Obj-C SDK Cleanup

This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.

A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.

The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.

The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.

Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}

BUILD.gn

@@ -320,6 +320,10 @@ config("common_config") {
 
 config("common_objc") {
   libs = [ "Foundation.framework" ]
+
+  if (rtc_use_metal_rendering) {
+    defines = [ "RTC_SUPPORTS_METAL" ]
+  }
 }
 
 if (!build_with_chromium) {

examples/BUILD.gn

@@ -205,19 +205,15 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
 
     if (is_ios) {
       # iOS must use WebRTC.framework which is dynamically linked.
-      # 'gn check' is disabled in order to avoid confusion and
-      # errors caused by multiple implementations.
-      check_includes = false
       deps = [
-        "../sdk:framework_objc",
+        "../sdk:framework_objc+link",
         "../system_wrappers:field_trial_default",
         "../system_wrappers:runtime_enabled_features_default",
       ]
     } else {
       deps = [
-        "../sdk:common_objc",
+        "../sdk:mac_framework_objc+link",
         "../system_wrappers:field_trial_default",
-        "../system_wrappers:metrics_default",
         "../system_wrappers:runtime_enabled_features_default",
       ]
     }
@@ -279,23 +275,12 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
     ]
     if (is_ios) {
       # iOS must use WebRTC.framework which is dynamically linked.
-      # 'gn check' is disabled in order to avoid confusion and
-      # errors caused by multiple implementations.
-      check_includes = false
       deps += [
-        ":AppRTCMobile_ios_frameworks",
+        "../sdk:framework_objc+link",
-        "../sdk:framework_objc",
+        "../sdk:ios_framework_bundle",
       ]
     } else {
-      deps += [
+      deps += [ "../sdk:mac_framework_objc+link" ]
-        "../sdk:common_objc",
-        "../sdk:mediaconstraints_objc",
-        "../sdk:peerconnectionfactory_base_objc",
-        "../sdk:videocapture_objc",
-        "../sdk:videocodec_objc",
-        "../sdk:videoframebuffer_objc",
-        "../sdk:videosource_objc",
-      ]
     }
     libs = [ "QuartzCore.framework" ]
   }
@@ -303,9 +288,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
   if (is_ios) {
     rtc_static_library("AppRTCMobile_lib") {
       # iOS must use WebRTC.framework which is dynamically linked.
-      # 'gn check' is disabled in order to avoid confusion and
-      # errors caused by multiple implementations.
-      check_includes = false
       testonly = true
       sources = [
         "objc/AppRTCMobile/ios/ARDAppDelegate.h",
@@ -331,10 +313,10 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
       ]
 
       deps = [
-        ":AppRTCMobile_ios_frameworks",
         ":apprtc_common",
         ":apprtc_signaling",
-        "../sdk:framework_objc",
+        "../sdk:framework_objc+link",
+        "../sdk:ios_framework_bundle",
       ]
     }
 
@@ -351,9 +333,9 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
 
       deps = [
         ":AppRTCMobile_ios_bundle_data",
-        ":AppRTCMobile_ios_frameworks",
         ":AppRTCMobile_lib",
         "../sdk:framework_objc",
+        "../sdk:ios_framework_bundle",
       ]
 
       if (rtc_apprtcmobile_broadcast_extension) {
@@ -396,7 +378,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
       }
 
       rtc_static_library("AppRTCMobileBroadcastUpload_lib") {
-        check_includes = false
         testonly = true
         sources = [
           "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h",
@@ -404,9 +385,9 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
         ]
 
         deps = [
-          ":AppRTCMobile_ios_frameworks",
           ":apprtc_signaling",
-          "../sdk:framework_objc",
+          "../sdk:framework_objc+link",
+          "../sdk:ios_framework_bundle",
         ]
 
         libs = [ "ReplayKit.framework" ]
@@ -441,18 +422,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
       }
     }
 
-    bundle_data("AppRTCMobile_ios_frameworks") {
-      deps = [
-        "../sdk:framework_objc+link",
-      ]
-      sources = [
-        "$root_build_dir/WebRTC.framework",
-      ]
-      outputs = [
-        "{{bundle_resources_dir}}/Frameworks/{{source_file_part}}",
-      ]
-    }
-
     bundle_data("AppRTCMobile_ios_bundle_data") {
       sources = [
         "objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf",
@@ -507,10 +476,12 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
         "../modules/audio_processing:audio_processing",
         "../pc:libjingle_peerconnection",
         "../rtc_base:rtc_base",
+        "../sdk:base_objc",
         "../sdk:default_codec_factory_objc",
-        "../sdk:framework_objc",
+        "../sdk:helpers_objc",
         "../sdk:native_api",
         "../sdk:ui_objc",
+        "../sdk:videocapture_objc",
         "../sdk:videotoolbox_objc",
         "../system_wrappers:field_trial_default",
         "../system_wrappers:metrics_default",
@@ -557,14 +528,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
       deps = [
         ":apprtc_common",
         ":apprtc_signaling",
-        "../sdk:default_codec_factory_objc",
+        "../sdk:mac_framework_objc+link",
-        "../sdk:metal_objc",
-        "../sdk:peerconnectionfactory_base_objc",
-        "../sdk:ui_objc",
-        "../sdk:videocapture_objc",
-        "../sdk:videocodec_objc",
-        "../sdk:videocodec_objc",
-        "../sdk:videotoolbox_objc",
       ]
     }
 
@@ -584,6 +548,8 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
 
       deps = [
         ":AppRTCMobile_lib",
+        "../sdk:mac_framework_bundle",
+        "../sdk:mac_framework_objc+link",
       ]
     }
   }
@@ -629,26 +595,22 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
     if (is_ios) {
       rtc_source_set("apprtcmobile_test_sources") {
         # iOS must use WebRTC.framework which is dynamically linked.
-        # 'gn check' is disabled in order to avoid confusion and
-        # errors caused by multiple implementations.
-        check_includes = false
         testonly = true
         include_dirs = [
           "objc/AppRTCMobile",
           "objc/AppRTCMobile/ios",
         ]
-        testonly = true
         sources = [
           "objc/AppRTCMobile/tests/ARDAppClient_xctest.mm",
           "objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm",
           "objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm",
         ]
         deps = [
-          ":AppRTCMobile_ios_frameworks",
           ":AppRTCMobile_lib",
           ":apprtc_signaling",
           "../rtc_base:rtc_base",
-          "../sdk:framework_objc",
+          "../sdk:framework_objc+link",
+          "../sdk:ios_framework_bundle",
           "//build/config/ios:xctest",
           "//third_party/ocmock",
         ]

examples/DEPS

@@ -1,5 +1,4 @@
 include_rules = [
-  "+WebRTC",
   "+api",
   "+common_video",
   "+media",
@@ -8,6 +7,6 @@ include_rules = [
   "+modules/audio_processing",
   "+p2p",
   "+pc",
-  "+sdk/objc/Framework/Native/api",
+  "+sdk/objc",
   "+third_party/libyuv",
 ]

examples/objc/AppRTCMobile/ARDAppClient+Internal.h

@@ -10,7 +10,7 @@
 
 #import "ARDAppClient.h"
 
-#import "WebRTC/RTCPeerConnection.h"
+#import <WebRTC/RTCPeerConnection.h>
 
 #import "ARDRoomServerClient.h"
 #import "ARDSignalingChannel.h"

examples/objc/AppRTCMobile/ARDAppClient.h

@@ -9,8 +9,8 @@
  */
 
 #import <Foundation/Foundation.h>
-#import "WebRTC/RTCPeerConnection.h"
+#import <WebRTC/RTCPeerConnection.h>
-#import "WebRTC/RTCVideoTrack.h"
+#import <WebRTC/RTCVideoTrack.h>
 
 typedef NS_ENUM(NSInteger, ARDAppClientState) {
   // Disconnected from servers.

examples/objc/AppRTCMobile/ARDAppClient.m

@@ -10,22 +10,23 @@
 
 #import "ARDAppClient+Internal.h"
 
-#import "WebRTC/RTCAudioTrack.h"
+#import <WebRTC/RTCAudioTrack.h>
-#import "WebRTC/RTCCameraVideoCapturer.h"
+#import <WebRTC/RTCCameraVideoCapturer.h>
-#import "WebRTC/RTCConfiguration.h"
+#import <WebRTC/RTCConfiguration.h>
-#import "WebRTC/RTCFileLogger.h"
+#import <WebRTC/RTCDefaultVideoDecoderFactory.h>
-#import "WebRTC/RTCFileVideoCapturer.h"
+#import <WebRTC/RTCDefaultVideoEncoderFactory.h>
-#import "WebRTC/RTCIceServer.h"
+#import <WebRTC/RTCFileLogger.h>
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCFileVideoCapturer.h>
-#import "WebRTC/RTCMediaConstraints.h"
+#import <WebRTC/RTCIceServer.h>
-#import "WebRTC/RTCMediaStream.h"
+#import <WebRTC/RTCLogging.h>
-#import "WebRTC/RTCPeerConnectionFactory.h"
+#import <WebRTC/RTCMediaConstraints.h>
-#import "WebRTC/RTCRtpSender.h"
+#import <WebRTC/RTCMediaStream.h>
-#import "WebRTC/RTCRtpTransceiver.h"
+#import <WebRTC/RTCPeerConnectionFactory.h>
-#import "WebRTC/RTCTracing.h"
+#import <WebRTC/RTCRtpSender.h>
-#import "WebRTC/RTCVideoCodecFactory.h"
+#import <WebRTC/RTCRtpTransceiver.h>
-#import "WebRTC/RTCVideoSource.h"
+#import <WebRTC/RTCTracing.h>
-#import "WebRTC/RTCVideoTrack.h"
+#import <WebRTC/RTCVideoSource.h>
+#import <WebRTC/RTCVideoTrack.h>
 
 #import "ARDAppEngineClient.h"
 #import "ARDExternalSampleCapturer.h"

examples/objc/AppRTCMobile/ARDAppEngineClient.m

@@ -10,7 +10,7 @@
 
 #import "ARDAppEngineClient.h"
 
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCLogging.h>
 
 #import "ARDJoinResponse.h"
 #import "ARDMessageResponse.h"

examples/objc/AppRTCMobile/ARDCaptureController.h

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <WebRTC/RTCCameraVideoCapturer.h>
+#import <WebRTC/RTCCameraVideoCapturer.h>
 
 @class ARDSettingsModel;
 

examples/objc/AppRTCMobile/ARDCaptureController.m

@@ -10,8 +10,9 @@
 
 #import "ARDCaptureController.h"
 
+#import <WebRTC/RTCLogging.h>
+
 #import "ARDSettingsModel.h"
-#import "WebRTC/RTCLogging.h"
 
 const Float64 kFramerateLimit = 30.0;
 

examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m

@@ -10,7 +10,8 @@
 
 #import "ARDExternalSampleCapturer.h"
 
-#import "WebRTC/RTCVideoFrameBuffer.h"
+#import <WebRTC/RTCCVPixelBuffer.h>
+#import <WebRTC/RTCVideoFrameBuffer.h>
 
 @implementation ARDExternalSampleCapturer
 

examples/objc/AppRTCMobile/ARDSettingsModel+Private.h

@@ -9,6 +9,7 @@
  */
 
 #import <Foundation/Foundation.h>
+
 #import "ARDSettingsModel.h"
 
 @class ARDSettingsStore;

examples/objc/AppRTCMobile/ARDSettingsModel.h

@@ -10,7 +10,7 @@
 
 #import <Foundation/Foundation.h>
 
-#import "WebRTC/RTCVideoCodec.h"
+#import <WebRTC/RTCVideoCodecInfo.h>
 
 NS_ASSUME_NONNULL_BEGIN
 

examples/objc/AppRTCMobile/ARDSettingsModel.m

@@ -10,9 +10,10 @@
 
 #import "ARDSettingsModel+Private.h"
 #import "ARDSettingsStore.h"
-#import "WebRTC/RTCCameraVideoCapturer.h"
+
-#import "WebRTC/RTCMediaConstraints.h"
+#import <WebRTC/RTCCameraVideoCapturer.h>
-#import "WebRTC/RTCVideoCodecFactory.h"
+#import <WebRTC/RTCDefaultVideoEncoderFactory.h>
+#import <WebRTC/RTCMediaConstraints.h>
 
 NS_ASSUME_NONNULL_BEGIN
 

examples/objc/AppRTCMobile/ARDSignalingMessage.h

@@ -10,8 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
-#import "WebRTC/RTCIceCandidate.h"
+#import <WebRTC/RTCIceCandidate.h>
-#import "WebRTC/RTCSessionDescription.h"
+#import <WebRTC/RTCSessionDescription.h>
 
 typedef enum {
   kARDSignalingMessageTypeCandidate,

examples/objc/AppRTCMobile/ARDSignalingMessage.m

@@ -10,7 +10,7 @@
 
 #import "ARDSignalingMessage.h"
 
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCLogging.h>
 
 #import "ARDUtilities.h"
 #import "RTCIceCandidate+JSON.h"

examples/objc/AppRTCMobile/ARDStatsBuilder.m

@@ -10,7 +10,7 @@
 
 #import "ARDStatsBuilder.h"
 
-#import "WebRTC/RTCLegacyStatsReport.h"
+#import <WebRTC/RTCLegacyStatsReport.h>
 
 #import "ARDBitrateTracker.h"
 #import "ARDUtilities.h"

examples/objc/AppRTCMobile/ARDWebSocketChannel.m

@@ -10,7 +10,7 @@
 
 #import "ARDWebSocketChannel.h"
 
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCLogging.h>
 #import "SRWebSocket.h"
 
 #import "ARDSignalingMessage.h"

examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCIceCandidate.h"
+#import <WebRTC/RTCIceCandidate.h>
 
 @interface RTCIceCandidate (JSON)
 

examples/objc/AppRTCMobile/RTCIceCandidate+JSON.m

@@ -10,7 +10,7 @@
 
 #import "RTCIceCandidate+JSON.h"
 
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCLogging.h>
 
 static NSString const *kRTCICECandidateTypeKey = @"type";
 static NSString const *kRTCICECandidateTypeValue = @"candidate";

examples/objc/AppRTCMobile/RTCIceServer+JSON.h

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCIceServer.h"
+#import <WebRTC/RTCIceServer.h>
 
 @interface RTCIceServer (JSON)
 

examples/objc/AppRTCMobile/RTCMediaConstraints+JSON.h

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCMediaConstraints.h"
+#import <WebRTC/RTCMediaConstraints.h>
 
 @interface RTCMediaConstraints (JSON)
 

examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCSessionDescription.h"
+#import <WebRTC/RTCSessionDescription.h>
 
 @interface RTCSessionDescription (JSON)
 

examples/objc/AppRTCMobile/common/ARDUtilities.m

@@ -12,7 +12,7 @@
 
 #import <mach/mach.h>
 
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCLogging.h>
 
 @implementation NSDictionary (ARDUtilites)
 

examples/objc/AppRTCMobile/ios/ARDAppDelegate.m

@@ -10,10 +10,10 @@
 
 #import "ARDAppDelegate.h"
 
-#import "WebRTC/RTCFieldTrials.h"
+#import <WebRTC/RTCFieldTrials.h>
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCLogging.h>
-#import "WebRTC/RTCSSLAdapter.h"
+#import <WebRTC/RTCSSLAdapter.h>
-#import "WebRTC/RTCTracing.h"
+#import <WebRTC/RTCTracing.h>
 
 #import "ARDMainViewController.h"
 

examples/objc/AppRTCMobile/ios/ARDFileCaptureController.m

@@ -10,7 +10,7 @@
 
 #import "ARDFileCaptureController.h"
 
-#import "WebRTC/RTCFileVideoCapturer.h"
+#import <WebRTC/RTCFileVideoCapturer.h>
 
 @interface ARDFileCaptureController ()
 

examples/objc/AppRTCMobile/ios/ARDMainViewController.m

@@ -12,11 +12,10 @@
 
 #import <AVFoundation/AVFoundation.h>
 
-#import "WebRTC/RTCAudioSession.h"
+#import <WebRTC/RTCAudioSession.h>
-#import "WebRTC/RTCAudioSessionConfiguration.h"
+#import <WebRTC/RTCAudioSessionConfiguration.h>
-#import "WebRTC/RTCDispatcher.h"
+#import <WebRTC/RTCDispatcher.h>
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCLogging.h>
-
 
 #import "ARDAppClient.h"
 #import "ARDMainView.h"

examples/objc/AppRTCMobile/ios/ARDStatsView.m

@@ -10,7 +10,7 @@
 
 #import "ARDStatsView.h"
 
-#import "WebRTC/RTCLegacyStatsReport.h"
+#import <WebRTC/RTCLegacyStatsReport.h>
 
 #import "ARDStatsBuilder.h"
 

examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m

@@ -10,17 +10,17 @@
 
 #import "ARDVideoCallViewController.h"
 
-#import "WebRTC/RTCAudioSession.h"
+#import <WebRTC/RTCAudioSession.h>
-#import "WebRTC/RTCCameraVideoCapturer.h"
+#import <WebRTC/RTCCameraVideoCapturer.h>
+#import <WebRTC/RTCDispatcher.h>
+#import <WebRTC/RTCLogging.h>
+#import <WebRTC/RTCMediaConstraints.h>
 
 #import "ARDAppClient.h"
 #import "ARDCaptureController.h"
 #import "ARDFileCaptureController.h"
 #import "ARDSettingsModel.h"
 #import "ARDVideoCallView.h"
-#import "WebRTC/RTCDispatcher.h"
-#import "WebRTC/RTCLogging.h"
-#import "WebRTC/RTCMediaConstraints.h"
 
 @interface ARDVideoCallViewController () <ARDAppClientDelegate,
                                           ARDVideoCallViewDelegate,

examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCVideoCodec.h"
+#import <WebRTC/RTCVideoCodecInfo.h>
 
 @interface RTCVideoCodecInfo (HumanReadable)
 

examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m

@@ -9,7 +9,8 @@
  */
 
 #import "RTCVideoCodecInfo+HumanReadable.h"
-#import "WebRTC/RTCVideoCodecH264.h"
+
+#import <WebRTC/RTCH264ProfileLevelId.h>
 
 @implementation RTCVideoCodecInfo (HumanReadable)
 

examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h

@@ -10,7 +10,7 @@
 
 #import <ReplayKit/ReplayKit.h>
 
-#import "WebRTC/RTCLogging.h"
+#import <WebRTC/RTCLogging.h>
 
 #import "ARDAppClient.h"
 

examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.m

@@ -10,7 +10,7 @@
 
 #import "APPRTCAppDelegate.h"
 #import "APPRTCViewController.h"
-#import "WebRTC/RTCSSLAdapter.h"
+#import <WebRTC/RTCSSLAdapter.h>
 
 @interface APPRTCAppDelegate () <NSWindowDelegate>
 @end

examples/objc/AppRTCMobile/mac/APPRTCViewController.m

@@ -12,9 +12,9 @@
 
 #import <AVFoundation/AVFoundation.h>
 
-#import "WebRTC/RTCMTLNSVideoView.h"
+#import <WebRTC/RTCMTLNSVideoView.h>
-#import "WebRTC/RTCNSGLVideoView.h"
+#import <WebRTC/RTCNSGLVideoView.h>
-#import "WebRTC/RTCVideoTrack.h"
+#import <WebRTC/RTCVideoTrack.h>
 
 #import "ARDAppClient.h"
 #import "ARDCaptureController.h"

examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm

@@ -15,8 +15,8 @@
 
 #include "rtc_base/ssladapter.h"
 
-#import "WebRTC/RTCMediaConstraints.h"
+#import <WebRTC/RTCMediaConstraints.h>
-#import "WebRTC/RTCPeerConnectionFactory.h"
+#import <WebRTC/RTCPeerConnectionFactory.h>
 
 #import "ARDAppClient+Internal.h"
 #import "ARDJoinResponse+Internal.h"

examples/objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm

@@ -14,7 +14,7 @@
 
 #import "ARDFileCaptureController.h"
 
-#import "WebRTC/RTCFileVideoCapturer.h"
+#import <WebRTC/RTCFileVideoCapturer.h>
 
 NS_CLASS_AVAILABLE_IOS(10)
 @interface ARDFileCaptureControllerTests : XCTestCase

examples/objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm

@@ -12,7 +12,7 @@
 #import <OCMock/OCMock.h>
 #import <XCTest/XCTest.h>
 
-#import "WebRTC/RTCMediaConstraints.h"
+#import <WebRTC/RTCMediaConstraints.h>
 
 #import "ARDSettingsModel+Private.h"
 #import "ARDSettingsStore.h"

examples/objcnativeapi/objc/NADViewController.mm

@@ -10,11 +10,14 @@
 
 #import "NADViewController.h"
 
-#import <WebRTC/RTCCameraPreviewView.h>
+#import "sdk/objc/base/RTCVideoRenderer.h"
-#import <WebRTC/RTCCameraVideoCapturer.h>
+#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h"
-#import <WebRTC/RTCEAGLVideoView.h>
+#if defined(RTC_SUPPORTS_METAL)
-#import <WebRTC/RTCMTLVideoView.h>
+#import "sdk/objc/components/renderer/metal/RTCMTLVideoView.h"  // nogncheck
-#import <WebRTC/RTCVideoRenderer.h>
+#endif
+#import "sdk/objc/components/renderer/opengl/RTCEAGLVideoView.h"
+#import "sdk/objc/helpers/RTCCameraPreviewView.h"
+
 #include <memory>
 
 #include "objccallclient.h"

examples/objcnativeapi/objc/objccallclient.mm

@@ -12,9 +12,10 @@
 
 #include <utility>
 
-#import <WebRTC/RTCCameraPreviewView.h>
+#import "sdk/objc/base/RTCVideoRenderer.h"
-#import <WebRTC/RTCVideoCodecFactory.h>
+#import "sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h"
-#import <WebRTC/RTCVideoRenderer.h>
+#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h"
+#import "sdk/objc/helpers/RTCCameraPreviewView.h"
 
 #include "absl/memory/memory.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -22,10 +23,10 @@
 #include "api/peerconnectioninterface.h"
 #include "media/engine/webrtcmediaengine.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "sdk/objc/Framework/Native/api/video_capturer.h"
+#include "sdk/objc/native/api/video_capturer.h"
-#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
+#include "sdk/objc/native/api/video_decoder_factory.h"
-#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
+#include "sdk/objc/native/api/video_encoder_factory.h"
-#include "sdk/objc/Framework/Native/api/video_renderer.h"
+#include "sdk/objc/native/api/video_renderer.h"
 
 namespace webrtc_examples {
 

modules/audio_device/BUILD.gn

@@ -69,8 +69,9 @@ if (rtc_include_internal_audio_device && is_ios) {
       "../../rtc_base:gtest_prod",
       "../../rtc_base:rtc_base",
       "../../rtc_base/system:fallthrough",
+      "../../sdk:audio_device",
       "../../sdk:audio_objc",
-      "../../sdk:common_objc",
+      "../../sdk:base_objc",
       "../../system_wrappers:metrics_api",
     ]
     if (!build_with_chromium && is_clang) {

modules/audio_device/DEPS

@@ -25,6 +25,9 @@ specific_include_rules = {
   "RTCAudioSessionDelegateAdapter\.h": [
     "+sdk/objc",
   ],
+  "RTCAudioSessionDelegateAdapter\.mm": [
+    "+sdk/objc",
+  ],
   "voice_processing_audio_unit\.mm": [
     "+sdk/objc",
   ],

modules/audio_device/ios/audio_device_ios.h

@@ -21,7 +21,7 @@
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/thread_checker.h"
-#include "sdk/objc/Framework/Headers/WebRTC/RTCMacros.h"
+#include "sdk/objc/base/RTCMacros.h"
 
 RTC_FWD_DECL_OBJC_CLASS(RTCAudioSessionDelegateAdapter);
 

modules/audio_device/ios/audio_device_ios.mm

@@ -25,14 +25,14 @@
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/timeutils.h"
-#include "sdk/objc/Framework/Classes/Common/helpers.h"
+#include "sdk/objc/native/src/audio/helpers.h"
 #include "system_wrappers/include/metrics.h"
 
-#import "WebRTC/RTCLogging.h"
 #import "modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h"
-#import "sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
+#import "sdk/objc/base/RTCLogging.h"
-#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
+#import "sdk/objc/components/audio/RTCAudioSession+Private.h"
-#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
+#import "sdk/objc/components/audio/RTCAudioSession.h"
+#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h"
 
 namespace webrtc {
 

modules/audio_device/ios/audio_device_unittest_ios.mm

@@ -31,8 +31,8 @@
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 
-#import "sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
+#import "sdk/objc/components/audio/RTCAudioSession+Private.h"
-#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
+#import "sdk/objc/components/audio/RTCAudioSession.h"
 
 using std::cout;
 using std::endl;

modules/audio_device/ios/objc/RTCAudioSession.h

@@ -8,4 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
+#import "sdk/objc/components/audio/RTCAudioSession.h"

modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h

@@ -8,4 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
+#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h"

modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
+#import "sdk/objc/components/audio/RTCAudioSession.h"
 
 namespace webrtc {
 class AudioSessionObserver;

modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm

@@ -12,7 +12,7 @@
 
 #include "modules/audio_device/ios/audio_session_observer.h"
 
-#import "WebRTC/RTCLogging.h"
+#import "sdk/objc/base/RTCLogging.h"
 
 @implementation RTCAudioSessionDelegateAdapter {
   webrtc::AudioSessionObserver *_observer;

modules/audio_device/ios/voice_processing_audio_unit.mm

@@ -14,8 +14,8 @@
 #include "rtc_base/system/fallthrough.h"
 #include "system_wrappers/include/metrics.h"
 
-#import "WebRTC/RTCLogging.h"
+#import "sdk/objc/base//RTCLogging.h"
-#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
+#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h"
 
 #if !defined(NDEBUG)
 static void LogStreamDescription(AudioStreamBasicDescription description) {

modules/desktop_capture/BUILD.gn

@@ -200,7 +200,7 @@ if (is_mac) {
       "../../rtc_base:rtc_base",
       "../../rtc_base:rtc_base_approved",
       "../../rtc_base/synchronization:rw_lock_wrapper",
-      "../../sdk:common_objc",
+      "../../sdk:helpers_objc",
       "//third_party/abseil-cpp/absl/memory",
     ]
     libs = [

modules/desktop_capture/mac/desktop_frame_cgimage.h

@@ -16,7 +16,7 @@
 #include <memory>
 
 #include "modules/desktop_capture/desktop_frame.h"
-#include "sdk/objc/Framework/Classes/Common/scoped_cftyperef.h"
+#include "sdk/objc/helpers/scoped_cftyperef.h"
 
 namespace webrtc {
 

modules/desktop_capture/mac/desktop_frame_iosurface.h

@@ -17,7 +17,7 @@
 #include <memory>
 
 #include "modules/desktop_capture/desktop_frame.h"
-#include "sdk/objc/Framework/Classes/Common/scoped_cftyperef.h"
+#include "sdk/objc/helpers/scoped_cftyperef.h"
 
 namespace webrtc {
 

modules/desktop_capture/mac/desktop_frame_provider.h

@@ -19,7 +19,7 @@
 
 #include "modules/desktop_capture/shared_desktop_frame.h"
 #include "rtc_base/thread_checker.h"
-#include "sdk/objc/Framework/Classes/Common/scoped_cftyperef.h"
+#include "sdk/objc/helpers/scoped_cftyperef.h"
 
 namespace webrtc {
 

modules/desktop_capture/mac/screen_capturer_mac.mm

@@ -18,7 +18,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/timeutils.h"
 #include "rtc_base/trace_event.h"
-#include "sdk/objc/Framework/Classes/Common/scoped_cftyperef.h"
+#include "sdk/objc/helpers/scoped_cftyperef.h"
 
 namespace webrtc {
 

modules/video_coding/BUILD.gn

@@ -536,7 +536,6 @@ if (rtc_include_tests) {
         "../../media:rtc_media_base",
         "../../modules:module_api",
         "../../rtc_base:rtc_base_approved",
-        "../../sdk:common_objc",
         "../../sdk:native_api",
         "../../sdk:peerconnectionfactory_base_objc",
         "../../sdk:videocodec_objc",

modules/video_coding/codecs/test/objc_codec_factory_helper.mm

@@ -10,9 +10,10 @@
 
 #include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"
 
-#import "WebRTC/RTCVideoCodecH264.h"
+#import "sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h"
-#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
+#import "sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h"
-#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
+#include "sdk/objc/native/api/video_decoder_factory.h"
+#include "sdk/objc/native/api/video_encoder_factory.h"
 
 namespace webrtc {
 namespace test {

sdk/objc/DEPS

@@ -1,12 +1,8 @@
 include_rules = [
-  "+WebRTC",
+  "+base",
-  "+Common",
+  "+components",
-  "+Video",
+  "+helpers",
-  "+Audio",
+  "+sdk",
-  "+UI",
-  "+PeerConnection",
-  "+VideoToolbox",
-  "+Metal",
   "+api",
   "+common_video/h264",
   "+common_video/include",
@@ -17,5 +13,6 @@ include_rules = [
   "+system_wrappers",
   "+modules/audio_device",
   "+modules/audio_processing",
+  "+native",
   "+third_party/libyuv",
 ]

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

@@ -8,19 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "helpers/NSString+StdString.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/RTCUIApplicationStatusObserver.h

@@ -8,18 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#if defined(WEBRTC_IOS)
+#import "helpers/RTCUIApplicationStatusObserver.h"
-
-#import <Foundation/Foundation.h>
-
-NS_EXTENSION_UNAVAILABLE_IOS("Application status not available in app extensions.")
-@interface RTCUIApplicationStatusObserver : NSObject
-
-+ (instancetype)sharedInstance;
-+ (void)prepareForUse;
-
-- (BOOL)isApplicationActive;
-
-@end
-
-#endif  // WEBRTC_IOS

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

@@ -9,108 +9,4 @@
  *
  */
 
-#ifndef WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_SCOPED_CFTYPEREF_H_
+#import "helpers/scoped_cftyperef.h"
-#define WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_SCOPED_CFTYPEREF_H_
-
-#include <CoreFoundation/CoreFoundation.h>
-namespace rtc {
-
-// RETAIN: ScopedTypeRef should retain the object when it takes
-// ownership.
-// ASSUME: Assume the object already has already been retained.
-// ScopedTypeRef takes over ownership.
-enum class RetainPolicy { RETAIN, ASSUME };
-
-namespace internal {
-template <typename T>
-struct CFTypeRefTraits {
-  static T InvalidValue() { return nullptr; }
-  static void Release(T ref) { CFRelease(ref); }
-  static T Retain(T ref) {
-    CFRetain(ref);
-    return ref;
-  }
-};
-
-template <typename T, typename Traits>
-class ScopedTypeRef {
- public:
-  ScopedTypeRef() : ptr_(Traits::InvalidValue()) {}
-  explicit ScopedTypeRef(T ptr) : ptr_(ptr) {}
-  ScopedTypeRef(T ptr, RetainPolicy policy) : ScopedTypeRef(ptr) {
-    if (ptr_ && policy == RetainPolicy::RETAIN)
-      Traits::Retain(ptr_);
-  }
-
-  ScopedTypeRef(const ScopedTypeRef<T, Traits>& rhs) : ptr_(rhs.ptr_) {
-    if (ptr_)
-      ptr_ = Traits::Retain(ptr_);
-  }
-
-  ~ScopedTypeRef() {
-    if (ptr_) {
-      Traits::Release(ptr_);
-    }
-  }
-
-  T get() const { return ptr_; }
-  T operator->() const { return ptr_; }
-  explicit operator bool() const { return ptr_; }
-
-  bool operator!() const { return !ptr_; }
-
-  ScopedTypeRef& operator=(const T& rhs) {
-    if (ptr_)
-      Traits::Release(ptr_);
-    ptr_ = rhs;
-    return *this;
-  }
-
-  ScopedTypeRef& operator=(const ScopedTypeRef<T, Traits>& rhs) {
-    reset(rhs.get(), RetainPolicy::RETAIN);
-    return *this;
-  }
-
-  // This is intended to take ownership of objects that are
-  // created by pass-by-pointer initializers.
-  T* InitializeInto() {
-    RTC_DCHECK(!ptr_);
-    return &ptr_;
-  }
-
-  void reset(T ptr, RetainPolicy policy = RetainPolicy::ASSUME) {
-    if (ptr && policy == RetainPolicy::RETAIN)
-      Traits::Retain(ptr);
-    if (ptr_)
-      Traits::Release(ptr_);
-    ptr_ = ptr;
-  }
-
-  T release() {
-    T temp = ptr_;
-    ptr_ = Traits::InvalidValue();
-    return temp;
-  }
-
- private:
-  T ptr_;
-};
-}  // namespace internal
-
-template <typename T>
-using ScopedCFTypeRef =
-    internal::ScopedTypeRef<T, internal::CFTypeRefTraits<T>>;
-
-template <typename T>
-static ScopedCFTypeRef<T> AdoptCF(T cftype) {
-  return ScopedCFTypeRef<T>(cftype, RetainPolicy::RETAIN);
-}
-
-template <typename T>
-static ScopedCFTypeRef<T> ScopedCF(T cftype) {
-  return ScopedCFTypeRef<T>(cftype);
-}
-
-}  // namespace rtc
-
-#endif  // WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_COMMON_SCOPED_CFTYPEREF_H_

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

@@ -8,21 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCConfiguration.h"
+#import "api/peerconnection/RTCConfiguration+Native.h"
-
-#include "api/peerconnectioninterface.h"
-
-NS_ASSUME_NONNULL_BEGIN
-
-@interface RTCConfiguration ()
-
-/** Optional TurnCustomizer.
- *  With this class one can modify outgoing TURN messages.
- *  The object passed in must remain valid until PeerConnection::Close() is
- * called.
- */
-@property(nonatomic, nullable) webrtc::TurnCustomizer* turnCustomizer;
-
-@end
-
-NS_ASSUME_NONNULL_END

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

@@ -1,83 +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 "WebRTC/RTCVideoCodec.h"
-
-#import "RTCVideoCodec+Private.h"
-
-#include "rtc_base/numerics/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_);
-    _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.SetTimestamp(_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/RTCPeerConnection+Native.h

@@ -8,27 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCPeerConnection.h"
+#import "api/peerconnection/RTCPeerConnection+Native.h"
-
-#include <memory>
-
-namespace rtc {
-class BitrateAllocationStrategy;
-}  // namespace rtc
-
-NS_ASSUME_NONNULL_BEGIN
-
-/**
- * This class extension exposes methods that work directly with injectable C++ components.
- */
-@interface RTCPeerConnection ()
-
-/** Sets current strategy. If not set default WebRTC allocator will be used. May be changed during
- *  an active session.
- */
-- (void)setBitrateAllocationStrategy:
-        (std::unique_ptr<rtc::BitrateAllocationStrategy>)bitrateAllocationStrategy;
-
-@end
-
-NS_ASSUME_NONNULL_END

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

@@ -8,47 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCPeerConnectionFactory.h"
+#import "api/peerconnection/RTCPeerConnectionFactory+Native.h"
-
-#include "rtc_base/scoped_ref_ptr.h"
-
-namespace webrtc {
-
-class AudioDeviceModule;
-class AudioEncoderFactory;
-class AudioDecoderFactory;
-class VideoEncoderFactory;
-class VideoDecoderFactory;
-class AudioProcessing;
-
-}  // namespace webrtc
-
-NS_ASSUME_NONNULL_BEGIN
-
-/**
- * This class extension exposes methods that work directly with injectable C++ components.
- */
-@interface RTCPeerConnectionFactory ()
-
-- (instancetype)initNative NS_DESIGNATED_INITIALIZER;
-
-/* Initializer used when WebRTC is compiled with no media support */
-- (instancetype)initWithNoMedia;
-
-/* 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:
-                            (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
-                        nativeVideoDecoderFactory:
-                            (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
-                                audioDeviceModule:
-                                    (nullable webrtc::AudioDeviceModule *)audioDeviceModule
-                            audioProcessingModule:
-                                (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
-
-@end
-
-NS_ASSUME_NONNULL_END

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

@@ -8,51 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCVideoCodec.h"
+#import "api/peerconnection/RTCEncodedImage+Private.h"
-
+#import "api/peerconnection/RTCRtpFragmentationHeader+Private.h"
-#import "WebRTC/RTCVideoCodecH264.h"
+#import "api/peerconnection/RTCVideoCodecInfo+Private.h"
-
+#import "api/peerconnection/RTCVideoEncoderSettings+Private.h"
-#include "api/video_codecs/sdp_video_format.h"
+#import "components/video_codec/RTCCodecSpecificInfoH264+Private.h"
-#include "common_video/include/video_frame.h"
-#include "media/base/codec.h"
-#include "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;
-- (webrtc::VideoCodec)nativeVideoCodec;
-
-@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)initWithNativeSdpVideoFormat:(webrtc::SdpVideoFormat)format;
-- (webrtc::SdpVideoFormat)nativeSdpVideoFormat;
-
-@end
-
-NS_ASSUME_NONNULL_END

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

@@ -1,167 +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 "WebRTC/RTCVideoCodec.h"
-
-#import "NSString+StdString.h"
-#import "RTCVideoCodec+Private.h"
-#if defined(WEBRTC_IOS)
-#import "UIDevice+H264Profile.h"
-#endif
-#import "WebRTC/RTCVideoCodecFactory.h"
-
-#include "media/base/mediaconstants.h"
-
-namespace {
-
-NSString *MaxSupportedProfileLevelConstrainedHigh();
-NSString *MaxSupportedProfileLevelConstrainedBaseline();
-
-}  // namespace
-
-NSString *const kRTCVideoCodecVp8Name = @(cricket::kVp8CodecName);
-NSString *const kRTCVideoCodecVp9Name = @(cricket::kVp9CodecName);
-NSString *const kRTCVideoCodecH264Name = @(cricket::kH264CodecName);
-NSString *const kRTCLevel31ConstrainedHigh = @"640c1f";
-NSString *const kRTCLevel31ConstrainedBaseline = @"42e01f";
-NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
-    MaxSupportedProfileLevelConstrainedHigh();
-NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
-    MaxSupportedProfileLevelConstrainedBaseline();
-
-namespace {
-
-#if defined(WEBRTC_IOS)
-
-using namespace webrtc::H264;
-
-NSString *MaxSupportedLevelForProfile(Profile profile) {
-  const absl::optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile];
-  if (profileLevelId && profileLevelId->profile >= profile) {
-    const absl::optional<std::string> profileString =
-        ProfileLevelIdToString(ProfileLevelId(profile, profileLevelId->level));
-    if (profileString) {
-      return [NSString stringForStdString:*profileString];
-    }
-  }
-  return nil;
-}
-#endif
-
-NSString *MaxSupportedProfileLevelConstrainedBaseline() {
-#if defined(WEBRTC_IOS)
-  NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedBaseline);
-  if (profile != nil) {
-    return profile;
-  }
-#endif
-  return kRTCLevel31ConstrainedBaseline;
-}
-
-NSString *MaxSupportedProfileLevelConstrainedHigh() {
-#if defined(WEBRTC_IOS)
-  NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedHigh);
-  if (profile != nil) {
-    return profile;
-  }
-#endif
-  return kRTCLevel31ConstrainedHigh;
-}
-
-}  // namespace
-
-@implementation RTCVideoCodecInfo
-
-@synthesize name = _name;
-@synthesize parameters = _parameters;
-
-- (instancetype)initWithName:(NSString *)name {
-  return [self initWithName:name parameters:nil];
-}
-
-- (instancetype)initWithName:(NSString *)name
-                  parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters {
-  if (self = [super init]) {
-    _name = name;
-    _parameters = (parameters ? parameters : @{});
-  }
-
-  return self;
-}
-
-- (instancetype)initWithNativeSdpVideoFormat:(webrtc::SdpVideoFormat)format {
-  NSMutableDictionary *params = [NSMutableDictionary dictionary];
-  for (auto it = format.parameters.begin(); it != format.parameters.end(); ++it) {
-    [params setObject:[NSString stringForStdString:it->second]
-               forKey:[NSString stringForStdString:it->first]];
-  }
-  return [self initWithName:[NSString stringForStdString:format.name] parameters:params];
-}
-
-- (BOOL)isEqualToCodecInfo:(RTCVideoCodecInfo *)info {
-  if (!info ||
-      ![self.name isEqualToString:info.name] ||
-      ![self.parameters isEqualToDictionary:info.parameters]) {
-    return NO;
-  }
-  return YES;
-}
-
-- (BOOL)isEqual:(id)object {
-  if (self == object)
-    return YES;
-  if (![object isKindOfClass:[self class]])
-    return NO;
-  return [self isEqualToCodecInfo:object];
-}
-
-- (NSUInteger)hash {
-  return [self.name hash] ^ [self.parameters hash];
-}
-
-- (webrtc::SdpVideoFormat)nativeSdpVideoFormat {
-  std::map<std::string, std::string> parameters;
-  for (NSString *paramKey in _parameters.allKeys) {
-    std::string key = [NSString stdStringForString:paramKey];
-    std::string value = [NSString stdStringForString:_parameters[paramKey]];
-    parameters[key] = value;
-  }
-
-  return webrtc::SdpVideoFormat([NSString stdStringForString:_name], parameters);
-}
-
-#pragma mark - NSCoding
-
-- (instancetype)initWithCoder:(NSCoder *)decoder {
-  return [self initWithName:[decoder decodeObjectForKey:@"name"]
-                 parameters:[decoder decodeObjectForKey:@"parameters"]];
-}
-
-- (void)encodeWithCoder:(NSCoder *)encoder {
-  [encoder encodeObject:_name forKey:@"name"];
-  [encoder encodeObject:_parameters forKey:@"parameters"];
-}
-
-@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/RTCVideoEncoderSettings.mm

@@ -1,66 +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 "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;
-}
-
-- (webrtc::VideoCodec)nativeVideoCodec {
-  webrtc::VideoCodec videoCodec;
-  videoCodec.width = _width;
-  videoCodec.height = _height;
-  videoCodec.startBitrate = _startBitrate;
-  videoCodec.maxBitrate = _maxBitrate;
-  videoCodec.minBitrate = _minBitrate;
-  videoCodec.targetBitrate = _targetBitrate;
-  videoCodec.maxBitrate = _maxBitrate;
-  videoCodec.qpMax = _qpMax;
-  videoCodec.mode = (webrtc::VideoCodecMode)_mode;
-
-  return videoCodec;
-}
-
-@end

sdk/objc/Framework/Classes/Video/RTCDefaultShader.h

@@ -8,16 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "WebRTC/RTCVideoViewShading.h"
+#import "components/renderer/opengl/RTCDefaultShader.h"
-
-NS_ASSUME_NONNULL_BEGIN
-
-/** Default RTCVideoViewShading that will be used in RTCNSGLVideoView and
- *  RTCEAGLVideoView if no external shader is specified. This shader will render
- *  the video in a rectangle without any color or geometric transformations.
- */
-@interface RTCDefaultShader : NSObject<RTCVideoViewShading>
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h

@@ -8,24 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <GLKit/GLKit.h>
+#import "components/renderer/opengl/RTCNV12TextureCache.h"
-
-@class RTCVideoFrame;
-
-NS_ASSUME_NONNULL_BEGIN
-
-@interface RTCNV12TextureCache : NSObject
-
-@property(nonatomic, readonly) GLuint yTexture;
-@property(nonatomic, readonly) GLuint uvTexture;
-
-- (instancetype)init NS_UNAVAILABLE;
-- (nullable instancetype)initWithContext:(EAGLContext *)context NS_DESIGNATED_INITIALIZER;
-
-- (BOOL)uploadFrameToTextures:(RTCVideoFrame *)frame;
-
-- (void)releaseTextures;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h

@@ -6,111 +6,6 @@
  *  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 SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_
+#import "components/video_codec/nalu_rewriter.h"
-#define SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_
-
-#include "modules/video_coding/codecs/h264/include/h264.h"
-
-#include <CoreMedia/CoreMedia.h>
-#include <vector>
-
-#include "common_video/h264/h264_common.h"
-#include "modules/include/module_common_types.h"
-#include "rtc_base/buffer.h"
-
-using webrtc::H264::NaluIndex;
-
-namespace webrtc {
-
-// Converts a sample buffer emitted from the VideoToolbox encoder into a buffer
-// suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer
-// needs to be in Annex B format. Data is written directly to |annexb_buffer|
-// and a new RTPFragmentationHeader is returned in |out_header|.
-bool H264CMSampleBufferToAnnexBBuffer(
-    CMSampleBufferRef avcc_sample_buffer,
-    bool is_keyframe,
-    rtc::Buffer* annexb_buffer,
-    std::unique_ptr<RTPFragmentationHeader>* out_header);
-
-// Converts a buffer received from RTP into a sample buffer suitable for the
-// VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
-// buffer is in avcc format.
-// If |is_keyframe| is true then |video_format| is ignored since the format will
-// be read from the buffer. Otherwise |video_format| must be provided.
-// Caller is responsible for releasing the created sample buffer.
-bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
-                                      size_t annexb_buffer_size,
-                                      CMVideoFormatDescriptionRef video_format,
-                                      CMSampleBufferRef* out_sample_buffer);
-
-// Returns a video format description created from the sps/pps information in
-// the Annex B buffer. If there is no such information, nullptr is returned.
-// The caller is responsible for releasing the description.
-CMVideoFormatDescriptionRef CreateVideoFormatDescription(
-    const uint8_t* annexb_buffer,
-    size_t annexb_buffer_size);
-
-// Helper class for reading NALUs from an RTP Annex B buffer.
-class AnnexBBufferReader final {
- public:
-  AnnexBBufferReader(const uint8_t* annexb_buffer, size_t length);
-  ~AnnexBBufferReader();
-  AnnexBBufferReader(const AnnexBBufferReader& other) = delete;
-  void operator=(const AnnexBBufferReader& other) = delete;
-
-  // Returns a pointer to the beginning of the next NALU slice without the
-  // header bytes and its length. Returns false if no more slices remain.
-  bool ReadNalu(const uint8_t** out_nalu, size_t* out_length);
-
-  // Returns the number of unread NALU bytes, including the size of the header.
-  // If the buffer has no remaining NALUs this will return zero.
-  size_t BytesRemaining() const;
-
-  // Reset the reader to start reading from the first NALU
-  void SeekToStart();
-
-  // Seek to the next position that holds a NALU of the desired type,
-  // or the end if no such NALU is found.
-  // Return true if a NALU of the desired type is found, false if we
-  // reached the end instead
-  bool SeekToNextNaluOfType(H264::NaluType type);
-
- private:
-  // Returns the the next offset that contains NALU data.
-  size_t FindNextNaluHeader(const uint8_t* start,
-                            size_t length,
-                            size_t offset) const;
-
-  const uint8_t* const start_;
-  std::vector<NaluIndex> offsets_;
-  std::vector<NaluIndex>::iterator offset_;
-  const size_t length_;
-};
-
-// Helper class for writing NALUs using avcc format into a buffer.
-class AvccBufferWriter final {
- public:
-  AvccBufferWriter(uint8_t* const avcc_buffer, size_t length);
-  ~AvccBufferWriter() {}
-  AvccBufferWriter(const AvccBufferWriter& other) = delete;
-  void operator=(const AvccBufferWriter& other) = delete;
-
-  // Writes the data slice into the buffer. Returns false if there isn't
-  // enough space left.
-  bool WriteNalu(const uint8_t* data, size_t data_size);
-
-  // Returns the unused bytes in the buffer.
-  size_t BytesRemaining() const;
-
- private:
-  uint8_t* const start_;
-  size_t offset_;
-  const size_t length_;
-};
-
-}  // namespace webrtc
-
-#endif  // SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_

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

@@ -8,242 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <AVFoundation/AVFoundation.h>
+#import "components/audio/RTCAudioSession.h"
-#import <Foundation/Foundation.h>
-
-#import <WebRTC/RTCMacros.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-extern NSString *const kRTCAudioSessionErrorDomain;
-/** Method that requires lock was called without lock. */
-extern NSInteger const kRTCAudioSessionErrorLockRequired;
-/** Unknown configuration error occurred. */
-extern NSInteger const kRTCAudioSessionErrorConfiguration;
-
-@class RTCAudioSession;
-@class RTCAudioSessionConfiguration;
-
-// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
-// from AVAudioSession and handle them before calling these delegate methods,
-// at which point applications can perform additional processing if required.
-RTC_EXPORT
-@protocol RTCAudioSessionDelegate <NSObject>
-
-@optional
-/** Called on a system notification thread when AVAudioSession starts an
- *  interruption event.
- */
-- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session;
-
-/** Called on a system notification thread when AVAudioSession ends an
- *  interruption event.
- */
-- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
-                   shouldResumeSession:(BOOL)shouldResumeSession;
-
-/** Called on a system notification thread when AVAudioSession changes the
- *  route.
- */
-- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
-                            reason:(AVAudioSessionRouteChangeReason)reason
-                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
-
-/** Called on a system notification thread when AVAudioSession media server
- *  terminates.
- */
-- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session;
-
-/** Called on a system notification thread when AVAudioSession media server
- *  restarts.
- */
-- (void)audioSessionMediaServerReset:(RTCAudioSession *)session;
-
-// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
-
-- (void)audioSession:(RTCAudioSession *)session didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
-
-/** Called on a WebRTC thread when the audio device is notified to begin
- *  playback or recording.
- */
-- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;
-
-/** Called on a WebRTC thread when the audio device is notified to stop
- *  playback or recording.
- */
-- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
-
-/** Called when the AVAudioSession output volume value changes. */
-- (void)audioSession:(RTCAudioSession *)audioSession didChangeOutputVolume:(float)outputVolume;
-
-/** Called when the audio device detects a playout glitch. The argument is the
- *  number of glitches detected so far in the current audio playout session.
- */
-- (void)audioSession:(RTCAudioSession *)audioSession
-    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
-
-/** Called when the audio session is about to change the active state.
- */
-- (void)audioSession:(RTCAudioSession *)audioSession willSetActive:(BOOL)active;
-
-/** Called after the audio session sucessfully changed the active state.
- */
-- (void)audioSession:(RTCAudioSession *)audioSession didSetActive:(BOOL)active;
-
-/** Called after the audio session failed to change the active state.
- */
-- (void)audioSession:(RTCAudioSession *)audioSession
-    failedToSetActive:(BOOL)active
-                error:(NSError *)error;
-
-@end
-
-/** This is a protocol used to inform RTCAudioSession when the audio session
- *  activation state has changed outside of RTCAudioSession. The current known use
- *  case of this is when CallKit activates the audio session for the application
- */
-RTC_EXPORT
-@protocol RTCAudioSessionActivationDelegate <NSObject>
-
-/** Called when the audio session is activated outside of the app by iOS. */
-- (void)audioSessionDidActivate:(AVAudioSession *)session;
-
-/** Called when the audio session is deactivated outside of the app by iOS. */
-- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
-
-@end
-
-/** Proxy class for AVAudioSession that adds a locking mechanism similar to
- *  AVCaptureDevice. This is used to that interleaving configurations between
- *  WebRTC and the application layer are avoided.
- *
- *  RTCAudioSession also coordinates activation so that the audio session is
- *  activated only once. See |setActive:error:|.
- */
-RTC_EXPORT
-@interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
-
-/** Convenience property to access the AVAudioSession singleton. Callers should
- *  not call setters on AVAudioSession directly, but other method invocations
- *  are fine.
- */
-@property(nonatomic, readonly) AVAudioSession *session;
-
-/** Our best guess at whether the session is active based on results of calls to
- *  AVAudioSession.
- */
-@property(nonatomic, readonly) BOOL isActive;
-/** Whether RTCAudioSession is currently locked for configuration. */
-@property(nonatomic, readonly) BOOL isLocked;
-
-/** If YES, WebRTC will not initialize the audio unit automatically when an
- *  audio track is ready for playout or recording. Instead, applications should
- *  call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
- *  as soon as an audio track is ready for playout or recording.
- */
-@property(nonatomic, assign) BOOL useManualAudio;
-
-/** This property is only effective if useManualAudio is YES.
- *  Represents permission for WebRTC to initialize the VoIP audio unit.
- *  When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
- *  stopped and uninitialized. This will stop incoming and outgoing audio.
- *  When set to YES, WebRTC will initialize and start the audio unit when it is
- *  needed (e.g. due to establishing an audio connection).
- *  This property was introduced to work around an issue where if an AVPlayer is
- *  playing audio while the VoIP audio unit is initialized, its audio would be
- *  either cut off completely or played at a reduced volume. By preventing
- *  the audio unit from being initialized until after the audio has completed,
- *  we are able to prevent the abrupt cutoff.
- */
-@property(nonatomic, assign) BOOL isAudioEnabled;
-
-// Proxy properties.
-@property(readonly) NSString *category;
-@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
-@property(readonly) NSString *mode;
-@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
-@property(readonly) AVAudioSessionRouteDescription *currentRoute;
-@property(readonly) NSInteger maximumInputNumberOfChannels;
-@property(readonly) NSInteger maximumOutputNumberOfChannels;
-@property(readonly) float inputGain;
-@property(readonly) BOOL inputGainSettable;
-@property(readonly) BOOL inputAvailable;
-@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
-@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
-@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
-@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
-@property(readonly) double sampleRate;
-@property(readonly) double preferredSampleRate;
-@property(readonly) NSInteger inputNumberOfChannels;
-@property(readonly) NSInteger outputNumberOfChannels;
-@property(readonly) float outputVolume;
-@property(readonly) NSTimeInterval inputLatency;
-@property(readonly) NSTimeInterval outputLatency;
-@property(readonly) NSTimeInterval IOBufferDuration;
-@property(readonly) NSTimeInterval preferredIOBufferDuration;
-
-/** Default constructor. */
-+ (instancetype)sharedInstance;
-- (instancetype)init NS_UNAVAILABLE;
-
-/** Adds a delegate, which is held weakly. */
-- (void)addDelegate:(id<RTCAudioSessionDelegate>)delegate;
-/** Removes an added delegate. */
-- (void)removeDelegate:(id<RTCAudioSessionDelegate>)delegate;
-
-/** Request exclusive access to the audio session for configuration. This call
- *  will block if the lock is held by another object.
- */
-- (void)lockForConfiguration;
-/** Relinquishes exclusive access to the audio session. */
-- (void)unlockForConfiguration;
-
-/** If |active|, activates the audio session if it isn't already active.
- *  Successful calls must be balanced with a setActive:NO when activation is no
- *  longer required. If not |active|, deactivates the audio session if one is
- *  active and this is the last balanced call. When deactivating, the
- *  AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
- *  AVAudioSession.
- */
-- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
-
-// The following methods are proxies for the associated methods on
-// AVAudioSession. |lockForConfiguration| must be called before using them
-// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
-
-- (BOOL)setCategory:(NSString *)category
-        withOptions:(AVAudioSessionCategoryOptions)options
-              error:(NSError **)outError;
-- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
-- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
-- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
-- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
-- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
-- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
-- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
-- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
-- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
-                     error:(NSError **)outError;
-- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
-                      error:(NSError **)outError;
-@end
-
-@interface RTCAudioSession (Configuration)
-
-/** Applies the configuration to the current session. Attempts to set all
- *  properties even if previous ones fail. Only the last error will be
- *  returned.
- *  |lockForConfiguration| must be called first.
- */
-- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration error:(NSError **)outError;
-
-/** Convenience method that calls both setConfiguration and setActive.
- *  |lockForConfiguration| must be called first.
- */
-- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
-                  active:(BOOL)active
-                   error:(NSError **)outError;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h

@@ -8,41 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <AVFoundation/AVFoundation.h>
+#import "components/audio/RTCAudioSessionConfiguration.h"
-#import <Foundation/Foundation.h>
-
-#import "WebRTC/RTCMacros.h"
-
-NS_ASSUME_NONNULL_BEGIN
-
-RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
-RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
-RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
-RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
-RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
-
-// Struct to hold configuration values.
-RTC_EXPORT
-@interface RTCAudioSessionConfiguration : NSObject
-
-@property(nonatomic, strong) NSString *category;
-@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
-@property(nonatomic, strong) NSString *mode;
-@property(nonatomic, assign) double sampleRate;
-@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
-@property(nonatomic, assign) NSInteger inputNumberOfChannels;
-@property(nonatomic, assign) NSInteger outputNumberOfChannels;
-
-/** Initializes configuration to defaults. */
-- (instancetype)init NS_DESIGNATED_INITIALIZER;
-
-/** Returns the current configuration of the audio session. */
-+ (instancetype)currentConfiguration;
-/** Returns the configuration that WebRTC needs. */
-+ (instancetype)webRTCConfiguration;
-/** Provide a way to override the default configuration. */
-+ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h

@@ -8,25 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCAudioSource.h"
-
-#import <WebRTC/RTCMacros.h>
-#import <WebRTC/RTCMediaSource.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-RTC_EXPORT
-@interface RTCAudioSource : RTCMediaSource
-
-- (instancetype)init NS_UNAVAILABLE;
-
-// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
-// [0, 10].
-// Temporary fix to be able to modify volume of remote audio tracks.
-// TODO(kthelgason): Property stays here temporarily until a proper volume-api
-// is available on the surface exposed by webrtc.
-@property(nonatomic, assign) double volume;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h

@@ -8,21 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <WebRTC/RTCMacros.h>
+#import "api/peerconnection/RTCAudioTrack.h"
-#import <WebRTC/RTCMediaStreamTrack.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-@class RTCAudioSource;
-
-RTC_EXPORT
-@interface RTCAudioTrack : RTCMediaStreamTrack
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/** The audio source for this audio track. */
-@property(nonatomic, readonly) RTCAudioSource *source;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCCVPixelBuffer.h

@@ -0,0 +1,11 @@
+/*
+ *  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 "components/video_frame_buffer/RTCCVPixelBuffer.h"

sdk/objc/Framework/Headers/WebRTC/RTCCallbackLogger.h

@@ -8,28 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/logging/RTCCallbackLogger.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

sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h

@@ -8,23 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "helpers/RTCCameraPreviewView.h"
-#import <UIKit/UIKit.h>
-
-#import <WebRTC/RTCMacros.h>
-
-@class AVCaptureSession;
-
-/** RTCCameraPreviewView is a view that renders local video from an
- *  AVCaptureSession.
- */
-RTC_EXPORT
-@interface RTCCameraPreviewView : UIView
-
-/** The capture session being rendered in the view. Capture session
- *  is assigned to AVCaptureVideoPreviewLayer async in the same
- *  queue that the AVCaptureSession is started/stopped.
- */
-@property(nonatomic, strong) AVCaptureSession* captureSession;
-
-@end

sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h

@@ -8,49 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <AVFoundation/AVFoundation.h>
+#import "components/capturer/RTCCameraVideoCapturer.h"
-#import <Foundation/Foundation.h>
-
-#import <WebRTC/RTCMacros.h>
-#import <WebRTC/RTCVideoCapturer.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-RTC_EXPORT
-// Camera capture that implements RTCVideoCapturer. Delivers frames to a RTCVideoCapturerDelegate
-// (usually RTCVideoSource).
-NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
-@interface RTCCameraVideoCapturer : RTCVideoCapturer
-
-// Capture session that is used for capturing. Valid from initialization to dealloc.
-@property(readonly, nonatomic) AVCaptureSession *captureSession;
-
-// Returns list of available capture devices that support video capture.
-+ (NSArray<AVCaptureDevice *> *)captureDevices;
-// Returns list of formats that are supported by this class for this device.
-+ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
-
-// 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;
-// Stops the capture session asynchronously.
-- (void)stopCapture;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCCertificate.h

@@ -8,37 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCCertificate.h"
-
-#import <WebRTC/RTCMacros.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-RTC_EXPORT
-@interface RTCCertificate : NSObject <NSCopying>
-
-/** Private key in PEM. */
-@property(nonatomic, readonly, copy) NSString *private_key;
-
-/** Public key in an x509 cert encoded in PEM. */
-@property(nonatomic, readonly, copy) NSString *certificate;
-
-/**
- * Initialize an RTCCertificate with PEM strings for private_key and certificate.
- */
-- (instancetype)initWithPrivateKey:(NSString *)private_key
-                       certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/** Generate a new certificate for 're' use.
- *
- *  Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
- *  provided.
- *  - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
- */
-+ (nullable RTCCertificate *)generateCertificateWithParams:(NSDictionary *)params;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h

@@ -8,167 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCConfiguration.h"
-
-#import <WebRTC/RTCCertificate.h>
-#import <WebRTC/RTCMacros.h>
-
-@class RTCIceServer;
-@class RTCIntervalRange;
-
-/**
- * Represents the ice transport policy. This exposes the same states in C++,
- * which include one more state than what exists in the W3C spec.
- */
-typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
-  RTCIceTransportPolicyNone,
-  RTCIceTransportPolicyRelay,
-  RTCIceTransportPolicyNoHost,
-  RTCIceTransportPolicyAll
-};
-
-/** Represents the bundle policy. */
-typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
-  RTCBundlePolicyBalanced,
-  RTCBundlePolicyMaxCompat,
-  RTCBundlePolicyMaxBundle
-};
-
-/** Represents the rtcp mux policy. */
-typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
-
-/** Represents the tcp candidate policy. */
-typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
-  RTCTcpCandidatePolicyEnabled,
-  RTCTcpCandidatePolicyDisabled
-};
-
-/** Represents the candidate network policy. */
-typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
-  RTCCandidateNetworkPolicyAll,
-  RTCCandidateNetworkPolicyLowCost
-};
-
-/** Represents the continual gathering policy. */
-typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
-  RTCContinualGatheringPolicyGatherOnce,
-  RTCContinualGatheringPolicyGatherContinually
-};
-
-/** Represents the encryption key type. */
-typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
-  RTCEncryptionKeyTypeRSA,
-  RTCEncryptionKeyTypeECDSA,
-};
-
-/** Represents the chosen SDP semantics for the RTCPeerConnection. */
-typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
-  RTCSdpSemanticsPlanB,
-  RTCSdpSemanticsUnifiedPlan,
-};
-
-NS_ASSUME_NONNULL_BEGIN
-RTC_EXPORT
-@interface RTCConfiguration : NSObject
-
-/** An array of Ice Servers available to be used by ICE. */
-@property(nonatomic, copy) NSArray<RTCIceServer *> *iceServers;
-
-/** An RTCCertificate for 're' use. */
-@property(nonatomic, nullable) RTCCertificate *certificate;
-
-/** Which candidates the ICE agent is allowed to use. The W3C calls it
- * |iceTransportPolicy|, while in C++ it is called |type|. */
-@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
-
-/** The media-bundling policy to use when gathering ICE candidates. */
-@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
-
-/** The rtcp-mux policy to use when gathering ICE candidates. */
-@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
-@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
-@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
-@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
-
-/** By default, the PeerConnection will use a limited number of IPv6 network
- *  interfaces, in order to avoid too many ICE candidate pairs being created
- *  and delaying ICE completion.
- *
- *  Can be set to INT_MAX to effectively disable the limit.
- */
-@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;
-@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
-
-/** Key type used to generate SSL identity. Default is ECDSA. */
-@property(nonatomic, assign) RTCEncryptionKeyType keyType;
-
-/** ICE candidate pool size as defined in JSEP. Default is 0. */
-@property(nonatomic, assign) int iceCandidatePoolSize;
-
-/** Prune turn ports on the same network to the same turn server.
- *  Default is NO.
- */
-@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
-
-/** If set to YES, this means the ICE transport should presume TURN-to-TURN
- *  candidate pairs will succeed, even before a binding response is received.
- */
-@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
-
-/** If set to non-nil, controls the minimal interval between consecutive ICE
- *  check packets.
- */
-@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
-
-/** ICE Periodic Regathering
- *  If set, WebRTC will periodically create and propose candidates without
- *  starting a new ICE generation. The regathering happens continuously with
- *  interval specified in milliseconds by the uniform distribution [a, b].
- */
-@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 wish to send multiple audio/video streams and need to stay
- *  interoperable with legacy WebRTC implementations or use legacy APIs,
- *  specify PlanB.
- *
- *  For all other users, specify UnifiedPlan.
- */
-@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
-
-/** Actively reset the SRTP parameters when the DTLS transports underneath are
- *  changed after offer/answer negotiation. This is only intended to be a
- *  workaround for crbug.com/835958
- */
-@property(nonatomic, assign) BOOL activeResetSrtpParams;
-
-- (instancetype)init;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h

@@ -8,123 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <AvailabilityMacros.h>
+#import "api/peerconnection/RTCDataChannel.h"
-#import <Foundation/Foundation.h>
-
-#import <WebRTC/RTCMacros.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-RTC_EXPORT
-@interface RTCDataBuffer : NSObject
-
-/** NSData representation of the underlying buffer. */
-@property(nonatomic, readonly) NSData *data;
-
-/** Indicates whether |data| contains UTF-8 or binary data. */
-@property(nonatomic, readonly) BOOL isBinary;
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/**
- * Initialize an RTCDataBuffer from NSData. |isBinary| indicates whether |data|
- * contains UTF-8 or binary data.
- */
-- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
-
-@end
-
-@class RTCDataChannel;
-RTC_EXPORT
-@protocol RTCDataChannelDelegate <NSObject>
-
-/** The data channel state changed. */
-- (void)dataChannelDidChangeState:(RTCDataChannel *)dataChannel;
-
-/** The data channel successfully received a data buffer. */
-- (void)dataChannel:(RTCDataChannel *)dataChannel
-    didReceiveMessageWithBuffer:(RTCDataBuffer *)buffer;
-
-@optional
-/** The data channel's |bufferedAmount| changed. */
-- (void)dataChannel:(RTCDataChannel *)dataChannel didChangeBufferedAmount:(uint64_t)amount;
-
-@end
-
-/** Represents the state of the data channel. */
-typedef NS_ENUM(NSInteger, RTCDataChannelState) {
-  RTCDataChannelStateConnecting,
-  RTCDataChannelStateOpen,
-  RTCDataChannelStateClosing,
-  RTCDataChannelStateClosed,
-};
-
-RTC_EXPORT
-@interface RTCDataChannel : NSObject
-
-/**
- * A label that can be used to distinguish this data channel from other data
- * channel objects.
- */
-@property(nonatomic, readonly) NSString *label;
-
-/** Whether the data channel can send messages in unreliable mode. */
-@property(nonatomic, readonly) BOOL isReliable DEPRECATED_ATTRIBUTE;
-
-/** Returns whether this data channel is ordered or not. */
-@property(nonatomic, readonly) BOOL isOrdered;
-
-/** Deprecated. Use maxPacketLifeTime. */
-@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
-
-/**
- * The length of the time window (in milliseconds) during which transmissions
- * and retransmissions may occur in unreliable mode.
- */
-@property(nonatomic, readonly) uint16_t maxPacketLifeTime;
-
-/**
- * The maximum number of retransmissions that are attempted in unreliable mode.
- */
-@property(nonatomic, readonly) uint16_t maxRetransmits;
-
-/**
- * The name of the sub-protocol used with this data channel, if any. Otherwise
- * this returns an empty string.
- */
-@property(nonatomic, readonly) NSString *protocol;
-
-/**
- * Returns whether this data channel was negotiated by the application or not.
- */
-@property(nonatomic, readonly) BOOL isNegotiated;
-
-/** Deprecated. Use channelId. */
-@property(nonatomic, readonly) NSInteger streamId DEPRECATED_ATTRIBUTE;
-
-/** The identifier for this data channel. */
-@property(nonatomic, readonly) int channelId;
-
-/** The state of the data channel. */
-@property(nonatomic, readonly) RTCDataChannelState readyState;
-
-/**
- * The number of bytes of application data that have been queued using
- * |sendData:| but that have not yet been transmitted to the network.
- */
-@property(nonatomic, readonly) uint64_t bufferedAmount;
-
-/** The delegate for this data channel. */
-@property(nonatomic, weak) id<RTCDataChannelDelegate> delegate;
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/** Closes the data channel. */
-- (void)close;
-
-/** Attempt to send |data| on this data channel's underlying data transport. */
-- (BOOL)sendData:(RTCDataBuffer *)data;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h

@@ -8,45 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <AvailabilityMacros.h>
+#import "api/peerconnection/RTCDataChannelConfiguration.h"
-#import <Foundation/Foundation.h>
-
-#import <WebRTC/RTCMacros.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-RTC_EXPORT
-@interface RTCDataChannelConfiguration : NSObject
-
-/** Set to YES if ordered delivery is required. */
-@property(nonatomic, assign) BOOL isOrdered;
-
-/** Deprecated. Use maxPacketLifeTime. */
-@property(nonatomic, assign) NSInteger maxRetransmitTimeMs DEPRECATED_ATTRIBUTE;
-
-/**
- * Max period in milliseconds in which retransmissions will be sent. After this
- * time, no more retransmissions will be sent. -1 if unset.
- */
-@property(nonatomic, assign) int maxPacketLifeTime;
-
-/** The max number of retransmissions. -1 if unset. */
-@property(nonatomic, assign) int maxRetransmits;
-
-/** Set to YES if the channel has been externally negotiated and we do not send
- * an in-band signalling in the form of an "open" message.
- */
-@property(nonatomic, assign) BOOL isNegotiated;
-
-/** Deprecated. Use channelId. */
-@property(nonatomic, assign) int streamId DEPRECATED_ATTRIBUTE;
-
-/** The id of the data channel. */
-@property(nonatomic, assign) int channelId;
-
-/** Set by the application and opaque to the WebRTC implementation. */
-@property(nonatomic) NSString* protocol;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCDefaultVideoDecoderFactory.h

@@ -0,0 +1,11 @@
+/*
+ *  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 "components/video_codec/RTCDefaultVideoDecoderFactory.h"

sdk/objc/Framework/Headers/WebRTC/RTCDefaultVideoEncoderFactory.h

@@ -0,0 +1,11 @@
+/*
+ *  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 "components/video_codec/RTCDefaultVideoEncoderFactory.h"

sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h

@@ -8,37 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "helpers/RTCDispatcher.h"
-
-#import <WebRTC/RTCMacros.h>
-
-typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
-  // Main dispatcher queue.
-  RTCDispatcherTypeMain,
-  // Used for starting/stopping AVCaptureSession, and assigning
-  // capture session to AVCaptureVideoPreviewLayer.
-  RTCDispatcherTypeCaptureSession,
-  // Used for operations on AVAudioSession.
-  RTCDispatcherTypeAudioSession,
-};
-
-/** Dispatcher that asynchronously dispatches blocks to a specific
- *  shared dispatch queue.
- */
-RTC_EXPORT
-@interface RTCDispatcher : NSObject
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/** Dispatch the block asynchronously on the queue for dispatchType.
- *  @param dispatchType The queue type to dispatch on.
- *  @param block The block to dispatch asynchronously.
- */
-+ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
-
-/** Returns YES if run on queue for the dispatchType otherwise NO.
- *  Useful for asserting that a method is run on a correct queue.
- */
-+ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType;
-
-@end

sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h

@@ -8,63 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCDtmfSender.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

sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h

@@ -8,37 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "components/renderer/opengl/RTCEAGLVideoView.h"
-#import <UIKit/UIKit.h>
-
-#import <WebRTC/RTCMacros.h>
-#import <WebRTC/RTCVideoRenderer.h>
-#import <WebRTC/RTCVideoViewShading.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-@class RTCEAGLVideoView;
-
-RTC_EXPORT
-@protocol RTCEAGLVideoViewDelegate <RTCVideoViewDelegate>
-@end
-
-/**
- * RTCEAGLVideoView is an RTCVideoRenderer which renders video frames in its
- * bounds using OpenGLES 2.0 or OpenGLES 3.0.
- */
-RTC_EXPORT
-NS_EXTENSION_UNAVAILABLE_IOS("Rendering not available in app extensions.")
-@interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
-
-@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
-
-- (instancetype)initWithFrame:(CGRect)frame
-                       shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
-
-- (instancetype)initWithCoder:(NSCoder *)aDecoder
-                       shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h

@@ -8,39 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCFieldTrials.h"
-
-#import <WebRTC/RTCMacros.h>
-
-/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
-RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweKey;
-RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweForVideoKey;
-RTC_EXTERN NSString * const kRTCFieldTrialAudioForceNoTWCCKey;
-RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey;
-RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey;
-RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
-RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
-RTC_EXTERN NSString * const kRTCFieldTrialImprovedBitrateEstimateKey;
-RTC_EXTERN NSString * const kRTCFieldTrialH264HighProfileKey;
-RTC_EXTERN NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey;
-
-/** The valid value for field trials above. */
-RTC_EXTERN NSString * const kRTCFieldTrialEnabledValue;
-
-/** Use a string returned by RTCFieldTrialMedianSlopeFilterValue as the value. */
-RTC_EXTERN NSString * const kRTCFieldTrialMedianSlopeFilterKey;
-RTC_EXTERN NSString *RTCFieldTrialMedianSlopeFilterValue(
-    size_t windowSize, double thresholdGain);
-
-/** Use a string returned by RTCFieldTrialTrendlineFilterValue as the value. */
-RTC_EXTERN NSString * const kRTCFieldTrialTrendlineFilterKey;
-/** Returns a valid value for kRTCFieldTrialTrendlineFilterKey. */
-RTC_EXTERN NSString *RTCFieldTrialTrendlineFilterValue(
-    size_t windowSize, double smoothingCoeff, double thresholdGain);
-
-/** Initialize field trials using a dictionary mapping field trial keys to their values. See above
- *  for valid keys and values.
- *  Must be called before any other call into WebRTC. See:
- *  webrtc/system_wrappers/include/field_trial_default.h
- */
-RTC_EXTERN void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials);

sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h

@@ -8,67 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCFileLogger.h"
-
-#import <WebRTC/RTCMacros.h>
-
-typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
-  RTCFileLoggerSeverityVerbose,
-  RTCFileLoggerSeverityInfo,
-  RTCFileLoggerSeverityWarning,
-  RTCFileLoggerSeverityError
-};
-
-typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
-  RTCFileLoggerTypeCall,
-  RTCFileLoggerTypeApp,
-};
-
-NS_ASSUME_NONNULL_BEGIN
-
-// This class intercepts WebRTC logs and saves them to a file. The file size
-// will not exceed the given maximum bytesize. When the maximum bytesize is
-// reached, logs are rotated according to the rotationType specified.
-// For kRTCFileLoggerTypeCall, logs from the beginning and the end
-// are preserved while the middle section is overwritten instead.
-// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
-// This class is not threadsafe.
-RTC_EXPORT
-@interface RTCFileLogger : NSObject
-
-// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
-@property(nonatomic, assign) RTCFileLoggerSeverity severity;
-
-// The rotation type for this file logger. The default is
-// kRTCFileLoggerTypeCall.
-@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
-
-// Disables buffering disk writes. Should be set before |start|. Buffering
-// is enabled by default for performance.
-@property(nonatomic, assign) BOOL shouldDisableBuffering;
-
-// Default constructor provides default settings for dir path, file size and
-// rotation type.
-- (instancetype)init;
-
-// Create file logger with default rotation type.
-- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
-
-- (instancetype)initWithDirPath:(NSString *)dirPath
-                    maxFileSize:(NSUInteger)maxFileSize
-                   rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
-
-// Starts writing WebRTC logs to disk if not already started. Overwrites any
-// existing file(s).
-- (void)start;
-
-// Stops writing WebRTC logs to disk. This method is also called on dealloc.
-- (void)stop;
-
-// Returns the current contents of the logs, or nil if start has been called
-// without a stop.
-- (nullable NSData *)logData;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h

@@ -8,43 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "components/capturer/RTCFileVideoCapturer.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

sdk/objc/Framework/Headers/WebRTC/RTCH264ProfileLevelId.h

@@ -0,0 +1,11 @@
+/*
+ *  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 "components/video_codec/RTCH264ProfileLevelId.h"

sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h

@@ -8,42 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCIceCandidate.h"
-
-#import <WebRTC/RTCMacros.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-RTC_EXPORT
-@interface RTCIceCandidate : NSObject
-
-/**
- * If present, the identifier of the "media stream identification" for the media
- * component this candidate is associated with.
- */
-@property(nonatomic, readonly, nullable) NSString *sdpMid;
-
-/**
- * The index (starting at zero) of the media description this candidate is
- * associated with in the SDP.
- */
-@property(nonatomic, readonly) int sdpMLineIndex;
-
-/** The SDP string for this candidate. */
-@property(nonatomic, readonly) NSString *sdp;
-
-/** The URL of the ICE server which this candidate is gathered from. */
-@property(nonatomic, readonly, nullable) NSString *serverUrl;
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/**
- * Initialize an RTCIceCandidate from SDP.
- */
-- (instancetype)initWithSdp:(NSString *)sdp
-              sdpMLineIndex:(int)sdpMLineIndex
-                     sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h

@@ -8,107 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCIceServer.h"
-
-#import <WebRTC/RTCMacros.h>
-
-typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
-  RTCTlsCertPolicySecure,
-  RTCTlsCertPolicyInsecureNoCheck
-};
-
-NS_ASSUME_NONNULL_BEGIN
-
-RTC_EXPORT
-@interface RTCIceServer : NSObject
-
-/** URI(s) for this server represented as NSStrings. */
-@property(nonatomic, readonly) NSArray<NSString *> *urlStrings;
-
-/** Username to use if this RTCIceServer object is a TURN server. */
-@property(nonatomic, readonly, nullable) NSString *username;
-
-/** Credential to use if this RTCIceServer object is a TURN server. */
-@property(nonatomic, readonly, nullable) NSString *credential;
-
-/**
- * TLS certificate policy to use if this RTCIceServer object is a TURN server.
- */
-@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
-
-/**
-  If the URIs in |urls| only contain IP addresses, this field can be used
-  to indicate the hostname, which may be necessary for TLS (using the SNI
-  extension). If |urls| itself contains the hostname, this isn't necessary.
- */
-@property(nonatomic, readonly, nullable) NSString *hostname;
-
-/** List of protocols to be used in the TLS ALPN extension. */
-@property(nonatomic, readonly) NSArray<NSString *> *tlsAlpnProtocols;
-
-/**
-  List elliptic curves to be used in the TLS elliptic curves extension.
-  Only curve names supported by OpenSSL should be used (eg. "P-256","X25519").
-  */
-@property(nonatomic, readonly) NSArray<NSString *> *tlsEllipticCurves;
-
-- (nonnull instancetype)init NS_UNAVAILABLE;
-
-/** Convenience initializer for a server with no authentication (e.g. STUN). */
-- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
-
-/**
- * Initialize an RTCIceServer with its associated URLs, optional username,
- * optional credential, and credentialType.
- */
-- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
-                          username:(nullable NSString *)username
-                        credential:(nullable NSString *)credential;
-
-/**
- * Initialize an RTCIceServer with its associated URLs, optional username,
- * optional credential, and TLS cert policy.
- */
-- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
-                          username:(nullable NSString *)username
-                        credential:(nullable NSString *)credential
-                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy;
-
-/**
- * Initialize an RTCIceServer with its associated URLs, optional username,
- * optional credential, TLS cert policy and hostname.
- */
-- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
-                          username:(nullable NSString *)username
-                        credential:(nullable NSString *)credential
-                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
-                          hostname:(nullable NSString *)hostname;
-
-/**
- * Initialize an RTCIceServer with its associated URLs, optional username,
- * optional credential, TLS cert policy, hostname and ALPN protocols.
- */
-- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
-                          username:(nullable NSString *)username
-                        credential:(nullable NSString *)credential
-                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
-                          hostname:(nullable NSString *)hostname
-                  tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols;
-
-/**
- * Initialize an RTCIceServer with its associated URLs, optional username,
- * optional credential, TLS cert policy, hostname, ALPN protocols and
- * elliptic curves.
- */
-- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
-                          username:(nullable NSString *)username
-                        credential:(nullable NSString *)credential
-                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
-                          hostname:(nullable NSString *)hostname
-                  tlsAlpnProtocols:(nullable NSArray<NSString *> *)tlsAlpnProtocols
-                 tlsEllipticCurves:(nullable NSArray<NSString *> *)tlsEllipticCurves
-    NS_DESIGNATED_INITIALIZER;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h

@@ -8,18 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCIntervalRange.h"
-
-NS_ASSUME_NONNULL_BEGIN
-
-@interface RTCIntervalRange : NSObject
-
-@property(nonatomic, readonly) NSInteger min;
-@property(nonatomic, readonly) NSInteger max;
-
-- (instancetype)init;
-- (instancetype)initWithMin:(NSInteger)min max:(NSInteger)max NS_DESIGNATED_INITIALIZER;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h

@@ -8,30 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCLegacyStatsReport.h"
-
-#import <WebRTC/RTCMacros.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-/** This does not currently conform to the spec. */
-RTC_EXPORT
-@interface RTCLegacyStatsReport : NSObject
-
-/** Time since 1970-01-01T00:00:00Z in milliseconds. */
-@property(nonatomic, readonly) CFTimeInterval timestamp;
-
-/** The type of stats held by this object. */
-@property(nonatomic, readonly) NSString *type;
-
-/** The identifier for this object. */
-@property(nonatomic, readonly) NSString *reportId;
-
-/** A dictionary holding the actual stats. */
-@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
-
-- (instancetype)init NS_UNAVAILABLE;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCLogging.h

@@ -8,59 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "base/RTCLogging.h"
-
-#import <WebRTC/RTCMacros.h>
-
-// Subset of rtc::LoggingSeverity.
-typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
-  RTCLoggingSeverityVerbose,
-  RTCLoggingSeverityInfo,
-  RTCLoggingSeverityWarning,
-  RTCLoggingSeverityError,
-};
-
-// 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);
-
-// Wrapper for rtc::LogMessage::LogToDebug.
-// Sets the minimum severity to be logged to console.
-RTC_EXTERN void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
-
-// Returns the filename with the path prefix removed.
-RTC_EXTERN NSString* RTCFileName(const char* filePath);
-
-// Some convenience macros.
-
-#define RTCLogString(format, ...)                                           \
-  [NSString stringWithFormat:@"(%@:%d %s): " format, RTCFileName(__FILE__), \
-                             __LINE__, __FUNCTION__, ##__VA_ARGS__]
-
-#define RTCLogFormat(severity, format, ...)                     \
-  do {                                                          \
-    NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
-    RTCLogEx(severity, log_string);                             \
-  } while (false)
-
-#define RTCLogVerbose(format, ...) \
-  RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
-
-#define RTCLogInfo(format, ...) \
-  RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
-
-#define RTCLogWarning(format, ...) \
-  RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
-
-#define RTCLogError(format, ...) \
-  RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
-
-#if !defined(NDEBUG)
-#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
-#else
-#define RTCLogDebug(format, ...) \
-  do {                           \
-  } while (false)
-#endif
-
-#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)

sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h

@@ -8,13 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Cocoa/Cocoa.h>
+#import "components/renderer/metal/RTCMTLNSVideoView.h"
-#import "WebRTC/RTCVideoRenderer.h"
-
-NS_AVAILABLE_MAC(10.11)
-@interface RTCMTLNSVideoView : NSView <RTCVideoRenderer>
-
-@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
-
-+ (BOOL)isMetalAvailable;
-@end

sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h

@@ -8,44 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "components/renderer/metal/RTCMTLVideoView.h"
-
-#import "WebRTC/RTCVideoFrame.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>
-
-@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
-
-@property(nonatomic) UIViewContentMode videoContentMode;
-
-/** @abstract Enables/disables rendering.
- */
-@property(nonatomic, getter=isEnabled) BOOL enabled;
-
-/** @abstract Wrapped RTCVideoRotation, or nil.
- */
-@property(nonatomic, nullable) NSValue* rotationOverride;
-
-@end
-
-NS_ASSUME_NONNULL_END

sdk/objc/Framework/Headers/WebRTC/RTCMacros.h

@@ -8,21 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_
+#import "base/RTCMacros.h"
-#define SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_
-
-#define RTC_EXPORT __attribute__((visibility("default")))
-
-#if defined(__cplusplus)
-#define RTC_EXTERN extern "C" RTC_EXPORT
-#else
-#define RTC_EXTERN extern RTC_EXPORT
-#endif
-
-#ifdef __OBJC__
-#define RTC_FWD_DECL_OBJC_CLASS(classname) @class classname
-#else
-#define RTC_FWD_DECL_OBJC_CLASS(classname) typedef struct objc_object classname
-#endif
-
-#endif  // SDK_OBJC_FRAMEWORK_HEADERS_WEBRTC_RTCMACROS_H_

sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h

@@ -8,47 +8,4 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
+#import "api/peerconnection/RTCMediaConstraints.h"
-
-#import <WebRTC/RTCMacros.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-/** Constraint keys for media sources. */
-RTC_EXTERN NSString *const kRTCMediaConstraintsMinAspectRatio;
-RTC_EXTERN NSString *const kRTCMediaConstraintsMaxAspectRatio;
-RTC_EXTERN NSString *const kRTCMediaConstraintsMaxWidth;
-RTC_EXTERN NSString *const kRTCMediaConstraintsMinWidth;
-RTC_EXTERN NSString *const kRTCMediaConstraintsMaxHeight;
-RTC_EXTERN NSString *const kRTCMediaConstraintsMinHeight;
-RTC_EXTERN NSString *const kRTCMediaConstraintsMaxFrameRate;
-RTC_EXTERN NSString *const kRTCMediaConstraintsMinFrameRate;
-/** The value for this key should be a base64 encoded string containing
- *  the data from the serialized configuration proto.
- */
-RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
-
-/** Constraint keys for generating offers and answers. */
-RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
-RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
-RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
-RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
-
-/** Constraint values for Boolean parameters. */
-RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
-RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
-
-RTC_EXPORT
-@interface RTCMediaConstraints : NSObject
-
-- (instancetype)init NS_UNAVAILABLE;
-
-/** Initialize with mandatory and/or optional constraints. */
-- (instancetype)
-    initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
-             optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
-    NS_DESIGNATED_INITIALIZER;
-
-@end
-
-NS_ASSUME_NONNULL_END