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d2a6686a10
This change adds classes so that we later can plumb information about received packets to each audio and video frame. It's not wired up to do anything yet. Bug: webrtc:10668 Change-Id: I962df493a76692f668314f78d6792d7636c5a31b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138203 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28138}
122 lines
3.6 KiB
C++
122 lines
3.6 KiB
C++
/*
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* Copyright (c) 2017 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 API_RTP_HEADERS_H_
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#define API_RTP_HEADERS_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <string>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/video/color_space.h"
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#include "api/video/video_content_type.h"
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#include "api/video/video_frame_marking.h"
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#include "api/video/video_rotation.h"
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#include "api/video/video_timing.h"
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#include "common_types.h" // NOLINT(build/include)
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namespace webrtc {
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struct FeedbackRequest {
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// Determines whether the recv delta as specified in
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// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
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// should be included.
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bool include_timestamps;
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// Include feedback of received packets in the range [sequence_number -
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// sequence_count + 1, sequence_number]. That is, no feedback will be sent if
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// sequence_count is zero.
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int sequence_count;
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};
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struct RTPHeaderExtension {
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RTPHeaderExtension();
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RTPHeaderExtension(const RTPHeaderExtension& other);
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RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
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bool hasTransmissionTimeOffset;
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int32_t transmissionTimeOffset;
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bool hasAbsoluteSendTime;
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uint32_t absoluteSendTime;
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bool hasTransportSequenceNumber;
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uint16_t transportSequenceNumber;
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absl::optional<FeedbackRequest> feedback_request;
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// Audio Level includes both level in dBov and voiced/unvoiced bit. See:
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// https://tools.ietf.org/html/rfc6464#section-3
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bool hasAudioLevel;
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bool voiceActivity;
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uint8_t audioLevel;
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// For Coordination of Video Orientation. See
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// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
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// ts_126114v120700p.pdf
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bool hasVideoRotation;
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VideoRotation videoRotation;
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// TODO(ilnik): Refactor this and one above to be absl::optional() and remove
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// a corresponding bool flag.
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bool hasVideoContentType;
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VideoContentType videoContentType;
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bool has_video_timing;
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VideoSendTiming video_timing;
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bool has_frame_marking;
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FrameMarking frame_marking;
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PlayoutDelay playout_delay = {-1, -1};
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// For identification of a stream when ssrc is not signaled. See
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// https://tools.ietf.org/html/draft-ietf-avtext-rid-09
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// TODO(danilchap): Update url from draft to release version.
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std::string stream_id;
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std::string repaired_stream_id;
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// For identifying the media section used to interpret this RTP packet. See
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// https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
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std::string mid;
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absl::optional<ColorSpace> color_space;
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};
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enum { kRtpCsrcSize = 15 }; // RFC 3550 page 13
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struct RTPHeader {
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RTPHeader();
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RTPHeader(const RTPHeader& other);
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RTPHeader& operator=(const RTPHeader& other);
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bool markerBit;
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uint8_t payloadType;
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uint16_t sequenceNumber;
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uint32_t timestamp;
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uint32_t ssrc;
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uint8_t numCSRCs;
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uint32_t arrOfCSRCs[kRtpCsrcSize];
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size_t paddingLength;
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size_t headerLength;
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int payload_type_frequency;
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RTPHeaderExtension extension;
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};
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// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
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// RTCP mode is described by RFC 5506.
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enum class RtcpMode { kOff, kCompound, kReducedSize };
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enum NetworkState {
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kNetworkUp,
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kNetworkDown,
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};
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} // namespace webrtc
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#endif // API_RTP_HEADERS_H_
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