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Merge commit 'upstream-main' into master

Browse Source 
Bug: 261600888
Test: none, build files to be updated in follow up cl
Change-Id: Ib520938290c6bbdee4a9f73b6419b6c947a96ec4
This commit is contained in:
Jorge E. Moreira
2022-12-06 16:34:41 -08:00
parent 28cc9bf0d7 e093c481bf
commit 16476ad1d2
5393 changed files with 541103 additions and 211666 deletions
Download Patch File Download Diff File Expand all files Collapse all files

15
sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h
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/*
* 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 "api/peerconnection/RTCEncodedImage+Private.h"
#import "api/peerconnection/RTCRtpFragmentationHeader+Private.h"
#import "api/peerconnection/RTCVideoCodecInfo+Private.h"
#import "api/peerconnection/RTCVideoEncoderSettings+Private.h"
#import "components/video_codec/RTCCodecSpecificInfoH264+Private.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "components/audio/RTCAudioSession.h"

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sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "components/audio/RTCAudioSessionConfiguration.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
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/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCAudioSource.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCAudioTrack.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCCVPixelBuffer.h
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/*
* 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"

11
sdk/objc/Framework/Headers/WebRTC/RTCCallbackLogger.h
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/*
* 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 "api/logging/RTCCallbackLogger.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
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/*
* Copyright 2015 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "helpers/RTCCameraPreviewView.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
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/*
* 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/capturer/RTCCameraVideoCapturer.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCCertificate.h
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/*
* 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 "api/peerconnection/RTCCertificate.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCConfiguration.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCDataChannel.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCDataChannelConfiguration.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCDefaultVideoEncoderFactory.h
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/*
* 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"

11
sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "helpers/RTCDispatcher.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
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/*
* 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 "api/peerconnection/RTCDtmfSender.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "components/renderer/opengl/RTCEAGLVideoView.h"

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sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h
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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCFieldTrials.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCFileLogger.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
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/*
* 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/capturer/RTCFileVideoCapturer.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCH264ProfileLevelId.h
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/*
* 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"

11
sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCIceCandidate.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCIceServer.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCLegacyStatsReport.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "base/RTCLogging.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
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/*
* 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/renderer/metal/RTCMTLNSVideoView.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
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/*
* 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/renderer/metal/RTCMTLVideoView.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMacros.h
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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "base/RTCMacros.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCMediaConstraints.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
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/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCMediaSource.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCMediaStream.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCMediaStreamTrack.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
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/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCMetrics.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
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/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCMetricsSampleInfo.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "components/renderer/opengl/RTCNSGLVideoView.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
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@ -1,11 +0,0 @@
/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCPeerConnection.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
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@ -1,11 +0,0 @@
/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCPeerConnectionFactory.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
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@ -1,11 +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 "api/peerconnection/RTCPeerConnectionFactoryOptions.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h
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@ -1,11 +0,0 @@
/*
* 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 "api/peerconnection/RTCRtcpParameters.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
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@ -1,11 +0,0 @@
/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCRtpCodecParameters.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
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@ -1,11 +0,0 @@
/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCRtpEncodingParameters.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCRtpHeaderExtension.h
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@ -1,11 +0,0 @@
/*
* 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 "api/peerconnection/RTCRtpHeaderExtension.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
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@ -1,11 +0,0 @@
/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCRtpParameters.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
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@ -1,11 +0,0 @@
/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCRtpReceiver.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
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@ -1,11 +0,0 @@
/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCRtpSender.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h
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@ -1,11 +0,0 @@
/*
* 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 "api/peerconnection/RTCRtpTransceiver.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h
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@ -1,11 +0,0 @@
/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCSSLAdapter.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
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@ -1,11 +0,0 @@
/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCSessionDescription.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCTracing.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
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@ -1,11 +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 "base/RTCVideoCapturer.h"

20
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
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@ -1,20 +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 "api/video_codec/RTCVideoCodecConstants.h"
#import "base/RTCCodecSpecificInfo.h"
#import "base/RTCEncodedImage.h"
#import "base/RTCRtpFragmentationHeader.h"
#import "base/RTCVideoCodecInfo.h"
#import "base/RTCVideoDecoder.h"
#import "base/RTCVideoEncoder.h"
#import "base/RTCVideoEncoderQpThresholds.h"
#import "base/RTCVideoEncoderSettings.h"
#import "components/video_codec/RTCH264ProfileLevelId.h"

16
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
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@ -1,16 +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 "components/video_codec/RTCCodecSpecificInfoH264.h"
#import "components/video_codec/RTCH264ProfileLevelId.h"
#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
#import "components/video_codec/RTCVideoDecoderH264.h"
#import "components/video_codec/RTCVideoEncoderFactoryH264.h"
#import "components/video_codec/RTCVideoEncoderH264.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecInfo.h
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@ -1,11 +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 "base/RTCVideoCodecInfo.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
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@ -1,11 +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 "api/video_codec/RTCVideoDecoderVP8.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
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@ -1,11 +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 "api/video_codec/RTCVideoDecoderVP9.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
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@ -1,11 +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 "api/video_codec/RTCVideoEncoderVP8.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
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@ -1,11 +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 "api/video_codec/RTCVideoEncoderVP9.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
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@ -1,11 +0,0 @@
/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "base/RTCVideoFrame.h"

18
sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
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@ -1,18 +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 "api/video_frame_buffer/RTCNativeI420Buffer.h"
#import "api/video_frame_buffer/RTCNativeMutableI420Buffer.h"
#import "base/RTCI420Buffer.h"
#import "base/RTCMutableI420Buffer.h"
#import "base/RTCMutableYUVPlanarBuffer.h"
#import "base/RTCVideoFrameBuffer.h"
#import "base/RTCYUVPlanarBuffer.h"
#import "components/video_frame_buffer/RTCCVPixelBuffer.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
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@ -1,11 +0,0 @@
/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "base/RTCVideoRenderer.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
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@ -1,11 +0,0 @@
/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCVideoSource.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
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@ -1,11 +0,0 @@
/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "api/peerconnection/RTCVideoTrack.h"

11
sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
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@ -1,11 +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 "components/renderer/opengl/RTCVideoViewShading.h"

11
sdk/objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h
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@ -1,11 +0,0 @@
/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "helpers/UIDevice+RTCDevice.h"

27
sdk/objc/api/logging/RTCCallbackLogger.mm
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@ -10,20 +10,29 @@
#import "RTCCallbackLogger.h"
#import "helpers/NSString+StdString.h"
#include <memory>
#include "absl/strings/string_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/log_sinks.h"
#include "rtc_base/logging.h"
class CallbackLogSink : public rtc::LogSink {
namespace {
class CallbackLogSink final : public rtc::LogSink {
public:
CallbackLogSink(RTCCallbackLoggerMessageHandler callbackHandler)
: callback_handler_(callbackHandler) {}
void OnLogMessage(const std::string &message) override {
void OnLogMessage(const std::string& message) override {
OnLogMessage(absl::string_view(message));
}
void OnLogMessage(absl::string_view message) override {
if (callback_handler_) {
callback_handler_([NSString stringWithUTF8String:message.c_str()]);
callback_handler_([NSString stringForAbslStringView:message]);
}
}
@ -31,17 +40,21 @@ class CallbackLogSink : public rtc::LogSink {
RTCCallbackLoggerMessageHandler callback_handler_;
};
class CallbackWithSeverityLogSink : public rtc::LogSink {
class CallbackWithSeverityLogSink final : public rtc::LogSink {
public:
CallbackWithSeverityLogSink(RTCCallbackLoggerMessageAndSeverityHandler callbackHandler)
: callback_handler_(callbackHandler) {}
void OnLogMessage(const std::string& message) override { RTC_NOTREACHED(); }
void OnLogMessage(const std::string& message) override { RTC_DCHECK_NOTREACHED(); }
void OnLogMessage(const std::string& message, rtc::LoggingSeverity severity) override {
OnLogMessage(absl::string_view(message), severity);
}
void OnLogMessage(absl::string_view message, rtc::LoggingSeverity severity) override {
if (callback_handler_) {
RTCLoggingSeverity loggingSeverity = NativeSeverityToObjcSeverity(severity);
callback_handler_([NSString stringWithUTF8String:message.c_str()], loggingSeverity);
callback_handler_([NSString stringForAbslStringView:message], loggingSeverity);
}
}
@ -64,6 +77,8 @@ class CallbackWithSeverityLogSink : public rtc::LogSink {
RTCCallbackLoggerMessageAndSeverityHandler callback_handler_;
};
}
@implementation RTC_OBJC_TYPE (RTCCallbackLogger) {
BOOL _hasStarted;
std::unique_ptr<rtc::LogSink> _logSink;

2
sdk/objc/api/peerconnection/RTCAudioSource.h
View File

@ -20,7 +20,7 @@ RTC_OBJC_EXPORT
- (instancetype)init NS_UNAVAILABLE;
// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
// 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

2
sdk/objc/api/peerconnection/RTCAudioSource.mm
View File

@ -35,7 +35,7 @@
- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type {
RTC_NOTREACHED();
RTC_DCHECK_NOTREACHED();
return nil;
}

10
sdk/objc/api/peerconnection/RTCAudioTrack.mm
View File

@ -30,7 +30,7 @@
std::string nativeId = [NSString stdStringForString:trackId];
rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource);
factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource.get());
if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
_source = source;
}
@ -48,11 +48,10 @@
- (RTC_OBJC_TYPE(RTCAudioSource) *)source {
if (!_source) {
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
self.nativeAudioTrack->GetSource();
rtc::scoped_refptr<webrtc::AudioSourceInterface> source(self.nativeAudioTrack->GetSource());
if (source) {
_source = [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self.factory
nativeAudioSource:source.get()];
nativeAudioSource:source];
}
}
return _source;
@ -61,7 +60,8 @@
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeAudioTrack {
return static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get());
return rtc::scoped_refptr<webrtc::AudioTrackInterface>(
static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get()));
}
@end

99
sdk/objc/api/peerconnection/RTCConfiguration.h
View File

@ -63,6 +63,7 @@ typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
/** Represents the chosen SDP semantics for the RTCPeerConnection. */
typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
// TODO(https://crbug.com/webrtc/13528): Remove support for Plan B.
RTCSdpSemanticsPlanB,
RTCSdpSemanticsUnifiedPlan,
};
@ -84,7 +85,7 @@ RTC_OBJC_EXPORT
@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate;
/** Which candidates the ICE agent is allowed to use. The W3C calls it
* |iceTransportPolicy|, while in C++ it is called |type|. */
* `iceTransportPolicy`, while in C++ it is called `type`. */
@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
/** The media-bundling policy to use when gathering ICE candidates. */
@ -96,11 +97,6 @@ RTC_OBJC_EXPORT
@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
/** If set to YES, don't gather IPv6 ICE candidates.
* Default is NO.
*/
@property(nonatomic, assign) BOOL disableIPV6;
/** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
* Only intended to be used on specific devices. Certain phones disable IPv6
* when the screen is turned off and it would be better to just disable the
@ -144,7 +140,7 @@ RTC_OBJC_EXPORT
*/
@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
/* This flag is only effective when |continualGatheringPolicy| is
/* This flag is only effective when `continualGatheringPolicy` is
* RTCContinualGatheringPolicyGatherContinually.
*
* If YES, after the ICE transport type is changed such that new types of
@ -161,27 +157,24 @@ RTC_OBJC_EXPORT
*/
@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
/** 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.
/**
* Configure the SDP semantics used by this PeerConnection. By default, this
* is RTCSdpSemanticsUnifiedPlan which is compliant to the WebRTC 1.0
* specification. It is possible to overrwite this to the deprecated
* RTCSdpSemanticsPlanB SDP format, but note that RTCSdpSemanticsPlanB will be
* deleted at some future date, see https://crbug.com/webrtc/13528.
*
* 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.
* RTCSdpSemanticsUnifiedPlan 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.
*
* 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.
* RTCSdpSemanticsPlanB 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.
*/
@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
@ -204,6 +197,12 @@ RTC_OBJC_EXPORT
*/
@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
/**
* An optional string that will be attached to the TURN_ALLOCATE_REQUEST which
* which can be used to correlate client logs with backend logs.
*/
@property(nonatomic, nullable, copy) NSString *turnLoggingId;
/**
* Time interval between audio RTCP reports.
*/
@ -214,6 +213,54 @@ RTC_OBJC_EXPORT
*/
@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
/**
* Allow implicit rollback of local description when remote description
* conflicts with local description.
* See: https://w3c.github.io/webrtc-pc/#dom-peerconnection-setremotedescription
*/
@property(nonatomic, assign) BOOL enableImplicitRollback;
/**
* Control if "a=extmap-allow-mixed" is included in the offer.
* See: https://www.chromestatus.com/feature/6269234631933952
*/
@property(nonatomic, assign) BOOL offerExtmapAllowMixed;
/**
* Defines the interval applied to ALL candidate pairs
* when ICE is strongly connected, and it overrides the
* default value of this interval in the ICE implementation;
*/
@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalStrongConnectivity;
/**
* Defines the counterpart for ALL pairs when ICE is
* weakly connected, and it overrides the default value of
* this interval in the ICE implementation
*/
@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalWeakConnectivity;
/**
* The min time period for which a candidate pair must wait for response to
* connectivity checks before it becomes unwritable. This parameter
* overrides the default value in the ICE implementation if set.
*/
@property(nonatomic, copy, nullable) NSNumber *iceUnwritableTimeout;
/**
* The min number of connectivity checks that a candidate pair must sent
* without receiving response before it becomes unwritable. This parameter
* overrides the default value in the ICE implementation if set.
*/
@property(nonatomic, copy, nullable) NSNumber *iceUnwritableMinChecks;
/**
* The min time period for which a candidate pair must wait for response to
* connectivity checks it becomes inactive. This parameter overrides the
* default value in the ICE implementation if set.
*/
@property(nonatomic, copy, nullable) NSNumber *iceInactiveTimeout;
- (instancetype)init;
@end

69
sdk/objc/api/peerconnection/RTCConfiguration.mm
View File

@ -17,6 +17,7 @@
#import "RTCIceServer+Private.h"
#import "base/RTCLogging.h"
#include "rtc_base/checks.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/ssl_identity.h"
@ -31,7 +32,6 @@
@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
@synthesize candidateNetworkPolicy = _candidateNetworkPolicy;
@synthesize continualGatheringPolicy = _continualGatheringPolicy;
@synthesize disableIPV6 = _disableIPV6;
@synthesize disableIPV6OnWiFi = _disableIPV6OnWiFi;
@synthesize maxIPv6Networks = _maxIPv6Networks;
@synthesize disableLinkLocalNetworks = _disableLinkLocalNetworks;
@ -53,12 +53,21 @@
@synthesize activeResetSrtpParams = _activeResetSrtpParams;
@synthesize allowCodecSwitching = _allowCodecSwitching;
@synthesize cryptoOptions = _cryptoOptions;
@synthesize turnLoggingId = _turnLoggingId;
@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
@synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
@synthesize enableImplicitRollback = _enableImplicitRollback;
@synthesize offerExtmapAllowMixed = _offerExtmapAllowMixed;
@synthesize iceCheckIntervalStrongConnectivity = _iceCheckIntervalStrongConnectivity;
@synthesize iceCheckIntervalWeakConnectivity = _iceCheckIntervalWeakConnectivity;
@synthesize iceUnwritableTimeout = _iceUnwritableTimeout;
@synthesize iceUnwritableMinChecks = _iceUnwritableMinChecks;
@synthesize iceInactiveTimeout = _iceInactiveTimeout;
- (instancetype)init {
// Copy defaults.
webrtc::PeerConnectionInterface::RTCConfiguration config;
config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
return [self initWithNativeConfiguration:config];
}
@ -93,9 +102,7 @@
candidateNetworkPolicyForNativePolicy:config.candidate_network_policy];
webrtc::PeerConnectionInterface::ContinualGatheringPolicy nativePolicy =
config.continual_gathering_policy;
_continualGatheringPolicy =
[[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
_disableIPV6 = config.disable_ipv6;
_continualGatheringPolicy = [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
_disableIPV6OnWiFi = config.disable_ipv6_on_wifi;
_maxIPv6Networks = config.max_ipv6_networks;
_disableLinkLocalNetworks = config.disable_link_local_networks;
@ -129,9 +136,28 @@
sframeRequireFrameEncryption:config.crypto_options->sframe
.require_frame_encryption];
}
_turnLoggingId = [NSString stringWithUTF8String:config.turn_logging_id.c_str()];
_rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
_rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
_allowCodecSwitching = config.allow_codec_switching.value_or(false);
_enableImplicitRollback = config.enable_implicit_rollback;
_offerExtmapAllowMixed = config.offer_extmap_allow_mixed;
_iceCheckIntervalStrongConnectivity =
config.ice_check_interval_strong_connectivity.has_value() ?
[NSNumber numberWithInt:*config.ice_check_interval_strong_connectivity] :
nil;
_iceCheckIntervalWeakConnectivity = config.ice_check_interval_weak_connectivity.has_value() ?
[NSNumber numberWithInt:*config.ice_check_interval_weak_connectivity] :
nil;
_iceUnwritableTimeout = config.ice_unwritable_timeout.has_value() ?
[NSNumber numberWithInt:*config.ice_unwritable_timeout] :
nil;
_iceUnwritableMinChecks = config.ice_unwritable_min_checks.has_value() ?
[NSNumber numberWithInt:*config.ice_unwritable_min_checks] :
nil;
_iceInactiveTimeout = config.ice_inactive_timeout.has_value() ?
[NSNumber numberWithInt:*config.ice_inactive_timeout] :
nil;
}
return self;
}
@ -139,7 +165,7 @@
- (NSString *)description {
static NSString *formatString = @"RTC_OBJC_TYPE(RTCConfiguration): "
@"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n"
@"%d\n%@\n%d\n%d\n%d\n%d\n%d\n%@\n}\n";
@"%d\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n}\n";
return [NSString
stringWithFormat:formatString,
@ -161,11 +187,11 @@
_shouldSurfaceIceCandidatesOnIceTransportTypeChanged,
_iceCheckMinInterval,
_disableLinkLocalNetworks,
_disableIPV6,
_disableIPV6OnWiFi,
_maxIPv6Networks,
_activeResetSrtpParams,
_enableDscp];
_enableDscp,
_enableImplicitRollback];
}
#pragma mark - Private
@ -190,9 +216,8 @@
[[self class] nativeTcpCandidatePolicyForPolicy:_tcpCandidatePolicy];
nativeConfig->candidate_network_policy = [[self class]
nativeCandidateNetworkPolicyForPolicy:_candidateNetworkPolicy];
nativeConfig->continual_gathering_policy = [[self class]
nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
nativeConfig->disable_ipv6 = _disableIPV6;
nativeConfig->continual_gathering_policy =
[[self class] nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
nativeConfig->disable_ipv6_on_wifi = _disableIPV6OnWiFi;
nativeConfig->max_ipv6_networks = _maxIPv6Networks;
nativeConfig->disable_link_local_networks = _disableLinkLocalNetworks;
@ -258,9 +283,29 @@
_cryptoOptions.sframeRequireFrameEncryption ? true : false;
nativeConfig->crypto_options = absl::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
}
nativeConfig->turn_logging_id = [_turnLoggingId UTF8String];
nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
nativeConfig->allow_codec_switching = _allowCodecSwitching;
nativeConfig->enable_implicit_rollback = _enableImplicitRollback;
nativeConfig->offer_extmap_allow_mixed = _offerExtmapAllowMixed;
if (_iceCheckIntervalStrongConnectivity != nil) {
nativeConfig->ice_check_interval_strong_connectivity =
absl::optional<int>(_iceCheckIntervalStrongConnectivity.intValue);
}
if (_iceCheckIntervalWeakConnectivity != nil) {
nativeConfig->ice_check_interval_weak_connectivity =
absl::optional<int>(_iceCheckIntervalWeakConnectivity.intValue);
}
if (_iceUnwritableTimeout != nil) {
nativeConfig->ice_unwritable_timeout = absl::optional<int>(_iceUnwritableTimeout.intValue);
}
if (_iceUnwritableMinChecks != nil) {
nativeConfig->ice_unwritable_min_checks = absl::optional<int>(_iceUnwritableMinChecks.intValue);
}
if (_iceInactiveTimeout != nil) {
nativeConfig->ice_inactive_timeout = absl::optional<int>(_iceInactiveTimeout.intValue);
}
return nativeConfig.release();
}
@ -472,7 +517,7 @@
+ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics {
switch (sdpSemantics) {
case RTCSdpSemanticsPlanB:
return webrtc::SdpSemantics::kPlanB;
return webrtc::SdpSemantics::kPlanB_DEPRECATED;
case RTCSdpSemanticsUnifiedPlan:
return webrtc::SdpSemantics::kUnifiedPlan;
}
@ -480,7 +525,7 @@
+ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics {
switch (sdpSemantics) {
case webrtc::SdpSemantics::kPlanB:
case webrtc::SdpSemantics::kPlanB_DEPRECATED:
return RTCSdpSemanticsPlanB;
case webrtc::SdpSemantics::kUnifiedPlan:
return RTCSdpSemanticsUnifiedPlan;

2
sdk/objc/api/peerconnection/RTCCryptoOptions.h
View File

@ -28,7 +28,7 @@ RTC_OBJC_EXPORT
@property(nonatomic, assign) BOOL srtpEnableGcmCryptoSuites;
/**
* If set to true, the (potentially insecure) crypto cipher
* SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
* kSrtpAes128CmSha1_32 will be included in the list of supported ciphers
* during negotiation. It will only be used if both peers support it and no
* other ciphers get preferred.
*/

10
sdk/objc/api/peerconnection/RTCDataChannel.h
View File

@ -21,13 +21,13 @@ RTC_OBJC_EXPORT
/** NSData representation of the underlying buffer. */
@property(nonatomic, readonly) NSData *data;
/** Indicates whether |data| contains UTF-8 or binary 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|
* Initialize an RTCDataBuffer from NSData. `isBinary` indicates whether `data`
* contains UTF-8 or binary data.
*/
- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
@ -47,7 +47,7 @@ RTC_OBJC_EXPORT
didReceiveMessageWithBuffer:(RTC_OBJC_TYPE(RTCDataBuffer) *)buffer;
@optional
/** The data channel's |bufferedAmount| changed. */
/** The data channel's `bufferedAmount` changed. */
- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
didChangeBufferedAmount:(uint64_t)amount;
@ -112,7 +112,7 @@ RTC_OBJC_EXPORT
/**
* The number of bytes of application data that have been queued using
* |sendData:| but that have not yet been transmitted to the network.
* `sendData:` but that have not yet been transmitted to the network.
*/
@property(nonatomic, readonly) uint64_t bufferedAmount;
@ -124,7 +124,7 @@ RTC_OBJC_EXPORT
/** Closes the data channel. */
- (void)close;
/** Attempt to send |data| on this data channel's underlying data transport. */
/** Attempt to send `data` on this data channel's underlying data transport. */
- (BOOL)sendData:(RTC_OBJC_TYPE(RTCDataBuffer) *)data;
@end

5
sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
View File

@ -13,14 +13,13 @@
#import <objc/runtime.h>
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/ref_counted_object.h"
namespace {
// An implementation of EncodedImageBufferInterface that doesn't perform any copies.
class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
public:
static rtc::scoped_refptr<ObjCEncodedImageBuffer> Create(NSData *data) {
return new rtc::RefCountedObject<ObjCEncodedImageBuffer>(data);
return rtc::make_ref_counted<ObjCEncodedImageBuffer>(data);
}
const uint8_t *data() const override { return static_cast<const uint8_t *>(data_.bytes); }
// TODO(bugs.webrtc.org/9378): delete this non-const data method.
@ -92,7 +91,6 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
self.encodeFinishMs = encodedImage.timing_.encode_finish_ms;
self.frameType = static_cast<RTCFrameType>(encodedImage._frameType);
self.rotation = static_cast<RTCVideoRotation>(encodedImage.rotation_);
self.completeFrame = encodedImage._completeFrame;
self.qp = @(encodedImage.qp_);
self.contentType = (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
RTCVideoContentTypeScreenshare :
@ -121,7 +119,6 @@ class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
encodedImage.timing_.encode_finish_ms = self.encodeFinishMs;
encodedImage._frameType = webrtc::VideoFrameType(self.frameType);
encodedImage.rotation_ = webrtc::VideoRotation(self.rotation);
encodedImage._completeFrame = self.completeFrame;
encodedImage.qp_ = self.qp ? self.qp.intValue : -1;
encodedImage.content_type_ = (self.contentType == RTCVideoContentTypeScreenshare) ?
webrtc::VideoContentType::SCREENSHARE :

6
sdk/objc/api/peerconnection/RTCFieldTrials.h
View File

@ -13,14 +13,12 @@
#import "RTCMacros.h"
/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweKey;
RTC_EXTERN NSString * const kRTCFieldTrialAudioForceNoTWCCKey;
RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey;
RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey;
RTC_EXTERN NSString *const kRTCFieldTrialAudioForceABWENoTWCCKey;
RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
RTC_EXTERN NSString * const kRTCFieldTrialH264HighProfileKey;
RTC_EXTERN NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey;
RTC_EXTERN NSString *const kRTCFieldTrialUseNWPathMonitor;
/** The valid value for field trials above. */
RTC_EXTERN NSString * const kRTCFieldTrialEnabledValue;

19
sdk/objc/api/peerconnection/RTCFieldTrials.mm
View File

@ -16,18 +16,18 @@
#include "system_wrappers/include/field_trial.h"
NSString * const kRTCFieldTrialAudioSendSideBweKey = @"WebRTC-Audio-SendSideBwe";
NSString * const kRTCFieldTrialAudioForceNoTWCCKey = @"WebRTC-Audio-ForceNoTWCC";
NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey = @"WebRTC-Audio-ABWENoTWCC";
NSString * const kRTCFieldTrialSendSideBweWithOverheadKey = @"WebRTC-SendSideBwe-WithOverhead";
NSString *const kRTCFieldTrialAudioForceABWENoTWCCKey = @"WebRTC-Audio-ABWENoTWCC";
NSString * const kRTCFieldTrialFlexFec03AdvertisedKey = @"WebRTC-FlexFEC-03-Advertised";
NSString * const kRTCFieldTrialFlexFec03Key = @"WebRTC-FlexFEC-03";
NSString * const kRTCFieldTrialH264HighProfileKey = @"WebRTC-H264HighProfile";
NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey =
@"WebRTC-Audio-MinimizeResamplingOnMobile";
NSString *const kRTCFieldTrialUseNWPathMonitor = @"WebRTC-Network-UseNWPathMonitor";
NSString * const kRTCFieldTrialEnabledValue = @"Enabled";
static std::unique_ptr<char[]> gFieldTrialInitString;
// InitFieldTrialsFromString stores the char*, so the char array must outlive
// the application.
static char *gFieldTrialInitString = nullptr;
void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials) {
if (!fieldTrials) {
@ -42,12 +42,15 @@ void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTria
[fieldTrialInitString appendString:fieldTrialEntry];
}
size_t len = fieldTrialInitString.length + 1;
gFieldTrialInitString.reset(new char[len]);
if (![fieldTrialInitString getCString:gFieldTrialInitString.get()
if (gFieldTrialInitString != nullptr) {
delete[] gFieldTrialInitString;
}
gFieldTrialInitString = new char[len];
if (![fieldTrialInitString getCString:gFieldTrialInitString
maxLength:len
encoding:NSUTF8StringEncoding]) {
RTCLogError(@"Failed to convert field trial string.");
return;
}
webrtc::field_trial::InitFieldTrialsFromString(gFieldTrialInitString.get());
webrtc::field_trial::InitFieldTrialsFromString(gFieldTrialInitString);
}

2
sdk/objc/api/peerconnection/RTCFileLogger.h
View File

@ -43,7 +43,7 @@ RTC_OBJC_EXPORT
// kRTCFileLoggerTypeCall.
@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
// Disables buffering disk writes. Should be set before |start|. Buffering
// Disables buffering disk writes. Should be set before `start`. Buffering
// is enabled by default for performance.
@property(nonatomic, assign) BOOL shouldDisableBuffering;

26
sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent+Private.h Normal file
View File

@ -0,0 +1,26 @@
/*
* Copyright (c) 2021 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 "RTCIceCandidateErrorEvent.h"
#include <string>
NS_ASSUME_NONNULL_BEGIN
@interface RTC_OBJC_TYPE (RTCIceCandidateErrorEvent)
()
- (instancetype)initWithAddress : (const std::string&)address port : (const int)port url
: (const std::string&)url errorCode : (const int)errorCode errorText
: (const std::string&)errorText;
@end
NS_ASSUME_NONNULL_END

42
sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.h Normal file
View File

@ -0,0 +1,42 @@
/*
* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import <Foundation/Foundation.h>
#import "RTCMacros.h"
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCIceCandidateErrorEvent) : NSObject
/** The local IP address used to communicate with the STUN or TURN server. */
@property(nonatomic, readonly) NSString *address;
/** The port used to communicate with the STUN or TURN server. */
@property(nonatomic, readonly) int port;
/** The STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred. */
@property(nonatomic, readonly) NSString *url;
/** The numeric STUN error code returned by the STUN or TURN server. If no host candidate can reach
* the server, errorCode will be set to the value 701 which is outside the STUN error code range.
* This error is only fired once per server URL while in the RTCIceGatheringState of "gathering". */
@property(nonatomic, readonly) int errorCode;
/** The STUN reason text returned by the STUN or TURN server. If the server could not be reached,
* errorText will be set to an implementation-specific value providing details about the error. */
@property(nonatomic, readonly) NSString *errorText;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

42
sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.mm Normal file
View File

@ -0,0 +1,42 @@
/*
* Copyright (c) 2021 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 "RTCIceCandidateErrorEvent+Private.h"
#import "helpers/NSString+StdString.h"
@implementation RTC_OBJC_TYPE (RTCIceCandidateErrorEvent)
@synthesize address = _address;
@synthesize port = _port;
@synthesize url = _url;
@synthesize errorCode = _errorCode;
@synthesize errorText = _errorText;
- (instancetype)init {
return [super init];
}
- (instancetype)initWithAddress:(const std::string&)address
port:(const int)port
url:(const std::string&)url
errorCode:(const int)errorCode
errorText:(const std::string&)errorText {
if (self = [self init]) {
_address = [NSString stringForStdString:address];
_port = port;
_url = [NSString stringForStdString:url];
_errorCode = errorCode;
_errorText = [NSString stringForStdString:errorText];
}
return self;
}
@end

4
sdk/objc/api/peerconnection/RTCIceServer.h
View File

@ -37,9 +37,9 @@ RTC_OBJC_EXPORT
@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
/**
If the URIs in |urls| only contain IP addresses, this field can be used
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.
extension). If `urls` itself contains the hostname, this isn't necessary.
*/
@property(nonatomic, readonly, nullable) NSString *hostname;

2
sdk/objc/api/peerconnection/RTCLegacyStatsReport+Private.h
View File

@ -10,7 +10,7 @@
#import "RTCLegacyStatsReport.h"
#include "api/stats_types.h"
#include "api/legacy_stats_types.h"
NS_ASSUME_NONNULL_BEGIN

49
sdk/objc/api/peerconnection/RTCMediaStream.mm
View File

@ -10,8 +10,6 @@
#import "RTCMediaStream+Private.h"
#include <vector>
#import "RTCAudioTrack+Private.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCPeerConnectionFactory+Private.h"
@ -20,8 +18,9 @@
@implementation RTC_OBJC_TYPE (RTCMediaStream) {
RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
NSMutableArray *_audioTracks;
NSMutableArray *_videoTracks;
rtc::Thread *_signalingThread;
NSMutableArray *_audioTracks /* accessed on _signalingThread */;
NSMutableArray *_videoTracks /* accessed on _signalingThread */;
rtc::scoped_refptr<webrtc::MediaStreamInterface> _nativeMediaStream;
}
@ -36,10 +35,16 @@
}
- (NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *)audioTracks {
if (!_signalingThread->IsCurrent()) {
return _signalingThread->BlockingCall([self]() { return self.audioTracks; });
}
return [_audioTracks copy];
}
- (NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *)videoTracks {
if (!_signalingThread->IsCurrent()) {
return _signalingThread->BlockingCall([self]() { return self.videoTracks; });
}
return [_videoTracks copy];
}
@ -48,33 +53,52 @@
}
- (void)addAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack {
if (!_signalingThread->IsCurrent()) {
return _signalingThread->BlockingCall(
[audioTrack, self]() { return [self addAudioTrack:audioTrack]; });
}
if (_nativeMediaStream->AddTrack(audioTrack.nativeAudioTrack)) {
[_audioTracks addObject:audioTrack];
}
}
- (void)addVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack {
if (!_signalingThread->IsCurrent()) {
return _signalingThread->BlockingCall(
[videoTrack, self]() { return [self addVideoTrack:videoTrack]; });
}
if (_nativeMediaStream->AddTrack(videoTrack.nativeVideoTrack)) {
[_videoTracks addObject:videoTrack];
}
}
- (void)removeAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack {
if (!_signalingThread->IsCurrent()) {
return _signalingThread->BlockingCall(
[audioTrack, self]() { return [self removeAudioTrack:audioTrack]; });
}
NSUInteger index = [_audioTracks indexOfObjectIdenticalTo:audioTrack];
NSAssert(index != NSNotFound,
@"|removeAudioTrack| called on unexpected RTC_OBJC_TYPE(RTCAudioTrack)");
if (index != NSNotFound &&
_nativeMediaStream->RemoveTrack(audioTrack.nativeAudioTrack)) {
if (index == NSNotFound) {
RTC_LOG(LS_INFO) << "|removeAudioTrack| called on unexpected RTC_OBJC_TYPE(RTCAudioTrack)";
return;
}
if (_nativeMediaStream->RemoveTrack(audioTrack.nativeAudioTrack)) {
[_audioTracks removeObjectAtIndex:index];
}
}
- (void)removeVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack {
if (!_signalingThread->IsCurrent()) {
return _signalingThread->BlockingCall(
[videoTrack, self]() { return [self removeVideoTrack:videoTrack]; });
}
NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:videoTrack];
NSAssert(index != NSNotFound,
@"|removeVideoTrack| called on unexpected RTC_OBJC_TYPE(RTCVideoTrack)");
if (index != NSNotFound &&
_nativeMediaStream->RemoveTrack(videoTrack.nativeVideoTrack)) {
if (index == NSNotFound) {
RTC_LOG(LS_INFO) << "|removeVideoTrack| called on unexpected RTC_OBJC_TYPE(RTCVideoTrack)";
return;
}
if (_nativeMediaStream->RemoveTrack(videoTrack.nativeVideoTrack)) {
[_videoTracks removeObjectAtIndex:index];
}
}
@ -98,6 +122,7 @@
NSParameterAssert(nativeMediaStream);
if (self = [super init]) {
_factory = factory;
_signalingThread = factory.signalingThread;
webrtc::AudioTrackVector audioTracks = nativeMediaStream->GetAudioTracks();
webrtc::VideoTrackVector videoTracks = nativeMediaStream->GetVideoTracks();

4
sdk/objc/api/peerconnection/RTCMetrics.mm
View File

@ -12,12 +12,14 @@
#import "RTCMetricsSampleInfo+Private.h"
#include "rtc_base/string_utils.h"
void RTCEnableMetrics(void) {
webrtc::metrics::Enable();
}
NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *> *RTCGetAndResetMetrics(void) {
std::map<std::string, std::unique_ptr<webrtc::metrics::SampleInfo>>
std::map<std::string, std::unique_ptr<webrtc::metrics::SampleInfo>, rtc::AbslStringViewCmp>
histograms;
webrtc::metrics::GetAndReset(&histograms);

8
sdk/objc/api/peerconnection/RTCPeerConnection+DataChannel.mm
View File

@ -23,14 +23,12 @@
std::string labelString = [NSString stdStringForString:label];
const webrtc::DataChannelInit nativeInit =
configuration.nativeDataChannelInit;
rtc::scoped_refptr<webrtc::DataChannelInterface> dataChannel =
self.nativePeerConnection->CreateDataChannel(labelString,
&nativeInit);
if (!dataChannel) {
auto result = self.nativePeerConnection->CreateDataChannelOrError(labelString, &nativeInit);
if (!result.ok()) {
return nil;
}
return [[RTC_OBJC_TYPE(RTCDataChannel) alloc] initWithFactory:self.factory
nativeDataChannel:dataChannel];
nativeDataChannel:result.MoveValue()];
}
@end

30
sdk/objc/api/peerconnection/RTCPeerConnection+Private.h
View File

@ -48,6 +48,12 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
void OnIceCandidate(const IceCandidateInterface *candidate) override;
void OnIceCandidateError(const std::string &address,
int port,
const std::string &url,
int error_code,
const std::string &error_text) override;
void OnIceCandidatesRemoved(const std::vector<cricket::Candidate> &candidates) override;
void OnIceSelectedCandidatePairChanged(const cricket::CandidatePairChangeEvent &event) override;
@ -62,6 +68,8 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
};
} // namespace webrtc
@protocol RTC_OBJC_TYPE
(RTCSSLCertificateVerifier);
@interface RTC_OBJC_TYPE (RTCPeerConnection)
()
@ -77,20 +85,22 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* delegate.
*/
- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
- (nullable instancetype)
initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
certificateVerifier:(nullable id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
/** Initialize an RTCPeerConnection with a configuration, constraints,
* delegate and PeerConnectionDependencies.
*/
- (instancetype)initWithDependencies:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
dependencies:
(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate
- (nullable instancetype)
initWithDependencies:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate
NS_DESIGNATED_INITIALIZER;
+ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:

21
sdk/objc/api/peerconnection/RTCPeerConnection+Stats.mm
View File

@ -69,35 +69,34 @@ class StatsObserverAdapter : public StatsObserver {
- (void)statisticsForSender : (RTC_OBJC_TYPE(RTCRtpSender) *)sender completionHandler
: (RTCStatisticsCompletionHandler)completionHandler {
rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector(
new rtc::RefCountedObject<webrtc::StatsCollectorCallbackAdapter>(completionHandler));
rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
self.nativePeerConnection->GetStats(sender.nativeRtpSender, collector);
}
- (void)statisticsForReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)receiver
completionHandler:(RTCStatisticsCompletionHandler)completionHandler {
rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector(
new rtc::RefCountedObject<webrtc::StatsCollectorCallbackAdapter>(completionHandler));
rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
self.nativePeerConnection->GetStats(receiver.nativeRtpReceiver, collector);
}
- (void)statisticsWithCompletionHandler:(RTCStatisticsCompletionHandler)completionHandler {
rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector(
new rtc::RefCountedObject<webrtc::StatsCollectorCallbackAdapter>(completionHandler));
self.nativePeerConnection->GetStats(collector);
rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
self.nativePeerConnection->GetStats(collector.get());
}
- (void)statsForTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack
statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
completionHandler:
(void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler {
rtc::scoped_refptr<webrtc::StatsObserverAdapter> observer(
new rtc::RefCountedObject<webrtc::StatsObserverAdapter>
(completionHandler));
rtc::scoped_refptr<webrtc::StatsObserverAdapter> observer =
rtc::make_ref_counted<webrtc::StatsObserverAdapter>(completionHandler);
webrtc::PeerConnectionInterface::StatsOutputLevel nativeOutputLevel =
[[self class] nativeStatsOutputLevelForLevel:statsOutputLevel];
self.nativePeerConnection->GetStats(
observer, mediaStreamTrack.nativeTrack, nativeOutputLevel);
observer.get(), mediaStreamTrack.nativeTrack.get(), nativeOutputLevel);
}
@end

68
sdk/objc/api/peerconnection/RTCPeerConnection.h
View File

@ -16,6 +16,7 @@
@class RTC_OBJC_TYPE(RTCDataChannel);
@class RTC_OBJC_TYPE(RTCDataChannelConfiguration);
@class RTC_OBJC_TYPE(RTCIceCandidate);
@class RTC_OBJC_TYPE(RTCIceCandidateErrorEvent);
@class RTC_OBJC_TYPE(RTCMediaConstraints);
@class RTC_OBJC_TYPE(RTCMediaStream);
@class RTC_OBJC_TYPE(RTCMediaStreamTrack);
@ -81,6 +82,12 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
RTCStatsOutputLevelDebug,
};
typedef void (^RTCCreateSessionDescriptionCompletionHandler)(RTC_OBJC_TYPE(RTCSessionDescription) *
_Nullable sdp,
NSError *_Nullable error);
typedef void (^RTCSetSessionDescriptionCompletionHandler)(NSError *_Nullable error);
@class RTC_OBJC_TYPE(RTCPeerConnection);
RTC_OBJC_EXPORT
@ -158,6 +165,10 @@ RTC_OBJC_EXPORT
lastReceivedMs:(int)lastDataReceivedMs
changeReason:(NSString *)reason;
/** Called when gathering of an ICE candidate failed. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didFailToGatherIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *)event;
@end
RTC_OBJC_EXPORT
@ -168,7 +179,7 @@ RTC_OBJC_EXPORT
*/
@property(nonatomic, weak, nullable) id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)> delegate;
/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
* |senders| instead.
* `senders` instead.
*/
@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * localDescription;
@ -201,7 +212,7 @@ RTC_OBJC_EXPORT
- (instancetype)init NS_UNAVAILABLE;
/** Sets the PeerConnection's global configuration to |configuration|.
/** Sets the PeerConnection's global configuration to `configuration`.
* Any changes to STUN/TURN servers or ICE candidate policy will affect the
* next gathering phase, and cause the next call to createOffer to generate
* new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
@ -213,7 +224,12 @@ RTC_OBJC_EXPORT
- (void)close;
/** Provide a remote candidate to the ICE Agent. */
- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
DEPRECATED_MSG_ATTRIBUTE("Please use addIceCandidate:completionHandler: instead");
/** Provide a remote candidate to the ICE Agent. */
- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
completionHandler:(void (^)(NSError *_Nullable error))completionHandler;
/** Remove a group of remote candidates from the ICE Agent. */
- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
@ -232,14 +248,14 @@ RTC_OBJC_EXPORT
/** Add a new media stream track to be sent on this peer connection, and return
* the newly created RTCRtpSender. The RTCRtpSender will be
* associated with the streams specified in the |streamIds| list.
* associated with the streams specified in the `streamIds` list.
*
* Errors: If an error occurs, returns nil. An error can occur if:
* - A sender already exists for the track.
* - The peer connection is closed.
*/
- (RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
streamIds:(NSArray<NSString *> *)streamIds;
- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
streamIds:(NSArray<NSString *> *)streamIds;
/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
*
@ -254,7 +270,7 @@ RTC_OBJC_EXPORT
* transceivers. Adding a transceiver will cause future calls to CreateOffer
* to add a media description for the corresponding transceiver.
*
* The initial value of |mid| in the returned transceiver is nil. Setting a
* The initial value of `mid` in the returned transceiver is nil. Setting a
* new session description may change it to a non-nil value.
*
* https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
@ -271,42 +287,50 @@ RTC_OBJC_EXPORT
* of the transceiver (and sender/receiver) will be derived from the kind of
* the track.
*/
- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track;
- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
* or RTCRtpMediaTypeVideo.
*/
- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType
init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)
init;
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
addTransceiverOfType:(RTCRtpMediaType)mediaType
init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
/** Tells the PeerConnection that ICE should be restarted. This triggers a need
* for negotiation and subsequent offerForConstraints:completionHandler call will act as if
* RTCOfferAnswerOptions::ice_restart is true.
*/
- (void)restartIce;
/** Generate an SDP offer. */
- (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
completionHandler:(nullable void (^)(RTC_OBJC_TYPE(RTCSessionDescription) * _Nullable sdp,
NSError *_Nullable error))completionHandler;
completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler;
/** Generate an SDP answer. */
- (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
completionHandler:
(nullable void (^)(RTC_OBJC_TYPE(RTCSessionDescription) * _Nullable sdp,
NSError *_Nullable error))completionHandler;
completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler;
/** Apply the supplied RTCSessionDescription as the local description. */
- (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler;
/** Creates an offer or answer (depending on current signaling state) and sets
* it as the local session description. */
- (void)setLocalDescriptionWithCompletionHandler:
(RTCSetSessionDescriptionCompletionHandler)completionHandler;
/** Apply the supplied RTCSessionDescription as the remote description. */
- (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler;
/** Limits the bandwidth allocated for all RTP streams sent by this
* PeerConnection. Nil parameters will be unchanged. Setting
* |currentBitrateBps| will force the available bitrate estimate to the given
* `currentBitrateBps` will force the available bitrate estimate to the given
* value. Returns YES if the parameters were successfully updated.
*/
- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
@ -346,7 +370,7 @@ typedef void (^RTCStatisticsCompletionHandler)(RTC_OBJC_TYPE(RTCStatisticsReport
@interface RTC_OBJC_TYPE (RTCPeerConnection)
(Stats)
/** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
/** Gather stats for the given RTCMediaStreamTrack. If `mediaStreamTrack` is nil
* statistics are gathered for all tracks.
*/
- (void)statsForTrack

238
sdk/objc/api/peerconnection/RTCPeerConnection.mm
View File

@ -13,6 +13,7 @@
#import "RTCConfiguration+Private.h"
#import "RTCDataChannel+Private.h"
#import "RTCIceCandidate+Private.h"
#import "RTCIceCandidateErrorEvent+Private.h"
#import "RTCLegacyStatsReport+Private.h"
#import "RTCMediaConstraints+Private.h"
#import "RTCMediaStream+Private.h"
@ -29,12 +30,52 @@
#include "api/jsep_ice_candidate.h"
#include "api/rtc_event_log_output_file.h"
#include "api/set_local_description_observer_interface.h"
#include "api/set_remote_description_observer_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "sdk/objc/native/api/ssl_certificate_verifier.h"
NSString *const kRTCPeerConnectionErrorDomain = @"org.webrtc.RTC_OBJC_TYPE(RTCPeerConnection)";
int const kRTCPeerConnnectionSessionDescriptionError = -1;
namespace {
class SetSessionDescriptionObserver : public webrtc::SetLocalDescriptionObserverInterface,
public webrtc::SetRemoteDescriptionObserverInterface {
public:
SetSessionDescriptionObserver(RTCSetSessionDescriptionCompletionHandler completionHandler) {
completion_handler_ = completionHandler;
}
virtual void OnSetLocalDescriptionComplete(webrtc::RTCError error) override {
OnCompelete(error);
}
virtual void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override {
OnCompelete(error);
}
private:
void OnCompelete(webrtc::RTCError error) {
RTC_DCHECK(completion_handler_ != nil);
if (error.ok()) {
completion_handler_(nil);
} else {
// TODO(hta): Add handling of error.type()
NSString *str = [NSString stringForStdString:error.message()];
NSError *err = [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
code:kRTCPeerConnnectionSessionDescriptionError
userInfo:@{NSLocalizedDescriptionKey : str}];
completion_handler_(err);
}
completion_handler_ = nil;
}
RTCSetSessionDescriptionCompletionHandler completion_handler_;
};
} // anonymous namespace
namespace webrtc {
class CreateSessionDescriptionObserverAdapter
@ -74,38 +115,6 @@ class CreateSessionDescriptionObserverAdapter
NSError *error);
};
class SetSessionDescriptionObserverAdapter :
public SetSessionDescriptionObserver {
public:
SetSessionDescriptionObserverAdapter(void (^completionHandler)
(NSError *error)) {
completion_handler_ = completionHandler;
}
~SetSessionDescriptionObserverAdapter() override { completion_handler_ = nil; }
void OnSuccess() override {
RTC_DCHECK(completion_handler_);
completion_handler_(nil);
completion_handler_ = nil;
}
void OnFailure(RTCError error) override {
RTC_DCHECK(completion_handler_);
// TODO(hta): Add handling of error.type()
NSString *str = [NSString stringForStdString:error.message()];
NSError* err =
[NSError errorWithDomain:kRTCPeerConnectionErrorDomain
code:kRTCPeerConnnectionSessionDescriptionError
userInfo:@{ NSLocalizedDescriptionKey : str }];
completion_handler_(err);
completion_handler_ = nil;
}
private:
void (^completion_handler_)(NSError *error);
};
PeerConnectionDelegateAdapter::PeerConnectionDelegateAdapter(RTC_OBJC_TYPE(RTCPeerConnection) *
peerConnection) {
peer_connection_ = peerConnection;
@ -219,6 +228,24 @@ void PeerConnectionDelegateAdapter::OnIceCandidate(
didGenerateIceCandidate:iceCandidate];
}
void PeerConnectionDelegateAdapter::OnIceCandidateError(const std::string &address,
int port,
const std::string &url,
int error_code,
const std::string &error_text) {
RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *event =
[[RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) alloc] initWithAddress:address
port:port
url:url
errorCode:error_code
errorText:error_text];
if ([peer_connection.delegate respondsToSelector:@selector(peerConnection:
didFailToGatherIceCandidate:)]) {
[peer_connection.delegate peerConnection:peer_connection didFailToGatherIceCandidate:event];
}
}
void PeerConnectionDelegateAdapter::OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
NSMutableArray* ice_candidates =
@ -307,13 +334,18 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
@synthesize delegate = _delegate;
@synthesize factory = _factory;
- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
- (nullable instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
certificateVerifier:
(nullable id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
NSParameterAssert(factory);
std::unique_ptr<webrtc::PeerConnectionDependencies> dependencies =
std::make_unique<webrtc::PeerConnectionDependencies>(nullptr);
if (certificateVerifier != nil) {
dependencies->tls_cert_verifier = webrtc::ObjCToNativeCertificateVerifier(certificateVerifier);
}
return [self initWithDependencies:factory
configuration:configuration
constraints:constraints
@ -321,12 +353,12 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
delegate:delegate];
}
- (instancetype)initWithDependencies:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
dependencies:
(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
- (nullable instancetype)
initWithDependencies:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
NSParameterAssert(factory);
NSParameterAssert(dependencies.get());
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
@ -341,11 +373,12 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
webrtc::PeerConnectionDependencies deps = std::move(*dependencies.release());
deps.observer = _observer.get();
_peerConnection = factory.nativeFactory->CreatePeerConnection(*config, std::move(deps));
auto result = factory.nativeFactory->CreatePeerConnectionOrError(*config, std::move(deps));
if (!_peerConnection) {
if (!result.ok()) {
return nil;
}
_peerConnection = result.MoveValue();
_factory = factory;
_localStreams = [[NSMutableArray alloc] init];
_delegate = delegate;
@ -358,19 +391,23 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
}
- (RTC_OBJC_TYPE(RTCSessionDescription) *)localDescription {
const webrtc::SessionDescriptionInterface *description =
_peerConnection->local_description();
return description ?
[[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
nil;
// It's only safe to operate on SessionDescriptionInterface on the signaling thread.
return _peerConnection->signaling_thread()->BlockingCall([self] {
const webrtc::SessionDescriptionInterface *description = _peerConnection->local_description();
return description ?
[[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
nil;
});
}
- (RTC_OBJC_TYPE(RTCSessionDescription) *)remoteDescription {
const webrtc::SessionDescriptionInterface *description =
_peerConnection->remote_description();
return description ?
[[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
nil;
// It's only safe to operate on SessionDescriptionInterface on the signaling thread.
return _peerConnection->signaling_thread()->BlockingCall([self] {
const webrtc::SessionDescriptionInterface *description = _peerConnection->remote_description();
return description ?
[[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
nil;
});
}
- (RTCSignalingState)signalingState {
@ -418,7 +455,22 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
candidate.nativeCandidate);
_peerConnection->AddIceCandidate(iceCandidate.get());
}
- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
completionHandler:(void (^)(NSError *_Nullable error))completionHandler {
RTC_DCHECK(completionHandler != nil);
_peerConnection->AddIceCandidate(
candidate.nativeCandidate, [completionHandler](const auto &error) {
if (error.ok()) {
completionHandler(nil);
} else {
NSString *str = [NSString stringForStdString:error.message()];
NSError *err = [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
code:static_cast<NSInteger>(error.type())
userInfo:@{NSLocalizedDescriptionKey : str}];
completionHandler(err);
}
});
}
- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)iceCandidates {
std::vector<cricket::Candidate> candidates;
for (RTC_OBJC_TYPE(RTCIceCandidate) * iceCandidate in iceCandidates) {
@ -436,7 +488,7 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
}
- (void)addStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream {
if (!_peerConnection->AddStream(stream.nativeMediaStream)) {
if (!_peerConnection->AddStream(stream.nativeMediaStream.get())) {
RTCLogError(@"Failed to add stream: %@", stream);
return;
}
@ -444,12 +496,12 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
}
- (void)removeStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream {
_peerConnection->RemoveStream(stream.nativeMediaStream);
_peerConnection->RemoveStream(stream.nativeMediaStream.get());
[_localStreams removeObject:stream];
}
- (RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
streamIds:(NSArray<NSString *> *)streamIds {
- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
streamIds:(NSArray<NSString *> *)streamIds {
std::vector<std::string> nativeStreamIds;
for (NSString *streamId in streamIds) {
nativeStreamIds.push_back([streamId UTF8String]);
@ -465,20 +517,20 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
}
- (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender {
bool result = _peerConnection->RemoveTrack(sender.nativeRtpSender);
bool result = _peerConnection->RemoveTrackOrError(sender.nativeRtpSender).ok();
if (!result) {
RTCLogError(@"Failed to remote track %@", sender);
}
return result;
}
- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track {
return [self addTransceiverWithTrack:track
init:[[RTC_OBJC_TYPE(RTCRtpTransceiverInit) alloc] init]];
}
- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init {
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
@ -493,14 +545,14 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
}
- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType {
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType {
return [self addTransceiverOfType:mediaType
init:[[RTC_OBJC_TYPE(RTCRtpTransceiverInit) alloc] init]];
}
- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType
init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)
init {
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
addTransceiverOfType:(RTCRtpMediaType)mediaType
init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init {
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
_peerConnection->AddTransceiver(
[RTC_OBJC_TYPE(RTCRtpReceiver) nativeMediaTypeForMediaType:mediaType], init.nativeInit);
@ -515,44 +567,54 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
}
- (void)restartIce {
_peerConnection->RestartIce();
}
- (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
completionHandler:(void (^)(RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription,
NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
observer(new rtc::RefCountedObject
<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler {
RTC_DCHECK(completionHandler != nil);
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter> observer =
rtc::make_ref_counted<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler);
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
_peerConnection->CreateOffer(observer, options);
_peerConnection->CreateOffer(observer.get(), options);
}
- (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
completionHandler:(void (^)(RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription,
NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
observer(new rtc::RefCountedObject
<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler {
RTC_DCHECK(completionHandler != nil);
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter> observer =
rtc::make_ref_counted<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler);
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
_peerConnection->CreateAnswer(observer, options);
_peerConnection->CreateAnswer(observer.get(), options);
}
- (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
completionHandler:(void (^)(NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
completionHandler));
_peerConnection->SetLocalDescription(observer, sdp.nativeDescription);
completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler {
RTC_DCHECK(completionHandler != nil);
rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface> observer =
rtc::make_ref_counted<::SetSessionDescriptionObserver>(completionHandler);
_peerConnection->SetLocalDescription(sdp.nativeDescription, observer);
}
- (void)setLocalDescriptionWithCompletionHandler:
(RTCSetSessionDescriptionCompletionHandler)completionHandler {
RTC_DCHECK(completionHandler != nil);
rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface> observer =
rtc::make_ref_counted<::SetSessionDescriptionObserver>(completionHandler);
_peerConnection->SetLocalDescription(observer);
}
- (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
completionHandler:(void (^)(NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
completionHandler));
_peerConnection->SetRemoteDescription(observer, sdp.nativeDescription);
completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler {
RTC_DCHECK(completionHandler != nil);
rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface> observer =
rtc::make_ref_counted<::SetSessionDescriptionObserver>(completionHandler);
_peerConnection->SetRemoteDescription(sdp.nativeDescription, observer);
}
- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps

2
sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
View File

@ -74,7 +74,7 @@ NS_ASSUME_NONNULL_BEGIN
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* dependencies.
*/
- (RTC_OBJC_TYPE(RTCPeerConnection) *)
- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithDependencies:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies

4
sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Private.h
View File

@ -12,6 +12,7 @@
#include "api/peer_connection_interface.h"
#include "api/scoped_refptr.h"
#include "rtc_base/thread.h"
NS_ASSUME_NONNULL_BEGIN
@ -26,6 +27,9 @@ NS_ASSUME_NONNULL_BEGIN
@property(nonatomic,
readonly) rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> nativeFactory;
@property(nonatomic, readonly) rtc::Thread* signalingThread;
@property(nonatomic, readonly) rtc::Thread* workerThread;
@end
NS_ASSUME_NONNULL_END

30
sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h
View File

@ -29,18 +29,28 @@ NS_ASSUME_NONNULL_BEGIN
(RTCVideoDecoderFactory);
@protocol RTC_OBJC_TYPE
(RTCVideoEncoderFactory);
@protocol RTC_OBJC_TYPE
(RTCSSLCertificateVerifier);
@protocol RTC_OBJC_TYPE
(RTCAudioDevice);
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCPeerConnectionFactory) : NSObject
/* Initialize object with default H264 video encoder/decoder factories */
/* Initialize object with default H264 video encoder/decoder factories and default ADM */
- (instancetype)init;
/* Initialize object with injectable video encoder/decoder factories */
/* Initialize object with injectable video encoder/decoder factories and default ADM */
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
/* Initialize object with injectable video encoder/decoder factories and injectable ADM */
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice;
/** Initialize an RTCAudioSource with constraints. */
- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
(nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints;
@ -60,6 +70,13 @@ RTC_OBJC_EXPORT
*/
- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource;
/** Initialize a generic RTCVideoSource with he posibility of marking
* it as usable for screen sharing. The RTCVideoSource should be
* passed to a RTCVideoCapturer implementation, e.g.
* RTCCameraVideoCapturer, in order to produce frames.
*/
- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast;
/** Initialize an RTCVideoTrack with a source and an id. */
- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
trackId:(NSString *)trackId;
@ -70,11 +87,18 @@ RTC_OBJC_EXPORT
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* delegate.
*/
- (RTC_OBJC_TYPE(RTCPeerConnection) *)
- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
certificateVerifier:
(id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
/** Set the options to be used for subsequently created RTCPeerConnections */
- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options;

106
sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
View File

@ -25,35 +25,32 @@
#import "base/RTCVideoDecoderFactory.h"
#import "base/RTCVideoEncoderFactory.h"
#import "helpers/NSString+StdString.h"
#ifndef HAVE_NO_MEDIA
#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
#import "components/video_codec/RTCVideoEncoderFactoryH264.h"
// The no-media version PeerConnectionFactory doesn't depend on these files, but the gn check tool
// is not smart enough to take the #ifdef into account.
#include "api/audio_codecs/builtin_audio_decoder_factory.h" // nogncheck
#include "api/audio_codecs/builtin_audio_encoder_factory.h" // nogncheck
#include "rtc_base/checks.h"
#include "sdk/objc/native/api/network_monitor_factory.h"
#include "sdk/objc/native/api/ssl_certificate_verifier.h"
#include "system_wrappers/include/field_trial.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "modules/audio_device/include/audio_device.h" // nogncheck
#include "modules/audio_processing/include/audio_processing.h" // nogncheck
#include "api/transport/field_trial_based_config.h"
#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
#import "components/video_codec/RTCVideoEncoderFactoryH264.h"
#include "media/engine/webrtc_media_engine.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "sdk/objc/native/api/objc_audio_device_module.h"
#include "sdk/objc/native/api/video_decoder_factory.h"
#include "sdk/objc/native/api/video_encoder_factory.h"
#include "sdk/objc/native/src/objc_video_decoder_factory.h"
#include "sdk/objc/native/src/objc_video_encoder_factory.h"
#endif
#if defined(WEBRTC_IOS)
#import "sdk/objc/native/api/audio_device_module.h"
#endif
// Adding the nogncheck to disable the including header check.
// The no-media version PeerConnectionFactory doesn't depend on media related
// C++ target.
// TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
// API layer.
#include "media/engine/webrtc_media_engine.h" // nogncheck
@implementation RTC_OBJC_TYPE (RTCPeerConnectionFactory) {
std::unique_ptr<rtc::Thread> _networkThread;
std::unique_ptr<rtc::Thread> _workerThread;
@ -72,9 +69,6 @@
}
- (instancetype)init {
#ifdef HAVE_NO_MEDIA
return [self initWithNoMedia];
#else
return [self
initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
@ -82,14 +76,20 @@
RTCVideoEncoderFactoryH264) alloc] init])
nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory([[RTC_OBJC_TYPE(
RTCVideoDecoderFactoryH264) alloc] init])
audioDeviceModule:[self audioDeviceModule]
audioDeviceModule:[self audioDeviceModule].get()
audioProcessingModule:nullptr];
#endif
}
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
return [self initWithEncoderFactory:encoderFactory decoderFactory:decoderFactory audioDevice:nil];
}
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice {
#ifdef HAVE_NO_MEDIA
return [self initWithNoMedia];
#else
@ -101,30 +101,37 @@
if (decoderFactory) {
native_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory);
}
rtc::scoped_refptr<webrtc::AudioDeviceModule> audio_device_module;
if (audioDevice) {
audio_device_module = webrtc::CreateAudioDeviceModule(audioDevice);
} else {
audio_device_module = [self audioDeviceModule];
}
return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
nativeVideoEncoderFactory:std::move(native_encoder_factory)
nativeVideoDecoderFactory:std::move(native_decoder_factory)
audioDeviceModule:[self audioDeviceModule]
audioDeviceModule:audio_device_module.get()
audioProcessingModule:nullptr];
#endif
}
- (instancetype)initNative {
if (self = [super init]) {
_networkThread = rtc::Thread::CreateWithSocketServer();
_networkThread->SetName("network_thread", _networkThread.get());
BOOL result = _networkThread->Start();
NSAssert(result, @"Failed to start network thread.");
RTC_DCHECK(result) << "Failed to start network thread.";
_workerThread = rtc::Thread::Create();
_workerThread->SetName("worker_thread", _workerThread.get());
result = _workerThread->Start();
NSAssert(result, @"Failed to start worker thread.");
RTC_DCHECK(result) << "Failed to start worker thread.";
_signalingThread = rtc::Thread::Create();
_signalingThread->SetName("signaling_thread", _signalingThread.get());
result = _signalingThread->Start();
NSAssert(result, @"Failed to start signaling thread.");
RTC_DCHECK(result) << "Failed to start signaling thread.";
}
return self;
}
@ -135,6 +142,9 @@
dependencies.network_thread = _networkThread.get();
dependencies.worker_thread = _workerThread.get();
dependencies.signaling_thread = _signalingThread.get();
if (webrtc::field_trial::IsEnabled("WebRTC-Network-UseNWPathMonitor")) {
dependencies.network_monitor_factory = webrtc::CreateNetworkMonitorFactory();
}
_nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
}
@ -179,8 +189,12 @@
dependencies.network_thread = _networkThread.get();
dependencies.worker_thread = _workerThread.get();
dependencies.signaling_thread = _signalingThread.get();
#ifndef HAVE_NO_MEDIA
dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
if (webrtc::field_trial::IsEnabled("WebRTC-Network-UseNWPathMonitor")) {
dependencies.network_monitor_factory = webrtc::CreateNetworkMonitorFactory();
}
dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
dependencies.task_queue_factory =
webrtc::CreateDefaultTaskQueueFactory(dependencies.trials.get());
cricket::MediaEngineDependencies media_deps;
media_deps.adm = std::move(audioDeviceModule);
media_deps.task_queue_factory = dependencies.task_queue_factory.get();
@ -193,12 +207,12 @@
} else {
media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
}
media_deps.trials = dependencies.trials.get();
dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
dependencies.call_factory = webrtc::CreateCallFactory();
dependencies.event_log_factory =
std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
dependencies.network_controller_factory = std::move(networkControllerFactory);
#endif
_nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
}
@ -235,6 +249,13 @@
workerThread:_workerThread.get()];
}
- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast {
return [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self
signalingThread:_signalingThread.get()
workerThread:_workerThread.get()
isScreenCast:forScreenCast];
}
- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
trackId:(NSString *)trackId {
return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:self source:source trackId:trackId];
@ -244,7 +265,7 @@
return [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:self streamId:streamId];
}
- (RTC_OBJC_TYPE(RTCPeerConnection) *)
- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
delegate:
@ -252,10 +273,25 @@
return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
configuration:configuration
constraints:constraints
certificateVerifier:nil
delegate:delegate];
}
- (RTC_OBJC_TYPE(RTCPeerConnection) *)
- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
certificateVerifier:
(id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
delegate:
(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
configuration:configuration
constraints:constraints
certificateVerifier:certificateVerifier
delegate:delegate];
}
- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithDependencies:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
@ -295,4 +331,12 @@
_hasStartedAecDump = NO;
}
- (rtc::Thread *)signalingThread {
return _signalingThread.get();
}
- (rtc::Thread *)workerThread {
return _workerThread.get();
}
@end

2
sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
View File

@ -38,7 +38,7 @@
nativeAudioDecoderFactory:_audioDecoderFactory
nativeVideoEncoderFactory:std::move(_videoEncoderFactory)
nativeVideoDecoderFactory:std::move(_videoDecoderFactory)
audioDeviceModule:_audioDeviceModule
audioDeviceModule:_audioDeviceModule.get()
audioProcessingModule:_audioProcessingModule];
}

3
sdk/objc/api/peerconnection/RTCRtcpParameters+Private.h
View File

@ -21,7 +21,8 @@ NS_ASSUME_NONNULL_BEGIN
@property(nonatomic, readonly) webrtc::RtcpParameters nativeParameters;
/** Initialize the object with a native RtcpParameters structure. */
- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters;
- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters
NS_DESIGNATED_INITIALIZER;
@end

2
sdk/objc/api/peerconnection/RTCRtcpParameters.h
View File

@ -23,7 +23,7 @@ RTC_OBJC_EXPORT
/** Whether reduced size RTCP is configured or compound RTCP. */
@property(nonatomic, assign) BOOL isReducedSize;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
- (instancetype)init;
@end

5
sdk/objc/api/peerconnection/RTCRtcpParameters.mm
View File

@ -18,11 +18,12 @@
@synthesize isReducedSize = _isReducedSize;
- (instancetype)init {
return [super init];
webrtc::RtcpParameters nativeParameters;
return [self initWithNativeParameters:nativeParameters];
}
- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters {
if (self = [self init]) {
if (self = [super init]) {
_cname = [NSString stringForStdString:nativeParameters.cname];
_isReducedSize = nativeParameters.reduced_size;
}

3
sdk/objc/api/peerconnection/RTCRtpCodecParameters+Private.h
View File

@ -21,7 +21,8 @@ NS_ASSUME_NONNULL_BEGIN
@property(nonatomic, readonly) webrtc::RtpCodecParameters nativeParameters;
/** Initialize the object with a native RtpCodecParameters structure. */
- (instancetype)initWithNativeParameters:(const webrtc::RtpCodecParameters &)nativeParameters;
- (instancetype)initWithNativeParameters:(const webrtc::RtpCodecParameters &)nativeParameters
NS_DESIGNATED_INITIALIZER;
@end

4
sdk/objc/api/peerconnection/RTCRtpCodecParameters.h
View File

@ -31,7 +31,7 @@ RTC_EXTERN const NSString *const kRTCVp8CodecName;
RTC_EXTERN const NSString *const kRTCVp9CodecName;
RTC_EXTERN const NSString *const kRTCH264CodecName;
/** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTC_OBJC_TYPE(RTCRtpCodecParameters) */
/** Defined in https://www.w3.org/TR/webrtc/#idl-def-rtcrtpcodecparameters */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpCodecParameters) : NSObject
@ -66,7 +66,7 @@ RTC_OBJC_EXPORT
/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
- (instancetype)init;
@end

12
sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
View File

@ -44,12 +44,13 @@ const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
@synthesize parameters = _parameters;
- (instancetype)init {
return [super init];
webrtc::RtpCodecParameters nativeParameters;
return [self initWithNativeParameters:nativeParameters];
}
- (instancetype)initWithNativeParameters:
(const webrtc::RtpCodecParameters &)nativeParameters {
if (self = [self init]) {
if (self = [super init]) {
_payloadType = nativeParameters.payload_type;
_name = [NSString stringForStdString:nativeParameters.name];
switch (nativeParameters.kind) {
@ -60,7 +61,10 @@ const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
_kind = kRTCMediaStreamTrackKindVideo;
break;
case cricket::MEDIA_TYPE_DATA:
RTC_NOTREACHED();
RTC_DCHECK_NOTREACHED();
break;
case cricket::MEDIA_TYPE_UNSUPPORTED:
RTC_DCHECK_NOTREACHED();
break;
}
if (nativeParameters.clock_rate) {
@ -90,7 +94,7 @@ const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
} else if (_kind == kRTCMediaStreamTrackKindVideo) {
parameters.kind = cricket::MEDIA_TYPE_VIDEO;
} else {
RTC_NOTREACHED();
RTC_DCHECK_NOTREACHED();
}
if (_clockRate != nil) {
parameters.clock_rate = absl::optional<int>(_clockRate.intValue);

3
sdk/objc/api/peerconnection/RTCRtpEncodingParameters+Private.h
View File

@ -21,7 +21,8 @@ NS_ASSUME_NONNULL_BEGIN
@property(nonatomic, readonly) webrtc::RtpEncodingParameters nativeParameters;
/** Initialize the object with a native RtpEncodingParameters structure. */
- (instancetype)initWithNativeParameters:(const webrtc::RtpEncodingParameters &)nativeParameters;
- (instancetype)initWithNativeParameters:(const webrtc::RtpEncodingParameters &)nativeParameters
NS_DESIGNATED_INITIALIZER;
@end

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