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authordanilchap <danilchap@webrtc.org>2015-12-28 10:18:46 -0800
committerCommit bot <commit-bot@chromium.org>2015-12-28 18:18:52 +0000
commitf6975f46131981f83e0c88d276dee6b6c5753180 (patch)
tree121d207849f903418c61f2cc4cdee70c373809eb /webrtc/modules/rtp_rtcp/source/rtp_utility.h
parente0d56a72250340d820823ecce72c2ab4f12433d9 (diff)
downloadwebrtc-f6975f46131981f83e0c88d276dee6b6c5753180.tar.gz
[rtp_rtcp] Lint errors cleaned from rtp_utility
R=åsapersson BUG=webrtc:5277 Review URL: https://codereview.webrtc.org/1539423003 Cr-Commit-Position: refs/heads/master@{#11131}
Diffstat (limited to 'webrtc/modules/rtp_rtcp/source/rtp_utility.h')
-rw-r--r--webrtc/modules/rtp_rtcp/source/rtp_utility.h95
1 files changed, 37 insertions, 58 deletions
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index bdcb11ccc2..57f54c1afc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -11,8 +11,6 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
-#include <stddef.h> // size_t, ptrdiff_t
-
#include <map>
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
@@ -31,62 +29,43 @@ RtpAudioFeedback* NullObjectRtpAudioFeedback();
ReceiveStatistics* NullObjectReceiveStatistics();
namespace RtpUtility {
- // January 1970, in NTP seconds.
- const uint32_t NTP_JAN_1970 = 2208988800UL;
-
- // Magic NTP fractional unit.
- const double NTP_FRAC = 4.294967296E+9;
-
- struct Payload
- {
- char name[RTP_PAYLOAD_NAME_SIZE];
- bool audio;
- PayloadUnion typeSpecific;
- };
-
- typedef std::map<int8_t, Payload*> PayloadTypeMap;
-
- uint32_t pow2(uint8_t exp);
-
- // Returns true if |newTimestamp| is older than |existingTimestamp|.
- // |wrapped| will be set to true if there has been a wraparound between the
- // two timestamps.
- bool OldTimestamp(uint32_t newTimestamp,
- uint32_t existingTimestamp,
- bool* wrapped);
-
- bool StringCompare(const char* str1,
- const char* str2,
- const uint32_t length);
-
- // 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,
- RtpHeaderExtensionMap* ptrExtensionMap = NULL) const;
-
- private:
- void ParseOneByteExtensionHeader(
- RTPHeader& parsedPacket,
- const RtpHeaderExtensionMap* ptrExtensionMap,
- const uint8_t* ptrRTPDataExtensionEnd,
- const uint8_t* ptr) const;
-
- uint8_t ParsePaddingBytes(
- const uint8_t* ptrRTPDataExtensionEnd,
- const uint8_t* ptr) const;
-
- const uint8_t* const _ptrRTPDataBegin;
- const uint8_t* const _ptrRTPDataEnd;
- };
+
+struct Payload {
+ char name[RTP_PAYLOAD_NAME_SIZE];
+ bool audio;
+ PayloadUnion typeSpecific;
+};
+
+typedef std::map<int8_t, Payload*> PayloadTypeMap;
+
+bool StringCompare(const char* str1, const char* str2, const uint32_t length);
+
+// 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,
+ RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
+
+ private:
+ void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
+ const RtpHeaderExtensionMap* ptrExtensionMap,
+ const uint8_t* ptrRTPDataExtensionEnd,
+ const uint8_t* ptr) const;
+
+ uint8_t ParsePaddingBytes(const uint8_t* ptrRTPDataExtensionEnd,
+ const uint8_t* ptr) const;
+
+ const uint8_t* const _ptrRTPDataBegin;
+ const uint8_t* const _ptrRTPDataEnd;
+};
} // namespace RtpUtility
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_