ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
100 lines
3.2 KiB
C++
100 lines
3.2 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 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 <stddef.h> // size_t, ptrdiff_t
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/include/receive_statistics.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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RtpAudioFeedback* NullObjectRtpAudioFeedback();
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ReceiveStatistics* NullObjectReceiveStatistics();
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namespace RtpUtility {
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// January 1970, in NTP seconds.
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const uint32_t NTP_JAN_1970 = 2208988800UL;
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// Magic NTP fractional unit.
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const double NTP_FRAC = 4.294967296E+9;
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struct Payload
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{
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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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// Return the current RTP timestamp from the NTP timestamp
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// returned by the specified clock.
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uint32_t GetCurrentRTP(Clock* clock, uint32_t freq);
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// Return the current RTP absolute timestamp.
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uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec,
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uint32_t NTPfrac,
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uint32_t freq);
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uint32_t pow2(uint8_t exp);
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// Returns true if |newTimestamp| is older than |existingTimestamp|.
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// |wrapped| will be set to true if there has been a wraparound between the
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// two timestamps.
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bool OldTimestamp(uint32_t newTimestamp,
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uint32_t existingTimestamp,
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bool* wrapped);
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bool StringCompare(const char* str1,
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const char* str2,
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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 = NULL) const;
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private:
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void ParseOneByteExtensionHeader(
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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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uint8_t ParsePaddingBytes(
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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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