Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

58 lines
1.5 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 MODULES_VIDEO_CODING_PACKET_H_
#define MODULES_VIDEO_CODING_PACKET_H_
#include "modules/include/module_common_types.h"
#include "modules/video_coding/jitter_buffer_common.h"
#include "typedefs.h"
namespace webrtc {
class VCMPacket {
public:
VCMPacket();
VCMPacket(const uint8_t* ptr,
const size_t size,
const WebRtcRTPHeader& rtpHeader);
void Reset();
uint8_t payloadType;
uint32_t timestamp;
// NTP time of the capture time in local timebase in milliseconds.
int64_t ntp_time_ms_;
uint16_t seqNum;
const uint8_t* dataPtr;
size_t sizeBytes;
bool markerBit;
int timesNacked;
FrameType frameType;
VideoCodecType codec;
bool is_first_packet_in_frame;
VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete.
bool insertStartCode; // True if a start code should be inserted before this
// packet.
int width;
int height;
RTPVideoHeader video_header;
int64_t receive_time_ms;
protected:
void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
};
} // namespace webrtc
#endif // MODULES_VIDEO_CODING_PACKET_H_