2016-01-11 09:47:07 -08:00
|
|
|
/*
|
|
|
|
|
* 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.
|
|
|
|
|
*/
|
|
|
|
|
|
2016-04-27 01:54:20 -07:00
|
|
|
#import "WebRTC/RTCVideoFrame.h"
|
2016-01-11 09:47:07 -08:00
|
|
|
|
2016-08-10 07:58:29 -07:00
|
|
|
#include "webrtc/common_video/rotation.h"
|
Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
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 in order to not
break Git history.
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.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media 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/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
|
|
|
#include "webrtc/media/base/videoframe.h"
|
2016-01-11 09:47:07 -08:00
|
|
|
|
|
|
|
|
NS_ASSUME_NONNULL_BEGIN
|
|
|
|
|
|
|
|
|
|
@interface RTCVideoFrame ()
|
|
|
|
|
|
2016-04-04 14:10:43 -07:00
|
|
|
@property(nonatomic, readonly)
|
|
|
|
|
rtc::scoped_refptr<webrtc::VideoFrameBuffer> i420Buffer;
|
|
|
|
|
|
2016-08-10 07:58:29 -07:00
|
|
|
- (instancetype)initWithVideoBuffer:
|
|
|
|
|
(rtc::scoped_refptr<webrtc::VideoFrameBuffer>)videoBuffer
|
|
|
|
|
rotation:(webrtc::VideoRotation)rotation
|
|
|
|
|
timeStampNs:(int64_t)timeStampNs
|
2016-01-11 09:47:07 -08:00
|
|
|
NS_DESIGNATED_INITIALIZER;
|
|
|
|
|
|
|
|
|
|
@end
|
|
|
|
|
|
|
|
|
|
NS_ASSUME_NONNULL_END
|