webrtc_m130/media/base/rtputils.h
Zhi Huang 95e7dbb7c7 Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport.""
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}
2018-03-29 02:45:17 +00:00

91 lines
3.2 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 MEDIA_BASE_RTPUTILS_H_
#define MEDIA_BASE_RTPUTILS_H_
#include "rtc_base/byteorder.h"
namespace rtc {
struct PacketTimeUpdateParams;
} // namespace rtc
namespace cricket {
const size_t kMinRtpPacketLen = 12;
const size_t kMaxRtpPacketLen = 2048;
const size_t kMinRtcpPacketLen = 4;
struct RtpHeader {
int payload_type;
int seq_num;
uint32_t timestamp;
uint32_t ssrc;
};
enum RtcpTypes {
kRtcpTypeSR = 200, // Sender report payload type.
kRtcpTypeRR = 201, // Receiver report payload type.
kRtcpTypeSDES = 202, // SDES payload type.
kRtcpTypeBye = 203, // BYE payload type.
kRtcpTypeApp = 204, // APP payload type.
kRtcpTypeRTPFB = 205, // Transport layer Feedback message payload type.
kRtcpTypePSFB = 206, // Payload-specific Feedback message payload type.
};
bool GetRtpPayloadType(const void* data, size_t len, int* value);
bool GetRtpSeqNum(const void* data, size_t len, int* value);
bool GetRtpTimestamp(const void* data, size_t len, uint32_t* value);
bool GetRtpSsrc(const void* data, size_t len, uint32_t* value);
bool GetRtpHeaderLen(const void* data, size_t len, size_t* value);
bool GetRtcpType(const void* data, size_t len, int* value);
bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value);
bool GetRtpHeader(const void* data, size_t len, RtpHeader* header);
bool SetRtpSsrc(void* data, size_t len, uint32_t value);
// Assumes version 2, no padding, no extensions, no csrcs.
bool SetRtpHeader(void* data, size_t len, const RtpHeader& header);
bool IsRtpPacket(const void* data, size_t len);
// True if |payload type| is 0-127.
bool IsValidRtpPayloadType(int payload_type);
// True if |size| is appropriate for the indicated packet type.
bool IsValidRtpRtcpPacketSize(bool rtcp, size_t size);
// TODO(zstein): Consider using an enum instead of a bool to differentiate
// between RTP and RTCP.
// Returns "RTCP" or "RTP" according to |rtcp|.
const char* RtpRtcpStringLiteral(bool rtcp);
// Verifies that a packet has a valid RTP header.
bool ValidateRtpHeader(const uint8_t* rtp,
size_t length,
size_t* header_length);
// Helper method which updates the absolute send time extension if present.
bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
size_t length,
int extension_id,
uint64_t time_us);
// Applies specified |options| to the packet. It updates the absolute send time
// extension header if it is present present then updates HMAC.
bool ApplyPacketOptions(uint8_t* data,
size_t length,
const rtc::PacketTimeUpdateParams& packet_time_params,
uint64_t time_us);
} // namespace cricket
#endif // MEDIA_BASE_RTPUTILS_H_