2013-07-10 00:45:36 +00:00
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/*
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2016-02-07 20:46:45 -08:00
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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2013-07-10 00:45:36 +00:00
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*
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2016-02-07 20:46:45 -08:00
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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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2013-07-10 00:45:36 +00:00
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*/
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2016-02-12 06:39:40 +01:00
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#ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOFRAME_H_
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#define WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOFRAME_H_
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2013-07-10 00:45:36 +00:00
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2016-02-26 03:00:35 -08:00
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#include <memory>
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2014-07-29 17:36:52 +00:00
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#include "webrtc/base/buffer.h"
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#include "webrtc/base/refcount.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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2013-07-10 00:45:36 +00:00
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#include "webrtc/common_types.h"
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2015-11-16 13:52:24 -08:00
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#include "webrtc/common_video/include/video_frame_buffer.h"
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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
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#include "webrtc/media/base/videoframe.h"
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2014-12-14 11:09:23 +00:00
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2013-07-10 00:45:36 +00:00
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namespace cricket {
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2016-10-08 22:21:35 -07:00
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// TODO(nisse): This class will be deleted when the cricket::VideoFrame and
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// webrtc::VideoFrame classes are merged. See
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// https://bugs.chromium.org/p/webrtc/issues/detail?id=5682. Try to use only the
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// preferred constructor, and the non-deprecated methods of the VideoFrame base
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// class.
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2013-07-10 00:45:36 +00:00
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class WebRtcVideoFrame : public VideoFrame {
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public:
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// TODO(nisse): Deprecated. Using the default constructor violates the
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// reasonable assumption that video_frame_buffer() returns a valid buffer.
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WebRtcVideoFrame();
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// Preferred constructor.
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2016-04-14 02:29:29 -07:00
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WebRtcVideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
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webrtc::VideoRotation rotation,
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int64_t timestamp_us,
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uint32_t transport_frame_id);
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// Alternative constructor, when not knowing or caring about the
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// transport_frame_id. Which is set to zero.
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2016-08-22 03:54:53 -07:00
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WebRtcVideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
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webrtc::VideoRotation rotation,
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int64_t timestamp_us);
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// TODO(nisse): Deprecated, delete as soon as all callers have switched to the
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// above constructor with microsecond timestamp.
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WebRtcVideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
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int64_t timestamp_ns,
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webrtc::VideoRotation rotation);
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~WebRtcVideoFrame();
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// TODO(nisse): Init (and its helpers Reset and Validate) are used
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// only by the LoadFrame function used in the VideoFrame unittests.
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// Rewrite tests, and delete this function.
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// Creates a frame from a raw sample with FourCC "format" and size "w" x "h".
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// "h" can be negative indicating a vertically flipped image.
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// "dh" is destination height if cropping is desired and is always positive.
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// Returns "true" if successful.
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bool Init(uint32_t format,
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int w,
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int h,
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int dw,
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int dh,
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uint8_t* sample,
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size_t sample_size,
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int64_t timestamp_ns,
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webrtc::VideoRotation rotation);
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void InitToEmptyBuffer(int w, int h);
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int width() const override;
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int height() const override;
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const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& video_frame_buffer()
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const override;
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uint32_t transport_frame_id() const override;
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int64_t timestamp_us() const override;
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void set_timestamp_us(int64_t time_us) override;
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webrtc::VideoRotation rotation() const override;
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protected:
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// Creates a frame from a raw sample with FourCC |format| and size |w| x |h|.
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// |h| can be negative indicating a vertically flipped image.
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// |dw| is destination width; can be less than |w| if cropping is desired.
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// |dh| is destination height, like |dw|, but must be a positive number.
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// Returns whether the function succeeded or failed.
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bool Reset(uint32_t format,
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int w,
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int h,
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int dw,
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int dh,
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uint8_t* sample,
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size_t sample_size,
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int64_t timestamp_us,
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webrtc::VideoRotation rotation,
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bool apply_rotation);
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private:
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// Tests mutate |rotation_|, so the base test class is a friend.
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friend class WebRtcVideoFrameTest;
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// An opaque reference counted handle that stores the pixel data.
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rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
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int64_t timestamp_us_;
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uint32_t transport_frame_id_;
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webrtc::VideoRotation rotation_;
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// This is mutable as the calculation is expensive but once calculated, it
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// remains const.
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mutable std::unique_ptr<VideoFrame> rotated_frame_;
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2013-07-10 00:45:36 +00:00
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};
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} // namespace cricket
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2016-02-12 06:39:40 +01:00
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#endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOFRAME_H_
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