| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h" |
| |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/logging.h" |
| |
| namespace webrtc { |
| |
| PlayoutDelayOracle::PlayoutDelayOracle() |
| : high_sequence_number_(0), |
| send_playout_delay_(false), |
| ssrc_(0), |
| playout_delay_{-1, -1} {} |
| |
| PlayoutDelayOracle::~PlayoutDelayOracle() {} |
| |
| void PlayoutDelayOracle::UpdateRequest(uint32_t ssrc, |
| PlayoutDelay playout_delay, |
| uint16_t seq_num) { |
| rtc::CritScope lock(&crit_sect_); |
| RTC_DCHECK_LE(playout_delay.min_ms, PlayoutDelayLimits::kMaxMs); |
| RTC_DCHECK_LE(playout_delay.max_ms, PlayoutDelayLimits::kMaxMs); |
| RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms); |
| int64_t unwrapped_seq_num = unwrapper_.Unwrap(seq_num); |
| if (playout_delay.min_ms >= 0 && |
| playout_delay.min_ms != playout_delay_.min_ms) { |
| send_playout_delay_ = true; |
| playout_delay_.min_ms = playout_delay.min_ms; |
| high_sequence_number_ = unwrapped_seq_num; |
| } |
| |
| if (playout_delay.max_ms >= 0 && |
| playout_delay.max_ms != playout_delay_.max_ms) { |
| send_playout_delay_ = true; |
| playout_delay_.max_ms = playout_delay.max_ms; |
| high_sequence_number_ = unwrapped_seq_num; |
| } |
| ssrc_ = ssrc; |
| } |
| |
| // If an ACK is received on the packet containing the playout delay extension, |
| // we stop sending the extension on future packets. |
| void PlayoutDelayOracle::OnReceivedRtcpReportBlocks( |
| const ReportBlockList& report_blocks) { |
| rtc::CritScope lock(&crit_sect_); |
| for (const RTCPReportBlock& report_block : report_blocks) { |
| if ((ssrc_ == report_block.source_ssrc) && send_playout_delay_ && |
| (report_block.extended_highest_sequence_number > |
| high_sequence_number_)) { |
| send_playout_delay_ = false; |
| } |
| } |
| } |
| |
| } // namespace webrtc |