This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e. > > > > Reason for revert: Broke chromium tests. > > Original change's description: > > > Replace BundleFilter with RtpDemuxer in RtpTransport. > > > > > > BundleFilter is replaced by RtpDemuxer in RtpTransport for payload > > > type-based demuxing. RtpTransport will support MID-based demuxing later. > > > > > > Each BaseChannel has its own RTP demuxing criteria and when connecting > > > to the RtpTransport, BaseChannel will register itself as a demuxer sink. > > > > > > The inheritance model is changed. New inheritance chain: > > > DtlsSrtpTransport->SrtpTransport->RtpTranpsort > > > > > > NOTE: > > > When RTCP packets are received, Call::DeliverRtcp will be called for > > > multiple times (webrtc:9035) which is an existing issue. With this CL, > > > it will become more of a problem and should be fixed. > > > > > > Bug: webrtc:8587 > > > Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0 > > > Reviewed-on: https://webrtc-review.googlesource.com/61360 > > > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#22613} > > > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > > > Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:8587 > > Reviewed-on: https://webrtc-review.googlesource.com/64860 > > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22614} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
91 lines
3.2 KiB
C++
91 lines
3.2 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 MEDIA_BASE_RTPUTILS_H_
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#define MEDIA_BASE_RTPUTILS_H_
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#include "rtc_base/byteorder.h"
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namespace rtc {
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struct PacketTimeUpdateParams;
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} // namespace rtc
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namespace cricket {
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const size_t kMinRtpPacketLen = 12;
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const size_t kMaxRtpPacketLen = 2048;
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const size_t kMinRtcpPacketLen = 4;
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struct RtpHeader {
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int payload_type;
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int seq_num;
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uint32_t timestamp;
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uint32_t ssrc;
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};
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enum RtcpTypes {
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kRtcpTypeSR = 200, // Sender report payload type.
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kRtcpTypeRR = 201, // Receiver report payload type.
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kRtcpTypeSDES = 202, // SDES payload type.
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kRtcpTypeBye = 203, // BYE payload type.
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kRtcpTypeApp = 204, // APP payload type.
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kRtcpTypeRTPFB = 205, // Transport layer Feedback message payload type.
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kRtcpTypePSFB = 206, // Payload-specific Feedback message payload type.
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};
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bool GetRtpPayloadType(const void* data, size_t len, int* value);
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bool GetRtpSeqNum(const void* data, size_t len, int* value);
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bool GetRtpTimestamp(const void* data, size_t len, uint32_t* value);
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bool GetRtpSsrc(const void* data, size_t len, uint32_t* value);
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bool GetRtpHeaderLen(const void* data, size_t len, size_t* value);
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bool GetRtcpType(const void* data, size_t len, int* value);
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bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value);
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bool GetRtpHeader(const void* data, size_t len, RtpHeader* header);
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bool SetRtpSsrc(void* data, size_t len, uint32_t value);
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// Assumes version 2, no padding, no extensions, no csrcs.
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bool SetRtpHeader(void* data, size_t len, const RtpHeader& header);
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bool IsRtpPacket(const void* data, size_t len);
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// True if |payload type| is 0-127.
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bool IsValidRtpPayloadType(int payload_type);
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// True if |size| is appropriate for the indicated packet type.
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bool IsValidRtpRtcpPacketSize(bool rtcp, size_t size);
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// TODO(zstein): Consider using an enum instead of a bool to differentiate
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// between RTP and RTCP.
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// Returns "RTCP" or "RTP" according to |rtcp|.
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const char* RtpRtcpStringLiteral(bool rtcp);
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// Verifies that a packet has a valid RTP header.
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bool ValidateRtpHeader(const uint8_t* rtp,
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size_t length,
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size_t* header_length);
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// Helper method which updates the absolute send time extension if present.
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bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
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size_t length,
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int extension_id,
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uint64_t time_us);
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// Applies specified |options| to the packet. It updates the absolute send time
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// extension header if it is present present then updates HMAC.
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bool ApplyPacketOptions(uint8_t* data,
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size_t length,
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const rtc::PacketTimeUpdateParams& packet_time_params,
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uint64_t time_us);
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} // namespace cricket
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#endif // MEDIA_BASE_RTPUTILS_H_
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