33c81d0561
Reason for revert: Breaks downstream projects. Original issue's description: > Remove RTPPayloadStrategy and simplify RTPPayloadRegistry > > This CL removes RTPPayloadStrategy that is currently used to handle > audio/video specific aspects of payload handling. Instead, the audio and > video specific aspects will now have different functions, with linear > code flow. > > This CL does not contain any functional changes, and is just a > preparation for future CL:s. > > The main purpose with this CL is to add this function: > bool PayloadIsCompatible(const RtpUtility::Payload& payload, > const webrtc::VideoCodec& video_codec); > that can easily be extended in a future CL to look at video codec > specific information. > > BUG=webrtc:6743 > > Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166 > Cr-Commit-Position: refs/heads/master@{#15232} TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6743 Review-Url: https://codereview.webrtc.org/2528993002 Cr-Commit-Position: refs/heads/master@{#15234}
69 lines
2.1 KiB
C++
69 lines
2.1 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
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#include <map>
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#include "webrtc/base/deprecation.h"
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#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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const uint8_t kRtpMarkerBitMask = 0x80;
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RtpData* NullObjectRtpData();
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RtpFeedback* NullObjectRtpFeedback();
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ReceiveStatistics* NullObjectReceiveStatistics();
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namespace RtpUtility {
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struct Payload {
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char name[RTP_PAYLOAD_NAME_SIZE];
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bool audio;
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PayloadUnion typeSpecific;
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};
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typedef std::map<int8_t, Payload*> PayloadTypeMap;
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bool StringCompare(const char* str1, const char* str2, const uint32_t length);
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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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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 // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
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