6b4b5f3770
This CL adds support for an extension on RTP frames to allow the sender to specify the minimum and maximum playout delay limits. The receiver makes a best-effort attempt to keep the capture-to-render delay within this range. This allows different types of application to specify different end-to-end delay goals. For example gaming can support rendering of frames as soon as received on receiver to minimize delay. A movie playback application can specify a minimum playout delay to allow fixed buffering in presence of network jitter. There are no tests at this time and most of testing is done with chromium webrtc prototype. On chromoting performance tests, this extension helps bring down end-to-end delay by about 150 ms on small frames. BUG=webrtc:5895 Review-Url: https://codereview.webrtc.org/2007743003 Cr-Commit-Position: refs/heads/master@{#13059}
61 lines
1.7 KiB
C++
61 lines
1.7 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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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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*/
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#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
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#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_coding/jitter_buffer_common.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class VCMPacket {
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public:
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VCMPacket();
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VCMPacket(const uint8_t* ptr,
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const size_t size,
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const WebRtcRTPHeader& rtpHeader);
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VCMPacket(const uint8_t* ptr,
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size_t size,
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uint16_t seqNum,
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uint32_t timestamp,
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bool markerBit);
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void Reset();
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uint8_t payloadType;
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uint32_t timestamp;
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// NTP time of the capture time in local timebase in milliseconds.
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int64_t ntp_time_ms_;
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uint16_t seqNum;
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const uint8_t* dataPtr;
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size_t sizeBytes;
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bool markerBit;
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int timesNacked;
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FrameType frameType;
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VideoCodecType codec;
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bool isFirstPacket; // Is this first packet in a frame.
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VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete.
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bool insertStartCode; // True if a start code should be inserted before this
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// packet.
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int width;
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int height;
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RTPVideoHeader video_header;
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protected:
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void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
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