Support for Signal calls.

Merge in RedPhone

// FREEBIE

jni/webrtc/modules/audio_coding/neteq/rtcp.h

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2011 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_AUDIO_CODING_NETEQ_RTCP_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Forward declaration.
+struct RTPHeader;
+
+class Rtcp {
+ public:
+  Rtcp() {
+    Init(0);
+  }
+
+  ~Rtcp() {}
+
+  // Resets the RTCP statistics, and sets the first received sequence number.
+  void Init(uint16_t start_sequence_number);
+
+  // Updates the RTCP statistics with a new received packet.
+  void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
+
+  // Returns the current RTCP statistics. If |no_reset| is true, the statistics
+  // are not reset, otherwise they are.
+  void GetStatistics(bool no_reset, RtcpStatistics* stats);
+
+ private:
+  uint16_t cycles_;  // The number of wrap-arounds for the sequence number.
+  uint16_t max_seq_no_;  // The maximum sequence number received. Starts over
+                         // from 0 after wrap-around.
+  uint16_t base_seq_no_;  // The sequence number of the first received packet.
+  uint32_t received_packets_;  // The number of packets that have been received.
+  uint32_t received_packets_prior_;  // Number of packets received when last
+                                     // report was generated.
+  uint32_t expected_prior_;  // Expected number of packets, at the time of the
+                             // last report.
+  uint32_t jitter_;  // Current jitter value.
+  int32_t transit_;  // Clock difference for previous packet.
+
+  DISALLOW_COPY_AND_ASSIGN(Rtcp);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_