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81d4bf7af6
This reverts commit 171df93262.
Reason for revert: Breaks downstream project
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,brandtr@webrtc.org,nisse@webrtc.org
Change-Id: I76489c29541827aaba72515a76db54bdb7495e28
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6883
Reviewed-on: https://webrtc-review.googlesource.com/c/119640
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26385}
64 lines
2 KiB
C++
64 lines
2 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 MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
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#include <stdint.h>
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#include <algorithm>
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#include "absl/strings/string_view.h"
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#include "api/rtp_headers.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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const uint8_t kRtpMarkerBitMask = 0x80;
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namespace RtpUtility {
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struct Payload {
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Payload(absl::string_view payload_name, const PayloadUnion& pu)
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: typeSpecific(pu) {
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size_t clipped_size = payload_name.copy(name, sizeof(name) - 1);
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name[clipped_size] = '\0';
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}
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char name[RTP_PAYLOAD_NAME_SIZE];
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PayloadUnion typeSpecific;
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};
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// Round up to the nearest size that is a multiple of 4.
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size_t Word32Align(size_t size);
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class RtpHeaderParser {
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public:
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RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
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~RtpHeaderParser();
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bool RTCP() const;
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bool ParseRtcp(RTPHeader* header) const;
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bool Parse(RTPHeader* parsedPacket,
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const RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
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private:
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void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
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const RtpHeaderExtensionMap* ptrExtensionMap,
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const uint8_t* ptrRTPDataExtensionEnd,
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const uint8_t* ptr) const;
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const uint8_t* const _ptrRTPDataBegin;
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const uint8_t* const _ptrRTPDataEnd;
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};
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} // namespace RtpUtility
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
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