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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
66 lines
2.3 KiB
C++
66 lines
2.3 KiB
C++
/*
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* Copyright (c) 2016 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 "modules/rtp_rtcp/source/playout_delay_oracle.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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PlayoutDelayOracle::PlayoutDelayOracle()
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: high_sequence_number_(0),
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send_playout_delay_(false),
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ssrc_(0),
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playout_delay_{-1, -1} {}
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PlayoutDelayOracle::~PlayoutDelayOracle() {}
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void PlayoutDelayOracle::UpdateRequest(uint32_t ssrc,
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PlayoutDelay playout_delay,
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uint16_t seq_num) {
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rtc::CritScope lock(&crit_sect_);
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RTC_DCHECK_LE(playout_delay.min_ms, PlayoutDelayLimits::kMaxMs);
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RTC_DCHECK_LE(playout_delay.max_ms, PlayoutDelayLimits::kMaxMs);
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RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms);
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int64_t unwrapped_seq_num = unwrapper_.Unwrap(seq_num);
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if (playout_delay.min_ms >= 0 &&
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playout_delay.min_ms != playout_delay_.min_ms) {
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send_playout_delay_ = true;
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playout_delay_.min_ms = playout_delay.min_ms;
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high_sequence_number_ = unwrapped_seq_num;
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}
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if (playout_delay.max_ms >= 0 &&
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playout_delay.max_ms != playout_delay_.max_ms) {
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send_playout_delay_ = true;
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playout_delay_.max_ms = playout_delay.max_ms;
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high_sequence_number_ = unwrapped_seq_num;
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}
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ssrc_ = ssrc;
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}
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// If an ACK is received on the packet containing the playout delay extension,
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// we stop sending the extension on future packets.
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void PlayoutDelayOracle::OnReceivedRtcpReportBlocks(
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const ReportBlockList& report_blocks) {
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rtc::CritScope lock(&crit_sect_);
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for (const RTCPReportBlock& report_block : report_blocks) {
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if ((ssrc_ == report_block.source_ssrc) && send_playout_delay_ &&
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(report_block.extended_highest_sequence_number >
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high_sequence_number_)) {
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send_playout_delay_ = false;
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}
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}
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}
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} // namespace webrtc
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