Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 46 insertions, 68 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index af20f97e82..23c175356a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -11,10 +11,11 @@
 #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/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/base/deprecation.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
@@ -29,71 +30,48 @@ 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;
-
-    // Return the current RTP timestamp from the NTP timestamp
-    // returned by the specified clock.
-    uint32_t GetCurrentRTP(Clock* clock, uint32_t freq);
-
-    // Return the current RTP absolute timestamp.
-    uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec,
-                                 uint32_t NTPfrac,
-                                 uint32_t freq);
-
-    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;
+  RTC_DEPRECATED bool Parse(
+      RTPHeader& parsedPacket,  // NOLINT(runtime/references)
+      RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const {
+    return Parse(&parsedPacket, ptrExtensionMap);
+  }
+
+ 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_