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}
135 lines
4.1 KiB
C++
135 lines
4.1 KiB
C++
/*
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* Copyright (c) 2012 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_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
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#include <map>
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
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const size_t kRtpOneByteHeaderLength = 4;
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const size_t kTransmissionTimeOffsetLength = 4;
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const size_t kAudioLevelLength = 2;
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const size_t kAbsoluteSendTimeLength = 4;
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const size_t kVideoRotationLength = 2;
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const size_t kTransportSequenceNumberLength = 3;
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const size_t kPlayoutDelayLength = 4;
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// Playout delay in milliseconds. A playout delay limit (min or max)
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// has 12 bits allocated. This allows a range of 0-4095 values which translates
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// to a range of 0-40950 in milliseconds.
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const int kPlayoutDelayGranularityMs = 10;
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// Maximum playout delay value in milliseconds.
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const int kPlayoutDelayMaxMs = 40950;
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struct HeaderExtension {
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explicit HeaderExtension(RTPExtensionType extension_type)
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: type(extension_type), length(0), active(true) {
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Init();
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}
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HeaderExtension(RTPExtensionType extension_type, bool active)
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: type(extension_type), length(0), active(active) {
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Init();
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}
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void Init() {
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// TODO(solenberg): Create handler classes for header extensions so we can
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// get rid of switches like these as well as handling code spread out all
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// over.
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switch (type) {
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case kRtpExtensionTransmissionTimeOffset:
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length = kTransmissionTimeOffsetLength;
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break;
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case kRtpExtensionAudioLevel:
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length = kAudioLevelLength;
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break;
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case kRtpExtensionAbsoluteSendTime:
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length = kAbsoluteSendTimeLength;
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break;
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case kRtpExtensionVideoRotation:
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length = kVideoRotationLength;
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break;
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case kRtpExtensionTransportSequenceNumber:
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length = kTransportSequenceNumberLength;
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break;
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case kRtpExtensionPlayoutDelay:
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length = kPlayoutDelayLength;
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break;
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default:
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assert(false);
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}
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}
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const RTPExtensionType type;
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uint8_t length;
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bool active;
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};
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class RtpHeaderExtensionMap {
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public:
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static constexpr RTPExtensionType kInvalidType = kRtpExtensionNone;
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static constexpr uint8_t kInvalidId = 0;
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RtpHeaderExtensionMap();
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~RtpHeaderExtensionMap();
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void Erase();
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int32_t Register(const RTPExtensionType type, const uint8_t id);
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// Active on an extension indicates whether it is currently being added on
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// on the RTP packets. The active/inactive status on an extension can change
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// dynamically depending on the need to convey new information.
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int32_t RegisterInactive(const RTPExtensionType type, const uint8_t id);
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bool SetActive(const RTPExtensionType type, bool active);
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int32_t Deregister(const RTPExtensionType type);
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bool IsRegistered(RTPExtensionType type) const;
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int32_t GetType(const uint8_t id, RTPExtensionType* type) const;
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// Return kInvalidType if not found.
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RTPExtensionType GetType(uint8_t id) const;
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int32_t GetId(const RTPExtensionType type, uint8_t* id) const;
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// Return kInvalidId if not found.
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uint8_t GetId(RTPExtensionType type) const;
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//
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// Methods below ignore any inactive rtp header extensions.
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//
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size_t GetTotalLengthInBytes() const;
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int32_t GetLengthUntilBlockStartInBytes(const RTPExtensionType type) const;
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void GetCopy(RtpHeaderExtensionMap* map) const;
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int32_t Size() const;
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RTPExtensionType First() const;
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RTPExtensionType Next(RTPExtensionType type) const;
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private:
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int32_t Register(const RTPExtensionType type, const uint8_t id, bool active);
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std::map<uint8_t, HeaderExtension*> extensionMap_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
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