webrtc/modules/rtp_rtcp/source/rtp_utility.h
Niels Möller 3ea55d56eb Reland "Delete RtpUtility::Payload, and refactor RTPSender to not use it"
This is a reland of 171df93262

Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
>
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
>
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}

Tbr: danilchap@webrtc.org
Bug: webrtc:6883
Change-Id: I30771b86bbe50de609353e23e80dc532dc884ad4
Reviewed-on: https://webrtc-review.googlesource.com/c/119661
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26394}
2019-01-24 16:35:00 +00:00

54 lines
1.7 KiB
C++

/*
* Copyright (c) 2012 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 MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
#include <stdint.h>
#include <algorithm>
#include "absl/strings/string_view.h"
#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
const uint8_t kRtpMarkerBitMask = 0x80;
namespace RtpUtility {
// Round up to the nearest size that is a multiple of 4.
size_t Word32Align(size_t size);
class RtpHeaderParser {
public:
RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
~RtpHeaderParser();
bool RTCP() const;
bool ParseRtcp(RTPHeader* header) const;
bool Parse(RTPHeader* parsedPacket,
const RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
private:
void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
const RtpHeaderExtensionMap* ptrExtensionMap,
const uint8_t* ptrRTPDataExtensionEnd,
const uint8_t* ptr) const;
const uint8_t* const _ptrRTPDataBegin;
const uint8_t* const _ptrRTPDataEnd;
};
} // namespace RtpUtility
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_