ilnik 29dbb1992a Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ )
Reason for revert:
Relanded by mistake.

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 fixes which break API
>
> 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: 64e739aeae
> >
> > 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: 5721866808
>
> 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/2811963002
> Cr-Commit-Position: refs/heads/master@{#17645}
> Committed: 4fa0c4f97f

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/2810923004
Cr-Commit-Position: refs/heads/master@{#17648}
2017-04-11 11:49:07 +00:00

129 lines
3.6 KiB
C++

/*
* 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_; }
/**
* 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_