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 #include "webrtc/modules/audio_coding/neteq/rtcp.h"
12 
13 #include <string.h>
14 
15 #include <algorithm>
16 
17 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
18 #include "webrtc/modules/include/module_common_types.h"
19 
20 namespace webrtc {
21 
Init(uint16_t start_sequence_number)22 void Rtcp::Init(uint16_t start_sequence_number) {
23   cycles_ = 0;
24   max_seq_no_ = start_sequence_number;
25   base_seq_no_ = start_sequence_number;
26   received_packets_ = 0;
27   received_packets_prior_ = 0;
28   expected_prior_ = 0;
29   jitter_ = 0;
30   transit_ = 0;
31 }
32 
Update(const RTPHeader & rtp_header,uint32_t receive_timestamp)33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) {
34   // Update number of received packets, and largest packet number received.
35   received_packets_++;
36   int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_;
37   if (sn_diff >= 0) {
38     if (rtp_header.sequenceNumber < max_seq_no_) {
39       // Wrap-around detected.
40       cycles_++;
41     }
42     max_seq_no_ = rtp_header.sequenceNumber;
43   }
44 
45   // Calculate jitter according to RFC 3550, and update previous timestamps.
46   // Note that the value in |jitter_| is in Q4.
47   if (received_packets_ > 1) {
48     int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_);
49     ts_diff = WEBRTC_SPL_ABS_W32(ts_diff);
50     int32_t jitter_diff = (ts_diff << 4) - jitter_;
51     // Calculate 15 * jitter_ / 16 + jitter_diff / 16 (with proper rounding).
52     jitter_ = jitter_ + ((jitter_diff + 8) >> 4);
53   }
54   transit_ = rtp_header.timestamp - receive_timestamp;
55 }
56 
GetStatistics(bool no_reset,RtcpStatistics * stats)57 void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) {
58   // Extended highest sequence number received.
59   stats->extended_max_sequence_number =
60       (static_cast<int>(cycles_) << 16) + max_seq_no_;
61 
62   // Calculate expected number of packets and compare it with the number of
63   // packets that were actually received. The cumulative number of lost packets
64   // can be extracted.
65   uint32_t expected_packets =
66       stats->extended_max_sequence_number - base_seq_no_ + 1;
67   if (received_packets_ == 0) {
68     // No packets received, assume none lost.
69     stats->cumulative_lost = 0;
70   } else if (expected_packets > received_packets_) {
71     stats->cumulative_lost = expected_packets - received_packets_;
72     if (stats->cumulative_lost > 0xFFFFFF) {
73       stats->cumulative_lost = 0xFFFFFF;
74     }
75   } else {
76     stats->cumulative_lost = 0;
77   }
78 
79   // Fraction lost since last report.
80   uint32_t expected_since_last = expected_packets - expected_prior_;
81   uint32_t received_since_last = received_packets_ - received_packets_prior_;
82   if (!no_reset) {
83     expected_prior_ = expected_packets;
84     received_packets_prior_ = received_packets_;
85   }
86   int32_t lost = expected_since_last - received_since_last;
87   if (expected_since_last == 0 || lost <= 0 || received_packets_ == 0) {
88     stats->fraction_lost = 0;
89   } else {
90     stats->fraction_lost = std::min(0xFFU, (lost << 8) / expected_since_last);
91   }
92 
93   stats->jitter = jitter_ >> 4;  // Scaling from Q4.
94 }
95 
96 }  // namespace webrtc
97