunction = fragmentFunction;
+  pipelineDescriptor.colorAttachments[0].pixelFormat = _view.colorPixelFormat;
+  pipelineDescriptor.depthAttachmentPixelFormat = MTLPixelFormatInvalid;
+  NSError *error = nil;
+  _pipelineState = [_device newRenderPipelineStateWithDescriptor:pipelineDescriptor error:&error];
+
+  if (!_pipelineState) {
+    RTCLogError(@"Metal: Failed to create pipeline state. %@", error);
+  }
+}
+
+- (void)setupBuffers {
+  _vertexBuffer = [_device newBufferWithBytes:cubeVertexData
+                                       length:sizeof(cubeVertexData)
+                                      options:MTLResourceOptionCPUCacheModeDefault];
+}
+
+- (void)render {
+  // Wait until the inflight (curently sent to GPU) command buffer
+  // has completed the GPU work.
+  dispatch_semaphore_wait(_inflight_semaphore, DISPATCH_TIME_FOREVER);
+
+  id<MTLCommandBuffer> commandBuffer = [_commandQueue commandBuffer];
+  commandBuffer.label = commandBufferLabel;
+
+  __block dispatch_semaphore_t block_semaphore = _inflight_semaphore;
+  [commandBuffer addCompletedHandler:^(id<MTLCommandBuffer> _Nonnull) {
+    // GPU work completed.
+    dispatch_semaphore_signal(block_semaphore);
+  }];
+
+  MTLRenderPassDescriptor *renderPassDescriptor = _view.currentRenderPassDescriptor;
+  if (renderPassDescriptor) {  // Valid drawable.
+    id<MTLRenderCommandEncoder> renderEncoder =
+        [commandBuffer renderCommandEncoderWithDescriptor:renderPassDescriptor];
+    renderEncoder.label = renderEncoderLabel;
+
+    // Set context state.
+    [renderEncoder pushDebugGroup:renderEncoderDebugGroup];
+    [renderEncoder setRenderPipelineState:_pipelineState];
+    [renderEncoder setVertexBuffer:_vertexBuffer offset:_offset * sizeof(float) atIndex:0];
+    [self uploadTexturesToRenderEncoder:renderEncoder];
+
+    [renderEncoder drawPrimitives:MTLPrimitiveTypeTriangleStrip
+                      vertexStart:0
+                      vertexCount:4
+                    instanceCount:1];
+    [renderEncoder popDebugGroup];
+    [renderEncoder endEncoding];
+
+    [commandBuffer presentDrawable:_view.currentDrawable];
+  }
+
+  // CPU work is completed, GPU work can be started.
+  [commandBuffer commit];
+}
+
+#pragma mark - RTCMTLRenderer
+
+- (void)drawFrame:(RTCVideoFrame *)frame {
+  @autoreleasepool {
+    if ([self setupTexturesForFrame:frame]) {
+      [self render];
+    }
+  }
+}
+
+@end

sdk/objc/Framework/Classes/Metal/RTCMTLVideoView.m

@@ -0,0 +1,152 @@
+/*
+ *  Copyright 2017 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/RTCMTLVideoView.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoFrame.h"
+#import "WebRTC/RTCVideoFrameBuffer.h"
+
+#import "RTCMTLI420Renderer.h"
+#import "RTCMTLNV12Renderer.h"
+
+// To avoid unreconized symbol linker errors, we're taking advantage of the objc runtime.
+// Linking errors occur when compiling for architectures that don't support Metal.
+#define MTKViewClass NSClassFromString(@"MTKView")
+#define RTCMTLNV12RendererClass NSClassFromString(@"RTCMTLNV12Renderer")
+#define RTCMTLI420RendererClass NSClassFromString(@"RTCMTLI420Renderer")
+
+@interface RTCMTLVideoView () <MTKViewDelegate>
+@property(nonatomic, strong) RTCMTLI420Renderer *rendererI420;
+@property(nonatomic, strong) RTCMTLNV12Renderer *rendererNV12;
+@property(nonatomic, strong) MTKView *metalView;
+@property(atomic, strong) RTCVideoFrame *videoFrame;
+@end
+
+@implementation RTCMTLVideoView
+
+@synthesize rendererI420 = _rendererI420;
+@synthesize rendererNV12 = _rendererNV12;
+@synthesize metalView = _metalView;
+@synthesize videoFrame = _videoFrame;
+
+- (instancetype)initWithFrame:(CGRect)frameRect {
+  self = [super initWithFrame:frameRect];
+  if (self) {
+    [self configure];
+  }
+  return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aCoder {
+  self = [super initWithCoder:aCoder];
+  if (self) {
+    [self configure];
+  }
+  return self;
+}
+
+#pragma mark - Private
+
++ (BOOL)isMetalAvailable {
+#if defined(RTC_SUPPORTS_METAL)
+  return YES;
+#else
+  return NO;
+#endif
+}
+
++ (MTKView *)createMetalView:(CGRect)frame {
+  MTKView *view = [[MTKViewClass alloc] initWithFrame:frame];
+  return view;
+}
+
++ (RTCMTLNV12Renderer *)createNV12Renderer {
+  return [[RTCMTLNV12RendererClass alloc] init];
+}
+
++ (RTCMTLI420Renderer *)createI420Renderer {
+  return [[RTCMTLI420RendererClass alloc] init];
+}
+
+- (void)configure {
+  NSAssert([RTCMTLVideoView isMetalAvailable], @"Metal not availiable on this device");
+
+  _metalView = [RTCMTLVideoView createMetalView:self.bounds];
+  [self configureMetalView];
+}
+
+- (void)configureMetalView {
+  if (_metalView) {
+    _metalView.delegate = self;
+    [self addSubview:_metalView];
+    _metalView.contentMode = UIViewContentModeScaleAspectFit;
+    _metalView.translatesAutoresizingMaskIntoConstraints = NO;
+    UILayoutGuide *margins = self.layoutMarginsGuide;
+    [_metalView.topAnchor constraintEqualToAnchor:margins.topAnchor].active = YES;
+    [_metalView.bottomAnchor constraintEqualToAnchor:margins.bottomAnchor].active = YES;
+    [_metalView.leftAnchor constraintEqualToAnchor:margins.leftAnchor].active = YES;
+    [_metalView.rightAnchor constraintEqualToAnchor:margins.rightAnchor].active = YES;
+  }
+}
+
+#pragma mark - MTKViewDelegate methods
+
+- (void)drawInMTKView:(nonnull MTKView *)view {
+  NSAssert(view == self.metalView, @"Receiving draw callbacks from foreign instance.");
+  if (!self.videoFrame) {
+    return;
+  }
+
+  id<RTCMTLRenderer> renderer = nil;
+  if ([self.videoFrame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
+    if (!self.rendererNV12) {
+      self.rendererNV12 = [RTCMTLVideoView createNV12Renderer];
+      if (![self.rendererNV12 addRenderingDestination:self.metalView]) {
+        self.rendererNV12 = nil;
+        RTCLogError(@"Failed to create NV12 renderer");
+      }
+    }
+    renderer = self.rendererNV12;
+  } else {
+    if (!self.rendererI420) {
+      self.rendererI420 = [RTCMTLVideoView createI420Renderer];
+      if (![self.rendererI420 addRenderingDestination:self.metalView]) {
+        self.rendererI420 = nil;
+        RTCLogError(@"Failed to create I420 renderer");
+      }
+    }
+    renderer = self.rendererI420;
+  }
+
+  [renderer drawFrame:self.videoFrame];
+}
+
+- (void)mtkView:(MTKView *)view drawableSizeWillChange:(CGSize)size {
+}
+
+#pragma mark - RTCVideoRenderer
+
+- (void)setSize:(CGSize)size {
+  self.metalView.drawableSize = size;
+}
+
+- (void)renderFrame:(nullable RTCVideoFrame *)frame {
+  if (frame == nil) {
+    RTCLogInfo(@"Incoming frame is nil. Exiting render callback.");
+    return;
+  }
+  self.videoFrame = frame;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/OWNERS

@@ -0,0 +1 @@
+deadbeef@webrtc.org

sdk/objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource+Private.h

@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2015 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/RTCAVFoundationVideoSource.h"
+
+#include "avfoundationvideocapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCAVFoundationVideoSource ()
+
+@property(nonatomic, readonly) webrtc::AVFoundationVideoCapturer *capturer;
+
+/** Initialize an RTCAVFoundationVideoSource with constraints. */
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                    constraints:(nullable RTCMediaConstraints *)constraints;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource.mm

@@ -0,0 +1,62 @@
+/*
+ *  Copyright 2015 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 "RTCAVFoundationVideoSource+Private.h"
+
+#import "RTCMediaConstraints+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCVideoSource+Private.h"
+
+@implementation RTCAVFoundationVideoSource {
+  webrtc::AVFoundationVideoCapturer *_capturer;
+}
+
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                    constraints:(RTCMediaConstraints *)constraints {
+  NSParameterAssert(factory);
+  // We pass ownership of the capturer to the source, but since we own
+  // the source, it should be ok to keep a raw pointer to the
+  // capturer.
+  _capturer = new webrtc::AVFoundationVideoCapturer();
+  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
+      factory.nativeFactory->CreateVideoSource(
+          std::unique_ptr<cricket::VideoCapturer>(_capturer),
+          constraints.nativeConstraints.get());
+
+  return [super initWithNativeVideoSource:source];
+}
+
+- (void)adaptOutputFormatToWidth:(int)width
+                          height:(int)height
+                             fps:(int)fps {
+  self.capturer->AdaptOutputFormat(width, height, fps);
+}
+
+- (BOOL)canUseBackCamera {
+  return self.capturer->CanUseBackCamera();
+}
+
+- (BOOL)useBackCamera {
+  return self.capturer->GetUseBackCamera();
+}
+
+- (void)setUseBackCamera:(BOOL)useBackCamera {
+  self.capturer->SetUseBackCamera(useBackCamera);
+}
+
+- (AVCaptureSession *)captureSession {
+  return self.capturer->GetCaptureSession();
+}
+
+- (webrtc::AVFoundationVideoCapturer *)capturer {
+  return _capturer;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h

@@ -0,0 +1,32 @@
+/*
+ *  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/RTCAudioSource.h"
+
+#import "RTCMediaSource+Private.h"
+
+@interface RTCAudioSource ()
+
+/**
+ * The AudioSourceInterface object passed to this RTCAudioSource during
+ * construction.
+ */
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::AudioSourceInterface> nativeAudioSource;
+
+/** Initialize an RTCAudioSource from a native AudioSourceInterface. */
+- (instancetype)initWithNativeAudioSource:
+    (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithNativeMediaSource:
+    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                                     type:(RTCMediaSourceType)type NS_UNAVAILABLE;
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource.mm

@@ -0,0 +1,48 @@
+/*
+ *  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 "RTCAudioSource+Private.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+@implementation RTCAudioSource {
+}
+
+@synthesize volume = _volume;
+@synthesize nativeAudioSource = _nativeAudioSource;
+
+- (instancetype)initWithNativeAudioSource:
+    (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource {
+  RTC_DCHECK(nativeAudioSource);
+  if (self = [super initWithNativeMediaSource:nativeAudioSource
+                                         type:RTCMediaSourceTypeAudio]) {
+    _nativeAudioSource = nativeAudioSource;
+  }
+  return self;
+}
+
+- (instancetype)initWithNativeMediaSource:
+    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                                     type:(RTCMediaSourceType)type {
+  RTC_NOTREACHED();
+  return nil;
+}
+
+- (NSString *)description {
+  NSString *stateString = [[self class] stringForState:self.state];
+  return [NSString stringWithFormat:@"RTCAudioSource( %p ): %@", self, stateString];
+}
+
+- (void)setVolume:(double)volume {
+  _volume = volume;
+  _nativeAudioSource->SetVolume(volume);
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h

@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2015 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/RTCAudioTrack.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCPeerConnectionFactory;
+@interface RTCAudioTrack ()
+
+/** AudioTrackInterface created or passed in at construction. */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeAudioTrack;
+
+/** Initialize an RTCAudioTrack with an id. */
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                         source:(RTCAudioSource *)source
+                        trackId:(NSString *)trackId;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm

@@ -0,0 +1,66 @@
+/*
+ *  Copyright 2015 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 "RTCAudioTrack+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCAudioSource+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+@implementation RTCAudioTrack
+
+@synthesize source = _source;
+
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                         source:(RTCAudioSource *)source
+                        trackId:(NSString *)trackId {
+  RTC_DCHECK(factory);
+  RTC_DCHECK(source);
+  RTC_DCHECK(trackId.length);
+
+  std::string nativeId = [NSString stdStringForString:trackId];
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+      factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource);
+  if ([self initWithNativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
+    _source = source;
+  }
+  return self;
+}
+
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                               type:(RTCMediaStreamTrackType)type {
+  NSParameterAssert(nativeTrack);
+  NSParameterAssert(type == RTCMediaStreamTrackTypeAudio);
+  return [super initWithNativeTrack:nativeTrack type:type];
+}
+
+
+- (RTCAudioSource *)source {
+  if (!_source) {
+    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+        self.nativeAudioTrack->GetSource();
+    if (source) {
+      _source = [[RTCAudioSource alloc] initWithNativeAudioSource:source.get()];
+    }
+  }
+  return _source;
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeAudioTrack {
+  return static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get());
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m

@@ -0,0 +1,447 @@
+/*
+ *  Copyright 2017 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 "WebRTC/RTCCameraVideoCapturer.h"
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoFrameBuffer.h"
+
+#if TARGET_OS_IPHONE
+#import "WebRTC/UIDevice+RTCDevice.h"
+#endif
+
+#import "AVCaptureSession+DevicePosition.h"
+#import "RTCDispatcher+Private.h"
+
+const int64_t kNanosecondsPerSecond = 1000000000;
+
+static inline BOOL IsMediaSubTypeSupported(FourCharCode mediaSubType) {
+  return (mediaSubType == kCVPixelFormatType_420YpCbCr8PlanarFullRange ||
+          mediaSubType == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange);
+}
+
+@interface RTCCameraVideoCapturer ()<AVCaptureVideoDataOutputSampleBufferDelegate>
+@property(nonatomic, readonly) dispatch_queue_t frameQueue;
+@end
+
+@implementation RTCCameraVideoCapturer {
+  AVCaptureVideoDataOutput *_videoDataOutput;
+  AVCaptureSession *_captureSession;
+  AVCaptureDevice *_currentDevice;
+  BOOL _hasRetriedOnFatalError;
+  BOOL _isRunning;
+  // Will the session be running once all asynchronous operations have been completed?
+  BOOL _willBeRunning;
+#if TARGET_OS_IPHONE
+  UIDeviceOrientation _orientation;
+#endif
+}
+
+@synthesize frameQueue = _frameQueue;
+@synthesize captureSession = _captureSession;
+
+- (instancetype)initWithDelegate:(__weak id<RTCVideoCapturerDelegate>)delegate {
+  if (self = [super initWithDelegate:delegate]) {
+    // Create the capture session and all relevant inputs and outputs. We need
+    // to do this in init because the application may want the capture session
+    // before we start the capturer for e.g. AVCapturePreviewLayer. All objects
+    // created here are retained until dealloc and never recreated.
+    if (![self setupCaptureSession]) {
+      return nil;
+    }
+    NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+#if TARGET_OS_IPHONE
+    _orientation = UIDeviceOrientationPortrait;
+    [center addObserver:self
+               selector:@selector(deviceOrientationDidChange:)
+                   name:UIDeviceOrientationDidChangeNotification
+                 object:nil];
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionInterruption:)
+                   name:AVCaptureSessionWasInterruptedNotification
+                 object:_captureSession];
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionInterruptionEnded:)
+                   name:AVCaptureSessionInterruptionEndedNotification
+                 object:_captureSession];
+    [center addObserver:self
+               selector:@selector(handleApplicationDidBecomeActive:)
+                   name:UIApplicationDidBecomeActiveNotification
+                 object:[UIApplication sharedApplication]];
+#endif
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionRuntimeError:)
+                   name:AVCaptureSessionRuntimeErrorNotification
+                 object:_captureSession];
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionDidStartRunning:)
+                   name:AVCaptureSessionDidStartRunningNotification
+                 object:_captureSession];
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionDidStopRunning:)
+                   name:AVCaptureSessionDidStopRunningNotification
+                 object:_captureSession];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  NSAssert(
+      !_willBeRunning,
+      @"Session was still running in RTCCameraVideoCapturer dealloc. Forgot to call stopCapture?");
+  [[NSNotificationCenter defaultCenter] removeObserver:self];
+}
+
++ (NSArray<AVCaptureDevice *> *)captureDevices {
+  return [AVCaptureDevice devicesWithMediaType:AVMediaTypeVideo];
+}
+
++ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device {
+  NSMutableArray<AVCaptureDeviceFormat *> *eligibleDeviceFormats = [NSMutableArray array];
+
+  for (AVCaptureDeviceFormat *format in device.formats) {
+    // Filter out subTypes that we currently don't support in the stack
+    FourCharCode mediaSubType = CMFormatDescriptionGetMediaSubType(format.formatDescription);
+    if (IsMediaSubTypeSupported(mediaSubType)) {
+      [eligibleDeviceFormats addObject:format];
+    }
+  }
+
+  return eligibleDeviceFormats;
+}
+
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps {
+  _willBeRunning = YES;
+  [RTCDispatcher
+      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                    block:^{
+                      RTCLogInfo("startCaptureWithDevice %@ @ %ld fps", format, (long)fps);
+
+#if TARGET_OS_IPHONE
+                      [[UIDevice currentDevice] beginGeneratingDeviceOrientationNotifications];
+#endif
+
+                      _currentDevice = device;
+
+                      NSError *error = nil;
+                      if (![_currentDevice lockForConfiguration:&error]) {
+                        RTCLogError(
+                            @"Failed to lock device %@. Error: %@", _currentDevice, error.userInfo);
+                        return;
+                      }
+                      [self reconfigureCaptureSessionInput];
+                      [self updateOrientation];
+                      [self updateDeviceCaptureFormat:format fps:fps];
+                      [_captureSession startRunning];
+                      [_currentDevice unlockForConfiguration];
+                      _isRunning = YES;
+                    }];
+}
+
+- (void)stopCapture {
+  _willBeRunning = NO;
+  [RTCDispatcher
+      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                    block:^{
+                      RTCLogInfo("Stop");
+                      _currentDevice = nil;
+                      for (AVCaptureDeviceInput *oldInput in [_captureSession.inputs copy]) {
+                        [_captureSession removeInput:oldInput];
+                      }
+                      [_captureSession stopRunning];
+
+#if TARGET_OS_IPHONE
+                      [[UIDevice currentDevice] endGeneratingDeviceOrientationNotifications];
+#endif
+                      _isRunning = NO;
+                    }];
+}
+
+#pragma mark iOS notifications
+
+#if TARGET_OS_IPHONE
+- (void)deviceOrientationDidChange:(NSNotification *)notification {
+  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                               block:^{
+                                 [self updateOrientation];
+                               }];
+}
+#endif
+
+#pragma mark AVCaptureVideoDataOutputSampleBufferDelegate
+
+- (void)captureOutput:(AVCaptureOutput *)captureOutput
+    didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
+           fromConnection:(AVCaptureConnection *)connection {
+  NSParameterAssert(captureOutput == _videoDataOutput);
+
+  if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) ||
+      !CMSampleBufferDataIsReady(sampleBuffer)) {
+    return;
+  }
+
+  CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
+  if (pixelBuffer == nil) {
+    return;
+  }
+
+#if TARGET_OS_IPHONE
+  // Default to portrait orientation on iPhone.
+  RTCVideoRotation rotation = RTCVideoRotation_90;
+  BOOL usingFrontCamera = NO;
+  // Check the image's EXIF for the camera the image came from as the image could have been
+  // delayed as we set alwaysDiscardsLateVideoFrames to NO.
+  AVCaptureDevicePosition cameraPosition =
+      [AVCaptureSession devicePositionForSampleBuffer:sampleBuffer];
+  if (cameraPosition != AVCaptureDevicePositionUnspecified) {
+    usingFrontCamera = AVCaptureDevicePositionFront == cameraPosition;
+  } else {
+    AVCaptureDeviceInput *deviceInput =
+        (AVCaptureDeviceInput *)((AVCaptureInputPort *)connection.inputPorts.firstObject).input;
+    usingFrontCamera = AVCaptureDevicePositionFront == deviceInput.device.position;
+  }
+  switch (_orientation) {
+    case UIDeviceOrientationPortrait:
+      rotation = RTCVideoRotation_90;
+      break;
+    case UIDeviceOrientationPortraitUpsideDown:
+      rotation = RTCVideoRotation_270;
+      break;
+    case UIDeviceOrientationLandscapeLeft:
+      rotation = usingFrontCamera ? RTCVideoRotation_180 : RTCVideoRotation_0;
+      break;
+    case UIDeviceOrientationLandscapeRight:
+      rotation = usingFrontCamera ? RTCVideoRotation_0 : RTCVideoRotation_180;
+      break;
+    case UIDeviceOrientationFaceUp:
+    case UIDeviceOrientationFaceDown:
+    case UIDeviceOrientationUnknown:
+      // Ignore.
+      break;
+  }
+#else
+  // No rotation on Mac.
+  RTCVideoRotation rotation = RTCVideoRotation_0;
+#endif
+
+  RTCCVPixelBuffer *rtcPixelBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer];
+  int64_t timeStampNs = CMTimeGetSeconds(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)) *
+      kNanosecondsPerSecond;
+  RTCVideoFrame *videoFrame = [[RTCVideoFrame alloc] initWithBuffer:rtcPixelBuffer
+                                                           rotation:rotation
+                                                        timeStampNs:timeStampNs];
+  [self.delegate capturer:self didCaptureVideoFrame:videoFrame];
+}
+
+- (void)captureOutput:(AVCaptureOutput *)captureOutput
+    didDropSampleBuffer:(CMSampleBufferRef)sampleBuffer
+         fromConnection:(AVCaptureConnection *)connection {
+  RTCLogError(@"Dropped sample buffer.");
+}
+
+#pragma mark - AVCaptureSession notifications
+
+- (void)handleCaptureSessionInterruption:(NSNotification *)notification {
+  NSString *reasonString = nil;
+#if defined(__IPHONE_9_0) && defined(__IPHONE_OS_VERSION_MAX_ALLOWED) && \
+    __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_9_0
+  if ([UIDevice isIOS9OrLater]) {
+    NSNumber *reason = notification.userInfo[AVCaptureSessionInterruptionReasonKey];
+    if (reason) {
+      switch (reason.intValue) {
+        case AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableInBackground:
+          reasonString = @"VideoDeviceNotAvailableInBackground";
+          break;
+        case AVCaptureSessionInterruptionReasonAudioDeviceInUseByAnotherClient:
+          reasonString = @"AudioDeviceInUseByAnotherClient";
+          break;
+        case AVCaptureSessionInterruptionReasonVideoDeviceInUseByAnotherClient:
+          reasonString = @"VideoDeviceInUseByAnotherClient";
+          break;
+        case AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableWithMultipleForegroundApps:
+          reasonString = @"VideoDeviceNotAvailableWithMultipleForegroundApps";
+          break;
+      }
+    }
+  }
+#endif
+  RTCLog(@"Capture session interrupted: %@", reasonString);
+}
+
+- (void)handleCaptureSessionInterruptionEnded:(NSNotification *)notification {
+  RTCLog(@"Capture session interruption ended.");
+}
+
+- (void)handleCaptureSessionRuntimeError:(NSNotification *)notification {
+  NSError *error = [notification.userInfo objectForKey:AVCaptureSessionErrorKey];
+  RTCLogError(@"Capture session runtime error: %@", error);
+
+  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                               block:^{
+#if TARGET_OS_IPHONE
+                                 if (error.code == AVErrorMediaServicesWereReset) {
+                                   [self handleNonFatalError];
+                                 } else {
+                                   [self handleFatalError];
+                                 }
+#else
+                                [self handleFatalError];
+#endif
+                               }];
+}
+
+- (void)handleCaptureSessionDidStartRunning:(NSNotification *)notification {
+  RTCLog(@"Capture session started.");
+
+  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                               block:^{
+                                 // If we successfully restarted after an unknown error,
+                                 // allow future retries on fatal errors.
+                                 _hasRetriedOnFatalError = NO;
+                               }];
+}
+
+- (void)handleCaptureSessionDidStopRunning:(NSNotification *)notification {
+  RTCLog(@"Capture session stopped.");
+}
+
+- (void)handleFatalError {
+  [RTCDispatcher
+      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                    block:^{
+                      if (!_hasRetriedOnFatalError) {
+                        RTCLogWarning(@"Attempting to recover from fatal capture error.");
+                        [self handleNonFatalError];
+                        _hasRetriedOnFatalError = YES;
+                      } else {
+                        RTCLogError(@"Previous fatal error recovery failed.");
+                      }
+                    }];
+}
+
+- (void)handleNonFatalError {
+  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                               block:^{
+                                 RTCLog(@"Restarting capture session after error.");
+                                 if (_isRunning) {
+                                   [_captureSession startRunning];
+                                 }
+                               }];
+}
+
+#if TARGET_OS_IPHONE
+
+#pragma mark - UIApplication notifications
+
+- (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
+  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                               block:^{
+                                 if (_isRunning && !_captureSession.isRunning) {
+                                   RTCLog(@"Restarting capture session on active.");
+                                   [_captureSession startRunning];
+                                 }
+                               }];
+}
+
+#endif  // TARGET_OS_IPHONE
+
+#pragma mark - Private
+
+- (dispatch_queue_t)frameQueue {
+  if (!_frameQueue) {
+    _frameQueue =
+        dispatch_queue_create("org.webrtc.avfoundationvideocapturer.video", DISPATCH_QUEUE_SERIAL);
+    dispatch_set_target_queue(_frameQueue,
+                              dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_HIGH, 0));
+  }
+  return _frameQueue;
+}
+
+- (BOOL)setupCaptureSession {
+  NSAssert(_captureSession == nil, @"Setup capture session called twice.");
+  _captureSession = [[AVCaptureSession alloc] init];
+#if defined(WEBRTC_IOS)
+  _captureSession.sessionPreset = AVCaptureSessionPresetInputPriority;
+  _captureSession.usesApplicationAudioSession = NO;
+#endif
+  [self setupVideoDataOutput];
+  // Add the output.
+  if (![_captureSession canAddOutput:_videoDataOutput]) {
+    RTCLogError(@"Video data output unsupported.");
+    return NO;
+  }
+  [_captureSession addOutput:_videoDataOutput];
+
+  return YES;
+}
+
+- (void)setupVideoDataOutput {
+  NSAssert(_videoDataOutput == nil, @"Setup video data output called twice.");
+  // Make the capturer output NV12. Ideally we want I420 but that's not
+  // currently supported on iPhone / iPad.
+  AVCaptureVideoDataOutput *videoDataOutput = [[AVCaptureVideoDataOutput alloc] init];
+  videoDataOutput.videoSettings = @{
+    (NSString *)
+    // TODO(denicija): Remove this color conversion and use the original capture format directly.
+    kCVPixelBufferPixelFormatTypeKey : @(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange)
+  };
+  videoDataOutput.alwaysDiscardsLateVideoFrames = NO;
+  [videoDataOutput setSampleBufferDelegate:self queue:self.frameQueue];
+  _videoDataOutput = videoDataOutput;
+}
+
+#pragma mark - Private, called inside capture queue
+
+- (void)updateDeviceCaptureFormat:(AVCaptureDeviceFormat *)format fps:(NSInteger)fps {
+  NSAssert([RTCDispatcher isOnQueueForType:RTCDispatcherTypeCaptureSession],
+           @"updateDeviceCaptureFormat must be called on the capture queue.");
+  @try {
+    _currentDevice.activeFormat = format;
+    _currentDevice.activeVideoMinFrameDuration = CMTimeMake(1, fps);
+  } @catch (NSException *exception) {
+    RTCLogError(@"Failed to set active format!\n User info:%@", exception.userInfo);
+    return;
+  }
+}
+
+- (void)reconfigureCaptureSessionInput {
+  NSAssert([RTCDispatcher isOnQueueForType:RTCDispatcherTypeCaptureSession],
+           @"reconfigureCaptureSessionInput must be called on the capture queue.");
+  NSError *error = nil;
+  AVCaptureDeviceInput *input =
+      [AVCaptureDeviceInput deviceInputWithDevice:_currentDevice error:&error];
+  if (!input) {
+    RTCLogError(@"Failed to create front camera input: %@", error.localizedDescription);
+    return;
+  }
+  [_captureSession beginConfiguration];
+  for (AVCaptureDeviceInput *oldInput in [_captureSession.inputs copy]) {
+    [_captureSession removeInput:oldInput];
+  }
+  if ([_captureSession canAddInput:input]) {
+    [_captureSession addInput:input];
+  } else {
+    RTCLogError(@"Cannot add camera as an input to the session.");
+  }
+  [_captureSession commitConfiguration];
+}
+
+- (void)updateOrientation {
+  NSAssert([RTCDispatcher isOnQueueForType:RTCDispatcherTypeCaptureSession],
+           @"updateOrientation must be called on the capture queue.");
+#if TARGET_OS_IPHONE
+  _orientation = [UIDevice currentDevice].orientation;
+#endif
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h

@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2015 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/RTCConfiguration.h"
+
+#include "webrtc/api/peerconnectioninterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCConfiguration ()
+
++ (webrtc::PeerConnectionInterface::IceTransportsType)
+    nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy;
+
++ (RTCIceTransportPolicy)transportPolicyForTransportsType:
+    (webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
+
++ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy;
+
++ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
+    (RTCBundlePolicy)policy;
+
++ (RTCBundlePolicy)bundlePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
+
++ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy;
+
++ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
+    (RTCRtcpMuxPolicy)policy;
+
++ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
+
++ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy;
+
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
+    nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy;
+
++ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
+
++ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
+
++ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)
+    nativeCandidateNetworkPolicyForPolicy:(RTCCandidateNetworkPolicy)policy;
+
++ (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy;
+
++ (NSString *)stringForCandidateNetworkPolicy:(RTCCandidateNetworkPolicy)policy;
+
++ (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:(RTCEncryptionKeyType)keyType;
+
+/**
+ * RTCConfiguration struct representation of this RTCConfiguration. This is
+ * needed to pass to the underlying C++ APIs.
+ */
+- (webrtc::PeerConnectionInterface::RTCConfiguration *)
+    createNativeConfiguration;
+
+- (instancetype)initWithNativeConfiguration:
+    (const webrtc::PeerConnectionInterface::RTCConfiguration &)config NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm

@@ -0,0 +1,390 @@
+/*
+ *  Copyright 2015 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 "RTCConfiguration+Private.h"
+
+#include <memory>
+
+#import "RTCIceServer+Private.h"
+#import "RTCIntervalRange+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/rtc_base/rtccertificategenerator.h"
+#include "webrtc/rtc_base/sslidentity.h"
+
+@implementation RTCConfiguration
+
+@synthesize iceServers = _iceServers;
+@synthesize iceTransportPolicy = _iceTransportPolicy;
+@synthesize bundlePolicy = _bundlePolicy;
+@synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
+@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
+@synthesize candidateNetworkPolicy = _candidateNetworkPolicy;
+@synthesize continualGatheringPolicy = _continualGatheringPolicy;
+@synthesize maxIPv6Networks = _maxIPv6Networks;
+@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
+@synthesize audioJitterBufferFastAccelerate = _audioJitterBufferFastAccelerate;
+@synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
+@synthesize iceBackupCandidatePairPingInterval =
+    _iceBackupCandidatePairPingInterval;
+@synthesize keyType = _keyType;
+@synthesize iceCandidatePoolSize = _iceCandidatePoolSize;
+@synthesize shouldPruneTurnPorts = _shouldPruneTurnPorts;
+@synthesize shouldPresumeWritableWhenFullyRelayed =
+    _shouldPresumeWritableWhenFullyRelayed;
+@synthesize iceCheckMinInterval = _iceCheckMinInterval;
+@synthesize iceRegatherIntervalRange = _iceRegatherIntervalRange;
+
+- (instancetype)init {
+  // Copy defaults.
+  webrtc::PeerConnectionInterface::RTCConfiguration config(
+    webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive);
+  return [self initWithNativeConfiguration:config];
+}
+
+- (instancetype)initWithNativeConfiguration:
+    (const webrtc::PeerConnectionInterface::RTCConfiguration &)config {
+  if (self = [super init]) {
+    NSMutableArray *iceServers = [NSMutableArray array];
+    for (const webrtc::PeerConnectionInterface::IceServer& server : config.servers) {
+      RTCIceServer *iceServer = [[RTCIceServer alloc] initWithNativeServer:server];
+      [iceServers addObject:iceServer];
+    }
+    _iceServers = iceServers;
+    _iceTransportPolicy =
+        [[self class] transportPolicyForTransportsType:config.type];
+    _bundlePolicy =
+        [[self class] bundlePolicyForNativePolicy:config.bundle_policy];
+    _rtcpMuxPolicy =
+        [[self class] rtcpMuxPolicyForNativePolicy:config.rtcp_mux_policy];
+    _tcpCandidatePolicy = [[self class] tcpCandidatePolicyForNativePolicy:
+        config.tcp_candidate_policy];
+    _candidateNetworkPolicy = [[self class]
+        candidateNetworkPolicyForNativePolicy:config.candidate_network_policy];
+    webrtc::PeerConnectionInterface::ContinualGatheringPolicy nativePolicy =
+    config.continual_gathering_policy;
+    _continualGatheringPolicy =
+        [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
+    _maxIPv6Networks = config.max_ipv6_networks;
+    _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
+    _audioJitterBufferFastAccelerate = config.audio_jitter_buffer_fast_accelerate;
+    _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
+    _iceBackupCandidatePairPingInterval =
+        config.ice_backup_candidate_pair_ping_interval;
+    _keyType = RTCEncryptionKeyTypeECDSA;
+    _iceCandidatePoolSize = config.ice_candidate_pool_size;
+    _shouldPruneTurnPorts = config.prune_turn_ports;
+    _shouldPresumeWritableWhenFullyRelayed =
+        config.presume_writable_when_fully_relayed;
+    if (config.ice_check_min_interval) {
+      _iceCheckMinInterval =
+          [NSNumber numberWithInt:*config.ice_check_min_interval];
+    }
+    if (config.ice_regather_interval_range) {
+      const rtc::IntervalRange &nativeIntervalRange = config.ice_regather_interval_range.value();
+      _iceRegatherIntervalRange =
+          [[RTCIntervalRange alloc] initWithNativeIntervalRange:nativeIntervalRange];
+    }
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return
+      [NSString stringWithFormat:
+                    @"RTCConfiguration: "
+                    @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n%@\n%@\n%d\n}\n",
+                    _iceServers,
+                    [[self class] stringForTransportPolicy:_iceTransportPolicy],
+                    [[self class] stringForBundlePolicy:_bundlePolicy],
+                    [[self class] stringForRtcpMuxPolicy:_rtcpMuxPolicy],
+                    [[self class] stringForTcpCandidatePolicy:_tcpCandidatePolicy],
+                    [[self class] stringForCandidateNetworkPolicy:_candidateNetworkPolicy],
+                    [[self class] stringForContinualGatheringPolicy:_continualGatheringPolicy],
+                    _audioJitterBufferMaxPackets,
+                    _audioJitterBufferFastAccelerate,
+                    _iceConnectionReceivingTimeout,
+                    _iceBackupCandidatePairPingInterval,
+                    _iceCandidatePoolSize,
+                    _shouldPruneTurnPorts,
+                    _shouldPresumeWritableWhenFullyRelayed,
+                    _iceCheckMinInterval,
+                    _iceRegatherIntervalRange,
+                    _maxIPv6Networks];
+}
+
+#pragma mark - Private
+
+- (webrtc::PeerConnectionInterface::RTCConfiguration *)
+    createNativeConfiguration {
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
+      nativeConfig(new webrtc::PeerConnectionInterface::RTCConfiguration(
+          webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive));
+
+  for (RTCIceServer *iceServer in _iceServers) {
+    nativeConfig->servers.push_back(iceServer.nativeServer);
+  }
+  nativeConfig->type =
+      [[self class] nativeTransportsTypeForTransportPolicy:_iceTransportPolicy];
+  nativeConfig->bundle_policy =
+      [[self class] nativeBundlePolicyForPolicy:_bundlePolicy];
+  nativeConfig->rtcp_mux_policy =
+      [[self class] nativeRtcpMuxPolicyForPolicy:_rtcpMuxPolicy];
+  nativeConfig->tcp_candidate_policy =
+      [[self class] nativeTcpCandidatePolicyForPolicy:_tcpCandidatePolicy];
+  nativeConfig->candidate_network_policy = [[self class]
+      nativeCandidateNetworkPolicyForPolicy:_candidateNetworkPolicy];
+  nativeConfig->continual_gathering_policy = [[self class]
+      nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
+  nativeConfig->max_ipv6_networks = _maxIPv6Networks;
+  nativeConfig->audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
+  nativeConfig->audio_jitter_buffer_fast_accelerate =
+      _audioJitterBufferFastAccelerate  ? true : false;
+  nativeConfig->ice_connection_receiving_timeout =
+      _iceConnectionReceivingTimeout;
+  nativeConfig->ice_backup_candidate_pair_ping_interval =
+      _iceBackupCandidatePairPingInterval;
+  rtc::KeyType keyType =
+      [[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
+  // Generate non-default certificate.
+  if (keyType != rtc::KT_DEFAULT) {
+    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
+        rtc::RTCCertificateGenerator::GenerateCertificate(
+            rtc::KeyParams(keyType), rtc::Optional<uint64_t>());
+    if (!certificate) {
+      RTCLogError(@"Failed to generate certificate.");
+      return nullptr;
+    }
+    nativeConfig->certificates.push_back(certificate);
+  }
+  nativeConfig->ice_candidate_pool_size = _iceCandidatePoolSize;
+  nativeConfig->prune_turn_ports = _shouldPruneTurnPorts ? true : false;
+  nativeConfig->presume_writable_when_fully_relayed =
+      _shouldPresumeWritableWhenFullyRelayed ? true : false;
+  if (_iceCheckMinInterval != nil) {
+    nativeConfig->ice_check_min_interval =
+        rtc::Optional<int>(_iceCheckMinInterval.intValue);
+  }
+  if (_iceRegatherIntervalRange != nil) {
+    std::unique_ptr<rtc::IntervalRange> nativeIntervalRange(
+        _iceRegatherIntervalRange.nativeIntervalRange);
+    nativeConfig->ice_regather_interval_range =
+        rtc::Optional<rtc::IntervalRange>(*nativeIntervalRange);
+  }
+
+  return nativeConfig.release();
+}
+
++ (webrtc::PeerConnectionInterface::IceTransportsType)
+    nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy {
+  switch (policy) {
+    case RTCIceTransportPolicyNone:
+      return webrtc::PeerConnectionInterface::kNone;
+    case RTCIceTransportPolicyRelay:
+      return webrtc::PeerConnectionInterface::kRelay;
+    case RTCIceTransportPolicyNoHost:
+      return webrtc::PeerConnectionInterface::kNoHost;
+    case RTCIceTransportPolicyAll:
+      return webrtc::PeerConnectionInterface::kAll;
+  }
+}
+
++ (RTCIceTransportPolicy)transportPolicyForTransportsType:
+    (webrtc::PeerConnectionInterface::IceTransportsType)nativeType {
+  switch (nativeType) {
+    case webrtc::PeerConnectionInterface::kNone:
+      return RTCIceTransportPolicyNone;
+    case webrtc::PeerConnectionInterface::kRelay:
+      return RTCIceTransportPolicyRelay;
+    case webrtc::PeerConnectionInterface::kNoHost:
+      return RTCIceTransportPolicyNoHost;
+    case webrtc::PeerConnectionInterface::kAll:
+      return RTCIceTransportPolicyAll;
+  }
+}
+
++ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy {
+  switch (policy) {
+    case RTCIceTransportPolicyNone:
+      return @"NONE";
+    case RTCIceTransportPolicyRelay:
+      return @"RELAY";
+    case RTCIceTransportPolicyNoHost:
+      return @"NO_HOST";
+    case RTCIceTransportPolicyAll:
+      return @"ALL";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
+    (RTCBundlePolicy)policy {
+  switch (policy) {
+    case RTCBundlePolicyBalanced:
+      return webrtc::PeerConnectionInterface::kBundlePolicyBalanced;
+    case RTCBundlePolicyMaxCompat:
+      return webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat;
+    case RTCBundlePolicyMaxBundle:
+      return webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
+  }
+}
+
++ (RTCBundlePolicy)bundlePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kBundlePolicyBalanced:
+      return RTCBundlePolicyBalanced;
+    case webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat:
+      return RTCBundlePolicyMaxCompat;
+    case webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle:
+      return RTCBundlePolicyMaxBundle;
+  }
+}
+
++ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy {
+  switch (policy) {
+    case RTCBundlePolicyBalanced:
+      return @"BALANCED";
+    case RTCBundlePolicyMaxCompat:
+      return @"MAX_COMPAT";
+    case RTCBundlePolicyMaxBundle:
+      return @"MAX_BUNDLE";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
+    (RTCRtcpMuxPolicy)policy {
+  switch (policy) {
+    case RTCRtcpMuxPolicyNegotiate:
+      return webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
+    case RTCRtcpMuxPolicyRequire:
+      return webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  }
+}
+
++ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate:
+      return RTCRtcpMuxPolicyNegotiate;
+    case webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire:
+      return RTCRtcpMuxPolicyRequire;
+  }
+}
+
++ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy {
+  switch (policy) {
+    case RTCRtcpMuxPolicyNegotiate:
+      return @"NEGOTIATE";
+    case RTCRtcpMuxPolicyRequire:
+      return @"REQUIRE";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
+    nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy {
+  switch (policy) {
+    case RTCTcpCandidatePolicyEnabled:
+      return webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled;
+    case RTCTcpCandidatePolicyDisabled:
+      return webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled;
+  }
+}
+
++ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)
+    nativeCandidateNetworkPolicyForPolicy:(RTCCandidateNetworkPolicy)policy {
+  switch (policy) {
+    case RTCCandidateNetworkPolicyAll:
+      return webrtc::PeerConnectionInterface::kCandidateNetworkPolicyAll;
+    case RTCCandidateNetworkPolicyLowCost:
+      return webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
+  }
+}
+
++ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled:
+      return RTCTcpCandidatePolicyEnabled;
+    case webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled:
+      return RTCTcpCandidatePolicyDisabled;
+  }
+}
+
++ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy {
+  switch (policy) {
+    case RTCTcpCandidatePolicyEnabled:
+      return @"TCP_ENABLED";
+    case RTCTcpCandidatePolicyDisabled:
+      return @"TCP_DISABLED";
+  }
+}
+
++ (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kCandidateNetworkPolicyAll:
+      return RTCCandidateNetworkPolicyAll;
+    case webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost:
+      return RTCCandidateNetworkPolicyLowCost;
+  }
+}
+
++ (NSString *)stringForCandidateNetworkPolicy:
+    (RTCCandidateNetworkPolicy)policy {
+  switch (policy) {
+    case RTCCandidateNetworkPolicyAll:
+      return @"CANDIDATE_ALL_NETWORKS";
+    case RTCCandidateNetworkPolicyLowCost:
+      return @"CANDIDATE_LOW_COST_NETWORKS";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::ContinualGatheringPolicy)
+    nativeContinualGatheringPolicyForPolicy:
+        (RTCContinualGatheringPolicy)policy {
+  switch (policy) {
+    case RTCContinualGatheringPolicyGatherOnce:
+      return webrtc::PeerConnectionInterface::GATHER_ONCE;
+    case RTCContinualGatheringPolicyGatherContinually:
+      return webrtc::PeerConnectionInterface::GATHER_CONTINUALLY;
+  }
+}
+
++ (RTCContinualGatheringPolicy)continualGatheringPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::ContinualGatheringPolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::GATHER_ONCE:
+      return RTCContinualGatheringPolicyGatherOnce;
+    case webrtc::PeerConnectionInterface::GATHER_CONTINUALLY:
+      return RTCContinualGatheringPolicyGatherContinually;
+  }
+}
+
++ (NSString *)stringForContinualGatheringPolicy:
+    (RTCContinualGatheringPolicy)policy {
+  switch (policy) {
+    case RTCContinualGatheringPolicyGatherOnce:
+      return @"GATHER_ONCE";
+    case RTCContinualGatheringPolicyGatherContinually:
+      return @"GATHER_CONTINUALLY";
+  }
+}
+
++ (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:
+    (RTCEncryptionKeyType)keyType {
+  switch (keyType) {
+    case RTCEncryptionKeyTypeRSA:
+      return rtc::KT_RSA;
+    case RTCEncryptionKeyTypeECDSA:
+      return rtc::KT_ECDSA;
+  }
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h

@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2015 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/RTCDataChannel.h"
+
+#include "webrtc/api/datachannelinterface.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCDataBuffer ()
+
+/**
+ * The native DataBuffer representation of this RTCDatabuffer object. This is
+ * needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly) const webrtc::DataBuffer *nativeDataBuffer;
+
+/** Initialize an RTCDataBuffer from a native DataBuffer. */
+- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer&)nativeBuffer;
+
+@end
+
+
+@interface RTCDataChannel ()
+
+/** Initialize an RTCDataChannel from a native DataChannelInterface. */
+- (instancetype)initWithNativeDataChannel:
+    (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel
+    NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::DataChannelInterface::DataState)
+    nativeDataChannelStateForState:(RTCDataChannelState)state;
+
++ (RTCDataChannelState)dataChannelStateForNativeState:
+    (webrtc::DataChannelInterface::DataState)nativeState;
+
++ (NSString *)stringForState:(RTCDataChannelState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel.mm

@@ -0,0 +1,220 @@
+/*
+ *  Copyright 2015 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 "RTCDataChannel+Private.h"
+
+#import "NSString+StdString.h"
+
+#include <memory>
+
+namespace webrtc {
+
+class DataChannelDelegateAdapter : public DataChannelObserver {
+ public:
+  DataChannelDelegateAdapter(RTCDataChannel *channel) { channel_ = channel; }
+
+  void OnStateChange() override {
+    [channel_.delegate dataChannelDidChangeState:channel_];
+  }
+
+  void OnMessage(const DataBuffer& buffer) override {
+    RTCDataBuffer *data_buffer =
+        [[RTCDataBuffer alloc] initWithNativeBuffer:buffer];
+    [channel_.delegate dataChannel:channel_
+       didReceiveMessageWithBuffer:data_buffer];
+  }
+
+  void OnBufferedAmountChange(uint64_t previousAmount) override {
+    id<RTCDataChannelDelegate> delegate = channel_.delegate;
+    SEL sel = @selector(dataChannel:didChangeBufferedAmount:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate dataChannel:channel_ didChangeBufferedAmount:previousAmount];
+    }
+  }
+
+ private:
+  __weak RTCDataChannel *channel_;
+};
+}
+
+
+@implementation RTCDataBuffer {
+  std::unique_ptr<webrtc::DataBuffer> _dataBuffer;
+}
+
+- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary {
+  NSParameterAssert(data);
+  if (self = [super init]) {
+    rtc::CopyOnWriteBuffer buffer(
+        reinterpret_cast<const uint8_t*>(data.bytes), data.length);
+    _dataBuffer.reset(new webrtc::DataBuffer(buffer, isBinary));
+  }
+  return self;
+}
+
+- (NSData *)data {
+  return [NSData dataWithBytes:_dataBuffer->data.data()
+                        length:_dataBuffer->data.size()];
+}
+
+- (BOOL)isBinary {
+  return _dataBuffer->binary;
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer&)nativeBuffer {
+  if (self = [super init]) {
+    _dataBuffer.reset(new webrtc::DataBuffer(nativeBuffer));
+  }
+  return self;
+}
+
+- (const webrtc::DataBuffer *)nativeDataBuffer {
+  return _dataBuffer.get();
+}
+
+@end
+
+
+@implementation RTCDataChannel {
+  rtc::scoped_refptr<webrtc::DataChannelInterface> _nativeDataChannel;
+  std::unique_ptr<webrtc::DataChannelDelegateAdapter> _observer;
+  BOOL _isObserverRegistered;
+}
+
+@synthesize delegate = _delegate;
+
+- (void)dealloc {
+  // Handles unregistering the observer properly. We need to do this because
+  // there may still be other references to the underlying data channel.
+  _nativeDataChannel->UnregisterObserver();
+}
+
+- (NSString *)label {
+  return [NSString stringForStdString:_nativeDataChannel->label()];
+}
+
+- (BOOL)isReliable {
+  return _nativeDataChannel->reliable();
+}
+
+- (BOOL)isOrdered {
+  return _nativeDataChannel->ordered();
+}
+
+- (NSUInteger)maxRetransmitTime {
+  return self.maxPacketLifeTime;
+}
+
+- (uint16_t)maxPacketLifeTime {
+  return _nativeDataChannel->maxRetransmitTime();
+}
+
+- (uint16_t)maxRetransmits {
+  return _nativeDataChannel->maxRetransmits();
+}
+
+- (NSString *)protocol {
+  return [NSString stringForStdString:_nativeDataChannel->protocol()];
+}
+
+- (BOOL)isNegotiated {
+  return _nativeDataChannel->negotiated();
+}
+
+- (NSInteger)streamId {
+  return self.channelId;
+}
+
+- (int)channelId {
+  return _nativeDataChannel->id();
+}
+
+- (RTCDataChannelState)readyState {
+  return [[self class] dataChannelStateForNativeState:
+      _nativeDataChannel->state()];
+}
+
+- (uint64_t)bufferedAmount {
+  return _nativeDataChannel->buffered_amount();
+}
+
+- (void)close {
+  _nativeDataChannel->Close();
+}
+
+- (BOOL)sendData:(RTCDataBuffer *)data {
+  return _nativeDataChannel->Send(*data.nativeDataBuffer);
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCDataChannel:\n%ld\n%@\n%@",
+                                    (long)self.channelId,
+                                    self.label,
+                                    [[self class]
+                                        stringForState:self.readyState]];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeDataChannel:
+    (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel {
+  NSParameterAssert(nativeDataChannel);
+  if (self = [super init]) {
+    _nativeDataChannel = nativeDataChannel;
+    _observer.reset(new webrtc::DataChannelDelegateAdapter(self));
+    _nativeDataChannel->RegisterObserver(_observer.get());
+  }
+  return self;
+}
+
++ (webrtc::DataChannelInterface::DataState)
+    nativeDataChannelStateForState:(RTCDataChannelState)state {
+  switch (state) {
+    case RTCDataChannelStateConnecting:
+      return webrtc::DataChannelInterface::DataState::kConnecting;
+    case RTCDataChannelStateOpen:
+      return webrtc::DataChannelInterface::DataState::kOpen;
+    case RTCDataChannelStateClosing:
+      return webrtc::DataChannelInterface::DataState::kClosing;
+    case RTCDataChannelStateClosed:
+      return webrtc::DataChannelInterface::DataState::kClosed;
+  }
+}
+
++ (RTCDataChannelState)dataChannelStateForNativeState:
+    (webrtc::DataChannelInterface::DataState)nativeState {
+  switch (nativeState) {
+    case webrtc::DataChannelInterface::DataState::kConnecting:
+      return RTCDataChannelStateConnecting;
+    case webrtc::DataChannelInterface::DataState::kOpen:
+      return RTCDataChannelStateOpen;
+    case webrtc::DataChannelInterface::DataState::kClosing:
+      return RTCDataChannelStateClosing;
+    case webrtc::DataChannelInterface::DataState::kClosed:
+      return RTCDataChannelStateClosed;
+  }
+}
+
++ (NSString *)stringForState:(RTCDataChannelState)state {
+  switch (state) {
+    case RTCDataChannelStateConnecting:
+      return @"Connecting";
+    case RTCDataChannelStateOpen:
+      return @"Open";
+    case RTCDataChannelStateClosing:
+      return @"Closing";
+    case RTCDataChannelStateClosed:
+      return @"Closed";
+  }
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCDataChannelConfiguration+Private.h

@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2015 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/RTCDataChannelConfiguration.h"
+
+#include "webrtc/api/datachannelinterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCDataChannelConfiguration ()
+
+@property(nonatomic, readonly) webrtc::DataChannelInit nativeDataChannelInit;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCDataChannelConfiguration.mm

@@ -0,0 +1,83 @@
+/*
+ *  Copyright 2015 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 "RTCDataChannelConfiguration+Private.h"
+
+#import "NSString+StdString.h"
+
+@implementation RTCDataChannelConfiguration
+
+@synthesize nativeDataChannelInit = _nativeDataChannelInit;
+
+- (BOOL)isOrdered {
+  return _nativeDataChannelInit.ordered;
+}
+
+- (void)setIsOrdered:(BOOL)isOrdered {
+  _nativeDataChannelInit.ordered = isOrdered;
+}
+
+- (NSInteger)maxRetransmitTimeMs {
+  return self.maxPacketLifeTime;
+}
+
+- (void)setMaxRetransmitTimeMs:(NSInteger)maxRetransmitTimeMs {
+  self.maxPacketLifeTime = maxRetransmitTimeMs;
+}
+
+- (int)maxPacketLifeTime {
+  return _nativeDataChannelInit.maxRetransmitTime;
+}
+
+- (void)setMaxPacketLifeTime:(int)maxPacketLifeTime {
+  _nativeDataChannelInit.maxRetransmitTime = maxPacketLifeTime;
+}
+
+- (int)maxRetransmits {
+  return _nativeDataChannelInit.maxRetransmits;
+}
+
+- (void)setMaxRetransmits:(int)maxRetransmits {
+  _nativeDataChannelInit.maxRetransmits = maxRetransmits;
+}
+
+- (NSString *)protocol {
+  return [NSString stringForStdString:_nativeDataChannelInit.protocol];
+}
+
+- (void)setProtocol:(NSString *)protocol {
+  _nativeDataChannelInit.protocol = [NSString stdStringForString:protocol];
+}
+
+- (BOOL)isNegotiated {
+  return _nativeDataChannelInit.negotiated;
+}
+
+- (void)setIsNegotiated:(BOOL)isNegotiated {
+  _nativeDataChannelInit.negotiated = isNegotiated;
+}
+
+- (int)streamId {
+  return self.channelId;
+}
+
+- (void)setStreamId:(int)streamId {
+  self.channelId = streamId;
+}
+
+- (int)channelId {
+  return _nativeDataChannelInit.id;
+}
+
+- (void)setChannelId:(int)channelId {
+  _nativeDataChannelInit.id = channelId;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCEncodedImage.mm

@@ -0,0 +1,83 @@
+/*
+ *  Copyright 2017 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/RTCVideoCodec.h"
+
+#import "RTCVideoCodec+Private.h"
+
+#include "webrtc/rtc_base/safe_conversions.h"
+
+@implementation RTCEncodedImage
+
+@synthesize buffer = _buffer;
+@synthesize encodedWidth = _encodedWidth;
+@synthesize encodedHeight = _encodedHeight;
+@synthesize timeStamp = _timeStamp;
+@synthesize captureTimeMs = _captureTimeMs;
+@synthesize ntpTimeMs = _ntpTimeMs;
+@synthesize flags = _flags;
+@synthesize encodeStartMs = _encodeStartMs;
+@synthesize encodeFinishMs = _encodeFinishMs;
+@synthesize frameType = _frameType;
+@synthesize rotation = _rotation;
+@synthesize completeFrame = _completeFrame;
+@synthesize qp = _qp;
+@synthesize contentType = _contentType;
+
+- (instancetype)initWithNativeEncodedImage:(webrtc::EncodedImage)encodedImage {
+  if (self = [super init]) {
+    // Wrap the buffer in NSData without copying, do not take ownership.
+    _buffer = [NSData dataWithBytesNoCopy:encodedImage._buffer
+                                   length:encodedImage._length
+                             freeWhenDone:NO];
+    _encodedWidth = rtc::dchecked_cast<int32_t>(encodedImage._encodedWidth);
+    _encodedHeight = rtc::dchecked_cast<int32_t>(encodedImage._encodedHeight);
+    _timeStamp = encodedImage._timeStamp;
+    _captureTimeMs = encodedImage.capture_time_ms_;
+    _ntpTimeMs = encodedImage.ntp_time_ms_;
+    _flags = encodedImage.timing_.flags;
+    _encodeStartMs = encodedImage.timing_.encode_start_ms;
+    _encodeFinishMs = encodedImage.timing_.encode_finish_ms;
+    _frameType = static_cast<RTCFrameType>(encodedImage._frameType);
+    _rotation = static_cast<RTCVideoRotation>(encodedImage.rotation_);
+    _completeFrame = encodedImage._completeFrame;
+    _qp = encodedImage.qp_ == -1 ? nil : @(encodedImage.qp_);
+    _contentType = (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
+        RTCVideoContentTypeScreenshare :
+        RTCVideoContentTypeUnspecified;
+  }
+
+  return self;
+}
+
+- (webrtc::EncodedImage)nativeEncodedImage {
+  // Return the pointer without copying.
+  webrtc::EncodedImage encodedImage(
+      (uint8_t *)_buffer.bytes, (size_t)_buffer.length, (size_t)_buffer.length);
+  encodedImage._encodedWidth = rtc::dchecked_cast<uint32_t>(_encodedWidth);
+  encodedImage._encodedHeight = rtc::dchecked_cast<uint32_t>(_encodedHeight);
+  encodedImage._timeStamp = _timeStamp;
+  encodedImage.capture_time_ms_ = _captureTimeMs;
+  encodedImage.ntp_time_ms_ = _ntpTimeMs;
+  encodedImage.timing_.flags = _flags;
+  encodedImage.timing_.encode_start_ms = _encodeStartMs;
+  encodedImage.timing_.encode_finish_ms = _encodeFinishMs;
+  encodedImage._frameType = webrtc::FrameType(_frameType);
+  encodedImage.rotation_ = webrtc::VideoRotation(_rotation);
+  encodedImage._completeFrame = _completeFrame;
+  encodedImage.qp_ = _qp ? _qp.intValue : -1;
+  encodedImage.content_type_ = (_contentType == RTCVideoContentTypeScreenshare) ?
+      webrtc::VideoContentType::SCREENSHARE :
+      webrtc::VideoContentType::UNSPECIFIED;
+
+  return encodedImage;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCFileVideoCapturer.h

@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 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 <WebRTC/RTCVideoCapturer.h>
+
+/**
+ * RTCVideoCapturer that reads buffers from file.
+ *
+ * Per design, the file capturer can only be run once and once stopped it cannot run again.
+ * To run another file capture session, create new instance of the class.
+ */
+@interface RTCFileVideoCapturer : RTCVideoCapturer
+
+- (void)startCapturingFromFileNamed:(NSString *)nameOfFile;
+- (void)stopCapture;
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCFileVideoCapturer.m

@@ -0,0 +1,163 @@
+/**
+ *  Copyright 2017 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 "RTCFileVideoCapturer.h"
+
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoFrameBuffer.h"
+
+@implementation RTCFileVideoCapturer {
+  AVAssetReader *_reader;
+  AVAssetReaderTrackOutput *_outTrack;
+  BOOL _capturerStopped;
+  CMTime _lastPresentationTime;
+  dispatch_queue_t _frameQueue;
+}
+
+- (void)startCapturingFromFileNamed:(NSString *)nameOfFile {
+  dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
+    if (_reader && _reader.status == AVAssetReaderStatusReading) {
+      RTCLog("Capturer exists and reads another file. Start capture request failed.");
+      return;
+    }
+    NSString *pathForFile = [self pathForFileName:nameOfFile];
+    if (!pathForFile) {
+      RTCLog("File %@ not found in bundle", nameOfFile);
+      return;
+    }
+
+    _lastPresentationTime = CMTimeMake(0, 0);
+
+    NSURL *URLForFile = [NSURL fileURLWithPath:pathForFile];
+    AVURLAsset *asset = [AVURLAsset URLAssetWithURL:URLForFile options:nil];
+
+    NSArray *allTracks = [asset tracksWithMediaType:AVMediaTypeVideo];
+    NSError *error = nil;
+    _reader = [[AVAssetReader alloc] initWithAsset:asset error:&error];
+    if (error) {
+      RTCLog("File reader failed with error: %@", error);
+      return;
+    }
+
+    NSDictionary *options = @{
+      (NSString *)
+      kCVPixelBufferPixelFormatTypeKey : @(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange)
+    };
+    _outTrack = [[AVAssetReaderTrackOutput alloc] initWithTrack:allTracks.firstObject
+                                                 outputSettings:options];
+    [_reader addOutput:_outTrack];
+
+    [_reader startReading];
+    RTCLog(@"File capturer started reading");
+    [self readNextBuffer];
+  });
+}
+
+- (void)stopCapture {
+  _capturerStopped = YES;
+  RTCLog(@"File capturer stopped.");
+}
+
+#pragma mark - Private
+
+- (nullable NSString *)pathForFileName:(NSString *)fileName {
+  NSArray *nameComponents = [fileName componentsSeparatedByString:@"."];
+  if (nameComponents.count != 2) {
+    return nil;
+  }
+
+  NSString *path =
+      [[NSBundle mainBundle] pathForResource:nameComponents[0] ofType:nameComponents[1]];
+  return path;
+}
+
+- (dispatch_queue_t)frameQueue {
+  if (!_frameQueue) {
+    _frameQueue = dispatch_queue_create("org.webrtc.filecapturer.video", DISPATCH_QUEUE_SERIAL);
+    dispatch_set_target_queue(_frameQueue,
+                              dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_BACKGROUND, 0));
+  }
+  return _frameQueue;
+}
+
+- (void)readNextBuffer {
+  if (_reader.status != AVAssetReaderStatusReading || _capturerStopped) {
+    [_reader cancelReading];
+    _reader = nil;
+    return;
+  }
+
+  CMSampleBufferRef sampleBuffer = [_outTrack copyNextSampleBuffer];
+  if (!sampleBuffer) {
+    [self readNextBuffer];
+    return;
+  }
+  if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) ||
+      !CMSampleBufferDataIsReady(sampleBuffer)) {
+    [self readNextBuffer];
+    return;
+  }
+
+  [self publishSampleBuffer:sampleBuffer];
+}
+
+- (void)publishSampleBuffer:(CMSampleBufferRef)sampleBuffer {
+  CMTime presentationTime = CMSampleBufferGetPresentationTimeStamp(sampleBuffer);
+  Float64 presentationDifference =
+      CMTimeGetSeconds(CMTimeSubtract(presentationTime, _lastPresentationTime));
+  _lastPresentationTime = presentationTime;
+  int64_t presentationDifferenceRound = lroundf(presentationDifference * NSEC_PER_SEC);
+
+  __block dispatch_source_t timer = [self createStrictTimer];
+  // Strict timer that will fire |presentationDifferenceRound| ns from now and never again.
+  dispatch_source_set_timer(timer,
+                            dispatch_time(DISPATCH_TIME_NOW, presentationDifferenceRound),
+                            DISPATCH_TIME_FOREVER,
+                            0);
+  dispatch_source_set_event_handler(timer, ^{
+    dispatch_source_cancel(timer);
+    timer = nil;
+
+    CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
+    if (!pixelBuffer) {
+      CFRelease(sampleBuffer);
+      dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
+        [self readNextBuffer];
+      });
+      return;
+    }
+
+    RTCCVPixelBuffer *rtcPixelBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer];
+    NSTimeInterval timeStampSeconds = CACurrentMediaTime();
+    int64_t timeStampNs = lroundf(timeStampSeconds * NSEC_PER_SEC);
+    RTCVideoFrame *videoFrame =
+        [[RTCVideoFrame alloc] initWithBuffer:rtcPixelBuffer rotation:0 timeStampNs:timeStampNs];
+    CFRelease(sampleBuffer);
+
+    dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
+      [self readNextBuffer];
+    });
+
+    [self.delegate capturer:self didCaptureVideoFrame:videoFrame];
+  });
+  dispatch_activate(timer);
+}
+
+- (dispatch_source_t)createStrictTimer {
+  dispatch_source_t timer = dispatch_source_create(
+      DISPATCH_SOURCE_TYPE_TIMER, 0, DISPATCH_TIMER_STRICT, [self frameQueue]);
+  return timer;
+}
+
+- (void)dealloc {
+  [self stopCapture];
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h

@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 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/RTCIceCandidate.h"
+
+#include <memory>
+
+#include "webrtc/api/jsep.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceCandidate ()
+
+/**
+ * The native IceCandidateInterface representation of this RTCIceCandidate
+ * object. This is needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly)
+    std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate;
+
+/**
+ * Initialize an RTCIceCandidate from a native IceCandidateInterface. No
+ * ownership is taken of the native candidate.
+ */
+- (instancetype)initWithNativeCandidate:
+    (const webrtc::IceCandidateInterface *)candidate;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate.mm

@@ -0,0 +1,76 @@
+/*
+ *  Copyright 2015 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 "RTCIceCandidate+Private.h"
+
+#include <memory>
+
+#import "NSString+StdString.h"
+#import "WebRTC/RTCLogging.h"
+
+@implementation RTCIceCandidate
+
+@synthesize sdpMid = _sdpMid;
+@synthesize sdpMLineIndex = _sdpMLineIndex;
+@synthesize sdp = _sdp;
+@synthesize serverUrl = _serverUrl;
+
+- (instancetype)initWithSdp:(NSString *)sdp
+              sdpMLineIndex:(int)sdpMLineIndex
+                     sdpMid:(NSString *)sdpMid {
+  NSParameterAssert(sdp.length);
+  if (self = [super init]) {
+    _sdpMid = [sdpMid copy];
+    _sdpMLineIndex = sdpMLineIndex;
+    _sdp = [sdp copy];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCIceCandidate:\n%@\n%d\n%@\n%@",
+                                    _sdpMid,
+                                    _sdpMLineIndex,
+                                    _sdp,
+                                    _serverUrl];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeCandidate:
+    (const webrtc::IceCandidateInterface *)candidate {
+  NSParameterAssert(candidate);
+  std::string sdp;
+  candidate->ToString(&sdp);
+
+  RTCIceCandidate *rtcCandidate =
+      [self initWithSdp:[NSString stringForStdString:sdp]
+          sdpMLineIndex:candidate->sdp_mline_index()
+                 sdpMid:[NSString stringForStdString:candidate->sdp_mid()]];
+  rtcCandidate->_serverUrl = [NSString stringForStdString:candidate->server_url()];
+  return rtcCandidate;
+}
+
+- (std::unique_ptr<webrtc::IceCandidateInterface>)nativeCandidate {
+  webrtc::SdpParseError error;
+
+  webrtc::IceCandidateInterface *candidate = webrtc::CreateIceCandidate(
+      _sdpMid.stdString, _sdpMLineIndex, _sdp.stdString, &error);
+
+  if (!candidate) {
+    RTCLog(@"Failed to create ICE candidate: %s\nline: %s",
+           error.description.c_str(),
+           error.line.c_str());
+  }
+
+  return std::unique_ptr<webrtc::IceCandidateInterface>(candidate);
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h

@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2015 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/RTCIceServer.h"
+
+#include "webrtc/api/peerconnectioninterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceServer ()
+
+/**
+ * IceServer struct representation of this RTCIceServer object's data.
+ * This is needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly)
+    webrtc::PeerConnectionInterface::IceServer nativeServer;
+
+/** Initialize an RTCIceServer from a native IceServer. */
+- (instancetype)initWithNativeServer:
+    (webrtc::PeerConnectionInterface::IceServer)nativeServer;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCIceServer.mm

@@ -0,0 +1,196 @@
+/*
+ *  Copyright 2015 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 "RTCIceServer+Private.h"
+
+#import "NSString+StdString.h"
+
+@implementation RTCIceServer
+
+@synthesize urlStrings = _urlStrings;
+@synthesize username = _username;
+@synthesize credential = _credential;
+@synthesize tlsCertPolicy = _tlsCertPolicy;
+@synthesize hostname = _hostname;
+@synthesize tlsAlpnProtocols = _tlsAlpnProtocols;
+@synthesize tlsEllipticCurves = _tlsEllipticCurves;
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings {
+  return [self initWithURLStrings:urlStrings
+                         username:nil
+                       credential:nil];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential {
+  return [self initWithURLStrings:urlStrings
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:RTCTlsCertPolicySecure];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy {
+  return [self initWithURLStrings:urlStrings
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:tlsCertPolicy
+                         hostname:nil];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(NSString *)hostname {
+  return [self initWithURLStrings:urlStrings
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:tlsCertPolicy
+                         hostname:hostname
+                 tlsAlpnProtocols:[NSArray array]];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(NSString *)hostname
+                  tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols {
+  return [self initWithURLStrings:urlStrings
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:tlsCertPolicy
+                         hostname:hostname
+                 tlsAlpnProtocols:tlsAlpnProtocols
+                tlsEllipticCurves:[NSArray array]];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(NSString *)hostname
+                  tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols
+                 tlsEllipticCurves:(NSArray<NSString *> *)tlsEllipticCurves {
+  NSParameterAssert(urlStrings.count);
+  if (self = [super init]) {
+    _urlStrings = [[NSArray alloc] initWithArray:urlStrings copyItems:YES];
+    _username = [username copy];
+    _credential = [credential copy];
+    _tlsCertPolicy = tlsCertPolicy;
+    _hostname = [hostname copy];
+    _tlsAlpnProtocols = [[NSArray alloc] initWithArray:tlsAlpnProtocols copyItems:YES];
+    _tlsEllipticCurves = [[NSArray alloc] initWithArray:tlsEllipticCurves copyItems:YES];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCIceServer:\n%@\n%@\n%@\n%@\n%@\n%@\n%@",
+                                    _urlStrings,
+                                    _username,
+                                    _credential,
+                                    [self stringForTlsCertPolicy:_tlsCertPolicy],
+                                    _hostname,
+                                    _tlsAlpnProtocols,
+                                    _tlsEllipticCurves];
+}
+
+#pragma mark - Private
+
+- (NSString *)stringForTlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy {
+  switch (tlsCertPolicy) {
+    case RTCTlsCertPolicySecure:
+      return @"RTCTlsCertPolicySecure";
+    case RTCTlsCertPolicyInsecureNoCheck:
+      return @"RTCTlsCertPolicyInsecureNoCheck";
+  }
+}
+
+- (webrtc::PeerConnectionInterface::IceServer)nativeServer {
+  __block webrtc::PeerConnectionInterface::IceServer iceServer;
+
+  iceServer.username = [NSString stdStringForString:_username];
+  iceServer.password = [NSString stdStringForString:_credential];
+  iceServer.hostname = [NSString stdStringForString:_hostname];
+
+  [_tlsAlpnProtocols enumerateObjectsUsingBlock:^(NSString *proto, NSUInteger idx, BOOL *stop) {
+    iceServer.tls_alpn_protocols.push_back(proto.stdString);
+  }];
+
+  [_tlsEllipticCurves enumerateObjectsUsingBlock:^(NSString *curve, NSUInteger idx, BOOL *stop) {
+    iceServer.tls_elliptic_curves.push_back(curve.stdString);
+  }];
+
+  [_urlStrings enumerateObjectsUsingBlock:^(NSString *url,
+                                            NSUInteger idx,
+                                            BOOL *stop) {
+    iceServer.urls.push_back(url.stdString);
+  }];
+
+  switch (_tlsCertPolicy) {
+    case RTCTlsCertPolicySecure:
+      iceServer.tls_cert_policy =
+          webrtc::PeerConnectionInterface::kTlsCertPolicySecure;
+      break;
+    case RTCTlsCertPolicyInsecureNoCheck:
+      iceServer.tls_cert_policy =
+          webrtc::PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck;
+      break;
+  }
+  return iceServer;
+}
+
+- (instancetype)initWithNativeServer:
+    (webrtc::PeerConnectionInterface::IceServer)nativeServer {
+  NSMutableArray *urls =
+      [NSMutableArray arrayWithCapacity:nativeServer.urls.size()];
+  for (auto const &url : nativeServer.urls) {
+    [urls addObject:[NSString stringForStdString:url]];
+  }
+  NSString *username = [NSString stringForStdString:nativeServer.username];
+  NSString *credential = [NSString stringForStdString:nativeServer.password];
+  NSString *hostname = [NSString stringForStdString:nativeServer.hostname];
+  NSMutableArray *tlsAlpnProtocols =
+      [NSMutableArray arrayWithCapacity:nativeServer.tls_alpn_protocols.size()];
+  for (auto const &proto : nativeServer.tls_alpn_protocols) {
+    [tlsAlpnProtocols addObject:[NSString stringForStdString:proto]];
+  }
+  NSMutableArray *tlsEllipticCurves =
+      [NSMutableArray arrayWithCapacity:nativeServer.tls_elliptic_curves.size()];
+  for (auto const &curve : nativeServer.tls_elliptic_curves) {
+    [tlsEllipticCurves addObject:[NSString stringForStdString:curve]];
+  }
+  RTCTlsCertPolicy tlsCertPolicy;
+
+  switch (nativeServer.tls_cert_policy) {
+    case webrtc::PeerConnectionInterface::kTlsCertPolicySecure:
+      tlsCertPolicy = RTCTlsCertPolicySecure;
+      break;
+    case webrtc::PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck:
+      tlsCertPolicy = RTCTlsCertPolicyInsecureNoCheck;
+      break;
+  }
+
+  self = [self initWithURLStrings:urls
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:tlsCertPolicy
+                         hostname:hostname
+                 tlsAlpnProtocols:tlsAlpnProtocols
+                tlsEllipticCurves:tlsEllipticCurves];
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h

@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2017 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/RTCIntervalRange.h"
+
+#include "webrtc/rtc_base/timeutils.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIntervalRange ()
+
+@property(nonatomic, readonly)
+    std::unique_ptr<rtc::IntervalRange> nativeIntervalRange;
+
+- (instancetype)initWithNativeIntervalRange:(const rtc::IntervalRange &)config;
+
+@end
+
+NS_ASSUME_NONNULL_END
+

sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange.mm

@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2017 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 "RTCIntervalRange+Private.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+@implementation RTCIntervalRange
+
+@synthesize min = _min;
+@synthesize max = _max;
+
+- (instancetype)init {
+  return [self initWithMin:0 max:0];
+}
+
+- (instancetype)initWithMin:(NSInteger)min
+                        max:(NSInteger)max {
+  RTC_DCHECK_LE(min, max);
+  if (self = [super init]) {
+    _min = min;
+    _max = max;
+  }
+  return self;
+}
+
+- (instancetype)initWithNativeIntervalRange:(const rtc::IntervalRange &)config {
+  return [self initWithMin:config.min() max:config.max()];
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"[%ld, %ld]", (long)_min, (long)_max];
+}
+
+#pragma mark - Private
+
+- (std::unique_ptr<rtc::IntervalRange>)nativeIntervalRange {
+  std::unique_ptr<rtc::IntervalRange> nativeIntervalRange(
+      new rtc::IntervalRange((int)_min, (int)_max));
+  return nativeIntervalRange;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCLegacyStatsReport+Private.h

@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2015 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/RTCLegacyStatsReport.h"
+
+#include "webrtc/api/statstypes.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCLegacyStatsReport ()
+
+/** Initialize an RTCLegacyStatsReport object from a native StatsReport. */
+- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCLegacyStatsReport.mm

@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2015 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 "RTCLegacyStatsReport+Private.h"
+
+#import "NSString+StdString.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+@implementation RTCLegacyStatsReport
+
+@synthesize timestamp = _timestamp;
+@synthesize type = _type;
+@synthesize reportId = _reportId;
+@synthesize values = _values;
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCLegacyStatsReport:\n%@\n%@\n%f\n%@",
+                                    _reportId,
+                                    _type,
+                                    _timestamp,
+                                    _values];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport {
+  if (self = [super init]) {
+    _timestamp = nativeReport.timestamp();
+    _type = [NSString stringForStdString:nativeReport.TypeToString()];
+    _reportId = [NSString stringForStdString:
+        nativeReport.id()->ToString()];
+
+    NSUInteger capacity = nativeReport.values().size();
+    NSMutableDictionary *values =
+        [NSMutableDictionary dictionaryWithCapacity:capacity];
+    for (auto const &valuePair : nativeReport.values()) {
+      NSString *key = [NSString stringForStdString:
+          valuePair.second->display_name()];
+      NSString *value = [NSString stringForStdString:
+          valuePair.second->ToString()];
+
+      // Not expecting duplicate keys.
+      RTC_DCHECK(![values objectForKey:key]);
+      [values setObject:value forKey:key];
+    }
+    _values = values;
+  }
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h

@@ -0,0 +1,54 @@
+/*
+ *  Copyright 2015 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/RTCMediaConstraints.h"
+
+#include <memory>
+
+#include "webrtc/api/mediaconstraintsinterface.h"
+
+namespace webrtc {
+
+class MediaConstraints : public MediaConstraintsInterface {
+ public:
+  virtual ~MediaConstraints();
+  MediaConstraints();
+  MediaConstraints(
+      const MediaConstraintsInterface::Constraints& mandatory,
+      const MediaConstraintsInterface::Constraints& optional);
+  virtual const Constraints& GetMandatory() const;
+  virtual const Constraints& GetOptional() const;
+
+ private:
+  MediaConstraintsInterface::Constraints mandatory_;
+  MediaConstraintsInterface::Constraints optional_;
+};
+
+}  // namespace webrtc
+
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaConstraints ()
+
+/**
+ * A MediaConstraints representation of this RTCMediaConstraints object. This is
+ * needed to pass to the underlying C++ APIs.
+ */
+- (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints;
+
+/** Return a native Constraints object representing these constraints */
++ (webrtc::MediaConstraintsInterface::Constraints)
+    nativeConstraintsForConstraints:
+        (NSDictionary<NSString *, NSString *> *)constraints;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints.mm

@@ -0,0 +1,138 @@
+/*
+ *  Copyright 2015 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 "RTCMediaConstraints+Private.h"
+
+#import "NSString+StdString.h"
+
+#include <memory>
+
+NSString * const kRTCMediaConstraintsMinAspectRatio =
+    @(webrtc::MediaConstraintsInterface::kMinAspectRatio);
+NSString * const kRTCMediaConstraintsMaxAspectRatio =
+    @(webrtc::MediaConstraintsInterface::kMaxAspectRatio);
+NSString * const kRTCMediaConstraintsMinWidth =
+    @(webrtc::MediaConstraintsInterface::kMinWidth);
+NSString * const kRTCMediaConstraintsMaxWidth =
+    @(webrtc::MediaConstraintsInterface::kMaxWidth);
+NSString * const kRTCMediaConstraintsMinHeight =
+    @(webrtc::MediaConstraintsInterface::kMinHeight);
+NSString * const kRTCMediaConstraintsMaxHeight =
+    @(webrtc::MediaConstraintsInterface::kMaxHeight);
+NSString * const kRTCMediaConstraintsMinFrameRate =
+    @(webrtc::MediaConstraintsInterface::kMinFrameRate);
+NSString * const kRTCMediaConstraintsMaxFrameRate =
+    @(webrtc::MediaConstraintsInterface::kMaxFrameRate);
+NSString * const kRTCMediaConstraintsLevelControl =
+    @(webrtc::MediaConstraintsInterface::kLevelControl);
+NSString * const kRTCMediaConstraintsAudioNetworkAdaptorConfig =
+    @(webrtc::MediaConstraintsInterface::kAudioNetworkAdaptorConfig);
+
+NSString * const kRTCMediaConstraintsIceRestart =
+    @(webrtc::MediaConstraintsInterface::kIceRestart);
+NSString * const kRTCMediaConstraintsOfferToReceiveAudio =
+    @(webrtc::MediaConstraintsInterface::kOfferToReceiveAudio);
+NSString * const kRTCMediaConstraintsOfferToReceiveVideo =
+    @(webrtc::MediaConstraintsInterface::kOfferToReceiveVideo);
+NSString * const kRTCMediaConstraintsVoiceActivityDetection =
+    @(webrtc::MediaConstraintsInterface::kVoiceActivityDetection);
+
+NSString * const kRTCMediaConstraintsValueTrue =
+    @(webrtc::MediaConstraintsInterface::kValueTrue);
+NSString * const kRTCMediaConstraintsValueFalse =
+    @(webrtc::MediaConstraintsInterface::kValueFalse);
+
+namespace webrtc {
+
+MediaConstraints::~MediaConstraints() {}
+
+MediaConstraints::MediaConstraints() {}
+
+MediaConstraints::MediaConstraints(
+    const MediaConstraintsInterface::Constraints& mandatory,
+    const MediaConstraintsInterface::Constraints& optional)
+    : mandatory_(mandatory), optional_(optional) {}
+
+const MediaConstraintsInterface::Constraints&
+MediaConstraints::GetMandatory() const {
+  return mandatory_;
+}
+
+const MediaConstraintsInterface::Constraints&
+MediaConstraints::GetOptional() const {
+  return optional_;
+}
+
+}  // namespace webrtc
+
+
+@implementation RTCMediaConstraints {
+  NSDictionary<NSString *, NSString *> *_mandatory;
+  NSDictionary<NSString *, NSString *> *_optional;
+}
+
+- (instancetype)initWithMandatoryConstraints:
+    (NSDictionary<NSString *, NSString *> *)mandatory
+                         optionalConstraints:
+    (NSDictionary<NSString *, NSString *> *)optional {
+  if (self = [super init]) {
+    _mandatory = [[NSDictionary alloc] initWithDictionary:mandatory
+                                                copyItems:YES];
+    _optional = [[NSDictionary alloc] initWithDictionary:optional
+                                               copyItems:YES];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCMediaConstraints:\n%@\n%@",
+                                    _mandatory,
+                                    _optional];
+}
+
+#pragma mark - Private
+
+- (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints {
+  webrtc::MediaConstraintsInterface::Constraints mandatory =
+      [[self class] nativeConstraintsForConstraints:_mandatory];
+  webrtc::MediaConstraintsInterface::Constraints optional =
+      [[self class] nativeConstraintsForConstraints:_optional];
+
+  webrtc::MediaConstraints *nativeConstraints =
+      new webrtc::MediaConstraints(mandatory, optional);
+  return std::unique_ptr<webrtc::MediaConstraints>(nativeConstraints);
+}
+
++ (webrtc::MediaConstraintsInterface::Constraints)
+    nativeConstraintsForConstraints:
+        (NSDictionary<NSString *, NSString *> *)constraints {
+  webrtc::MediaConstraintsInterface::Constraints nativeConstraints;
+  for (NSString *key in constraints) {
+    NSAssert([key isKindOfClass:[NSString class]],
+             @"%@ is not an NSString.", key);
+    NSString *value = [constraints objectForKey:key];
+    NSAssert([value isKindOfClass:[NSString class]],
+             @"%@ is not an NSString.", value);
+    if ([kRTCMediaConstraintsAudioNetworkAdaptorConfig isEqualToString:key]) {
+      // This value is base64 encoded.
+      NSData *charData = [[NSData alloc] initWithBase64EncodedString:value options:0];
+      std::string configValue =
+          std::string(reinterpret_cast<const char *>(charData.bytes), charData.length);
+      nativeConstraints.push_back(webrtc::MediaConstraintsInterface::Constraint(
+          key.stdString, configValue));
+    } else {
+      nativeConstraints.push_back(webrtc::MediaConstraintsInterface::Constraint(
+          key.stdString, value.stdString));
+    }
+  }
+  return nativeConstraints;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h

@@ -0,0 +1,42 @@
+/*
+ *  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/RTCMediaSource.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(NSInteger, RTCMediaSourceType) {
+  RTCMediaSourceTypeAudio,
+  RTCMediaSourceTypeVideo,
+};
+
+@interface RTCMediaSource ()
+
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
+
+- (instancetype)initWithNativeMediaSource:
+   (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                                     type:(RTCMediaSourceType)type
+    NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
+    (RTCSourceState)state;
+
++ (RTCSourceState)sourceStateForNativeState:
+    (webrtc::MediaSourceInterface::SourceState)nativeState;
+
++ (NSString *)stringForState:(RTCSourceState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource.mm

@@ -0,0 +1,79 @@
+/*
+ *  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 "RTCMediaSource+Private.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+@implementation RTCMediaSource {
+  RTCMediaSourceType _type;
+}
+
+@synthesize nativeMediaSource = _nativeMediaSource;
+
+- (instancetype)initWithNativeMediaSource:
+   (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                                     type:(RTCMediaSourceType)type {
+  RTC_DCHECK(nativeMediaSource);
+  if (self = [super init]) {
+    _nativeMediaSource = nativeMediaSource;
+    _type = type;
+  }
+  return self;
+}
+
+- (RTCSourceState)state {
+  return [[self class] sourceStateForNativeState:_nativeMediaSource->state()];
+}
+
+#pragma mark - Private
+
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
+    (RTCSourceState)state {
+  switch (state) {
+    case RTCSourceStateInitializing:
+      return webrtc::MediaSourceInterface::kInitializing;
+    case RTCSourceStateLive:
+      return webrtc::MediaSourceInterface::kLive;
+    case RTCSourceStateEnded:
+      return webrtc::MediaSourceInterface::kEnded;
+    case RTCSourceStateMuted:
+      return webrtc::MediaSourceInterface::kMuted;
+  }
+}
+
++ (RTCSourceState)sourceStateForNativeState:
+    (webrtc::MediaSourceInterface::SourceState)nativeState {
+  switch (nativeState) {
+    case webrtc::MediaSourceInterface::kInitializing:
+      return RTCSourceStateInitializing;
+    case webrtc::MediaSourceInterface::kLive:
+      return RTCSourceStateLive;
+    case webrtc::MediaSourceInterface::kEnded:
+      return RTCSourceStateEnded;
+    case webrtc::MediaSourceInterface::kMuted:
+      return RTCSourceStateMuted;
+  }
+}
+
++ (NSString *)stringForState:(RTCSourceState)state {
+  switch (state) {
+    case RTCSourceStateInitializing:
+      return @"Initializing";
+    case RTCSourceStateLive:
+      return @"Live";
+    case RTCSourceStateEnded:
+      return @"Ended";
+    case RTCSourceStateMuted:
+      return @"Muted";
+  }
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h

@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2015 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/RTCMediaStream.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaStream ()
+
+/**
+ * MediaStreamInterface representation of this RTCMediaStream object. This is
+ * needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
+
+/** Initialize an RTCMediaStream with an id. */
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                       streamId:(NSString *)streamId;
+
+/** Initialize an RTCMediaStream from a native MediaStreamInterface. */
+- (instancetype)initWithNativeMediaStream:
+    (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream.mm

@@ -0,0 +1,122 @@
+/*
+ *  Copyright 2015 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 "RTCMediaStream+Private.h"
+
+#include <vector>
+
+#import "NSString+StdString.h"
+#import "RTCAudioTrack+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCVideoTrack+Private.h"
+
+@implementation RTCMediaStream {
+  NSMutableArray *_audioTracks;
+  NSMutableArray *_videoTracks;
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> _nativeMediaStream;
+}
+
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                       streamId:(NSString *)streamId {
+  NSParameterAssert(factory);
+  NSParameterAssert(streamId.length);
+  std::string nativeId = [NSString stdStringForString:streamId];
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+      factory.nativeFactory->CreateLocalMediaStream(nativeId);
+  return [self initWithNativeMediaStream:stream];
+}
+
+- (NSArray<RTCAudioTrack *> *)audioTracks {
+  return [_audioTracks copy];
+}
+
+- (NSArray<RTCVideoTrack *> *)videoTracks {
+  return [_videoTracks copy];
+}
+
+- (NSString *)streamId {
+  return [NSString stringForStdString:_nativeMediaStream->label()];
+}
+
+- (void)addAudioTrack:(RTCAudioTrack *)audioTrack {
+  if (_nativeMediaStream->AddTrack(audioTrack.nativeAudioTrack)) {
+    [_audioTracks addObject:audioTrack];
+  }
+}
+
+- (void)addVideoTrack:(RTCVideoTrack *)videoTrack {
+  if (_nativeMediaStream->AddTrack(videoTrack.nativeVideoTrack)) {
+    [_videoTracks addObject:videoTrack];
+  }
+}
+
+- (void)removeAudioTrack:(RTCAudioTrack *)audioTrack {
+  NSUInteger index = [_audioTracks indexOfObjectIdenticalTo:audioTrack];
+  NSAssert(index != NSNotFound,
+           @"|removeAudioTrack| called on unexpected RTCAudioTrack");
+  if (index != NSNotFound &&
+      _nativeMediaStream->RemoveTrack(audioTrack.nativeAudioTrack)) {
+    [_audioTracks removeObjectAtIndex:index];
+  }
+}
+
+- (void)removeVideoTrack:(RTCVideoTrack *)videoTrack {
+  NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:videoTrack];
+  NSAssert(index != NSNotFound,
+           @"|removeVideoTrack| called on unexpected RTCVideoTrack");
+  if (index != NSNotFound &&
+      _nativeMediaStream->RemoveTrack(videoTrack.nativeVideoTrack)) {
+    [_videoTracks removeObjectAtIndex:index];
+  }
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCMediaStream:\n%@\nA=%lu\nV=%lu",
+                                    self.streamId,
+                                    (unsigned long)self.audioTracks.count,
+                                    (unsigned long)self.videoTracks.count];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
+  return _nativeMediaStream;
+}
+
+- (instancetype)initWithNativeMediaStream:
+    (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
+  NSParameterAssert(nativeMediaStream);
+  if (self = [super init]) {
+    webrtc::AudioTrackVector audioTracks = nativeMediaStream->GetAudioTracks();
+    webrtc::VideoTrackVector videoTracks = nativeMediaStream->GetVideoTracks();
+
+    _audioTracks = [NSMutableArray arrayWithCapacity:audioTracks.size()];
+    _videoTracks = [NSMutableArray arrayWithCapacity:videoTracks.size()];
+    _nativeMediaStream = nativeMediaStream;
+
+    for (auto &track : audioTracks) {
+      RTCMediaStreamTrackType type = RTCMediaStreamTrackTypeAudio;
+      RTCAudioTrack *audioTrack =
+          [[RTCAudioTrack alloc] initWithNativeTrack:track type:type];
+      [_audioTracks addObject:audioTrack];
+    }
+
+    for (auto &track : videoTracks) {
+      RTCMediaStreamTrackType type = RTCMediaStreamTrackTypeVideo;
+      RTCVideoTrack *videoTrack =
+          [[RTCVideoTrack alloc] initWithNativeTrack:track type:type];
+      [_videoTracks addObject:videoTrack];
+    }
+  }
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h

@@ -0,0 +1,54 @@
+/*
+ *  Copyright 2015 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/RTCMediaStreamTrack.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+typedef NS_ENUM(NSInteger, RTCMediaStreamTrackType) {
+  RTCMediaStreamTrackTypeAudio,
+  RTCMediaStreamTrackTypeVideo,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaStreamTrack ()
+
+/**
+ * The native MediaStreamTrackInterface passed in or created during
+ * construction.
+ */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
+
+/**
+ * Initialize an RTCMediaStreamTrack from a native MediaStreamTrackInterface.
+ */
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                               type:(RTCMediaStreamTrackType)type
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
+
+- (BOOL)isEqualToTrack:(RTCMediaStreamTrack *)track;
+
++ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
+    (RTCMediaStreamTrackState)state;
+
++ (RTCMediaStreamTrackState)trackStateForNativeState:
+    (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
+
++ (NSString *)stringForState:(RTCMediaStreamTrackState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack.mm

@@ -0,0 +1,137 @@
+/*
+ *  Copyright 2015 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 "RTCMediaStreamTrack+Private.h"
+
+#import "NSString+StdString.h"
+
+NSString * const kRTCMediaStreamTrackKindAudio =
+    @(webrtc::MediaStreamTrackInterface::kAudioKind);
+NSString * const kRTCMediaStreamTrackKindVideo =
+    @(webrtc::MediaStreamTrackInterface::kVideoKind);
+
+@implementation RTCMediaStreamTrack {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> _nativeTrack;
+  RTCMediaStreamTrackType _type;
+}
+
+- (NSString *)kind {
+  return [NSString stringForStdString:_nativeTrack->kind()];
+}
+
+- (NSString *)trackId {
+  return [NSString stringForStdString:_nativeTrack->id()];
+}
+
+- (BOOL)isEnabled {
+  return _nativeTrack->enabled();
+}
+
+- (void)setIsEnabled:(BOOL)isEnabled {
+  _nativeTrack->set_enabled(isEnabled);
+}
+
+- (RTCMediaStreamTrackState)readyState {
+  return [[self class] trackStateForNativeState:_nativeTrack->state()];
+}
+
+- (NSString *)description {
+  NSString *readyState = [[self class] stringForState:self.readyState];
+  return [NSString stringWithFormat:@"RTCMediaStreamTrack:\n%@\n%@\n%@\n%@",
+                                    self.kind,
+                                    self.trackId,
+                                    self.isEnabled ? @"enabled" : @"disabled",
+                                    readyState];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  return [self isEqualToTrack:(RTCMediaStreamTrack *)object];
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeTrack.get();
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+  return _nativeTrack;
+}
+
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                               type:(RTCMediaStreamTrackType)type {
+  NSParameterAssert(nativeTrack);
+  if (self = [super init]) {
+    _nativeTrack = nativeTrack;
+    _type = type;
+  }
+  return self;
+}
+
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+  NSParameterAssert(nativeTrack);
+  if (nativeTrack->kind() ==
+      std::string(webrtc::MediaStreamTrackInterface::kAudioKind)) {
+    return [self initWithNativeTrack:nativeTrack
+                                type:RTCMediaStreamTrackTypeAudio];
+  }
+  if (nativeTrack->kind() ==
+      std::string(webrtc::MediaStreamTrackInterface::kVideoKind)) {
+    return [self initWithNativeTrack:nativeTrack
+                                type:RTCMediaStreamTrackTypeVideo];
+  }
+  return nil;
+}
+
+- (BOOL)isEqualToTrack:(RTCMediaStreamTrack *)track {
+  if (!track) {
+    return NO;
+  }
+  return _nativeTrack == track.nativeTrack;
+}
+
++ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
+    (RTCMediaStreamTrackState)state {
+  switch (state) {
+    case RTCMediaStreamTrackStateLive:
+      return webrtc::MediaStreamTrackInterface::kLive;
+    case RTCMediaStreamTrackStateEnded:
+      return webrtc::MediaStreamTrackInterface::kEnded;
+  }
+}
+
++ (RTCMediaStreamTrackState)trackStateForNativeState:
+    (webrtc::MediaStreamTrackInterface::TrackState)nativeState {
+  switch (nativeState) {
+    case webrtc::MediaStreamTrackInterface::kLive:
+      return RTCMediaStreamTrackStateLive;
+    case webrtc::MediaStreamTrackInterface::kEnded:
+      return RTCMediaStreamTrackStateEnded;
+  }
+}
+
++ (NSString *)stringForState:(RTCMediaStreamTrackState)state {
+  switch (state) {
+    case RTCMediaStreamTrackStateLive:
+      return @"Live";
+    case RTCMediaStreamTrackStateEnded:
+      return @"Ended";
+  }
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCMetrics.mm

@@ -0,0 +1,32 @@
+/*
+ *  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/RTCMetrics.h"
+
+#import "RTCMetricsSampleInfo+Private.h"
+
+void RTCEnableMetrics() {
+  webrtc::metrics::Enable();
+}
+
+NSArray<RTCMetricsSampleInfo *> *RTCGetAndResetMetrics() {
+  std::map<std::string, std::unique_ptr<webrtc::metrics::SampleInfo>>
+      histograms;
+  webrtc::metrics::GetAndReset(&histograms);
+
+  NSMutableArray *metrics =
+      [NSMutableArray arrayWithCapacity:histograms.size()];
+  for (auto const &histogram : histograms) {
+    RTCMetricsSampleInfo *metric = [[RTCMetricsSampleInfo alloc]
+        initWithNativeSampleInfo:*histogram.second];
+    [metrics addObject:metric];
+  }
+  return metrics;
+}

sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h

@@ -0,0 +1,28 @@
+/*
+ *  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/RTCMetricsSampleInfo.h"
+
+// Adding 'nogncheck' to disable the gn include headers check.
+// We don't want to depend on 'system_wrappers:metrics_default' because
+// clients should be able to provide their own implementation.
+#include "webrtc/system_wrappers/include/metrics_default.h"  // nogncheck
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMetricsSampleInfo ()
+
+/** Initialize an RTCMetricsSampleInfo object from native SampleInfo. */
+- (instancetype)initWithNativeSampleInfo:
+    (const webrtc::metrics::SampleInfo &)info;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo.mm

@@ -0,0 +1,43 @@
+/*
+ *  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 "RTCMetricsSampleInfo+Private.h"
+
+#import "NSString+StdString.h"
+
+@implementation RTCMetricsSampleInfo
+
+@synthesize name = _name;
+@synthesize min = _min;
+@synthesize max = _max;
+@synthesize bucketCount = _bucketCount;
+@synthesize samples = _samples;
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeSampleInfo:
+    (const webrtc::metrics::SampleInfo &)info {
+  if (self = [super init]) {
+    _name = [NSString stringForStdString:info.name];
+    _min = info.min;
+    _max = info.max;
+    _bucketCount = info.bucket_count;
+
+    NSMutableDictionary *samples =
+        [NSMutableDictionary dictionaryWithCapacity:info.samples.size()];
+    for (auto const &sample : info.samples) {
+      [samples setObject:@(sample.second) forKey:@(sample.first)];
+    }
+    _samples = samples;
+  }
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+DataChannel.mm

@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2015 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 "RTCPeerConnection+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCDataChannel+Private.h"
+#import "RTCDataChannelConfiguration+Private.h"
+
+@implementation RTCPeerConnection (DataChannel)
+
+- (RTCDataChannel *)dataChannelForLabel:(NSString *)label
+                          configuration:
+    (RTCDataChannelConfiguration *)configuration {
+  std::string labelString = [NSString stdStringForString:label];
+  const webrtc::DataChannelInit nativeInit =
+      configuration.nativeDataChannelInit;
+  rtc::scoped_refptr<webrtc::DataChannelInterface> dataChannel =
+      self.nativePeerConnection->CreateDataChannel(labelString,
+                                                   &nativeInit);
+  if (!dataChannel) {
+    return nil;
+  }
+  return [[RTCDataChannel alloc] initWithNativeDataChannel:dataChannel];
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h

@@ -0,0 +1,107 @@
+/*
+ *  Copyright 2015 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/RTCPeerConnection.h"
+
+#include "webrtc/api/peerconnectioninterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+namespace webrtc {
+
+/**
+ * These objects are created by RTCPeerConnectionFactory to wrap an
+ * id<RTCPeerConnectionDelegate> and call methods on that interface.
+ */
+class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
+
+ public:
+  PeerConnectionDelegateAdapter(RTCPeerConnection *peerConnection);
+  virtual ~PeerConnectionDelegateAdapter();
+
+  void OnSignalingChange(
+      PeerConnectionInterface::SignalingState new_state) override;
+
+  void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
+
+  void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
+
+  void OnDataChannel(
+      rtc::scoped_refptr<DataChannelInterface> data_channel) override;
+
+  void OnRenegotiationNeeded() override;
+
+  void OnIceConnectionChange(
+      PeerConnectionInterface::IceConnectionState new_state) override;
+
+  void OnIceGatheringChange(
+      PeerConnectionInterface::IceGatheringState new_state) override;
+
+  void OnIceCandidate(const IceCandidateInterface *candidate) override;
+
+  void OnIceCandidatesRemoved(
+      const std::vector<cricket::Candidate>& candidates) override;
+
+ private:
+  __weak RTCPeerConnection *peer_connection_;
+};
+
+} // namespace webrtc
+
+
+@interface RTCPeerConnection ()
+
+/** The native PeerConnectionInterface created during construction. */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::PeerConnectionInterface> nativePeerConnection;
+
+/** Initialize an RTCPeerConnection with a configuration, constraints, and
+ *  delegate.
+ */
+- (instancetype)initWithFactory:
+    (RTCPeerConnectionFactory *)factory
+                  configuration:
+    (RTCConfiguration *)configuration
+                    constraints:
+    (RTCMediaConstraints *)constraints
+                       delegate:
+    (nullable id<RTCPeerConnectionDelegate>)delegate
+    NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
+    (RTCSignalingState)state;
+
++ (RTCSignalingState)signalingStateForNativeState:
+    (webrtc::PeerConnectionInterface::SignalingState)nativeState;
+
++ (NSString *)stringForSignalingState:(RTCSignalingState)state;
+
++ (webrtc::PeerConnectionInterface::IceConnectionState)
+    nativeIceConnectionStateForState:(RTCIceConnectionState)state;
+
++ (RTCIceConnectionState)iceConnectionStateForNativeState:
+    (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
+
++ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state;
+
++ (webrtc::PeerConnectionInterface::IceGatheringState)
+    nativeIceGatheringStateForState:(RTCIceGatheringState)state;
+
++ (RTCIceGatheringState)iceGatheringStateForNativeState:
+    (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
+
++ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state;
+
++ (webrtc::PeerConnectionInterface::StatsOutputLevel)
+    nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm

@@ -0,0 +1,64 @@
+/*
+ *  Copyright 2015 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 "RTCPeerConnection+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCLegacyStatsReport+Private.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+namespace webrtc {
+
+class StatsObserverAdapter : public StatsObserver {
+ public:
+  StatsObserverAdapter(void (^completionHandler)
+      (NSArray<RTCLegacyStatsReport *> *stats)) {
+    completion_handler_ = completionHandler;
+  }
+
+  ~StatsObserverAdapter() {
+    completion_handler_ = nil;
+  }
+
+  void OnComplete(const StatsReports& reports) override {
+    RTC_DCHECK(completion_handler_);
+    NSMutableArray *stats = [NSMutableArray arrayWithCapacity:reports.size()];
+    for (const auto* report : reports) {
+      RTCLegacyStatsReport *statsReport =
+          [[RTCLegacyStatsReport alloc] initWithNativeReport:*report];
+      [stats addObject:statsReport];
+    }
+    completion_handler_(stats);
+    completion_handler_ = nil;
+  }
+
+ private:
+  void (^completion_handler_)(NSArray<RTCLegacyStatsReport *> *stats);
+};
+}  // namespace webrtc
+
+@implementation RTCPeerConnection (Stats)
+
+- (void)statsForTrack:(RTCMediaStreamTrack *)mediaStreamTrack
+     statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
+    completionHandler:
+    (void (^)(NSArray<RTCLegacyStatsReport *> *stats))completionHandler {
+  rtc::scoped_refptr<webrtc::StatsObserverAdapter> observer(
+      new rtc::RefCountedObject<webrtc::StatsObserverAdapter>
+          (completionHandler));
+  webrtc::PeerConnectionInterface::StatsOutputLevel nativeOutputLevel =
+      [[self class] nativeStatsOutputLevelForLevel:statsOutputLevel];
+  self.nativePeerConnection->GetStats(
+      observer, mediaStreamTrack.nativeTrack, nativeOutputLevel);
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm

@@ -0,0 +1,617 @@
+/*
+ *  Copyright 2015 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 "RTCPeerConnection+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCConfiguration+Private.h"
+#import "RTCDataChannel+Private.h"
+#import "RTCIceCandidate+Private.h"
+#import "RTCLegacyStatsReport+Private.h"
+#import "RTCMediaConstraints+Private.h"
+#import "RTCMediaStream+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCRtpReceiver+Private.h"
+#import "RTCRtpSender+Private.h"
+#import "RTCSessionDescription+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include <memory>
+
+#include "webrtc/api/jsepicecandidate.h"
+#include "webrtc/rtc_base/checks.h"
+
+NSString * const kRTCPeerConnectionErrorDomain =
+    @"org.webrtc.RTCPeerConnection";
+int const kRTCPeerConnnectionSessionDescriptionError = -1;
+
+namespace webrtc {
+
+class CreateSessionDescriptionObserverAdapter
+    : public CreateSessionDescriptionObserver {
+ public:
+  CreateSessionDescriptionObserverAdapter(
+      void (^completionHandler)(RTCSessionDescription *sessionDescription,
+                                NSError *error)) {
+    completion_handler_ = completionHandler;
+  }
+
+  ~CreateSessionDescriptionObserverAdapter() {
+    completion_handler_ = nil;
+  }
+
+  void OnSuccess(SessionDescriptionInterface *desc) override {
+    RTC_DCHECK(completion_handler_);
+    std::unique_ptr<webrtc::SessionDescriptionInterface> description =
+        std::unique_ptr<webrtc::SessionDescriptionInterface>(desc);
+    RTCSessionDescription* session =
+        [[RTCSessionDescription alloc] initWithNativeDescription:
+            description.get()];
+    completion_handler_(session, nil);
+    completion_handler_ = nil;
+  }
+
+  void OnFailure(const std::string& error) override {
+    RTC_DCHECK(completion_handler_);
+    NSString* str = [NSString stringForStdString:error];
+    NSError* err =
+        [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
+                            code:kRTCPeerConnnectionSessionDescriptionError
+                        userInfo:@{ NSLocalizedDescriptionKey : str }];
+    completion_handler_(nil, err);
+    completion_handler_ = nil;
+  }
+
+ private:
+  void (^completion_handler_)
+      (RTCSessionDescription *sessionDescription, NSError *error);
+};
+
+class SetSessionDescriptionObserverAdapter :
+    public SetSessionDescriptionObserver {
+ public:
+  SetSessionDescriptionObserverAdapter(void (^completionHandler)
+      (NSError *error)) {
+    completion_handler_ = completionHandler;
+  }
+
+  ~SetSessionDescriptionObserverAdapter() {
+    completion_handler_ = nil;
+  }
+
+  void OnSuccess() override {
+    RTC_DCHECK(completion_handler_);
+    completion_handler_(nil);
+    completion_handler_ = nil;
+  }
+
+  void OnFailure(const std::string& error) override {
+    RTC_DCHECK(completion_handler_);
+    NSString* str = [NSString stringForStdString:error];
+    NSError* err =
+        [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
+                            code:kRTCPeerConnnectionSessionDescriptionError
+                        userInfo:@{ NSLocalizedDescriptionKey : str }];
+    completion_handler_(err);
+    completion_handler_ = nil;
+  }
+
+ private:
+  void (^completion_handler_)(NSError *error);
+};
+
+PeerConnectionDelegateAdapter::PeerConnectionDelegateAdapter(
+    RTCPeerConnection *peerConnection) {
+  peer_connection_ = peerConnection;
+}
+
+PeerConnectionDelegateAdapter::~PeerConnectionDelegateAdapter() {
+  peer_connection_ = nil;
+}
+
+void PeerConnectionDelegateAdapter::OnSignalingChange(
+    PeerConnectionInterface::SignalingState new_state) {
+  RTCSignalingState state =
+      [[RTCPeerConnection class] signalingStateForNativeState:new_state];
+  RTCPeerConnection *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                   didChangeSignalingState:state];
+}
+
+void PeerConnectionDelegateAdapter::OnAddStream(
+    rtc::scoped_refptr<MediaStreamInterface> stream) {
+  RTCMediaStream *mediaStream =
+      [[RTCMediaStream alloc] initWithNativeMediaStream:stream];
+  RTCPeerConnection *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                              didAddStream:mediaStream];
+}
+
+void PeerConnectionDelegateAdapter::OnRemoveStream(
+    rtc::scoped_refptr<MediaStreamInterface> stream) {
+  RTCMediaStream *mediaStream =
+      [[RTCMediaStream alloc] initWithNativeMediaStream:stream];
+  RTCPeerConnection *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                           didRemoveStream:mediaStream];
+}
+
+void PeerConnectionDelegateAdapter::OnDataChannel(
+    rtc::scoped_refptr<DataChannelInterface> data_channel) {
+  RTCDataChannel *dataChannel =
+      [[RTCDataChannel alloc] initWithNativeDataChannel:data_channel];
+  RTCPeerConnection *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                        didOpenDataChannel:dataChannel];
+}
+
+void PeerConnectionDelegateAdapter::OnRenegotiationNeeded() {
+  RTCPeerConnection *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnectionShouldNegotiate:peer_connection];
+}
+
+void PeerConnectionDelegateAdapter::OnIceConnectionChange(
+    PeerConnectionInterface::IceConnectionState new_state) {
+  RTCIceConnectionState state =
+      [[RTCPeerConnection class] iceConnectionStateForNativeState:new_state];
+  RTCPeerConnection *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+               didChangeIceConnectionState:state];
+}
+
+void PeerConnectionDelegateAdapter::OnIceGatheringChange(
+    PeerConnectionInterface::IceGatheringState new_state) {
+  RTCIceGatheringState state =
+      [[RTCPeerConnection class] iceGatheringStateForNativeState:new_state];
+  RTCPeerConnection *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                didChangeIceGatheringState:state];
+}
+
+void PeerConnectionDelegateAdapter::OnIceCandidate(
+    const IceCandidateInterface *candidate) {
+  RTCIceCandidate *iceCandidate =
+      [[RTCIceCandidate alloc] initWithNativeCandidate:candidate];
+  RTCPeerConnection *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                   didGenerateIceCandidate:iceCandidate];
+}
+
+void PeerConnectionDelegateAdapter::OnIceCandidatesRemoved(
+    const std::vector<cricket::Candidate>& candidates) {
+  NSMutableArray* ice_candidates =
+      [NSMutableArray arrayWithCapacity:candidates.size()];
+  for (const auto& candidate : candidates) {
+    std::unique_ptr<JsepIceCandidate> candidate_wrapper(
+        new JsepIceCandidate(candidate.transport_name(), -1, candidate));
+    RTCIceCandidate* ice_candidate = [[RTCIceCandidate alloc]
+        initWithNativeCandidate:candidate_wrapper.get()];
+    [ice_candidates addObject:ice_candidate];
+  }
+  RTCPeerConnection* peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                    didRemoveIceCandidates:ice_candidates];
+}
+
+}  // namespace webrtc
+
+
+@implementation RTCPeerConnection {
+  NSMutableArray<RTCMediaStream *> *_localStreams;
+  std::unique_ptr<webrtc::PeerConnectionDelegateAdapter> _observer;
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection;
+  std::unique_ptr<webrtc::MediaConstraints> _nativeConstraints;
+  BOOL _hasStartedRtcEventLog;
+}
+
+@synthesize delegate = _delegate;
+
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                  configuration:(RTCConfiguration *)configuration
+                    constraints:(RTCMediaConstraints *)constraints
+                       delegate:(id<RTCPeerConnectionDelegate>)delegate {
+  NSParameterAssert(factory);
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
+      [configuration createNativeConfiguration]);
+  if (!config) {
+    return nil;
+  }
+  if (self = [super init]) {
+    _observer.reset(new webrtc::PeerConnectionDelegateAdapter(self));
+    _nativeConstraints = constraints.nativeConstraints;
+    CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(),
+                                        config.get());
+    _peerConnection =
+        factory.nativeFactory->CreatePeerConnection(*config,
+                                                    nullptr,
+                                                    nullptr,
+                                                    _observer.get());
+    if (!_peerConnection) {
+      return nil;
+    }
+    _localStreams = [[NSMutableArray alloc] init];
+    _delegate = delegate;
+  }
+  return self;
+}
+
+- (NSArray<RTCMediaStream *> *)localStreams {
+  return [_localStreams copy];
+}
+
+- (RTCSessionDescription *)localDescription {
+  const webrtc::SessionDescriptionInterface *description =
+      _peerConnection->local_description();
+  return description ?
+      [[RTCSessionDescription alloc] initWithNativeDescription:description]
+          : nil;
+}
+
+- (RTCSessionDescription *)remoteDescription {
+  const webrtc::SessionDescriptionInterface *description =
+      _peerConnection->remote_description();
+  return description ?
+      [[RTCSessionDescription alloc] initWithNativeDescription:description]
+          : nil;
+}
+
+- (RTCSignalingState)signalingState {
+  return [[self class]
+      signalingStateForNativeState:_peerConnection->signaling_state()];
+}
+
+- (RTCIceConnectionState)iceConnectionState {
+  return [[self class] iceConnectionStateForNativeState:
+      _peerConnection->ice_connection_state()];
+}
+
+- (RTCIceGatheringState)iceGatheringState {
+  return [[self class] iceGatheringStateForNativeState:
+      _peerConnection->ice_gathering_state()];
+}
+
+- (BOOL)setConfiguration:(RTCConfiguration *)configuration {
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
+      [configuration createNativeConfiguration]);
+  if (!config) {
+    return NO;
+  }
+  CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(),
+                                      config.get());
+  return _peerConnection->SetConfiguration(*config);
+}
+
+- (RTCConfiguration *)configuration {
+  webrtc::PeerConnectionInterface::RTCConfiguration config =
+    _peerConnection->GetConfiguration();
+  return [[RTCConfiguration alloc] initWithNativeConfiguration:config];
+}
+
+- (void)close {
+  _peerConnection->Close();
+}
+
+- (void)addIceCandidate:(RTCIceCandidate *)candidate {
+  std::unique_ptr<const webrtc::IceCandidateInterface> iceCandidate(
+      candidate.nativeCandidate);
+  _peerConnection->AddIceCandidate(iceCandidate.get());
+}
+
+- (void)removeIceCandidates:(NSArray<RTCIceCandidate *> *)iceCandidates {
+  std::vector<cricket::Candidate> candidates;
+  for (RTCIceCandidate *iceCandidate in iceCandidates) {
+    std::unique_ptr<const webrtc::IceCandidateInterface> candidate(
+        iceCandidate.nativeCandidate);
+    if (candidate) {
+      candidates.push_back(candidate->candidate());
+      // Need to fill the transport name from the sdp_mid.
+      candidates.back().set_transport_name(candidate->sdp_mid());
+    }
+  }
+  if (!candidates.empty()) {
+    _peerConnection->RemoveIceCandidates(candidates);
+  }
+}
+
+- (void)addStream:(RTCMediaStream *)stream {
+  if (!_peerConnection->AddStream(stream.nativeMediaStream)) {
+    RTCLogError(@"Failed to add stream: %@", stream);
+    return;
+  }
+  [_localStreams addObject:stream];
+}
+
+- (void)removeStream:(RTCMediaStream *)stream {
+  _peerConnection->RemoveStream(stream.nativeMediaStream);
+  [_localStreams removeObject:stream];
+}
+
+- (void)offerForConstraints:(RTCMediaConstraints *)constraints
+          completionHandler:
+    (void (^)(RTCSessionDescription *sessionDescription,
+              NSError *error))completionHandler {
+  rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
+      observer(new rtc::RefCountedObject
+          <webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
+  _peerConnection->CreateOffer(observer, constraints.nativeConstraints.get());
+}
+
+- (void)answerForConstraints:(RTCMediaConstraints *)constraints
+           completionHandler:
+    (void (^)(RTCSessionDescription *sessionDescription,
+              NSError *error))completionHandler {
+  rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
+      observer(new rtc::RefCountedObject
+          <webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
+  _peerConnection->CreateAnswer(observer, constraints.nativeConstraints.get());
+}
+
+- (void)setLocalDescription:(RTCSessionDescription *)sdp
+          completionHandler:(void (^)(NSError *error))completionHandler {
+  rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
+      new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
+          completionHandler));
+  _peerConnection->SetLocalDescription(observer, sdp.nativeDescription);
+}
+
+- (void)setRemoteDescription:(RTCSessionDescription *)sdp
+           completionHandler:(void (^)(NSError *error))completionHandler {
+  rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
+      new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
+          completionHandler));
+  _peerConnection->SetRemoteDescription(observer, sdp.nativeDescription);
+}
+
+- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
+          currentBitrateBps:(nullable NSNumber *)currentBitrateBps
+              maxBitrateBps:(nullable NSNumber *)maxBitrateBps {
+  webrtc::PeerConnectionInterface::BitrateParameters params;
+  if (minBitrateBps != nil) {
+    params.min_bitrate_bps = rtc::Optional<int>(minBitrateBps.intValue);
+  }
+  if (currentBitrateBps != nil) {
+    params.current_bitrate_bps = rtc::Optional<int>(currentBitrateBps.intValue);
+  }
+  if (maxBitrateBps != nil) {
+    params.max_bitrate_bps = rtc::Optional<int>(maxBitrateBps.intValue);
+  }
+  return _peerConnection->SetBitrate(params).ok();
+}
+
+- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
+                      maxSizeInBytes:(int64_t)maxSizeInBytes {
+  RTC_DCHECK(filePath.length);
+  RTC_DCHECK_GT(maxSizeInBytes, 0);
+  RTC_DCHECK(!_hasStartedRtcEventLog);
+  if (_hasStartedRtcEventLog) {
+    RTCLogError(@"Event logging already started.");
+    return NO;
+  }
+  int fd = open(filePath.UTF8String, O_WRONLY | O_CREAT | O_TRUNC,
+                S_IRUSR | S_IWUSR);
+  if (fd < 0) {
+    RTCLogError(@"Error opening file: %@. Error: %d", filePath, errno);
+    return NO;
+  }
+  _hasStartedRtcEventLog =
+      _peerConnection->StartRtcEventLog(fd, maxSizeInBytes);
+  return _hasStartedRtcEventLog;
+}
+
+- (void)stopRtcEventLog {
+  _peerConnection->StopRtcEventLog();
+  _hasStartedRtcEventLog = NO;
+}
+
+- (RTCRtpSender *)senderWithKind:(NSString *)kind
+                        streamId:(NSString *)streamId {
+  std::string nativeKind = [NSString stdStringForString:kind];
+  std::string nativeStreamId = [NSString stdStringForString:streamId];
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender(
+      _peerConnection->CreateSender(nativeKind, nativeStreamId));
+  return nativeSender ?
+      [[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender]
+      : nil;
+}
+
+- (NSArray<RTCRtpSender *> *)senders {
+  std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenders(
+      _peerConnection->GetSenders());
+  NSMutableArray *senders = [[NSMutableArray alloc] init];
+  for (const auto &nativeSender : nativeSenders) {
+    RTCRtpSender *sender =
+        [[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender];
+    [senders addObject:sender];
+  }
+  return senders;
+}
+
+- (NSArray<RTCRtpReceiver *> *)receivers {
+  std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> nativeReceivers(
+      _peerConnection->GetReceivers());
+  NSMutableArray *receivers = [[NSMutableArray alloc] init];
+  for (const auto &nativeReceiver : nativeReceivers) {
+    RTCRtpReceiver *receiver =
+        [[RTCRtpReceiver alloc] initWithNativeRtpReceiver:nativeReceiver];
+    [receivers addObject:receiver];
+  }
+  return receivers;
+}
+
+#pragma mark - Private
+
++ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
+    (RTCSignalingState)state {
+  switch (state) {
+    case RTCSignalingStateStable:
+      return webrtc::PeerConnectionInterface::kStable;
+    case RTCSignalingStateHaveLocalOffer:
+      return webrtc::PeerConnectionInterface::kHaveLocalOffer;
+    case RTCSignalingStateHaveLocalPrAnswer:
+      return webrtc::PeerConnectionInterface::kHaveLocalPrAnswer;
+    case RTCSignalingStateHaveRemoteOffer:
+      return webrtc::PeerConnectionInterface::kHaveRemoteOffer;
+    case RTCSignalingStateHaveRemotePrAnswer:
+      return webrtc::PeerConnectionInterface::kHaveRemotePrAnswer;
+    case RTCSignalingStateClosed:
+      return webrtc::PeerConnectionInterface::kClosed;
+  }
+}
+
++ (RTCSignalingState)signalingStateForNativeState:
+    (webrtc::PeerConnectionInterface::SignalingState)nativeState {
+  switch (nativeState) {
+    case webrtc::PeerConnectionInterface::kStable:
+      return RTCSignalingStateStable;
+    case webrtc::PeerConnectionInterface::kHaveLocalOffer:
+      return RTCSignalingStateHaveLocalOffer;
+    case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer:
+      return RTCSignalingStateHaveLocalPrAnswer;
+    case webrtc::PeerConnectionInterface::kHaveRemoteOffer:
+      return RTCSignalingStateHaveRemoteOffer;
+    case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer:
+      return RTCSignalingStateHaveRemotePrAnswer;
+    case webrtc::PeerConnectionInterface::kClosed:
+      return RTCSignalingStateClosed;
+  }
+}
+
++ (NSString *)stringForSignalingState:(RTCSignalingState)state {
+  switch (state) {
+    case RTCSignalingStateStable:
+      return @"STABLE";
+    case RTCSignalingStateHaveLocalOffer:
+      return @"HAVE_LOCAL_OFFER";
+    case RTCSignalingStateHaveLocalPrAnswer:
+      return @"HAVE_LOCAL_PRANSWER";
+    case RTCSignalingStateHaveRemoteOffer:
+      return @"HAVE_REMOTE_OFFER";
+    case RTCSignalingStateHaveRemotePrAnswer:
+      return @"HAVE_REMOTE_PRANSWER";
+    case RTCSignalingStateClosed:
+      return @"CLOSED";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::IceConnectionState)
+    nativeIceConnectionStateForState:(RTCIceConnectionState)state {
+  switch (state) {
+    case RTCIceConnectionStateNew:
+      return webrtc::PeerConnectionInterface::kIceConnectionNew;
+    case RTCIceConnectionStateChecking:
+      return webrtc::PeerConnectionInterface::kIceConnectionChecking;
+    case RTCIceConnectionStateConnected:
+      return webrtc::PeerConnectionInterface::kIceConnectionConnected;
+    case RTCIceConnectionStateCompleted:
+      return webrtc::PeerConnectionInterface::kIceConnectionCompleted;
+    case RTCIceConnectionStateFailed:
+      return webrtc::PeerConnectionInterface::kIceConnectionFailed;
+    case RTCIceConnectionStateDisconnected:
+      return webrtc::PeerConnectionInterface::kIceConnectionDisconnected;
+    case RTCIceConnectionStateClosed:
+      return webrtc::PeerConnectionInterface::kIceConnectionClosed;
+    case RTCIceConnectionStateCount:
+      return webrtc::PeerConnectionInterface::kIceConnectionMax;
+  }
+}
+
++ (RTCIceConnectionState)iceConnectionStateForNativeState:
+    (webrtc::PeerConnectionInterface::IceConnectionState)nativeState {
+  switch (nativeState) {
+    case webrtc::PeerConnectionInterface::kIceConnectionNew:
+      return RTCIceConnectionStateNew;
+    case webrtc::PeerConnectionInterface::kIceConnectionChecking:
+      return RTCIceConnectionStateChecking;
+    case webrtc::PeerConnectionInterface::kIceConnectionConnected:
+      return RTCIceConnectionStateConnected;
+    case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
+      return RTCIceConnectionStateCompleted;
+    case webrtc::PeerConnectionInterface::kIceConnectionFailed:
+      return RTCIceConnectionStateFailed;
+    case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
+      return RTCIceConnectionStateDisconnected;
+    case webrtc::PeerConnectionInterface::kIceConnectionClosed:
+      return RTCIceConnectionStateClosed;
+    case webrtc::PeerConnectionInterface::kIceConnectionMax:
+      return RTCIceConnectionStateCount;
+  }
+}
+
++ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state {
+  switch (state) {
+    case RTCIceConnectionStateNew:
+      return @"NEW";
+    case RTCIceConnectionStateChecking:
+      return @"CHECKING";
+    case RTCIceConnectionStateConnected:
+      return @"CONNECTED";
+    case RTCIceConnectionStateCompleted:
+      return @"COMPLETED";
+    case RTCIceConnectionStateFailed:
+      return @"FAILED";
+    case RTCIceConnectionStateDisconnected:
+      return @"DISCONNECTED";
+    case RTCIceConnectionStateClosed:
+      return @"CLOSED";
+    case RTCIceConnectionStateCount:
+      return @"COUNT";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::IceGatheringState)
+    nativeIceGatheringStateForState:(RTCIceGatheringState)state {
+  switch (state) {
+    case RTCIceGatheringStateNew:
+      return webrtc::PeerConnectionInterface::kIceGatheringNew;
+    case RTCIceGatheringStateGathering:
+      return webrtc::PeerConnectionInterface::kIceGatheringGathering;
+    case RTCIceGatheringStateComplete:
+      return webrtc::PeerConnectionInterface::kIceGatheringComplete;
+  }
+}
+
++ (RTCIceGatheringState)iceGatheringStateForNativeState:
+    (webrtc::PeerConnectionInterface::IceGatheringState)nativeState {
+  switch (nativeState) {
+    case webrtc::PeerConnectionInterface::kIceGatheringNew:
+      return RTCIceGatheringStateNew;
+    case webrtc::PeerConnectionInterface::kIceGatheringGathering:
+      return RTCIceGatheringStateGathering;
+    case webrtc::PeerConnectionInterface::kIceGatheringComplete:
+      return RTCIceGatheringStateComplete;
+  }
+}
+
++ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state {
+  switch (state) {
+    case RTCIceGatheringStateNew:
+      return @"NEW";
+    case RTCIceGatheringStateGathering:
+      return @"GATHERING";
+    case RTCIceGatheringStateComplete:
+      return @"COMPLETE";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::StatsOutputLevel)
+    nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level {
+  switch (level) {
+    case RTCStatsOutputLevelStandard:
+      return webrtc::PeerConnectionInterface::kStatsOutputLevelStandard;
+    case RTCStatsOutputLevelDebug:
+      return webrtc::PeerConnectionInterface::kStatsOutputLevelDebug;
+  }
+}
+
+- (rtc::scoped_refptr<webrtc::PeerConnectionInterface>)nativePeerConnection {
+  return _peerConnection;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h

@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2017 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/RTCPeerConnectionFactory.h"
+
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+class AudioEncoderFactory;
+class AudioDecoderFactory;
+
+}  // namespace webrtc
+
+namespace cricket {
+
+class WebRtcVideoEncoderFactory;
+class WebRtcVideoDecoderFactory;
+
+}  // namespace cricket
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * This class extension exposes methods that work directly with injectable C++ components.
+ */
+@interface RTCPeerConnectionFactory ()
+
+/* Initialize object with injectable native audio/video encoder/decoder factories */
+- (instancetype)initWithNativeAudioEncoderFactory:
+                    (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+                        nativeAudioDecoderFactory:
+                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                        nativeVideoEncoderFactory:
+                            (nullable cricket::WebRtcVideoEncoderFactory *)videoEncoderFactory
+                        nativeVideoDecoderFactory:
+                            (nullable cricket::WebRtcVideoDecoderFactory *)videoDecoderFactory
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2015 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/RTCPeerConnectionFactory.h"
+
+#include "webrtc/api/peerconnectioninterface.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCPeerConnectionFactory ()
+
+/**
+ * PeerConnectionFactoryInterface created and held by this
+ * RTCPeerConnectionFactory object. This is needed to pass to the underlying
+ * C++ APIs.
+ */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> nativeFactory;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm

@@ -0,0 +1,226 @@
+/*
+ *  Copyright 2015 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 "RTCPeerConnectionFactory+Native.h"
+#import "RTCPeerConnectionFactory+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCAVFoundationVideoSource+Private.h"
+#import "RTCAudioSource+Private.h"
+#import "RTCAudioTrack+Private.h"
+#import "RTCMediaConstraints+Private.h"
+#import "RTCMediaStream+Private.h"
+#import "RTCPeerConnection+Private.h"
+#import "RTCVideoSource+Private.h"
+#import "RTCVideoTrack+Private.h"
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoCodecFactory.h"
+#ifndef HAVE_NO_MEDIA
+#include "VideoToolbox/objc_video_decoder_factory.h"
+#include "VideoToolbox/objc_video_encoder_factory.h"
+#import "WebRTC/RTCVideoCodecH264.h"
+// The no-media version PeerConnectionFactory doesn't depend on these files, but the gn check tool
+// is not smart enough to take the #ifdef into account.
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"  // nogncheck
+#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"  // nogncheck
+#endif
+
+#include "Video/objcvideotracksource.h"
+#include "webrtc/api/videosourceproxy.h"
+// Adding the nogncheck to disable the including header check.
+// The no-media version PeerConnectionFactory doesn't depend on media related
+// C++ target.
+// TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
+// API layer.
+#include "webrtc/media/engine/webrtcmediaengine.h"  // nogncheck
+
+@implementation RTCPeerConnectionFactory {
+  std::unique_ptr<rtc::Thread> _networkThread;
+  std::unique_ptr<rtc::Thread> _workerThread;
+  std::unique_ptr<rtc::Thread> _signalingThread;
+  BOOL _hasStartedAecDump;
+}
+
+@synthesize nativeFactory = _nativeFactory;
+
+- (instancetype)init {
+#ifdef HAVE_NO_MEDIA
+  return [self initWithNativeAudioEncoderFactory:nil
+                       nativeAudioDecoderFactory:nil
+                       nativeVideoEncoderFactory:nil
+                       nativeVideoDecoderFactory:nil];
+#else
+  return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
+                       nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
+                       nativeVideoEncoderFactory:new webrtc::ObjCVideoEncoderFactory(
+                                                     [[RTCVideoEncoderFactoryH264 alloc] init])
+                       nativeVideoDecoderFactory:new webrtc::ObjCVideoDecoderFactory(
+                                                     [[RTCVideoDecoderFactoryH264 alloc] init])];
+#endif
+}
+
+- (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
+                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory {
+#ifdef HAVE_NO_MEDIA
+  return [self initWithNativeAudioEncoderFactory:nil
+                       nativeAudioDecoderFactory:nil
+                       nativeVideoEncoderFactory:nil
+                       nativeVideoDecoderFactory:nil];
+#else
+  cricket::WebRtcVideoEncoderFactory *native_encoder_factory =
+      encoderFactory ? new webrtc::ObjCVideoEncoderFactory(encoderFactory) : nullptr;
+  cricket::WebRtcVideoDecoderFactory *native_decoder_factory =
+      decoderFactory ? new webrtc::ObjCVideoDecoderFactory(decoderFactory) : nullptr;
+  return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
+                       nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
+                       nativeVideoEncoderFactory:native_encoder_factory
+                       nativeVideoDecoderFactory:native_decoder_factory];
+#endif
+}
+
+- (instancetype)initWithNativeAudioEncoderFactory:
+                    (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+                        nativeAudioDecoderFactory:
+                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                        nativeVideoEncoderFactory:
+                            (nullable cricket::WebRtcVideoEncoderFactory *)videoEncoderFactory
+                        nativeVideoDecoderFactory:
+                            (nullable cricket::WebRtcVideoDecoderFactory *)videoDecoderFactory {
+  if (self = [super init]) {
+    _networkThread = rtc::Thread::CreateWithSocketServer();
+    BOOL result = _networkThread->Start();
+    NSAssert(result, @"Failed to start network thread.");
+
+    _workerThread = rtc::Thread::Create();
+    result = _workerThread->Start();
+    NSAssert(result, @"Failed to start worker thread.");
+
+    _signalingThread = rtc::Thread::Create();
+    result = _signalingThread->Start();
+    NSAssert(result, @"Failed to start signaling thread.");
+#ifdef HAVE_NO_MEDIA
+    _nativeFactory = webrtc::CreateModularPeerConnectionFactory(
+        _networkThread.get(),
+        _workerThread.get(),
+        _signalingThread.get(),
+        nullptr,  // default_adm
+        nullptr,  // audio_encoder_factory
+        nullptr,  // audio_decoder_factory
+        nullptr,  // video_encoder_factory
+        nullptr,  // video_decoder_factory
+        nullptr,  // audio_mixer
+        std::unique_ptr<cricket::MediaEngineInterface>(),
+        std::unique_ptr<webrtc::CallFactoryInterface>(),
+        std::unique_ptr<webrtc::RtcEventLogFactoryInterface>());
+#else
+    // Ownership of encoder/decoder factories is passed on to the
+    // peerconnectionfactory, that handles deleting them.
+    _nativeFactory = webrtc::CreatePeerConnectionFactory(_networkThread.get(),
+                                                         _workerThread.get(),
+                                                         _signalingThread.get(),
+                                                         nullptr,  // audio device module
+                                                         audioEncoderFactory,
+                                                         audioDecoderFactory,
+                                                         videoEncoderFactory,
+                                                         videoDecoderFactory);
+#endif
+    NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
+  }
+  return self;
+}
+
+- (RTCAudioSource *)audioSourceWithConstraints:(nullable RTCMediaConstraints *)constraints {
+  std::unique_ptr<webrtc::MediaConstraints> nativeConstraints;
+  if (constraints) {
+    nativeConstraints = constraints.nativeConstraints;
+  }
+  rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+      _nativeFactory->CreateAudioSource(nativeConstraints.get());
+  return [[RTCAudioSource alloc] initWithNativeAudioSource:source];
+}
+
+- (RTCAudioTrack *)audioTrackWithTrackId:(NSString *)trackId {
+  RTCAudioSource *audioSource = [self audioSourceWithConstraints:nil];
+  return [self audioTrackWithSource:audioSource trackId:trackId];
+}
+
+- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source
+                                trackId:(NSString *)trackId {
+  return [[RTCAudioTrack alloc] initWithFactory:self
+                                         source:source
+                                        trackId:trackId];
+}
+
+- (RTCAVFoundationVideoSource *)avFoundationVideoSourceWithConstraints:
+    (nullable RTCMediaConstraints *)constraints {
+#ifdef HAVE_NO_MEDIA
+  return nil;
+#else
+  return [[RTCAVFoundationVideoSource alloc] initWithFactory:self constraints:constraints];
+#endif
+}
+
+- (RTCVideoSource *)videoSource {
+  rtc::scoped_refptr<webrtc::ObjcVideoTrackSource> objcVideoTrackSource(
+      new rtc::RefCountedObject<webrtc::ObjcVideoTrackSource>());
+  return [[RTCVideoSource alloc]
+      initWithNativeVideoSource:webrtc::VideoTrackSourceProxy::Create(_signalingThread.get(),
+                                                                      _workerThread.get(),
+                                                                      objcVideoTrackSource)];
+}
+
+- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source
+                                trackId:(NSString *)trackId {
+  return [[RTCVideoTrack alloc] initWithFactory:self
+                                         source:source
+                                        trackId:trackId];
+}
+
+- (RTCMediaStream *)mediaStreamWithStreamId:(NSString *)streamId {
+  return [[RTCMediaStream alloc] initWithFactory:self
+                                        streamId:streamId];
+}
+
+- (RTCPeerConnection *)peerConnectionWithConfiguration:
+    (RTCConfiguration *)configuration
+                                           constraints:
+    (RTCMediaConstraints *)constraints
+                                              delegate:
+    (nullable id<RTCPeerConnectionDelegate>)delegate {
+  return [[RTCPeerConnection alloc] initWithFactory:self
+                                      configuration:configuration
+                                        constraints:constraints
+                                           delegate:delegate];
+}
+
+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath
+                  maxSizeInBytes:(int64_t)maxSizeInBytes {
+  RTC_DCHECK(filePath.length);
+  RTC_DCHECK_GT(maxSizeInBytes, 0);
+
+  if (_hasStartedAecDump) {
+    RTCLogError(@"Aec dump already started.");
+    return NO;
+  }
+  int fd = open(filePath.UTF8String, O_WRONLY | O_CREAT | O_TRUNC, S_IRUSR | S_IWUSR);
+  if (fd < 0) {
+    RTCLogError(@"Error opening file: %@. Error: %d", filePath, errno);
+    return NO;
+  }
+  _hasStartedAecDump = _nativeFactory->StartAecDump(fd, maxSizeInBytes);
+  return _hasStartedAecDump;
+}
+
+- (void)stopAecDump {
+  _nativeFactory->StopAecDump();
+  _hasStartedAecDump = NO;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h

@@ -0,0 +1,28 @@
+/*
+ *  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/RTCRtpCodecParameters.h"
+
+#include "webrtc/api/rtpparameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpCodecParameters ()
+
+/** Returns the equivalent native RtpCodecParameters structure. */
+@property(nonatomic, readonly) webrtc::RtpCodecParameters nativeParameters;
+
+/** Initialize the object with a native RtpCodecParameters structure. */
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpCodecParameters &)nativeParameters;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm

@@ -0,0 +1,97 @@
+/*
+ *  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 "RTCRtpCodecParameters+Private.h"
+
+#import "NSString+StdString.h"
+#import "WebRTC/RTCMediaStreamTrack.h"  // For "kind" strings.
+
+#include "webrtc/media/base/mediaconstants.h"
+#include "webrtc/rtc_base/checks.h"
+
+const NSString * const kRTCRtxCodecName = @(cricket::kRtxCodecName);
+const NSString * const kRTCRedCodecName = @(cricket::kRedCodecName);
+const NSString * const kRTCUlpfecCodecName = @(cricket::kUlpfecCodecName);
+const NSString * const kRTCFlexfecCodecName = @(cricket::kFlexfecCodecName);
+const NSString * const kRTCOpusCodecName = @(cricket::kOpusCodecName);
+const NSString * const kRTCIsacCodecName = @(cricket::kIsacCodecName);
+const NSString * const kRTCL16CodecName  = @(cricket::kL16CodecName);
+const NSString * const kRTCG722CodecName = @(cricket::kG722CodecName);
+const NSString * const kRTCIlbcCodecName = @(cricket::kIlbcCodecName);
+const NSString * const kRTCPcmuCodecName = @(cricket::kPcmuCodecName);
+const NSString * const kRTCPcmaCodecName = @(cricket::kPcmaCodecName);
+const NSString * const kRTCDtmfCodecName = @(cricket::kDtmfCodecName);
+const NSString * const kRTCComfortNoiseCodecName =
+    @(cricket::kComfortNoiseCodecName);
+const NSString * const kRTCVp8CodecName = @(cricket::kVp8CodecName);
+const NSString * const kRTCVp9CodecName = @(cricket::kVp9CodecName);
+const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
+
+@implementation RTCRtpCodecParameters
+
+@synthesize payloadType = _payloadType;
+@synthesize name = _name;
+@synthesize kind = _kind;
+@synthesize clockRate = _clockRate;
+@synthesize numChannels = _numChannels;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpCodecParameters &)nativeParameters {
+  if (self = [self init]) {
+    _payloadType = nativeParameters.payload_type;
+    _name = [NSString stringForStdString:nativeParameters.name];
+    switch (nativeParameters.kind) {
+      case cricket::MEDIA_TYPE_AUDIO:
+        _kind = kRTCMediaStreamTrackKindAudio;
+        break;
+      case cricket::MEDIA_TYPE_VIDEO:
+        _kind = kRTCMediaStreamTrackKindVideo;
+        break;
+      case cricket::MEDIA_TYPE_DATA:
+        RTC_NOTREACHED();
+        break;
+    }
+    if (nativeParameters.clock_rate) {
+      _clockRate = [NSNumber numberWithInt:*nativeParameters.clock_rate];
+    }
+    if (nativeParameters.num_channels) {
+      _numChannels = [NSNumber numberWithInt:*nativeParameters.num_channels];
+    }
+  }
+  return self;
+}
+
+- (webrtc::RtpCodecParameters)nativeParameters {
+  webrtc::RtpCodecParameters parameters;
+  parameters.payload_type = _payloadType;
+  parameters.name = [NSString stdStringForString:_name];
+  // NSString pointer comparison is safe here since "kind" is readonly and only
+  // populated above.
+  if (_kind == kRTCMediaStreamTrackKindAudio) {
+    parameters.kind = cricket::MEDIA_TYPE_AUDIO;
+  } else if (_kind == kRTCMediaStreamTrackKindVideo) {
+    parameters.kind = cricket::MEDIA_TYPE_VIDEO;
+  } else {
+    RTC_NOTREACHED();
+  }
+  if (_clockRate != nil) {
+    parameters.clock_rate = rtc::Optional<int>(_clockRate.intValue);
+  }
+  if (_numChannels != nil) {
+    parameters.num_channels = rtc::Optional<int>(_numChannels.intValue);
+  }
+  return parameters;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h

@@ -0,0 +1,28 @@
+/*
+ *  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/RTCRtpEncodingParameters.h"
+
+#include "webrtc/api/rtpparameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpEncodingParameters ()
+
+/** Returns the equivalent native RtpEncodingParameters structure. */
+@property(nonatomic, readonly) webrtc::RtpEncodingParameters nativeParameters;
+
+/** Initialize the object with a native RtpEncodingParameters structure. */
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpEncodingParameters &)nativeParameters;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters.mm

@@ -0,0 +1,50 @@
+/*
+ *  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 "RTCRtpEncodingParameters+Private.h"
+
+@implementation RTCRtpEncodingParameters
+
+@synthesize isActive = _isActive;
+@synthesize maxBitrateBps = _maxBitrateBps;
+@synthesize ssrc = _ssrc;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpEncodingParameters &)nativeParameters {
+  if (self = [self init]) {
+    _isActive = nativeParameters.active;
+    if (nativeParameters.max_bitrate_bps) {
+      _maxBitrateBps =
+          [NSNumber numberWithInt:*nativeParameters.max_bitrate_bps];
+    }
+    if (nativeParameters.ssrc) {
+      _ssrc = [NSNumber numberWithUnsignedLong:*nativeParameters.ssrc];
+    }
+  }
+  return self;
+}
+
+- (webrtc::RtpEncodingParameters)nativeParameters {
+  webrtc::RtpEncodingParameters parameters;
+  parameters.active = _isActive;
+  if (_maxBitrateBps != nil) {
+    parameters.max_bitrate_bps = rtc::Optional<int>(_maxBitrateBps.intValue);
+  }
+  if (_ssrc != nil) {
+    parameters.ssrc = rtc::Optional<uint32_t>(_ssrc.unsignedLongValue);
+  }
+  return parameters;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCRtpFragmentationHeader.mm

@@ -0,0 +1,62 @@
+/*
+ *  Copyright 2017 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/RTCVideoCodec.h"
+
+#include "webrtc/modules/include/module_common_types.h"
+
+@implementation RTCRtpFragmentationHeader
+
+@synthesize fragmentationOffset = _fragmentationOffset;
+@synthesize fragmentationLength = _fragmentationLength;
+@synthesize fragmentationTimeDiff = _fragmentationTimeDiff;
+@synthesize fragmentationPlType = _fragmentationPlType;
+
+- (instancetype)initWithNativeFragmentationHeader:
+        (const webrtc::RTPFragmentationHeader *)fragmentationHeader {
+  if (self = [super init]) {
+    if (fragmentationHeader) {
+      int count = fragmentationHeader->fragmentationVectorSize;
+      NSMutableArray *offsets = [NSMutableArray array];
+      NSMutableArray *lengths = [NSMutableArray array];
+      NSMutableArray *timeDiffs = [NSMutableArray array];
+      NSMutableArray *plTypes = [NSMutableArray array];
+      for (int i = 0; i < count; ++i) {
+        [offsets addObject:@(fragmentationHeader->fragmentationOffset[i])];
+        [lengths addObject:@(fragmentationHeader->fragmentationLength[i])];
+        [timeDiffs addObject:@(fragmentationHeader->fragmentationTimeDiff[i])];
+        [plTypes addObject:@(fragmentationHeader->fragmentationPlType[i])];
+      }
+      _fragmentationOffset = [offsets copy];
+      _fragmentationLength = [lengths copy];
+      _fragmentationTimeDiff = [timeDiffs copy];
+      _fragmentationPlType = [plTypes copy];
+    }
+  }
+
+  return self;
+}
+
+- (std::unique_ptr<webrtc::RTPFragmentationHeader>)createNativeFragmentationHeader {
+  auto fragmentationHeader =
+      std::unique_ptr<webrtc::RTPFragmentationHeader>(new webrtc::RTPFragmentationHeader);
+  fragmentationHeader->VerifyAndAllocateFragmentationHeader(_fragmentationOffset.count);
+  for (NSUInteger i = 0; i < _fragmentationOffset.count; ++i) {
+    fragmentationHeader->fragmentationOffset[i] = (size_t)_fragmentationOffset[i].unsignedIntValue;
+    fragmentationHeader->fragmentationLength[i] = (size_t)_fragmentationLength[i].unsignedIntValue;
+    fragmentationHeader->fragmentationTimeDiff[i] =
+        (uint16_t)_fragmentationOffset[i].unsignedIntValue;
+    fragmentationHeader->fragmentationPlType[i] = (uint8_t)_fragmentationOffset[i].unsignedIntValue;
+  }
+
+  return fragmentationHeader;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h

@@ -0,0 +1,28 @@
+/*
+ *  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/RTCRtpParameters.h"
+
+#include "webrtc/api/rtpparameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpParameters ()
+
+/** Returns the equivalent native RtpParameters structure. */
+@property(nonatomic, readonly) webrtc::RtpParameters nativeParameters;
+
+/** Initialize the object with a native RtpParameters structure. */
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpParameters &)nativeParameters;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm

@@ -0,0 +1,56 @@
+/*
+ *  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 "RTCRtpParameters+Private.h"
+
+#import "RTCRtpCodecParameters+Private.h"
+#import "RTCRtpEncodingParameters+Private.h"
+
+@implementation RTCRtpParameters
+
+@synthesize encodings = _encodings;
+@synthesize codecs = _codecs;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpParameters &)nativeParameters {
+  if (self = [self init]) {
+    NSMutableArray *encodings = [[NSMutableArray alloc] init];
+    for (const auto &encoding : nativeParameters.encodings) {
+      [encodings addObject:[[RTCRtpEncodingParameters alloc]
+                               initWithNativeParameters:encoding]];
+    }
+    _encodings = encodings;
+
+    NSMutableArray *codecs = [[NSMutableArray alloc] init];
+    for (const auto &codec : nativeParameters.codecs) {
+      [codecs addObject:[[RTCRtpCodecParameters alloc]
+                            initWithNativeParameters:codec]];
+    }
+    _codecs = codecs;
+  }
+  return self;
+}
+
+- (webrtc::RtpParameters)nativeParameters {
+    webrtc::RtpParameters parameters;
+  for (RTCRtpEncodingParameters *encoding in _encodings) {
+    parameters.encodings.push_back(encoding.nativeParameters);
+  }
+  for (RTCRtpCodecParameters *codec in _codecs) {
+    parameters.codecs.push_back(codec.nativeParameters);
+  }
+  return parameters;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h

@@ -0,0 +1,45 @@
+/*
+ *  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/RTCRtpReceiver.h"
+
+#include "webrtc/api/rtpreceiverinterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+namespace webrtc {
+
+class RtpReceiverDelegateAdapter : public RtpReceiverObserverInterface {
+ public:
+  RtpReceiverDelegateAdapter(RTCRtpReceiver* receiver);
+
+  void OnFirstPacketReceived(cricket::MediaType media_type) override;
+
+ private:
+  __weak RTCRtpReceiver* receiver_;
+};
+
+}  // namespace webrtc
+
+@interface RTCRtpReceiver ()
+
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
+
+/** Initialize an RTCRtpReceiver with a native RtpReceiverInterface. */
+- (instancetype)initWithNativeRtpReceiver:
+    (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
+    NS_DESIGNATED_INITIALIZER;
+
++ (RTCRtpMediaType)mediaTypeForNativeMediaType:(cricket::MediaType)nativeMediaType;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver.mm

@@ -0,0 +1,123 @@
+/*
+ *  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 "RTCRtpReceiver+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCRtpParameters+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+namespace webrtc {
+
+RtpReceiverDelegateAdapter::RtpReceiverDelegateAdapter(
+    RTCRtpReceiver *receiver) {
+  RTC_CHECK(receiver);
+  receiver_ = receiver;
+}
+
+void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
+    cricket::MediaType media_type) {
+  RTCRtpMediaType packet_media_type =
+      [RTCRtpReceiver mediaTypeForNativeMediaType:media_type];
+  RTCRtpReceiver *receiver = receiver_;
+  [receiver.delegate rtpReceiver:receiver didReceiveFirstPacketForMediaType:packet_media_type];
+}
+
+}  // namespace webrtc
+
+@implementation RTCRtpReceiver {
+  rtc::scoped_refptr<webrtc::RtpReceiverInterface> _nativeRtpReceiver;
+  std::unique_ptr<webrtc::RtpReceiverDelegateAdapter> _observer;
+}
+
+@synthesize delegate = _delegate;
+
+- (NSString *)receiverId {
+  return [NSString stringForStdString:_nativeRtpReceiver->id()];
+}
+
+- (RTCRtpParameters *)parameters {
+  return [[RTCRtpParameters alloc]
+      initWithNativeParameters:_nativeRtpReceiver->GetParameters()];
+}
+
+- (void)setParameters:(RTCRtpParameters *)parameters {
+  if (!_nativeRtpReceiver->SetParameters(parameters.nativeParameters)) {
+    RTCLogError(@"RTCRtpReceiver(%p): Failed to set parameters: %@", self,
+        parameters);
+  }
+}
+
+- (RTCMediaStreamTrack *)track {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
+    _nativeRtpReceiver->track());
+  if (nativeTrack) {
+    return [[RTCMediaStreamTrack alloc] initWithNativeTrack:nativeTrack];
+  }
+  return nil;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCRtpReceiver {\n  receiverId: %@\n}",
+      self.receiverId];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (object == nil) {
+    return NO;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  RTCRtpReceiver *receiver = (RTCRtpReceiver *)object;
+  return _nativeRtpReceiver == receiver.nativeRtpReceiver;
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeRtpReceiver.get();
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
+  return _nativeRtpReceiver;
+}
+
+- (instancetype)initWithNativeRtpReceiver:
+    (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
+  if (self = [super init]) {
+    _nativeRtpReceiver = nativeRtpReceiver;
+    RTCLogInfo(
+        @"RTCRtpReceiver(%p): created receiver: %@", self, self.description);
+    _observer.reset(new webrtc::RtpReceiverDelegateAdapter(self));
+    _nativeRtpReceiver->SetObserver(_observer.get());
+  }
+  return self;
+}
+
++ (RTCRtpMediaType)mediaTypeForNativeMediaType:
+    (cricket::MediaType)nativeMediaType {
+  switch (nativeMediaType) {
+    case cricket::MEDIA_TYPE_AUDIO:
+      return RTCRtpMediaTypeAudio;
+    case cricket::MEDIA_TYPE_VIDEO:
+      return RTCRtpMediaTypeVideo;
+    case cricket::MEDIA_TYPE_DATA:
+      return RTCRtpMediaTypeData;
+  }
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h

@@ -0,0 +1,29 @@
+/*
+ *  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/RTCRtpSender.h"
+
+#include "webrtc/api/rtpsenderinterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpSender ()
+
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeRtpSender;
+
+/** Initialize an RTCRtpSender with a native RtpSenderInterface. */
+- (instancetype)initWithNativeRtpSender:
+    (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender.mm

@@ -0,0 +1,94 @@
+/*
+ *  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 "RTCRtpSender+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCRtpParameters+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+@implementation RTCRtpSender {
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> _nativeRtpSender;
+}
+
+- (NSString *)senderId {
+  return [NSString stringForStdString:_nativeRtpSender->id()];
+}
+
+- (RTCRtpParameters *)parameters {
+  return [[RTCRtpParameters alloc]
+      initWithNativeParameters:_nativeRtpSender->GetParameters()];
+}
+
+- (void)setParameters:(RTCRtpParameters *)parameters {
+  if (!_nativeRtpSender->SetParameters(parameters.nativeParameters)) {
+    RTCLogError(@"RTCRtpSender(%p): Failed to set parameters: %@", self,
+        parameters);
+  }
+}
+
+- (RTCMediaStreamTrack *)track {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
+    _nativeRtpSender->track());
+  if (nativeTrack) {
+    return [[RTCMediaStreamTrack alloc] initWithNativeTrack:nativeTrack];
+  }
+  return nil;
+}
+
+- (void)setTrack:(RTCMediaStreamTrack *)track {
+  if (!_nativeRtpSender->SetTrack(track.nativeTrack)) {
+    RTCLogError(@"RTCRtpSender(%p): Failed to set track %@", self, track);
+  }
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCRtpSender {\n  senderId: %@\n}",
+      self.senderId];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (object == nil) {
+    return NO;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  RTCRtpSender *sender = (RTCRtpSender *)object;
+  return _nativeRtpSender == sender.nativeRtpSender;
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeRtpSender.get();
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+  return _nativeRtpSender;
+}
+
+- (instancetype)initWithNativeRtpSender:
+    (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+  NSParameterAssert(nativeRtpSender);
+  if (self = [super init]) {
+    _nativeRtpSender = nativeRtpSender;
+    RTCLogInfo(@"RTCRtpSender(%p): created sender: %@", self, self.description);
+  }
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCSSLAdapter.mm

@@ -0,0 +1,26 @@
+/*
+ *  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/RTCSSLAdapter.h"
+
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/ssladapter.h"
+
+BOOL RTCInitializeSSL() {
+  BOOL initialized = rtc::InitializeSSL();
+  RTC_DCHECK(initialized);
+  return initialized;
+}
+
+BOOL RTCCleanupSSL() {
+  BOOL cleanedUp = rtc::CleanupSSL();
+  RTC_DCHECK(cleanedUp);
+  return cleanedUp;
+}

sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h

@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2015 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/RTCSessionDescription.h"
+
+#include "webrtc/api/jsep.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCSessionDescription ()
+
+/**
+ * The native SessionDescriptionInterface representation of this
+ * RTCSessionDescription object. This is needed to pass to the underlying C++
+ * APIs.
+ */
+@property(nonatomic, readonly)
+    webrtc::SessionDescriptionInterface *nativeDescription;
+
+/**
+ * Initialize an RTCSessionDescription from a native
+ * SessionDescriptionInterface. No ownership is taken of the native session
+ * description.
+ */
+- (instancetype)initWithNativeDescription:
+    (const webrtc::SessionDescriptionInterface *)nativeDescription;
+
++ (std::string)stdStringForType:(RTCSdpType)type;
+
++ (RTCSdpType)typeForStdString:(const std::string &)string;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription.mm

@@ -0,0 +1,102 @@
+/*
+ *  Copyright 2015 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 "RTCSessionDescription+Private.h"
+
+#import "NSString+StdString.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/rtc_base/checks.h"
+
+@implementation RTCSessionDescription
+
+@synthesize type = _type;
+@synthesize sdp = _sdp;
+
++ (NSString *)stringForType:(RTCSdpType)type {
+  std::string string = [[self class] stdStringForType:type];
+  return [NSString stringForStdString:string];
+}
+
++ (RTCSdpType)typeForString:(NSString *)string {
+  std::string typeString = string.stdString;
+  return [[self class] typeForStdString:typeString];
+}
+
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp {
+  NSParameterAssert(sdp.length);
+  if (self = [super init]) {
+    _type = type;
+    _sdp = [sdp copy];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCSessionDescription:\n%@\n%@",
+                                    [[self class] stringForType:_type],
+                                    _sdp];
+}
+
+#pragma mark - Private
+
+- (webrtc::SessionDescriptionInterface *)nativeDescription {
+  webrtc::SdpParseError error;
+
+  webrtc::SessionDescriptionInterface *description =
+      webrtc::CreateSessionDescription([[self class] stdStringForType:_type],
+                                       _sdp.stdString,
+                                       &error);
+
+  if (!description) {
+    RTCLogError(@"Failed to create session description: %s\nline: %s",
+                error.description.c_str(),
+                error.line.c_str());
+  }
+
+  return description;
+}
+
+- (instancetype)initWithNativeDescription:
+    (const webrtc::SessionDescriptionInterface *)nativeDescription {
+  NSParameterAssert(nativeDescription);
+  std::string sdp;
+  nativeDescription->ToString(&sdp);
+  RTCSdpType type = [[self class] typeForStdString:nativeDescription->type()];
+
+  return [self initWithType:type
+                        sdp:[NSString stringForStdString:sdp]];
+}
+
++ (std::string)stdStringForType:(RTCSdpType)type {
+  switch (type) {
+    case RTCSdpTypeOffer:
+      return webrtc::SessionDescriptionInterface::kOffer;
+    case RTCSdpTypePrAnswer:
+      return webrtc::SessionDescriptionInterface::kPrAnswer;
+    case RTCSdpTypeAnswer:
+      return webrtc::SessionDescriptionInterface::kAnswer;
+  }
+}
+
++ (RTCSdpType)typeForStdString:(const std::string &)string {
+  if (string == webrtc::SessionDescriptionInterface::kOffer) {
+    return RTCSdpTypeOffer;
+  } else if (string == webrtc::SessionDescriptionInterface::kPrAnswer) {
+    return RTCSdpTypePrAnswer;
+  } else if (string == webrtc::SessionDescriptionInterface::kAnswer) {
+    return RTCSdpTypeAnswer;
+  } else {
+    RTC_NOTREACHED();
+    return RTCSdpTypeOffer;
+  }
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCTracing.mm

@@ -0,0 +1,29 @@
+/*
+ *  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/RTCTracing.h"
+
+#include "webrtc/rtc_base/event_tracer.h"
+
+void RTCSetupInternalTracer() {
+  rtc::tracing::SetupInternalTracer();
+}
+
+BOOL RTCStartInternalCapture(NSString *filePath) {
+  return rtc::tracing::StartInternalCapture(filePath.UTF8String);
+}
+
+void RTCStopInternalCapture() {
+  rtc::tracing::StopInternalCapture();
+}
+
+void RTCShutdownInternalTracer() {
+  rtc::tracing::ShutdownInternalTracer();
+}

sdk/objc/Framework/Classes/PeerConnection/RTCVideoCapturer.m

@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 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/RTCVideoCapturer.h"
+
+@implementation RTCVideoCapturer
+
+@synthesize delegate = _delegate;
+
+- (instancetype)initWithDelegate:(id<RTCVideoCapturerDelegate>)delegate {
+  NSAssert(delegate != nil, @"delegate cannot be nil");
+  if (self = [super init]) {
+    _delegate = delegate;
+  }
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h

@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2017 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/RTCVideoCodec.h"
+
+#import "WebRTC/RTCVideoCodecH264.h"
+
+#include "webrtc/common_video/include/video_frame.h"
+#include "webrtc/media/base/codec.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/* Interfaces for converting to/from internal C++ formats. */
+@interface RTCEncodedImage ()
+
+- (instancetype)initWithNativeEncodedImage:(webrtc::EncodedImage)encodedImage;
+- (webrtc::EncodedImage)nativeEncodedImage;
+
+@end
+
+@interface RTCVideoEncoderSettings ()
+
+- (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *__nullable)videoCodec;
+
+@end
+
+@interface RTCCodecSpecificInfoH264 ()
+
+- (webrtc::CodecSpecificInfo)nativeCodecSpecificInfo;
+
+@end
+
+@interface RTCRtpFragmentationHeader ()
+
+- (instancetype)initWithNativeFragmentationHeader:
+        (const webrtc::RTPFragmentationHeader *__nullable)fragmentationHeader;
+- (std::unique_ptr<webrtc::RTPFragmentationHeader>)createNativeFragmentationHeader;
+
+@end
+
+@interface RTCVideoCodecInfo ()
+
+- (instancetype)initWithNativeVideoCodec:(cricket::VideoCodec)videoCodec;
+- (instancetype)initWithPayload:(NSInteger)payload
+                           name:(NSString *)name
+                     parameters:(NSDictionary<NSString *, NSString *> *)parameters;
+- (cricket::VideoCodec)nativeVideoCodec;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm

@@ -0,0 +1,80 @@
+/*
+ *  Copyright 2017 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/RTCVideoCodec.h"
+
+#import "NSString+StdString.h"
+#import "RTCVideoCodec+Private.h"
+#import "WebRTC/RTCVideoCodecFactory.h"
+
+@implementation RTCVideoCodecInfo
+
+@synthesize payload = _payload;
+@synthesize name = _name;
+@synthesize parameters = _parameters;
+
+- (instancetype)initWithName:(NSString *)name
+                  parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters {
+  if (self = [super init]) {
+    _payload = 0;
+    _name = name;
+    _parameters = (parameters ? parameters : @{});
+  }
+
+  return self;
+}
+
+- (instancetype)initWithNativeVideoCodec:(cricket::VideoCodec)videoCodec {
+  NSMutableDictionary *params = [NSMutableDictionary dictionary];
+  for (auto it = videoCodec.params.begin(); it != videoCodec.params.end(); ++it) {
+    [params setObject:[NSString stringForStdString:it->second]
+               forKey:[NSString stringForStdString:it->first]];
+  }
+  return [self initWithPayload:videoCodec.id
+                          name:[NSString stringForStdString:videoCodec.name]
+                    parameters:params];
+}
+
+- (instancetype)initWithPayload:(NSInteger)payload
+                           name:(NSString *)name
+                     parameters:(NSDictionary<NSString *, NSString *> *)parameters {
+  if (self = [self initWithName:name parameters:parameters]) {
+    _payload = payload;
+  }
+
+  return self;
+}
+
+- (cricket::VideoCodec)nativeVideoCodec {
+  cricket::VideoCodec codec([NSString stdStringForString:_name]);
+  for (NSString *paramKey in _parameters.allKeys) {
+    codec.SetParam([NSString stdStringForString:paramKey],
+                   [NSString stdStringForString:_parameters[paramKey]]);
+  }
+
+  return codec;
+}
+
+@end
+
+@implementation RTCVideoEncoderQpThresholds
+
+@synthesize low = _low;
+@synthesize high = _high;
+
+- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high {
+  if (self = [super init]) {
+    _low = low;
+    _high = high;
+  }
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm

@@ -0,0 +1,95 @@
+/*
+ *  Copyright 2017 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/RTCVideoCodecH264.h"
+
+#include <vector>
+
+#import "RTCVideoCodec+Private.h"
+#import "WebRTC/RTCVideoCodec.h"
+
+#include "webrtc/rtc_base/timeutils.h"
+#include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
+
+const char kHighProfileExperiment[] = "WebRTC-H264HighProfile";
+static NSString *kLevel31ConstrainedHigh = @"640c1f";
+static NSString *kLevel31ConstrainedBaseline = @"42e01f";
+
+bool IsHighProfileEnabled() {
+  return webrtc::field_trial::IsEnabled(kHighProfileExperiment);
+}
+
+// H264 specific settings.
+@implementation RTCCodecSpecificInfoH264
+
+@synthesize packetizationMode = _packetizationMode;
+
+- (webrtc::CodecSpecificInfo)nativeCodecSpecificInfo {
+  webrtc::CodecSpecificInfo codecSpecificInfo;
+  codecSpecificInfo.codecType = webrtc::kVideoCodecH264;
+  codecSpecificInfo.codec_name = "H264";
+  codecSpecificInfo.codecSpecific.H264.packetization_mode =
+      (webrtc::H264PacketizationMode)_packetizationMode;
+
+  return codecSpecificInfo;
+}
+
+@end
+
+// Encoder factory.
+@implementation RTCVideoEncoderFactoryH264
+
+- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
+  NSMutableArray<RTCVideoCodecInfo *> *codecs = [NSMutableArray array];
+  NSString *codecName = [NSString stringWithUTF8String:cricket::kH264CodecName];
+
+  if (IsHighProfileEnabled()) {
+    NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
+      @"profile-level-id" : kLevel31ConstrainedHigh,
+      @"level-asymmetry-allowed" : @"1",
+      @"packetization-mode" : @"1",
+    };
+    RTCVideoCodecInfo *constrainedHighInfo =
+        [[RTCVideoCodecInfo alloc] initWithName:codecName parameters:constrainedHighParams];
+    [codecs addObject:constrainedHighInfo];
+  }
+
+  NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
+    @"profile-level-id" : kLevel31ConstrainedBaseline,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTCVideoCodecInfo *constrainedBaselineInfo =
+      [[RTCVideoCodecInfo alloc] initWithName:codecName parameters:constrainedBaselineParams];
+  [codecs addObject:constrainedBaselineInfo];
+
+  return [codecs copy];
+}
+
+- (id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info {
+  return [[RTCVideoEncoderH264 alloc] initWithCodecInfo:info];
+}
+
+@end
+
+// Decoder factory.
+@implementation RTCVideoDecoderFactoryH264
+
+- (id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info {
+  return [[RTCVideoDecoderH264 alloc] init];
+}
+
+- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
+  NSString *codecName = [NSString stringWithUTF8String:cricket::kH264CodecName];
+  return @[ [[RTCVideoCodecInfo alloc] initWithName:codecName parameters:nil] ];
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCVideoEncoderSettings.mm

@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2017 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/RTCVideoCodec.h"
+
+#import "NSString+StdString.h"
+#import "RTCVideoCodec+Private.h"
+#import "WebRTC/RTCVideoCodecFactory.h"
+
+@implementation RTCVideoEncoderSettings
+
+@synthesize name = _name;
+@synthesize width = _width;
+@synthesize height = _height;
+@synthesize startBitrate = _startBitrate;
+@synthesize maxBitrate = _maxBitrate;
+@synthesize minBitrate = _minBitrate;
+@synthesize targetBitrate = _targetBitrate;
+@synthesize maxFramerate = _maxFramerate;
+@synthesize qpMax = _qpMax;
+@synthesize mode = _mode;
+
+- (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec {
+  if (self = [super init]) {
+    if (videoCodec) {
+      const char *codecName = CodecTypeToPayloadString(videoCodec->codecType);
+      _name = [NSString stringWithUTF8String:codecName];
+
+      _width = videoCodec->width;
+      _height = videoCodec->height;
+      _startBitrate = videoCodec->startBitrate;
+      _maxBitrate = videoCodec->maxBitrate;
+      _minBitrate = videoCodec->minBitrate;
+      _targetBitrate = videoCodec->targetBitrate;
+      _maxFramerate = videoCodec->maxFramerate;
+      _qpMax = videoCodec->qpMax;
+      _mode = (RTCVideoCodecMode)videoCodec->mode;
+    }
+  }
+
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCVideoFrame.mm

@@ -0,0 +1,83 @@
+/*
+ *  Copyright 2015 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/RTCVideoFrame.h"
+#import "WebRTC/RTCVideoFrameBuffer.h"
+
+@implementation RTCVideoFrame {
+  RTCVideoRotation _rotation;
+  int64_t _timeStampNs;
+}
+
+@synthesize buffer = _buffer;
+@synthesize timeStamp;
+
+- (int)width {
+  return _buffer.width;
+}
+
+- (int)height {
+  return _buffer.height;
+}
+
+- (RTCVideoRotation)rotation {
+  return _rotation;
+}
+
+- (int64_t)timeStampNs {
+  return _timeStampNs;
+}
+
+- (RTCVideoFrame *)newI420VideoFrame {
+  return [[RTCVideoFrame alloc] initWithBuffer:[_buffer toI420]
+                                      rotation:_rotation
+                                   timeStampNs:_timeStampNs];
+}
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs {
+  return [self initWithBuffer:[[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer]
+                     rotation:rotation
+                  timeStampNs:timeStampNs];
+}
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                        scaledWidth:(int)scaledWidth
+                       scaledHeight:(int)scaledHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs {
+  RTCCVPixelBuffer *rtcPixelBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer
+                                                                      adaptedWidth:scaledWidth
+                                                                     adaptedHeight:scaledHeight
+                                                                         cropWidth:cropWidth
+                                                                        cropHeight:cropHeight
+                                                                             cropX:cropX
+                                                                             cropY:cropY];
+  return [self initWithBuffer:rtcPixelBuffer rotation:rotation timeStampNs:timeStampNs];
+}
+
+- (instancetype)initWithBuffer:(id<RTCVideoFrameBuffer>)buffer
+                      rotation:(RTCVideoRotation)rotation
+                   timeStampNs:(int64_t)timeStampNs {
+  if (self = [super init]) {
+    _buffer = buffer;
+    _rotation = rotation;
+    _timeStampNs = timeStampNs;
+  }
+
+  return self;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2015 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 "RTCVideoRendererAdapter.h"
+
+#import "WebRTC/RTCVideoRenderer.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCVideoRendererAdapter ()
+
+/**
+ * The Objective-C video renderer passed to this adapter during construction.
+ * Calls made to the webrtc::VideoRenderInterface will be adapted and passed to
+ * this video renderer.
+ */
+@property(nonatomic, readonly) id<RTCVideoRenderer> videoRenderer;
+
+/**
+ * The native VideoSinkInterface surface exposed by this adapter. Calls made
+ * to this interface will be adapted and passed to the RTCVideoRenderer supplied
+ * during construction. This pointer is unsafe and owned by this class.
+ */
+@property(nonatomic, readonly)
+    rtc::VideoSinkInterface<webrtc::VideoFrame> *nativeVideoRenderer;
+
+/** Initialize an RTCVideoRendererAdapter with an RTCVideoRenderer. */
+- (instancetype)initWithNativeRenderer:(id<RTCVideoRenderer>)videoRenderer
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h

@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2015 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>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/*
+ * Creates a rtc::VideoSinkInterface surface for an RTCVideoRenderer. The
+ * rtc::VideoSinkInterface is used by WebRTC rendering code - this
+ * adapter adapts calls made to that interface to the RTCVideoRenderer supplied
+ * during construction.
+ */
+@interface RTCVideoRendererAdapter : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.mm

@@ -0,0 +1,80 @@
+/*
+ *  Copyright 2015 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 "RTCI420Buffer+Private.h"
+#import "RTCVideoRendererAdapter+Private.h"
+#import "WebRTC/RTCVideoFrame.h"
+#import "WebRTC/RTCVideoFrameBuffer.h"
+#import "objc_frame_buffer.h"
+
+#include <memory>
+
+namespace webrtc {
+
+class VideoRendererAdapter
+    : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  VideoRendererAdapter(RTCVideoRendererAdapter* adapter) {
+    adapter_ = adapter;
+    size_ = CGSizeZero;
+  }
+
+  void OnFrame(const webrtc::VideoFrame& nativeVideoFrame) override {
+    rtc::scoped_refptr<VideoFrameBuffer> video_frame_buffer = nativeVideoFrame.video_frame_buffer();
+    id<RTCVideoFrameBuffer> rtc_frame_buffer;
+    if (video_frame_buffer->type() == VideoFrameBuffer::Type::kNative) {
+      rtc::scoped_refptr<ObjCFrameBuffer> objc_frame_buffer(
+          static_cast<ObjCFrameBuffer*>(video_frame_buffer.get()));
+      rtc_frame_buffer = (id<RTCVideoFrameBuffer>)objc_frame_buffer->wrapped_frame_buffer();
+    } else {
+      rtc_frame_buffer = [[RTCI420Buffer alloc] initWithFrameBuffer:video_frame_buffer->ToI420()];
+    }
+    RTCVideoFrame* videoFrame = [[RTCVideoFrame alloc]
+        initWithBuffer:rtc_frame_buffer
+              rotation:static_cast<RTCVideoRotation>(nativeVideoFrame.rotation())
+           timeStampNs:nativeVideoFrame.timestamp_us() * rtc::kNumNanosecsPerMicrosec];
+
+    CGSize current_size = (videoFrame.rotation % 180 == 0)
+                              ? CGSizeMake(videoFrame.width, videoFrame.height)
+                              : CGSizeMake(videoFrame.height, videoFrame.width);
+
+    if (!CGSizeEqualToSize(size_, current_size)) {
+      size_ = current_size;
+      [adapter_.videoRenderer setSize:size_];
+    }
+    [adapter_.videoRenderer renderFrame:videoFrame];
+  }
+
+ private:
+  __weak RTCVideoRendererAdapter *adapter_;
+  CGSize size_;
+};
+}
+
+@implementation RTCVideoRendererAdapter {
+  std::unique_ptr<webrtc::VideoRendererAdapter> _adapter;
+}
+
+@synthesize videoRenderer = _videoRenderer;
+
+- (instancetype)initWithNativeRenderer:(id<RTCVideoRenderer>)videoRenderer {
+  NSParameterAssert(videoRenderer);
+  if (self = [super init]) {
+    _videoRenderer = videoRenderer;
+    _adapter.reset(new webrtc::VideoRendererAdapter(self));
+  }
+  return self;
+}
+
+- (rtc::VideoSinkInterface<webrtc::VideoFrame> *)nativeVideoRenderer {
+  return _adapter.get();
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2015 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/RTCVideoSource.h"
+
+#import "RTCMediaSource+Private.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCVideoSource ()
+
+/**
+ * The VideoTrackSourceInterface object passed to this RTCVideoSource during
+ * construction.
+ */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>
+        nativeVideoSource;
+
+/** Initialize an RTCVideoSource from a native VideoTrackSourceInterface. */
+- (instancetype)initWithNativeVideoSource:
+    (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithNativeMediaSource:
+    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                                     type:(RTCMediaSourceType)type NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource.mm

@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2015 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 "RTCVideoSource+Private.h"
+
+#include "webrtc/api/videosourceproxy.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/sdk/objc/Framework/Classes/Video/objcvideotracksource.h"
+
+static webrtc::ObjcVideoTrackSource *getObjcVideoSource(
+    const rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> nativeSource) {
+  webrtc::VideoTrackSourceProxy *proxy_source =
+      static_cast<webrtc::VideoTrackSourceProxy *>(nativeSource.get());
+  return static_cast<webrtc::ObjcVideoTrackSource *>(proxy_source->internal());
+}
+
+// TODO(magjed): Refactor this class and target ObjcVideoTrackSource only once
+// RTCAVFoundationVideoSource is gone. See http://crbug/webrtc/7177 for more
+// info.
+@implementation RTCVideoSource {
+  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> _nativeVideoSource;
+}
+
+- (instancetype)initWithNativeVideoSource:
+    (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
+  RTC_DCHECK(nativeVideoSource);
+  if (self = [super initWithNativeMediaSource:nativeVideoSource
+                                         type:RTCMediaSourceTypeVideo]) {
+    _nativeVideoSource = nativeVideoSource;
+  }
+  return self;
+}
+
+- (instancetype)initWithNativeMediaSource:
+    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                                     type:(RTCMediaSourceType)type {
+  RTC_NOTREACHED();
+  return nil;
+}
+
+- (NSString *)description {
+  NSString *stateString = [[self class] stringForState:self.state];
+  return [NSString stringWithFormat:@"RTCVideoSource( %p ): %@", self, stateString];
+}
+
+- (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame {
+  getObjcVideoSource(_nativeVideoSource)->OnCapturedFrame(frame);
+}
+
+- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps {
+  getObjcVideoSource(_nativeVideoSource)->OnOutputFormatRequest(width, height, fps);
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
+  return _nativeVideoSource;
+}
+
+@end

sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2015 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/RTCVideoTrack.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCVideoTrack ()
+
+/** VideoTrackInterface created or passed in at construction. */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
+
+/** Initialize an RTCVideoTrack with its source and an id. */
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                         source:(RTCVideoSource *)source
+                        trackId:(NSString *)trackId;
+
+@end
+
+NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack.mm

@@ -0,0 +1,110 @@
+/*
+ *  Copyright 2015 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 "RTCVideoTrack+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCVideoRendererAdapter+Private.h"
+#import "RTCVideoSource+Private.h"
+
+@implementation RTCVideoTrack {
+  NSMutableArray *_adapters;
+}
+
+@synthesize source = _source;
+
+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
+                         source:(RTCVideoSource *)source
+                        trackId:(NSString *)trackId {
+  NSParameterAssert(factory);
+  NSParameterAssert(source);
+  NSParameterAssert(trackId.length);
+  std::string nativeId = [NSString stdStringForString:trackId];
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
+      factory.nativeFactory->CreateVideoTrack(nativeId,
+                                              source.nativeVideoSource);
+  if ([self initWithNativeTrack:track type:RTCMediaStreamTrackTypeVideo]) {
+    _source = source;
+  }
+  return self;
+}
+
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeMediaTrack
+                               type:(RTCMediaStreamTrackType)type {
+  NSParameterAssert(nativeMediaTrack);
+  NSParameterAssert(type == RTCMediaStreamTrackTypeVideo);
+  if (self = [super initWithNativeTrack:nativeMediaTrack type:type]) {
+    _adapters = [NSMutableArray array];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  for (RTCVideoRendererAdapter *adapter in _adapters) {
+    self.nativeVideoTrack->RemoveSink(adapter.nativeVideoRenderer);
+  }
+}
+
+- (RTCVideoSource *)source {
+  if (!_source) {
+    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
+        self.nativeVideoTrack->GetSource();
+    if (source) {
+      _source = [[RTCVideoSource alloc] initWithNativeVideoSource:source.get()];
+    }
+  }
+  return _source;
+}
+
+- (void)addRenderer:(id<RTCVideoRenderer>)renderer {
+  // Make sure we don't have this renderer yet.
+  for (RTCVideoRendererAdapter *adapter in _adapters) {
+    if (adapter.videoRenderer == renderer) {
+      NSAssert(NO, @"|renderer| is already attached to this track");
+      return;
+    }
+  }
+  // Create a wrapper that provides a native pointer for us.
+  RTCVideoRendererAdapter* adapter =
+      [[RTCVideoRendererAdapter alloc] initWithNativeRenderer:renderer];
+  [_adapters addObject:adapter];
+  self.nativeVideoTrack->AddOrUpdateSink(adapter.nativeVideoRenderer,
+                                         rtc::VideoSinkWants());
+}
+
+- (void)removeRenderer:(id<RTCVideoRenderer>)renderer {
+  __block NSUInteger indexToRemove = NSNotFound;
+  [_adapters enumerateObjectsUsingBlock:^(RTCVideoRendererAdapter *adapter,
+                                          NSUInteger idx,
+                                          BOOL *stop) {
+    if (adapter.videoRenderer == renderer) {
+      indexToRemove = idx;
+      *stop = YES;
+    }
+  }];
+  if (indexToRemove == NSNotFound) {
+    return;
+  }
+  RTCVideoRendererAdapter *adapterToRemove =
+      [_adapters objectAtIndex:indexToRemove];
+  self.nativeVideoTrack->RemoveSink(adapterToRemove.nativeVideoRenderer);
+  [_adapters removeObjectAtIndex:indexToRemove];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::VideoTrackInterface>)nativeVideoTrack {
+  return static_cast<webrtc::VideoTrackInterface *>(self.nativeTrack.get());
+}
+
+@end

sdk/objc/Framework/Classes/UI/RTCCameraPreviewView.m

@@ -0,0 +1,78 @@
+/*
+ *  Copyright 2015 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/RTCCameraPreviewView.h"
+
+#import <AVFoundation/AVFoundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCDispatcher+Private.h"
+
+@implementation RTCCameraPreviewView
+
+@synthesize captureSession = _captureSession;
+
++ (Class)layerClass {
+  return [AVCaptureVideoPreviewLayer class];
+}
+
+- (void)setCaptureSession:(AVCaptureSession *)captureSession {
+  if (_captureSession == captureSession) {
+    return;
+  }
+  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeMain
+                               block:^{
+    _captureSession = captureSession;
+    AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer];
+    [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                                 block:^{
+      previewLayer.session = captureSession;
+    }];
+  }];
+}
+
+- (void)layoutSubviews {
+  [super layoutSubviews];
+
+  // Update the video orientation based on the device orientation.
+  [self setCorrectVideoOrientation];
+}
+
+- (void)setCorrectVideoOrientation {
+  // Get current device orientation.
+  UIDeviceOrientation deviceOrientation = [UIDevice currentDevice].orientation;
+  AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer];
+
+  // First check if we are allowed to set the video orientation.
+  if (previewLayer.connection.isVideoOrientationSupported) {
+    // Set the video orientation based on device orientation.
+    if (deviceOrientation == UIInterfaceOrientationPortraitUpsideDown) {
+      previewLayer.connection.videoOrientation =
+          AVCaptureVideoOrientationPortraitUpsideDown;
+    } else if (deviceOrientation == UIInterfaceOrientationLandscapeRight) {
+      previewLayer.connection.videoOrientation =
+          AVCaptureVideoOrientationLandscapeRight;
+    } else if (deviceOrientation == UIInterfaceOrientationLandscapeLeft) {
+      previewLayer.connection.videoOrientation =
+          AVCaptureVideoOrientationLandscapeLeft;
+    } else {
+      previewLayer.connection.videoOrientation =
+          AVCaptureVideoOrientationPortrait;
+    }
+  }
+}
+
+#pragma mark - Private
+
+- (AVCaptureVideoPreviewLayer *)previewLayer {
+  return (AVCaptureVideoPreviewLayer *)self.layer;
+}
+
+@end

sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m

@@ -0,0 +1,328 @@
+/*
+ *  Copyright 2015 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/RTCEAGLVideoView.h"
+
+#import <GLKit/GLKit.h>
+
+#import "RTCDefaultShader.h"
+#import "RTCI420TextureCache.h"
+#import "RTCNV12TextureCache.h"
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoFrame.h"
+#import "WebRTC/RTCVideoFrameBuffer.h"
+
+// RTCDisplayLinkTimer wraps a CADisplayLink and is set to fire every two screen
+// refreshes, which should be 30fps. We wrap the display link in order to avoid
+// a retain cycle since CADisplayLink takes a strong reference onto its target.
+// The timer is paused by default.
+@interface RTCDisplayLinkTimer : NSObject
+
+@property(nonatomic) BOOL isPaused;
+
+- (instancetype)initWithTimerHandler:(void (^)(void))timerHandler;
+- (void)invalidate;
+
+@end
+
+@implementation RTCDisplayLinkTimer {
+  CADisplayLink *_displayLink;
+  void (^_timerHandler)(void);
+}
+
+- (instancetype)initWithTimerHandler:(void (^)(void))timerHandler {
+  NSParameterAssert(timerHandler);
+  if (self = [super init]) {
+    _timerHandler = timerHandler;
+    _displayLink =
+        [CADisplayLink displayLinkWithTarget:self
+                                    selector:@selector(displayLinkDidFire:)];
+    _displayLink.paused = YES;
+    // Set to half of screen refresh, which should be 30fps.
+    [_displayLink setFrameInterval:2];
+    [_displayLink addToRunLoop:[NSRunLoop currentRunLoop]
+                       forMode:NSRunLoopCommonModes];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self invalidate];
+}
+
+- (BOOL)isPaused {
+  return _displayLink.paused;
+}
+
+- (void)setIsPaused:(BOOL)isPaused {
+  _displayLink.paused = isPaused;
+}
+
+- (void)invalidate {
+  [_displayLink invalidate];
+}
+
+- (void)displayLinkDidFire:(CADisplayLink *)displayLink {
+  _timerHandler();
+}
+
+@end
+
+// RTCEAGLVideoView wraps a GLKView which is setup with
+// enableSetNeedsDisplay = NO for the purpose of gaining control of
+// exactly when to call -[GLKView display]. This need for extra
+// control is required to avoid triggering method calls on GLKView
+// that results in attempting to bind the underlying render buffer
+// when the drawable size would be empty which would result in the
+// error GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT. -[GLKView display] is
+// the method that will trigger the binding of the render
+// buffer. Because the standard behaviour of -[UIView setNeedsDisplay]
+// is disabled for the reasons above, the RTCEAGLVideoView maintains
+// its own |isDirty| flag.
+
+@interface RTCEAGLVideoView () <GLKViewDelegate>
+// |videoFrame| is set when we receive a frame from a worker thread and is read
+// from the display link callback so atomicity is required.
+@property(atomic, strong) RTCVideoFrame *videoFrame;
+@property(nonatomic, readonly) GLKView *glkView;
+@end
+
+@implementation RTCEAGLVideoView {
+  RTCDisplayLinkTimer *_timer;
+  EAGLContext *_glContext;
+  // This flag should only be set and read on the main thread (e.g. by
+  // setNeedsDisplay)
+  BOOL _isDirty;
+  id<RTCVideoViewShading> _shader;
+  RTCNV12TextureCache *_nv12TextureCache;
+  RTCI420TextureCache *_i420TextureCache;
+  RTCVideoFrame *_lastDrawnFrame;
+}
+
+@synthesize delegate = _delegate;
+@synthesize videoFrame = _videoFrame;
+@synthesize glkView = _glkView;
+
+- (instancetype)initWithFrame:(CGRect)frame {
+  return [self initWithFrame:frame shader:[[RTCDefaultShader alloc] init]];
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder {
+  return [self initWithCoder:aDecoder shader:[[RTCDefaultShader alloc] init]];
+}
+
+- (instancetype)initWithFrame:(CGRect)frame shader:(id<RTCVideoViewShading>)shader {
+  if (self = [super initWithFrame:frame]) {
+    _shader = shader;
+    if (![self configure]) {
+      return nil;
+    }
+  }
+  return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder shader:(id<RTCVideoViewShading>)shader {
+  if (self = [super initWithCoder:aDecoder]) {
+    _shader = shader;
+    if (![self configure]) {
+      return nil;
+    }
+  }
+  return self;
+}
+
+- (BOOL)configure {
+  EAGLContext *glContext =
+    [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES3];
+  if (!glContext) {
+    glContext = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2];
+  }
+  if (!glContext) {
+    RTCLogError(@"Failed to create EAGLContext");
+    return NO;
+  }
+  _glContext = glContext;
+
+  // GLKView manages a framebuffer for us.
+  _glkView = [[GLKView alloc] initWithFrame:CGRectZero
+                                    context:_glContext];
+  _glkView.drawableColorFormat = GLKViewDrawableColorFormatRGBA8888;
+  _glkView.drawableDepthFormat = GLKViewDrawableDepthFormatNone;
+  _glkView.drawableStencilFormat = GLKViewDrawableStencilFormatNone;
+  _glkView.drawableMultisample = GLKViewDrawableMultisampleNone;
+  _glkView.delegate = self;
+  _glkView.layer.masksToBounds = YES;
+  _glkView.enableSetNeedsDisplay = NO;
+  [self addSubview:_glkView];
+
+  // Listen to application state in order to clean up OpenGL before app goes
+  // away.
+  NSNotificationCenter *notificationCenter =
+    [NSNotificationCenter defaultCenter];
+  [notificationCenter addObserver:self
+                         selector:@selector(willResignActive)
+                             name:UIApplicationWillResignActiveNotification
+                           object:nil];
+  [notificationCenter addObserver:self
+                         selector:@selector(didBecomeActive)
+                             name:UIApplicationDidBecomeActiveNotification
+                           object:nil];
+
+  // Frames are received on a separate thread, so we poll for current frame
+  // using a refresh rate proportional to screen refresh frequency. This
+  // occurs on the main thread.
+  __weak RTCEAGLVideoView *weakSelf = self;
+  _timer = [[RTCDisplayLinkTimer alloc] initWithTimerHandler:^{
+      RTCEAGLVideoView *strongSelf = weakSelf;
+      [strongSelf displayLinkTimerDidFire];
+    }];
+  [self setupGL];
+  return YES;
+}
+
+- (void)dealloc {
+  [[NSNotificationCenter defaultCenter] removeObserver:self];
+  UIApplicationState appState =
+      [UIApplication sharedApplication].applicationState;
+  if (appState == UIApplicationStateActive) {
+    [self teardownGL];
+  }
+  [_timer invalidate];
+  if (_glContext && [EAGLContext currentContext] == _glContext) {
+    [EAGLContext setCurrentContext:nil];
+  }
+}
+
+#pragma mark - UIView
+
+- (void)setNeedsDisplay {
+  [super setNeedsDisplay];
+  _isDirty = YES;
+}
+
+- (void)setNeedsDisplayInRect:(CGRect)rect {
+  [super setNeedsDisplayInRect:rect];
+  _isDirty = YES;
+}
+
+- (void)layoutSubviews {
+  [super layoutSubviews];
+  _glkView.frame = self.bounds;
+}
+
+#pragma mark - GLKViewDelegate
+
+// This method is called when the GLKView's content is dirty and needs to be
+// redrawn. This occurs on main thread.
+- (void)glkView:(GLKView *)view drawInRect:(CGRect)rect {
+  // The renderer will draw the frame to the framebuffer corresponding to the
+  // one used by |view|.
+  RTCVideoFrame *frame = self.videoFrame;
+  if (!frame || frame == _lastDrawnFrame) {
+    return;
+  }
+  [self ensureGLContext];
+  glClear(GL_COLOR_BUFFER_BIT);
+  if ([frame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
+    if (!_nv12TextureCache) {
+      _nv12TextureCache = [[RTCNV12TextureCache alloc] initWithContext:_glContext];
+    }
+    if (_nv12TextureCache) {
+      [_nv12TextureCache uploadFrameToTextures:frame];
+      [_shader applyShadingForFrameWithWidth:frame.width
+                                      height:frame.height
+                                    rotation:frame.rotation
+                                      yPlane:_nv12TextureCache.yTexture
+                                     uvPlane:_nv12TextureCache.uvTexture];
+      [_nv12TextureCache releaseTextures];
+    }
+  } else {
+    if (!_i420TextureCache) {
+      _i420TextureCache = [[RTCI420TextureCache alloc] initWithContext:_glContext];
+    }
+    [_i420TextureCache uploadFrameToTextures:frame];
+    [_shader applyShadingForFrameWithWidth:frame.width
+                                    height:frame.height
+                                  rotation:frame.rotation
+                                    yPlane:_i420TextureCache.yTexture
+                                    uPlane:_i420TextureCache.uTexture
+                                    vPlane:_i420TextureCache.vTexture];
+  }
+}
+
+#pragma mark - RTCVideoRenderer
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+  __weak RTCEAGLVideoView *weakSelf = self;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    RTCEAGLVideoView *strongSelf = weakSelf;
+    [strongSelf.delegate videoView:strongSelf didChangeVideoSize:size];
+  });
+}
+
+- (void)renderFrame:(RTCVideoFrame *)frame {
+  self.videoFrame = frame;
+}
+
+#pragma mark - Private
+
+- (void)displayLinkTimerDidFire {
+  // Don't render unless video frame have changed or the view content
+  // has explicitly been marked dirty.
+  if (!_isDirty && _lastDrawnFrame == self.videoFrame) {
+    return;
+  }
+
+  // Always reset isDirty at this point, even if -[GLKView display]
+  // won't be called in the case the drawable size is empty.
+  _isDirty = NO;
+
+  // Only call -[GLKView display] if the drawable size is
+  // non-empty. Calling display will make the GLKView setup its
+  // render buffer if necessary, but that will fail with error
+  // GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT if size is empty.
+  if (self.bounds.size.width > 0 && self.bounds.size.height > 0) {
+    [_glkView display];
+  }
+}
+
+- (void)setupGL {
+  self.videoFrame = nil;
+  [self ensureGLContext];
+  glDisable(GL_DITHER);
+  _timer.isPaused = NO;
+}
+
+- (void)teardownGL {
+  self.videoFrame = nil;
+  _timer.isPaused = YES;
+  [_glkView deleteDrawable];
+  [self ensureGLContext];
+  _nv12TextureCache = nil;
+  _i420TextureCache = nil;
+}
+
+- (void)didBecomeActive {
+  [self setupGL];
+}
+
+- (void)willResignActive {
+  [self teardownGL];
+}
+
+- (void)ensureGLContext {
+  NSAssert(_glContext, @"context shouldn't be nil");
+  if ([EAGLContext currentContext] != _glContext) {
+    [EAGLContext setCurrentContext:_glContext];
+  }
+}
+
+@end

sdk/objc/Framework/Classes/UI/RTCNSGLVideoView.m

@@ -0,0 +1,196 @@
+/*
+ *  Copyright 2015 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>
+
+#if !TARGET_OS_IPHONE
+
+#import "WebRTC/RTCNSGLVideoView.h"
+
+#import <AppKit/NSOpenGL.h>
+#import <CoreVideo/CVDisplayLink.h>
+#import <OpenGL/gl3.h>
+
+#import "RTCDefaultShader.h"
+#import "RTCI420TextureCache.h"
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCVideoFrame.h"
+
+@interface RTCNSGLVideoView ()
+// |videoFrame| is set when we receive a frame from a worker thread and is read
+// from the display link callback so atomicity is required.
+@property(atomic, strong) RTCVideoFrame *videoFrame;
+@property(atomic, strong) RTCI420TextureCache *i420TextureCache;
+
+- (void)drawFrame;
+@end
+
+static CVReturn OnDisplayLinkFired(CVDisplayLinkRef displayLink,
+                                   const CVTimeStamp *now,
+                                   const CVTimeStamp *outputTime,
+                                   CVOptionFlags flagsIn,
+                                   CVOptionFlags *flagsOut,
+                                   void *displayLinkContext) {
+  RTCNSGLVideoView *view = (__bridge RTCNSGLVideoView *)displayLinkContext;
+  [view drawFrame];
+  return kCVReturnSuccess;
+}
+
+@implementation RTCNSGLVideoView {
+  CVDisplayLinkRef _displayLink;
+  RTCVideoFrame *_lastDrawnFrame;
+  id<RTCVideoViewShading> _shader;
+}
+
+@synthesize delegate = _delegate;
+@synthesize videoFrame = _videoFrame;
+@synthesize i420TextureCache = _i420TextureCache;
+
+- (instancetype)initWithFrame:(NSRect)frame pixelFormat:(NSOpenGLPixelFormat *)format {
+  return [self initWithFrame:frame pixelFormat:format shader:[[RTCDefaultShader alloc] init]];
+}
+
+- (instancetype)initWithFrame:(NSRect)frame
+                  pixelFormat:(NSOpenGLPixelFormat *)format
+                       shader:(id<RTCVideoViewShading>)shader {
+  if (self = [super initWithFrame:frame pixelFormat:format]) {
+    _shader = shader;
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self teardownDisplayLink];
+}
+
+- (void)drawRect:(NSRect)rect {
+  [self drawFrame];
+}
+
+- (void)reshape {
+  [super reshape];
+  NSRect frame = [self frame];
+  [self ensureGLContext];
+  CGLLockContext([[self openGLContext] CGLContextObj]);
+  glViewport(0, 0, frame.size.width, frame.size.height);
+  CGLUnlockContext([[self openGLContext] CGLContextObj]);
+}
+
+- (void)lockFocus {
+  NSOpenGLContext *context = [self openGLContext];
+  [super lockFocus];
+  if ([context view] != self) {
+    [context setView:self];
+  }
+  [context makeCurrentContext];
+}
+
+- (void)prepareOpenGL {
+  [super prepareOpenGL];
+  [self ensureGLContext];
+  glDisable(GL_DITHER);
+  [self setupDisplayLink];
+}
+
+- (void)clearGLContext {
+  [self ensureGLContext];
+  self.i420TextureCache = nil;
+  [super clearGLContext];
+}
+
+#pragma mark - RTCVideoRenderer
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    [self.delegate videoView:self didChangeVideoSize:size];
+  });
+}
+
+- (void)renderFrame:(RTCVideoFrame *)frame {
+  self.videoFrame = frame;
+}
+
+#pragma mark - Private
+
+- (void)drawFrame {
+  RTCVideoFrame *frame = self.videoFrame;
+  if (!frame || frame == _lastDrawnFrame) {
+    return;
+  }
+  // This method may be called from CVDisplayLink callback which isn't on the
+  // main thread so we have to lock the GL context before drawing.
+  NSOpenGLContext *context = [self openGLContext];
+  CGLLockContext([context CGLContextObj]);
+
+  [self ensureGLContext];
+  glClear(GL_COLOR_BUFFER_BIT);
+
+  // Rendering native CVPixelBuffer is not supported on OS X.
+  // TODO(magjed): Add support for NV12 texture cache on OS X.
+  frame = [frame newI420VideoFrame];
+  if (!self.i420TextureCache) {
+    self.i420TextureCache = [[RTCI420TextureCache alloc] initWithContext:context];
+  }
+  RTCI420TextureCache *i420TextureCache = self.i420TextureCache;
+  if (i420TextureCache) {
+    [i420TextureCache uploadFrameToTextures:frame];
+    [_shader applyShadingForFrameWithWidth:frame.width
+                                    height:frame.height
+                                  rotation:frame.rotation
+                                    yPlane:i420TextureCache.yTexture
+                                    uPlane:i420TextureCache.uTexture
+                                    vPlane:i420TextureCache.vTexture];
+    [context flushBuffer];
+    _lastDrawnFrame = frame;
+  }
+  CGLUnlockContext([context CGLContextObj]);
+}
+
+- (void)setupDisplayLink {
+  if (_displayLink) {
+    return;
+  }
+  // Synchronize buffer swaps with vertical refresh rate.
+  GLint swapInt = 1;
+  [[self openGLContext] setValues:&swapInt forParameter:NSOpenGLCPSwapInterval];
+
+  // Create display link.
+  CVDisplayLinkCreateWithActiveCGDisplays(&_displayLink);
+  CVDisplayLinkSetOutputCallback(_displayLink,
+                                 &OnDisplayLinkFired,
+                                 (__bridge void *)self);
+  // Set the display link for the current renderer.
+  CGLContextObj cglContext = [[self openGLContext] CGLContextObj];
+  CGLPixelFormatObj cglPixelFormat = [[self pixelFormat] CGLPixelFormatObj];
+  CVDisplayLinkSetCurrentCGDisplayFromOpenGLContext(
+      _displayLink, cglContext, cglPixelFormat);
+  CVDisplayLinkStart(_displayLink);
+}
+
+- (void)teardownDisplayLink {
+  if (!_displayLink) {
+    return;
+  }
+  CVDisplayLinkRelease(_displayLink);
+  _displayLink = NULL;
+}
+
+- (void)ensureGLContext {
+  NSOpenGLContext* context = [self openGLContext];
+  NSAssert(context, @"context shouldn't be nil");
+  if ([NSOpenGLContext currentContext] != context) {
+    [context makeCurrentContext];
+  }
+}
+
+@end
+
+#endif  // !TARGET_OS_IPHONE

sdk/objc/Framework/Classes/Video/AVCaptureSession+DevicePosition.h

@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2017 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 <AVFoundation/AVFoundation.h>
+#import <CoreMedia/CoreMedia.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface AVCaptureSession (DevicePosition)
+
+// Check the image's EXIF for the camera the image came from.
++ (AVCaptureDevicePosition)devicePositionForSampleBuffer:(CMSampleBufferRef)sampleBuffer;
+
+@end
+
+NS_ASSUME_NONNULL_END