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/*
* 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_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/common_video/include/video_image.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
namespace webrtc {
class VCMEncodedFrame : protected EncodedImage {
public:
VCMEncodedFrame();
explicit VCMEncodedFrame(const webrtc::EncodedImage& rhs);
VCMEncodedFrame(const VCMEncodedFrame& rhs);
~VCMEncodedFrame();
/**
* Delete VideoFrame and resets members to zero
*/
void Free();
/**
* Set render time in milliseconds
*/
void SetRenderTime(const int64_t renderTimeMs) {
_renderTimeMs = renderTimeMs;
}
/**
* Set the encoded frame size
*/
void SetEncodedSize(uint32_t width, uint32_t height) {
_encodedWidth = width;
_encodedHeight = height;
}
/**
* Get the encoded image
*/
const webrtc::EncodedImage& EncodedImage() const {
return static_cast<const webrtc::EncodedImage&>(*this);
}
/**
* Get pointer to frame buffer
*/
const uint8_t* Buffer() const { return _buffer; }
/**
* Get frame length
*/
size_t Length() const { return _length; }
/**
* Get frame timestamp (90kHz)
*/
uint32_t TimeStamp() const { return _timeStamp; }
/**
* Get render time in milliseconds
*/
int64_t RenderTimeMs() const { return _renderTimeMs; }
/**
* Get frame type
*/
webrtc::FrameType FrameType() const { return _frameType; }
/**
* Get frame rotation
*/
VideoRotation rotation() const { return rotation_; }
/**
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ ) Reason for revert: Breaks android buildbots. Original issue's description: > Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) > > Reason for revert: > Reland with appropriate changes to API to not break depending projects. > > Original issue's description: > > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ ) > > > > Reason for revert: > > Breaks dependent projects. > > > > Original issue's description: > > > Add content type information to Encoded Images and add corresponding RTP extension header. > > > Use it to separate UMA e2e delay metric between screenshare from video. > > > Content type extension is set based on encoder settings and processed and decoders. > > > > > > Also, > > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested. > > > > > > BUG=webrtc:7420 > > > > > > Review-Url: https://codereview.webrtc.org/2772033002 > > > Cr-Commit-Position: refs/heads/master@{#17640} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/64e739aeae5629cbbebf2a19e1d3e6b452bb6d0b > > > > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > NOTRY=true > > BUG=webrtc:7420 > > > > Review-Url: https://codereview.webrtc.org/2816463002 > > Cr-Commit-Position: refs/heads/master@{#17644} > > Committed: https://chromium.googlesource.com/external/webrtc/+/57218668083fce36a55efb8c8f1d31785253b8c0 > > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7420 > > Review-Url: https://codereview.webrtc.org/2812913002 > Cr-Commit-Position: refs/heads/master@{#17651} > Committed: https://chromium.googlesource.com/external/webrtc/+/774f6b4b968b46d6bd95c42eec0ce3202b47f0b2 TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7420 Review-Url: https://codereview.webrtc.org/2809653004 Cr-Commit-Position: refs/heads/master@{#17653}
2017-04-11 06:20:05 -07:00
* True if this frame is complete, false otherwise
*/
bool Complete() const { return _completeFrame; }
/**
* True if there's a frame missing before this frame
*/
bool MissingFrame() const { return _missingFrame; }
/**
* Payload type of the encoded payload
*/
uint8_t PayloadType() const { return _payloadType; }
/**
* Get codec specific info.
* The returned pointer is only valid as long as the VCMEncodedFrame
* is valid. Also, VCMEncodedFrame owns the pointer and will delete
* the object.
*/
const CodecSpecificInfo* CodecSpecific() const { return &_codecSpecificInfo; }
protected:
/**
* Verifies that current allocated buffer size is larger than or equal to the
* input size.
* If the current buffer size is smaller, a new allocation is made and the old
* buffer data
* is copied to the new buffer.
* Buffer size is updated to minimumSize.
*/
void VerifyAndAllocate(size_t minimumSize);
void Reset();
void CopyCodecSpecific(const RTPVideoHeader* header);
int64_t _renderTimeMs;
uint8_t _payloadType;
bool _missingFrame;
CodecSpecificInfo _codecSpecificInfo;
webrtc::VideoCodecType _codec;
// Video rotation is only set along with the last packet for each frame
// (same as marker bit). This |_rotation_set| is only for debugging purpose
// to ensure we don't set it twice for a frame.
bool _rotation_set;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_