Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 99 insertions, 0 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
new file mode 100644
index 0000000000..af20f97e82
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#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 "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.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"
+
+namespace webrtc {
+
+const uint8_t kRtpMarkerBitMask = 0x80;
+
+RtpData* NullObjectRtpData();
+RtpFeedback* NullObjectRtpFeedback();
+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;
+    };
+}  // namespace RtpUtility
+}  // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_