mirror of
https://github.com/mollyim/webrtc.git
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d264df587f
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $dir -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}
86 lines
3 KiB
C++
86 lines
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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#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
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#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
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#include <vector>
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/source/rtcp_packet.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
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namespace webrtc {
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namespace rtcp {
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class CommonHeader;
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// From RFC 3611: RTP Control Protocol Extended Reports (RTCP XR).
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class ExtendedReports : public RtcpPacket {
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public:
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static constexpr uint8_t kPacketType = 207;
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static constexpr size_t kMaxNumberOfDlrrItems = 50;
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ExtendedReports();
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~ExtendedReports() override;
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// Parse assumes header is already parsed and validated.
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bool Parse(const CommonHeader& packet);
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void SetSenderSsrc(uint32_t ssrc) { sender_ssrc_ = ssrc; }
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void SetRrtr(const Rrtr& rrtr);
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bool AddDlrrItem(const ReceiveTimeInfo& time_info);
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void SetVoipMetric(const VoipMetric& voip_metric);
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void SetTargetBitrate(const TargetBitrate& target_bitrate);
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uint32_t sender_ssrc() const { return sender_ssrc_; }
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const absl::optional<Rrtr>& rrtr() const { return rrtr_block_; }
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const Dlrr& dlrr() const { return dlrr_block_; }
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const absl::optional<VoipMetric>& voip_metric() const {
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return voip_metric_block_;
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}
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const absl::optional<TargetBitrate>& target_bitrate() const {
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return target_bitrate_;
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}
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size_t BlockLength() const override;
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bool Create(uint8_t* packet,
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size_t* index,
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size_t max_length,
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PacketReadyCallback callback) const override;
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private:
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static constexpr size_t kXrBaseLength = 4;
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size_t RrtrLength() const { return rrtr_block_ ? Rrtr::kLength : 0; }
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size_t DlrrLength() const { return dlrr_block_.BlockLength(); }
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size_t VoipMetricLength() const {
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return voip_metric_block_ ? VoipMetric::kLength : 0;
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}
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size_t TargetBitrateLength() const;
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void ParseRrtrBlock(const uint8_t* block, uint16_t block_length);
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void ParseDlrrBlock(const uint8_t* block, uint16_t block_length);
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void ParseVoipMetricBlock(const uint8_t* block, uint16_t block_length);
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void ParseTargetBitrateBlock(const uint8_t* block, uint16_t block_length);
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uint32_t sender_ssrc_;
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absl::optional<Rrtr> rrtr_block_;
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Dlrr dlrr_block_; // Dlrr without items treated same as no dlrr block.
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absl::optional<VoipMetric> voip_metric_block_;
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absl::optional<TargetBitrate> target_bitrate_;
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};
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} // namespace rtcp
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
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