webrtc_m130/webrtc/api/objc/RTCMediaStreamTrack+Private.h
Henrik Kjellander 15583c19d7 Move talk/app/webrtc to webrtc/api
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}
2016-02-10 09:53:26 +00:00

51 lines
1.5 KiB
Objective-C

/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#import "RTCMediaStreamTrack.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/base/scoped_ptr.h"
typedef NS_ENUM(NSInteger, RTCMediaStreamTrackType) {
RTCMediaStreamTrackTypeAudio,
RTCMediaStreamTrackTypeVideo,
};
NS_ASSUME_NONNULL_BEGIN
@interface RTCMediaStreamTrack ()
/**
* The native MediaStreamTrackInterface passed in or created during
* construction.
*/
@property(nonatomic, readonly)
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
/**
* Initialize an RTCMediaStreamTrack from a native MediaStreamTrackInterface.
*/
- (instancetype)initWithNativeTrack:
(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
type:(RTCMediaStreamTrackType)type
NS_DESIGNATED_INITIALIZER;
+ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
(RTCMediaStreamTrackState)state;
+ (RTCMediaStreamTrackState)trackStateForNativeState:
(webrtc::MediaStreamTrackInterface::TrackState)nativeState;
+ (NSString *)stringForState:(RTCMediaStreamTrackState)state;
@end
NS_ASSUME_NONNULL_END