ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
97 lines
3.3 KiB
C++
97 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/neteq/rtcp.h"
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#include <string.h>
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#include <algorithm>
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/modules/include/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_sequence_number =
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(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 =
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stats->extended_max_sequence_number - 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 = std::min(0xFFU, (lost << 8) / expected_since_last);
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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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