d94659dc27
This is the first public upload of the new NetEq, version 4. It has been through extensive internal review during the course of the project. TEST=trybots Review URL: https://webrtc-codereview.appspot.com/1073005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
96 lines
3.3 KiB
C++
96 lines
3.3 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/neteq4/rtcp.h"
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#include <string.h>
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/modules/interface/module_common_types.h"
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namespace webrtc {
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void Rtcp::Init(uint16_t start_sequence_number) {
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cycles_ = 0;
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max_seq_no_ = start_sequence_number;
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base_seq_no_ = start_sequence_number;
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received_packets_ = 0;
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received_packets_prior_ = 0;
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expected_prior_ = 0;
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jitter_ = 0;
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transit_ = 0;
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}
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void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) {
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// Update number of received packets, and largest packet number received.
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received_packets_++;
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int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_;
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if (sn_diff >= 0) {
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if (rtp_header.sequenceNumber < max_seq_no_) {
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// Wrap-around detected.
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cycles_++;
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}
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max_seq_no_ = rtp_header.sequenceNumber;
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}
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// Calculate jitter according to RFC 3550, and update previous timestamps.
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// Note that the value in |jitter_| is in Q4.
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if (received_packets_ > 1) {
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int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_);
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ts_diff = WEBRTC_SPL_ABS_W32(ts_diff);
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int32_t jitter_diff = (ts_diff << 4) - jitter_;
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// Calculate 15 * jitter_ / 16 + jitter_diff / 16 (with proper rounding).
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jitter_ = jitter_ + ((jitter_diff + 8) >> 4);
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}
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transit_ = rtp_header.timestamp - receive_timestamp;
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}
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void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) {
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// Extended highest sequence number received.
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stats->extended_max = (static_cast<int>(cycles_) << 16) + max_seq_no_;
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// Calculate expected number of packets and compare it with the number of
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// packets that were actually received. The cumulative number of lost packets
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// can be extracted.
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uint32_t expected_packets = stats->extended_max - base_seq_no_ + 1;
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if (received_packets_ == 0) {
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// No packets received, assume none lost.
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stats->cumulative_lost = 0;
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} else if (expected_packets > received_packets_) {
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stats->cumulative_lost = expected_packets - received_packets_;
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if (stats->cumulative_lost > 0xFFFFFF) {
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stats->cumulative_lost = 0xFFFFFF;
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}
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} else {
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stats->cumulative_lost = 0;
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}
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// Fraction lost since last report.
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uint32_t expected_since_last = expected_packets - expected_prior_;
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uint32_t received_since_last = received_packets_ - received_packets_prior_;
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if (!no_reset) {
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expected_prior_ = expected_packets;
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received_packets_prior_ = received_packets_;
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}
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int32_t lost = expected_since_last - received_since_last;
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if (expected_since_last == 0 || lost <= 0 || received_packets_ == 0) {
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stats->fraction_lost = 0;
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} else {
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stats->fraction_lost = (lost << 8) / expected_since_last;
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}
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if (stats->fraction_lost > 0xFF) {
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stats->fraction_lost = 0xFF;
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}
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stats->jitter = jitter_ >> 4; // Scaling from Q4.
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}
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} // namespace webrtc
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