15583c19d7
The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc The HAVE_SCTP define was added for the peerconnection_unittests target in api_tests.gyp. I also checked that none of SRTP_RELATIVE_PATH HAVE_SRTP HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE were used by the talk/app/webrtc code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1615433002 BUG=webrtc:5418 NOPRESUBMIT=True R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1610243002 . Cr-Commit-Position: refs/heads/master@{#11545}
61 lines
2.0 KiB
Objective-C
61 lines
2.0 KiB
Objective-C
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import "RTCConfiguration.h"
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#include "webrtc/api/peerconnectioninterface.h"
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NS_ASSUME_NONNULL_BEGIN
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@interface RTCConfiguration ()
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/**
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* RTCConfiguration struct representation of this RTCConfiguration. This is
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* needed to pass to the underlying C++ APIs.
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*/
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@property(nonatomic, readonly)
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webrtc::PeerConnectionInterface::RTCConfiguration nativeConfiguration;
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+ (webrtc::PeerConnectionInterface::IceTransportsType)
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nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy;
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+ (RTCIceTransportPolicy)transportPolicyForTransportsType:
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(webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
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+ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy;
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+ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
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(RTCBundlePolicy)policy;
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+ (RTCBundlePolicy)bundlePolicyForNativePolicy:
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(webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
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+ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy;
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+ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
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(RTCRtcpMuxPolicy)policy;
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+ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
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(webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
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+ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy;
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+ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
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nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy;
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+ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
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(webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
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+ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
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@end
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NS_ASSUME_NONNULL_END
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