webrtc_m130/webrtc/media/engine/webrtcvideoframe.h

124 lines
4.4 KiB
C
Raw Normal View History

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