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/interface/module_common_types.h" 19 20 namespace webrtc { 21 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 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 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
