WebRTC M66

Build/iOS/WebRTC.framework/Headers/RTCAVFoundationVideoSource.h

@@ -1,55 +0,0 @@
-/*
- *  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/RTCMacros.h>
-#import <WebRTC/RTCVideoSource.h>
-
-@class AVCaptureSession;
-@class RTCMediaConstraints;
-@class RTCPeerConnectionFactory;
-
-NS_ASSUME_NONNULL_BEGIN
-
-/**
- * DEPRECATED Use RTCCameraVideoCapturer instead.
- *
- * RTCAVFoundationVideoSource is a video source that uses
- * webrtc::AVFoundationVideoCapturer. We do not currently provide a wrapper for
- * that capturer because cricket::VideoCapturer is not ref counted and we cannot
- * guarantee its lifetime. Instead, we expose its properties through the ref
- * counted video source interface.
- */
-RTC_EXPORT
-@interface RTCAVFoundationVideoSource : RTCVideoSource
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/**
- * Calling this function will cause frames to be scaled down to the
- * requested resolution. Also, frames will be cropped to match the
- * requested aspect ratio, and frames will be dropped to match the
- * requested fps. The requested aspect ratio is orientation agnostic and
- * will be adjusted to maintain the input orientation, so it doesn't
- * matter if e.g. 1280x720 or 720x1280 is requested.
- */
-- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps;
-
-/** Returns whether rear-facing camera is available for use. */
-@property(nonatomic, readonly) BOOL canUseBackCamera;
-
-/** Switches the camera being used (either front or back). */
-@property(nonatomic, assign) BOOL useBackCamera;
-
-/** Returns the active capture session. */
-@property(nonatomic, readonly) AVCaptureSession *captureSession;
-
-@end
-
-NS_ASSUME_NONNULL_END

Build/iOS/WebRTC.framework/Headers/RTCCallbackLogger.h

@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2018 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/RTCLogging.h>
+#import <WebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+// This class intercepts WebRTC logs and forwards them to a registered block.
+// This class is not threadsafe.
+RTC_EXPORT
+@interface RTCCallbackLogger : NSObject
+
+// The severity level to capture. The default is kRTCLoggingSeverityInfo.
+@property(nonatomic, assign) RTCLoggingSeverity severity;
+
+// The callback will be called on the same thread that does the logging, so
+// if the logging callback can be slow it may be a good idea to implement
+// dispatching to some other queue.
+- (void)start:(nullable void (^)(NSString*))callback;
+
+- (void)stop;
+
+@end
+
+NS_ASSUME_NONNULL_END

Build/iOS/WebRTC.framework/Headers/RTCCameraPreviewView.h

@@ -14,7 +14,6 @@
 #import <WebRTC/RTCMacros.h>
 
 @class AVCaptureSession;
-@class RTCAVFoundationVideoSource;
 
 /** RTCCameraPreviewView is a view that renders local video from an
  *  AVCaptureSession.

Build/iOS/WebRTC.framework/Headers/RTCCameraVideoCapturer.h

@@ -29,7 +29,21 @@ RTC_EXPORT
 // Returns list of formats that are supported by this class for this device.
 + (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
 
-// Starts and stops the capture session asynchronously.
+// Returns the most efficient supported output pixel format for this capturer.
+- (FourCharCode)preferredOutputPixelFormat;
+
+// Starts the capture session asynchronously and notifies callback on completion.
+// The device will capture video in the format given in the `format` parameter. If the pixel format
+// in `format` is supported by the WebRTC pipeline, the same pixel format will be used for the
+// output. Otherwise, the format returned by `preferredOutputPixelFormat` will be used.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps
+             completionHandler:(nullable void (^)(NSError *))completionHandler;
+// Stops the capture session asynchronously and notifies callback on completion.
+- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
+
+// Starts the capture session asynchronously.
 - (void)startCaptureWithDevice:(AVCaptureDevice *)device
                         format:(AVCaptureDeviceFormat *)format
                            fps:(NSInteger)fps;

Build/iOS/WebRTC.framework/Headers/RTCConfiguration.h

@@ -63,6 +63,13 @@ typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
   RTCEncryptionKeyTypeECDSA,
 };
 
+/** Represents the chosen SDP semantics for the RTCPeerConnection. */
+typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
+  RTCSdpSemanticsDefault,
+  RTCSdpSemanticsPlanB,
+  RTCSdpSemanticsUnifiedPlan,
+};
+
 NS_ASSUME_NONNULL_BEGIN
 
 RTC_EXPORT
@@ -93,6 +100,12 @@ RTC_EXPORT
  */
 @property(nonatomic, assign) int maxIPv6Networks;
 
+/** Exclude link-local network interfaces
+ *  from considertaion for gathering ICE candidates.
+ *  Defaults to NO.
+ */
+@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
+
 @property(nonatomic, assign) int audioJitterBufferMaxPackets;
 @property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
 @property(nonatomic, assign) int iceConnectionReceivingTimeout;
@@ -126,6 +139,33 @@ RTC_EXPORT
  */
 @property(nonatomic, strong, nullable) RTCIntervalRange *iceRegatherIntervalRange;
 
+/** Configure the SDP semantics used by this PeerConnection. Note that the
+ *  WebRTC 1.0 specification requires UnifiedPlan semantics. The
+ *  RTCRtpTransceiver API is only available with UnifiedPlan semantics.
+ *
+ *  PlanB will cause RTCPeerConnection to create offers and answers with at
+ *  most one audio and one video m= section with multiple RTCRtpSenders and
+ *  RTCRtpReceivers specified as multiple a=ssrc lines within the section. This
+ *  will also cause RTCPeerConnection to ignore all but the first m= section of
+ *  the same media type.
+ *
+ *  UnifiedPlan will cause RTCPeerConnection to create offers and answers with
+ *  multiple m= sections where each m= section maps to one RTCRtpSender and one
+ *  RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both video. This
+ *  will also cause RTCPeerConnection to ignore all but the first a=ssrc lines
+ *  that form a Plan B stream.
+ *
+ *  For users who only send at most one audio and one video track, this
+ *  choice does not matter and should be left as Default.
+ *
+ *  For users who wish to send multiple audio/video streams and need to stay
+ *  interoperable with legacy WebRTC implementations, specify PlanB.
+ *
+ *  For users who wish to send multiple audio/video streams and/or wish to
+ *  use the new RTCRtpTransceiver API, specify UnifiedPlan.
+ */
+@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
+
 - (instancetype)init;
 
 @end

Build/iOS/WebRTC.framework/Headers/RTCDtmfSender.h

@@ -0,0 +1,70 @@
+/*
+ *  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/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@protocol RTCDtmfSender <NSObject>
+
+/**
+ * Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
+ * returns false. To be able to send DTMF, the associated RTCRtpSender must be
+ * able to send packets, and a "telephone-event" codec must be negotiated.
+ */
+@property(nonatomic, readonly) BOOL canInsertDtmf;
+
+/**
+ * Queues a task that sends the DTMF tones. The tones parameter is treated
+ * as a series of characters. The characters 0 through 9, A through D, #, and *
+ * generate the associated DTMF tones. The characters a to d are equivalent
+ * to A to D. The character ',' indicates a delay of 2 seconds before
+ * processing the next character in the tones parameter.
+ *
+ * Unrecognized characters are ignored.
+ *
+ * @param duration The parameter indicates the duration to use for each
+ * character passed in the tones parameter. The duration cannot be more
+ * than 6000 or less than 70 ms.
+ *
+ * @param interToneGap The parameter indicates the gap between tones.
+ * This parameter must be at least 50 ms but should be as short as
+ * possible.
+ *
+ * If InsertDtmf is called on the same object while an existing task for this
+ * object to generate DTMF is still running, the previous task is canceled.
+ * Returns true on success and false on failure.
+ */
+- (BOOL)insertDtmf:(nonnull NSString *)tones
+          duration:(NSTimeInterval)duration
+      interToneGap:(NSTimeInterval)interToneGap;
+
+/** The tones remaining to be played out */
+- (nonnull NSString *)remainingTones;
+
+/**
+ * The current tone duration value. This value will be the value last set via the
+ * insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
+ */
+- (NSTimeInterval)duration;
+
+/**
+ * The current value of the between-tone gap. This value will be the value last set
+ * via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
+ * called.
+ */
+- (NSTimeInterval)interToneGap;
+
+@end
+
+NS_ASSUME_NONNULL_END

Build/iOS/WebRTC.framework/Headers/RTCFileLogger.h

@@ -69,7 +69,7 @@ RTC_EXPORT
 
 // Returns the current contents of the logs, or nil if start has been called
 // without a stop.
-- (NSData *)logData;
+- (nullable NSData *)logData;
 
 @end
 

Build/iOS/WebRTC.framework/Headers/RTCFileVideoCapturer.h

@@ -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 <Foundation/Foundation.h>
+#import <WebRTC/RTCVideoCapturer.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Error passing block.
+ */
+typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
+
+/**
+ * Captures buffers from bundled video file.
+ *
+ * See @c RTCVideoCapturer for more info on capturers.
+ */
+RTC_EXPORT
+
+NS_CLASS_AVAILABLE_IOS(10)
+@interface RTCFileVideoCapturer : RTCVideoCapturer
+
+/**
+ * Starts asynchronous capture of frames from video file.
+ *
+ * Capturing is not started if error occurs. Underlying error will be
+ * relayed in the errorBlock if one is provided.
+ * Successfully captured video frames will be passed to the delegate.
+ *
+ * @param nameOfFile The name of the bundled video file to be read.
+ * @errorBlock block to be executed upon error.
+ */
+- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
+                            onError:(__nullable RTCFileVideoCapturerErrorBlock)errorBlock;
+
+/**
+ * Immediately stops capture.
+ */
+- (void)stopCapture;
+@end
+
+NS_ASSUME_NONNULL_END

Build/iOS/WebRTC.framework/Headers/RTCLogging.h

@@ -20,7 +20,7 @@ typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
   RTCLoggingSeverityError,
 };
 
-// Wrapper for C++ LOG(sev) macros.
+// Wrapper for C++ RTC_LOG(sev) macros.
 // Logs the log string to the webrtc logstream for the given severity.
 RTC_EXTERN void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
 

Build/iOS/WebRTC.framework/Headers/RTCMTLVideoView.h

@@ -1,37 +0,0 @@
-/*
- *  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/RTCVideoRenderer.h"
-
-// Check if metal is supported in WebRTC.
-// NOTE: Currently arm64 == Metal.
-#if defined(__aarch64__)
-#define RTC_SUPPORTS_METAL
-#endif
-
-NS_ASSUME_NONNULL_BEGIN
-
-/**
- * RTCMTLVideoView is thin wrapper around MTKView.
- *
- * It has id<RTCVideoRenderer> property that renders video frames in the view's
- * bounds using Metal.
- * NOTE: always check if metal is available on the running device via
- * RTC_SUPPORTS_METAL macro before initializing this class.
- */
-NS_CLASS_AVAILABLE_IOS(9)
-
-RTC_EXPORT
-@interface RTCMTLVideoView : UIView <RTCVideoRenderer>
-
-@end
-NS_ASSUME_NONNULL_END

Build/iOS/WebRTC.framework/Headers/RTCMetrics.h

@@ -17,7 +17,8 @@
  * Enables gathering of metrics (which can be fetched with
  * RTCGetAndResetMetrics). Must be called before any other call into WebRTC.
  */
-RTC_EXTERN void RTCEnableMetrics();
+RTC_EXTERN void RTCEnableMetrics(void);
 
 /** Gets and clears native histograms. */
-RTC_EXTERN NSArray<RTCMetricsSampleInfo *> *RTCGetAndResetMetrics();
+RTC_EXTERN NSArray<RTCMetricsSampleInfo *> *RTCGetAndResetMetrics(void);
+

Build/iOS/WebRTC.framework/Headers/RTCPeerConnection.h

@@ -22,9 +22,13 @@
 @class RTCPeerConnectionFactory;
 @class RTCRtpReceiver;
 @class RTCRtpSender;
+@class RTCRtpTransceiver;
+@class RTCRtpTransceiverInit;
 @class RTCSessionDescription;
 @class RTCLegacyStatsReport;
 
+typedef NS_ENUM(NSInteger, RTCRtpMediaType);
+
 NS_ASSUME_NONNULL_BEGIN
 
 extern NSString * const kRTCPeerConnectionErrorDomain;
@@ -79,7 +83,9 @@ RTC_EXPORT
 - (void)peerConnection:(RTCPeerConnection *)peerConnection
           didAddStream:(RTCMediaStream *)stream;
 
-/** Called when a remote peer closes a stream. */
+/** Called when a remote peer closes a stream.
+ *  This is not called when RTCSdpSemanticsUnifiedPlan is specified.
+ */
 - (void)peerConnection:(RTCPeerConnection *)peerConnection
        didRemoveStream:(RTCMediaStream *)stream;
 
@@ -106,6 +112,14 @@ RTC_EXPORT
 - (void)peerConnection:(RTCPeerConnection *)peerConnection
     didOpenDataChannel:(RTCDataChannel *)dataChannel;
 
+/** Called when signaling indicates a transceiver will be receiving media from
+ *  the remote endpoint.
+ *  This is only called with RTCSdpSemanticsUnifiedPlan specified.
+ */
+@optional
+- (void)peerConnection:(RTCPeerConnection *)peerConnection
+    didStartReceivingOnTransceiver:(RTCRtpTransceiver *)transceiver;
+
 @end
 
 RTC_EXPORT
@@ -115,6 +129,9 @@ RTC_EXPORT
  *  streams being added or removed.
  */
 @property(nonatomic, weak, nullable) id<RTCPeerConnectionDelegate> delegate;
+/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
+ *  |senders| instead.
+ */
 @property(nonatomic, readonly) NSArray<RTCMediaStream *> *localStreams;
 @property(nonatomic, readonly, nullable)
     RTCSessionDescription *localDescription;
@@ -137,6 +154,14 @@ RTC_EXPORT
  */
 @property(nonatomic, readonly) NSArray<RTCRtpReceiver *> *receivers;
 
+/** Gets all RTCRtpTransceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of
+ *  RTCRtpTransceiver. Use isEqual: instead of == to compare RTCRtpTransceiver
+ *  instances.
+ *  This is only available with RTCSdpSemanticsUnifiedPlan specified.
+ */
+@property(nonatomic, readonly) NSArray<RTCRtpTransceiver *> *transceivers;
+
 - (instancetype)init NS_UNAVAILABLE;
 
 /** Sets the PeerConnection's global configuration to |configuration|.
@@ -156,12 +181,70 @@ RTC_EXPORT
 /** Remove a group of remote candidates from the ICE Agent. */
 - (void)removeIceCandidates:(NSArray<RTCIceCandidate *> *)candidates;
 
-/** Add a new media stream to be sent on this peer connection. */
+/** Add a new media stream to be sent on this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  addTrack instead.
+ */
 - (void)addStream:(RTCMediaStream *)stream;
 
-/** Remove the given media stream from this peer connection. */
+/** Remove the given media stream from this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  removeTrack instead.
+ */
 - (void)removeStream:(RTCMediaStream *)stream;
 
+/** Add a new media stream track to be sent on this peer connection, and return
+ *  the newly created RTCRtpSender. The RTCRtpSender will be associated with
+ *  the streams specified in the |streamLabels| list.
+ *
+ *  Errors: If an error occurs, returns nil. An error can occur if:
+ *  - A sender already exists for the track.
+ *  - The peer connection is closed.
+ */
+- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track
+              streamLabels:(NSArray<NSString *> *)streamLabels;
+
+/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
+ *
+ *  With UnifiedPlan semantics, sets sender's track to null and removes the
+ *  send component from the associated RTCRtpTransceiver's direction.
+ *
+ *  Returns YES on success.
+ */
+- (BOOL)removeTrack:(RTCRtpSender *)sender;
+
+/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
+ *  transceivers. Adding a transceiver will cause future calls to CreateOffer
+ *  to add a media description for the corresponding transceiver.
+ *
+ *  The initial value of |mid| in the returned transceiver is nil. Setting a
+ *  new session description may change it to a non-nil value.
+ *
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
+ *
+ *  Optionally, an RtpTransceiverInit structure can be specified to configure
+ *  the transceiver from construction. If not specified, the transceiver will
+ *  default to having a direction of kSendRecv and not be part of any streams.
+ *
+ *  These methods are only available when Unified Plan is enabled (see
+ *  RTCConfiguration).
+ */
+
+/** Adds a transceiver with a sender set to transmit the given track. The kind
+ *  of the transceiver (and sender/receiver) will be derived from the kind of
+ *  the track.
+ */
+- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track;
+- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track
+                                          init:(RTCRtpTransceiverInit *)init;
+
+/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
+ *  or RTCRtpMediaTypeVideo.
+ */
+- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
+- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType
+                                       init:(RTCRtpTransceiverInit *)init;
+
 /** Generate an SDP offer. */
 - (void)offerForConstraints:(RTCMediaConstraints *)constraints
           completionHandler:(nullable void (^)
@@ -204,6 +287,8 @@ RTC_EXPORT
 
 /** Create an RTCRtpSender with the specified kind and media stream ID.
  *  See RTCMediaStreamTrack.h for available kinds.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  addTransceiver instead.
  */
 - (RTCRtpSender *)senderWithKind:(NSString *)kind streamId:(NSString *)streamId;
 
@@ -212,8 +297,8 @@ RTC_EXPORT
 @interface RTCPeerConnection (DataChannel)
 
 /** Create a new data channel with the given label and configuration. */
-- (RTCDataChannel *)dataChannelForLabel:(NSString *)label
-    configuration:(RTCDataChannelConfiguration *)configuration;
+- (nullable RTCDataChannel *)dataChannelForLabel:(NSString *)label
+                                   configuration:(RTCDataChannelConfiguration *)configuration;
 
 @end
 

Build/iOS/WebRTC.framework/Headers/RTCPeerConnectionFactory.h

@@ -14,7 +14,6 @@
 
 NS_ASSUME_NONNULL_BEGIN
 
-@class RTCAVFoundationVideoSource;
 @class RTCAudioSource;
 @class RTCAudioTrack;
 @class RTCConfiguration;
@@ -23,6 +22,7 @@ NS_ASSUME_NONNULL_BEGIN
 @class RTCPeerConnection;
 @class RTCVideoSource;
 @class RTCVideoTrack;
+@class RTCPeerConnectionFactoryOptions;
 @protocol RTCPeerConnectionDelegate;
 @protocol RTCVideoDecoderFactory;
 @protocol RTCVideoEncoderFactory;
@@ -49,10 +49,6 @@ RTC_EXPORT
 - (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source
                                 trackId:(NSString *)trackId;
 
-/** Initialize an RTCAVFoundationVideoSource with constraints. */
-- (RTCAVFoundationVideoSource *)avFoundationVideoSourceWithConstraints:
-    (nullable RTCMediaConstraints *)constraints;
-
 /** Initialize a generic RTCVideoSource. The RTCVideoSource should be passed to a RTCVideoCapturer
  *  implementation, e.g. RTCCameraVideoCapturer, in order to produce frames.
  */
@@ -75,6 +71,9 @@ RTC_EXPORT
                                               delegate:
     (nullable id<RTCPeerConnectionDelegate>)delegate;
 
+/** Set the options to be used for subsequently created RTCPeerConnections */
+- (void)setOptions:(nonnull RTCPeerConnectionFactoryOptions *)options;
+
 /** Start an AecDump recording. This API call will likely change in the future. */
 - (BOOL)startAecDumpWithFilePath:(NSString *)filePath
                   maxSizeInBytes:(int64_t)maxSizeInBytes;

Build/iOS/WebRTC.framework/Headers/RTCPeerConnectionFactoryOptions.h

@@ -0,0 +1,38 @@
+/*
+ *  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/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@interface RTCPeerConnectionFactoryOptions : NSObject
+
+@property(nonatomic, assign) BOOL disableEncryption;
+
+@property(nonatomic, assign) BOOL disableNetworkMonitor;
+
+@property(nonatomic, assign) BOOL ignoreLoopbackNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreVPNNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreCellularNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreWiFiNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END

Build/iOS/WebRTC.framework/Headers/RTCRtpReceiver.h

@@ -65,7 +65,7 @@ RTC_EXPORT
  *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
  *  RTCMediaStreamTrack instances.
  */
-@property(nonatomic, readonly) RTCMediaStreamTrack *track;
+@property(nonatomic, readonly, nullable) RTCMediaStreamTrack *track;
 
 /** The delegate for this RtpReceiver. */
 @property(nonatomic, weak) id<RTCRtpReceiverDelegate> delegate;

Build/iOS/WebRTC.framework/Headers/RTCRtpSender.h

@@ -10,6 +10,7 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCDtmfSender.h>
 #import <WebRTC/RTCMacros.h>
 #import <WebRTC/RTCMediaStreamTrack.h>
 #import <WebRTC/RTCRtpParameters.h>
@@ -34,6 +35,9 @@ RTC_EXPORT
  */
 @property(nonatomic, copy, nullable) RTCMediaStreamTrack *track;
 
+/** The RTCDtmfSender accociated with the RTP sender. */
+@property(nonatomic, readonly, nullable) id<RTCDtmfSender> dtmfSender;
+
 @end
 
 RTC_EXPORT

Build/iOS/WebRTC.framework/Headers/RTCRtpTransceiver.h

@@ -0,0 +1,128 @@
+/*
+ *  Copyright 2018 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/RTCMacros.h>
+#import <WebRTC/RTCRtpReceiver.h>
+#import <WebRTC/RTCRtpSender.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection */
+typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
+  RTCRtpTransceiverDirectionSendRecv,
+  RTCRtpTransceiverDirectionSendOnly,
+  RTCRtpTransceiverDirectionRecvOnly,
+  RTCRtpTransceiverDirectionInactive,
+};
+
+/** Structure for initializing an RTCRtpTransceiver in a call to
+ *  RTCPeerConnection.addTransceiver.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
+ */
+@interface RTCRtpTransceiverInit : NSObject
+
+/** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
+@property(nonatomic) RTCRtpTransceiverDirection direction;
+
+/** The added RTCRtpTransceiver will be added to these streams. */
+@property(nonatomic) NSArray<NSString *> *streamLabels;
+
+/** TODO(bugs.webrtc.org/7600): Not implemented. */
+@property(nonatomic) NSArray<RTCRtpEncodingParameters *> *sendEncodings;
+
+@end
+
+@class RTCRtpTransceiver;
+
+/** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the WebRTC
+ *  specification. A transceiver represents a combination of an RTCRtpSender
+ *  and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
+ *  RTCRtpTransceiver is said to be associated with a media description if its
+ *  mid property is non-nil; otherwise, it is said to be disassociated.
+ *  JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
+ *
+ *  Note that RTCRtpTransceivers are only supported when using
+ *  RTCPeerConnection with Unified Plan SDP.
+ *
+ *  WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
+ */
+RTC_EXPORT
+@protocol RTCRtpTransceiver <NSObject>
+
+/** Media type of the transceiver. The sender and receiver will also have this
+ *  type.
+ */
+@property(nonatomic, readonly) RTCRtpMediaType mediaType;
+
+/** The mid attribute is the mid negotiated and present in the local and
+ *  remote descriptions. Before negotiation is complete, the mid value may be
+ *  nil. After rollbacks, the value may change from a non-nil value to nil.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
+ */
+@property(nonatomic, readonly) NSString *mid;
+
+/** The sender attribute exposes the RTCRtpSender corresponding to the RTP
+ *  media that may be sent with the transceiver's mid. The sender is always
+ *  present, regardless of the direction of media.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
+ */
+@property(nonatomic, readonly) RTCRtpSender *sender;
+
+/** The receiver attribute exposes the RTCRtpReceiver corresponding to the RTP
+ *  media that may be received with the transceiver's mid. The receiver is
+ *  always present, regardless of the direction of media.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
+ */
+@property(nonatomic, readonly) RTCRtpReceiver *receiver;
+
+/** The isStopped attribute indicates that the sender of this transceiver will
+ *  no longer send, and that the receiver will no longer receive. It is true if
+ *  either stop has been called or if setting the local or remote description
+ *  has caused the RTCRtpTransceiver to be stopped.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
+ */
+@property(nonatomic, readonly) BOOL isStopped;
+
+/** The direction attribute indicates the preferred direction of this
+ *  transceiver, which will be used in calls to createOffer and createAnswer.
+ *  An update of directionality does not take effect immediately. Instead,
+ *  future calls to createOffer and createAnswer mark the corresponding media
+ *  descriptions as sendrecv, sendonly, recvonly, or inactive.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
+ */
+@property(nonatomic) RTCRtpTransceiverDirection direction;
+
+/** The currentDirection attribute indicates the current direction negotiated
+ *  for this transceiver. If this transceiver has never been represented in an
+ *  offer/answer exchange, or if the transceiver is stopped, the value is not
+ *  present and this method returns NO.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
+ */
+- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut;
+
+/** The stop method irreversibly stops the RTCRtpTransceiver. The sender of
+ *  this transceiver will no longer send, the receiver will no longer receive.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
+ */
+- (void)stop;
+
+@end
+
+RTC_EXPORT
+@interface RTCRtpTransceiver : NSObject <RTCRtpTransceiver>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END

Build/iOS/WebRTC.framework/Headers/RTCSSLAdapter.h

@@ -16,5 +16,5 @@
  * Initialize and clean up the SSL library. Failure is fatal. These call the
  * corresponding functions in webrtc/rtc_base/ssladapter.h.
  */
-RTC_EXTERN BOOL RTCInitializeSSL();
-RTC_EXTERN BOOL RTCCleanupSSL();
+RTC_EXTERN BOOL RTCInitializeSSL(void);
+RTC_EXTERN BOOL RTCCleanupSSL(void);

Build/iOS/WebRTC.framework/Headers/RTCTracing.h

@@ -12,10 +12,10 @@
 
 #import <WebRTC/RTCMacros.h>
 
-RTC_EXTERN void RTCSetupInternalTracer();
+RTC_EXTERN void RTCSetupInternalTracer(void);
 /** Starts capture to specified file. Must be a valid writable path.
  *  Returns YES if capture starts.
  */
 RTC_EXTERN BOOL RTCStartInternalCapture(NSString *filePath);
-RTC_EXTERN void RTCStopInternalCapture();
-RTC_EXTERN void RTCShutdownInternalTracer();
+RTC_EXTERN void RTCStopInternalCapture(void);
+RTC_EXTERN void RTCShutdownInternalTracer(void);

Build/iOS/WebRTC.framework/Headers/RTCVideoCodec.h

@@ -15,6 +15,12 @@
 
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT extern NSString *const kRTCVideoCodecVp8Name;
+RTC_EXPORT extern NSString *const kRTCVideoCodecVp9Name;
+RTC_EXPORT extern NSString *const kRTCVideoCodecH264Name;
+RTC_EXPORT extern NSString *const kRTCLevel31ConstrainedHigh;
+RTC_EXPORT extern NSString *const kRTCLevel31ConstrainedBaseline;
+
 /** Represents an encoded frame's type. */
 typedef NS_ENUM(NSUInteger, RTCFrameType) {
   RTCFrameTypeEmptyFrame = 0,
@@ -84,7 +90,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
 
 /** Holds information to identify a codec. Corresponds to cricket::VideoCodec. */
 RTC_EXPORT
-@interface RTCVideoCodecInfo : NSObject
+@interface RTCVideoCodecInfo : NSObject <NSCoding>
 
 - (instancetype)init NS_UNAVAILABLE;
 
@@ -142,7 +148,7 @@ RTC_EXPORT
                        numberOfCores:(int)numberOfCores;
 - (NSInteger)releaseEncoder;
 - (NSInteger)encode:(RTCVideoFrame *)frame
-    codecSpecificInfo:(id<RTCCodecSpecificInfo>)info
+    codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
            frameTypes:(NSArray<NSNumber *> *)frameTypes;
 - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
 - (NSString *)implementationName;
@@ -165,7 +171,7 @@ RTC_EXPORT
 - (NSInteger)decode:(RTCEncodedImage *)encodedImage
           missingFrames:(BOOL)missingFrames
     fragmentationHeader:(RTCRtpFragmentationHeader *)fragmentationHeader
-      codecSpecificInfo:(__nullable id<RTCCodecSpecificInfo>)info
+      codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
            renderTimeMs:(int64_t)renderTimeMs;
 - (NSString *)implementationName;
 

Build/iOS/WebRTC.framework/Headers/RTCVideoCodecFactory.h

@@ -19,7 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
 RTC_EXPORT
 @protocol RTCVideoEncoderFactory <NSObject>
 
-- (id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
+- (nullable id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
 - (NSArray<RTCVideoCodecInfo *> *)supportedCodecs;  // TODO(andersc): "supportedFormats" instead?
 
 @end
@@ -28,9 +28,27 @@ RTC_EXPORT
 RTC_EXPORT
 @protocol RTCVideoDecoderFactory <NSObject>
 
-- (id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
+- (nullable id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
 - (NSArray<RTCVideoCodecInfo *> *)supportedCodecs;  // TODO(andersc): "supportedFormats" instead?
 
 @end
 
+#pragma mark - Default factories
+
+/** These codec factories include support for all codecs bundled with WebRTC. If using custom
+ *  codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
+ */
+RTC_EXPORT
+@interface RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
+
+@property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
+
++ (NSArray<RTCVideoCodecInfo *> *)supportedCodecs;
+
+@end
+
+RTC_EXPORT
+@interface RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
+@end
+
 NS_ASSUME_NONNULL_END

Build/iOS/WebRTC.framework/Headers/RTCVideoFrameBuffer.h

@@ -72,6 +72,8 @@ RTC_EXPORT
 @property(nonatomic, readonly) int cropX;
 @property(nonatomic, readonly) int cropY;
 
++ (NSSet<NSNumber *> *)supportedPixelFormats;
+
 - (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
 - (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
                        adaptedWidth:(int)adaptedWidth

Build/iOS/WebRTC.framework/Headers/UIDevice+RTCDevice.h

@@ -28,6 +28,11 @@ typedef NS_ENUM(NSInteger, RTCDeviceType) {
   RTCDeviceTypeIPhone6,
   RTCDeviceTypeIPhone6S,
   RTCDeviceTypeIPhone6SPlus,
+  RTCDeviceTypeIPhone7,
+  RTCDeviceTypeIPhone7Plus,
+  RTCDeviceTypeIPhone8,
+  RTCDeviceTypeIPhone8Plus,
+  RTCDeviceTypeIPhoneX,
   RTCDeviceTypeIPodTouch1G,
   RTCDeviceTypeIPodTouch2G,
   RTCDeviceTypeIPodTouch3G,
@@ -58,8 +63,6 @@ typedef NS_ENUM(NSInteger, RTCDeviceType) {
 @interface UIDevice (RTCDevice)
 
 + (RTCDeviceType)deviceType;
-+ (NSString *)stringForDeviceType:(RTCDeviceType)deviceType;
-+ (BOOL)isIOS9OrLater;
 + (BOOL)isIOS11OrLater;
 
 @end

Build/iOS/WebRTC.framework/Headers/WebRTC.h

@@ -1,36 +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.
- */
+     *  Copyright 2018 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>
-#if TARGET_OS_IPHONE
 #import <WebRTC/RTCAudioSession.h>
+#import <WebRTC/RTCVideoCodec.h>
+#import <WebRTC/RTCVideoCodecFactory.h>
 #import <WebRTC/RTCAudioSessionConfiguration.h>
-#endif
 #import <WebRTC/RTCAudioSource.h>
 #import <WebRTC/RTCAudioTrack.h>
 #import <WebRTC/RTCCameraVideoCapturer.h>
-#import <WebRTC/RTCVideoCapturer.h>
-#if TARGET_OS_IPHONE
 #import <WebRTC/RTCCameraPreviewView.h>
-#endif
 #import <WebRTC/RTCConfiguration.h>
 #import <WebRTC/RTCDataChannel.h>
 #import <WebRTC/RTCDataChannelConfiguration.h>
 #import <WebRTC/RTCDispatcher.h>
-#if TARGET_OS_IPHONE
 #import <WebRTC/RTCEAGLVideoView.h>
-#import <WebRTC/RTCMTLVideoView.h>
-#import <WebRTC/RTCVideoViewShading.h>
-#endif
 #import <WebRTC/RTCFieldTrials.h>
-#import <WebRTC/RTCFileLogger.h>
+#import <WebRTC/RTCFileVideoCapturer.h>
 #import <WebRTC/RTCIceCandidate.h>
 #import <WebRTC/RTCIceServer.h>
 #import <WebRTC/RTCIntervalRange.h>
@@ -45,22 +37,29 @@
 #import <WebRTC/RTCMetricsSampleInfo.h>
 #import <WebRTC/RTCPeerConnection.h>
 #import <WebRTC/RTCPeerConnectionFactory.h>
+#import <WebRTC/RTCPeerConnectionFactoryOptions.h>
 #import <WebRTC/RTCRtpCodecParameters.h>
 #import <WebRTC/RTCRtpEncodingParameters.h>
 #import <WebRTC/RTCRtpParameters.h>
 #import <WebRTC/RTCRtpReceiver.h>
 #import <WebRTC/RTCRtpSender.h>
+#import <WebRTC/RTCRtpTransceiver.h>
+#import <WebRTC/RTCDtmfSender.h>
 #import <WebRTC/RTCSSLAdapter.h>
 #import <WebRTC/RTCSessionDescription.h>
 #import <WebRTC/RTCTracing.h>
-#import <WebRTC/RTCVideoCodec.h>
-#import <WebRTC/RTCVideoCodecFactory.h>
+#import <WebRTC/RTCVideoCapturer.h>
 #import <WebRTC/RTCVideoCodecH264.h>
 #import <WebRTC/RTCVideoFrame.h>
 #import <WebRTC/RTCVideoFrameBuffer.h>
 #import <WebRTC/RTCVideoRenderer.h>
 #import <WebRTC/RTCVideoSource.h>
 #import <WebRTC/RTCVideoTrack.h>
-#if TARGET_OS_IPHONE
+#import <WebRTC/RTCVideoViewShading.h>
 #import <WebRTC/UIDevice+RTCDevice.h>
-#endif
+#import <WebRTC/RTCVideoDecoderVP8.h>
+#import <WebRTC/RTCVideoDecoderVP9.h>
+#import <WebRTC/RTCVideoEncoderVP8.h>
+#import <WebRTC/RTCVideoEncoderVP9.h>
+#import <WebRTC/RTCCallbackLogger.h>
+#import <WebRTC/RTCFileLogger.h>

Build/iOS/WebRTC.framework/LICENSE.md

@@ -36,10 +36,6 @@ party licenses. Paths to the files and associated licenses are collected here.
 Files governed by third party licenses:
 base/base64.cc
 base/base64.h
-base/md5.cc
-base/md5.h
-base/sha1.cc
-base/sha1.h
 base/sigslot.cc
 base/sigslot.h
 common_audio/fft4g.c
@@ -81,34 +77,6 @@ License:
 //*********************************************************************
 -------------------------------------------------------------------------------
 Files:
-base/md5.cc
-base/md5.h
-
-License:
-/*
- * This code implements the MD5 message-digest algorithm.
- * The algorithm is due to Ron Rivest.  This code was
- * written by Colin Plumb in 1993, no copyright is claimed.
- * This code is in the public domain; do with it what you wish.
- *
--------------------------------------------------------------------------------
-Files:
-base/sha1.cc
-base/sha1.h
-
-License:
-/*
- * SHA-1 in C
- * By Steve Reid <sreid@sea-to-sky.net>
- * 100% Public Domain
- *
- * -----------------
- * Modified 7/98
- * By James H. Brown <jbrown@burgoyne.com>
- * Still 100% Public Domain
- *
--------------------------------------------------------------------------------
-Files:
 base/sigslot.cc
 base/sigslot.h
 
@@ -500,8 +468,9 @@ license. This license is reproduced at the bottom of this file.
 Contributors to BoringSSL are required to follow the CLA rules for Chromium:
 https://cla.developers.google.com/clas
 
-Some files from Intel are under yet another license, which is also included
-underneath.
+Files in third_party/ have their own licenses, as described therein. The MIT
+license, for third_party/fiat, which, unlike other third_party directories, is
+compiled into non-test libraries, is included below.
 
 The OpenSSL toolkit stays under a dual license, i.e. both the conditions of the
 OpenSSL License and the original SSLeay license apply to the toolkit. See below
@@ -651,40 +620,55 @@ ISC license used for completely new code in BoringSSL:
  * CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. */
 
 
-Some files from Intel carry the following license:
+The code in third_party/fiat carries the MIT license:
 
-# Copyright (c) 2012, Intel Corporation
-#
-# All rights reserved.
-#
-# Redistribution and use in source and binary forms, with or without
-# modification, are permitted provided that the following conditions are
-# met:
-#
-# *  Redistributions of source code must retain the above copyright
-#    notice, this list of conditions and the following disclaimer.
-#
-# *  Redistributions in binary form must reproduce the above copyright
-#    notice, this list of conditions and the following disclaimer in the
-#    documentation and/or other materials provided with the
-#    distribution.
-#
-# *  Neither the name of the Intel Corporation nor the names of its
-#    contributors may be used to endorse or promote products derived from
-#    this software without specific prior written permission.
-#
-#
-# THIS SOFTWARE IS PROVIDED BY INTEL CORPORATION ""AS IS"" AND ANY
-# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
-# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
-# PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL INTEL CORPORATION OR
-# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-# LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-# NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-# SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+Copyright (c) 2015-2016 the fiat-crypto authors (see
+https://github.com/mit-plv/fiat-crypto/blob/master/AUTHORS).
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.
+
+```
+
+# fiat
+```
+The MIT License (MIT)
+
+Copyright (c) 2015-2016 the fiat-crypto authors (see
+https://github.com/mit-plv/fiat-crypto/blob/master/AUTHORS).
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.
 
 ```
 
@@ -929,3 +913,57 @@ SUCH DAMAGE.
 
 ```
 
+# zlib
+```
+Copyright (c) 2011, The WebRTC project authors. All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+  * Redistributions of source code must retain the above copyright
+    notice, this list of conditions and the following disclaimer.
+
+  * Redistributions in binary form must reproduce the above copyright
+    notice, this list of conditions and the following disclaimer in
+    the documentation and/or other materials provided with the
+    distribution.
+
+  * Neither the name of Google nor the names of its contributors may
+    be used to endorse or promote products derived from this software
+    without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+version 1.2.11, January 15th, 2017
+
+Copyright (C) 1995-2017 Jean-loup Gailly and Mark Adler
+
+This software is provided 'as-is', without any express or implied
+warranty.  In no event will the authors be held liable for any damages
+arising from the use of this software.
+
+Permission is granted to anyone to use this software for any purpose,
+including commercial applications, and to alter it and redistribute it
+freely, subject to the following restrictions:
+
+1. The origin of this software must not be misrepresented; you must not
+   claim that you wrote the original software. If you use this software
+   in a product, an acknowledgment in the product documentation would be
+   appreciated but is not required.
+2. Altered source versions must be plainly marked as such, and must not be
+   misrepresented as being the original software.
+3. This notice may not be removed or altered from any source distribution.
+
+```
+

Build/iOS/WebRTC.framework/build_env.txt

@@ -1,35 +0,0 @@
-## WebRTC Build Details
-
-To track down potential future issues, we log some of our build environment details.
-
-webrtc git branch:
-(HEAD detached at dc5904661)
-
-webrtc git sha:
-dc5904661c4c8b357e19595d80b2881db3f4a71b
-
-build_script git sha:
-2a793074f4cfd6f9ed121724b52963de68134372
-
-xcodebuild -version:
-Xcode 9.2
-Build version 9C40b
-
-xcode-select -p:
-/Applications/Xcode.app/Contents/Developer
-
-gcc -v:
-Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1
-Apple LLVM version 9.0.0 (clang-900.0.39.2)
-Target: x86_64-apple-darwin16.7.0
-Thread model: posix
-InstalledDir: /Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin
-
-osx_version_details:
-ProductName:	Mac OS X
-ProductVersion:	10.12.6
-BuildVersion:	16G1212
-
-hostname:
-oak.local
-