Line data Source code
1 : /*
2 : * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
3 : *
4 : * Use of this source code is governed by a BSD-style license
5 : * that can be found in the LICENSE file in the root of the source
6 : * tree. An additional intellectual property rights grant can be found
7 : * in the file PATENTS. All contributing project authors may
8 : * be found in the AUTHORS file in the root of the source tree.
9 : */
10 :
11 : #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
12 : #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
13 :
14 : #include "webrtc/base/constructormagic.h"
15 : #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
16 : #include "webrtc/typedefs.h"
17 :
18 : namespace webrtc {
19 :
20 : // Forward declaration.
21 : struct RTPHeader;
22 :
23 : class Rtcp {
24 : public:
25 0 : Rtcp() {
26 0 : Init(0);
27 0 : }
28 :
29 0 : ~Rtcp() {}
30 :
31 : // Resets the RTCP statistics, and sets the first received sequence number.
32 : void Init(uint16_t start_sequence_number);
33 :
34 : // Updates the RTCP statistics with a new received packet.
35 : void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
36 :
37 : // Returns the current RTCP statistics. If |no_reset| is true, the statistics
38 : // are not reset, otherwise they are.
39 : void GetStatistics(bool no_reset, RtcpStatistics* stats);
40 :
41 : private:
42 : uint16_t cycles_; // The number of wrap-arounds for the sequence number.
43 : uint16_t max_seq_no_; // The maximum sequence number received. Starts over
44 : // from 0 after wrap-around.
45 : uint16_t base_seq_no_; // The sequence number of the first received packet.
46 : uint32_t received_packets_; // The number of packets that have been received.
47 : uint32_t received_packets_prior_; // Number of packets received when last
48 : // report was generated.
49 : uint32_t expected_prior_; // Expected number of packets, at the time of the
50 : // last report.
51 : int64_t jitter_; // Current jitter value in Q4.
52 : int32_t transit_; // Clock difference for previous packet.
53 :
54 : RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp);
55 : };
56 :
57 : } // namespace webrtc
58 : #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
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