With this CL, the decision if an RTP packet should be sent as ect(1) is made in RtpControllerSend depending on if RFC 8888 has been negotiated and if CCFB is received with ECN enabled. Since webrtc does not yet adapt to ECN feedback, packets are sent as ECT(1) until the first feedback is received. Change-Id: Iddf63849328afbe54a7c8f921f2e8db134aeff6a Bug: webrtc:42225697 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367388 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43609}
307 lines
10 KiB
C++
307 lines
10 KiB
C++
/*
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* Copyright (c) 2018 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 "media/base/media_channel_impl.h"
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#include <map>
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#include <string>
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#include <type_traits>
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#include <utility>
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#include "absl/functional/any_invocable.h"
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#include "api/audio_options.h"
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#include "api/media_stream_interface.h"
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#include "api/rtc_error.h"
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#include "api/rtp_sender_interface.h"
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#include "api/units/time_delta.h"
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#include "api/video/video_timing.h"
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#include "api/video_codecs/scalability_mode.h"
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#include "common_video/include/quality_limitation_reason.h"
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#include "media/base/codec.h"
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#include "media/base/media_channel.h"
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#include "media/base/rtp_utils.h"
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#include "media/base/stream_params.h"
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#include "modules/rtp_rtcp/include/report_block_data.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
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RTCError error) {
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if (callback) {
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std::move(callback)(error);
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callback = nullptr;
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}
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return error;
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}
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} // namespace webrtc
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namespace cricket {
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using webrtc::FrameDecryptorInterface;
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using webrtc::FrameEncryptorInterface;
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using webrtc::FrameTransformerInterface;
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using webrtc::PendingTaskSafetyFlag;
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using webrtc::SafeTask;
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using webrtc::TaskQueueBase;
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using webrtc::VideoTrackInterface;
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VideoOptions::VideoOptions()
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: content_hint(VideoTrackInterface::ContentHint::kNone) {}
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VideoOptions::~VideoOptions() = default;
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MediaChannelUtil::MediaChannelUtil(TaskQueueBase* network_thread,
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bool enable_dscp)
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: transport_(network_thread, enable_dscp) {}
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MediaChannelUtil::~MediaChannelUtil() {}
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void MediaChannelUtil::SetInterface(MediaChannelNetworkInterface* iface) {
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transport_.SetInterface(iface);
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}
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int MediaChannelUtil::GetRtpSendTimeExtnId() const {
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return -1;
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}
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bool MediaChannelUtil::SendPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options) {
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return transport_.DoSendPacket(packet, false, options);
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}
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bool MediaChannelUtil::SendRtcp(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options) {
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return transport_.DoSendPacket(packet, true, options);
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}
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int MediaChannelUtil::SetOption(MediaChannelNetworkInterface::SocketType type,
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rtc::Socket::Option opt,
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int option) {
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return transport_.SetOption(type, opt, option);
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}
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// Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
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// Set to true if it's allowed to mix one- and two-byte RTP header extensions
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// in the same stream. The setter and getter must only be called from
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// worker_thread.
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void MediaChannelUtil::SetExtmapAllowMixed(bool extmap_allow_mixed) {
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extmap_allow_mixed_ = extmap_allow_mixed;
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}
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bool MediaChannelUtil::ExtmapAllowMixed() const {
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return extmap_allow_mixed_;
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}
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bool MediaChannelUtil::HasNetworkInterface() const {
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return transport_.HasNetworkInterface();
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}
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bool MediaChannelUtil::DscpEnabled() const {
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return transport_.DscpEnabled();
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}
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void MediaChannelUtil::SetPreferredDscp(rtc::DiffServCodePoint new_dscp) {
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transport_.SetPreferredDscp(new_dscp);
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}
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MediaSenderInfo::MediaSenderInfo() = default;
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MediaSenderInfo::~MediaSenderInfo() = default;
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MediaReceiverInfo::MediaReceiverInfo() = default;
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MediaReceiverInfo::~MediaReceiverInfo() = default;
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VoiceSenderInfo::VoiceSenderInfo() = default;
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VoiceSenderInfo::~VoiceSenderInfo() = default;
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VoiceReceiverInfo::VoiceReceiverInfo() = default;
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VoiceReceiverInfo::~VoiceReceiverInfo() = default;
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VideoSenderInfo::VideoSenderInfo() = default;
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VideoSenderInfo::~VideoSenderInfo() = default;
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VideoReceiverInfo::VideoReceiverInfo() = default;
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VideoReceiverInfo::~VideoReceiverInfo() = default;
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VoiceMediaInfo::VoiceMediaInfo() = default;
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VoiceMediaInfo::~VoiceMediaInfo() = default;
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VideoMediaInfo::VideoMediaInfo() = default;
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VideoMediaInfo::~VideoMediaInfo() = default;
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VideoMediaSendInfo::VideoMediaSendInfo() = default;
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VideoMediaSendInfo::~VideoMediaSendInfo() = default;
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VoiceMediaSendInfo::VoiceMediaSendInfo() = default;
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VoiceMediaSendInfo::~VoiceMediaSendInfo() = default;
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VideoMediaReceiveInfo::VideoMediaReceiveInfo() = default;
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VideoMediaReceiveInfo::~VideoMediaReceiveInfo() = default;
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VoiceMediaReceiveInfo::VoiceMediaReceiveInfo() = default;
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VoiceMediaReceiveInfo::~VoiceMediaReceiveInfo() = default;
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AudioSenderParameter::AudioSenderParameter() = default;
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AudioSenderParameter::~AudioSenderParameter() = default;
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std::map<std::string, std::string> AudioSenderParameter::ToStringMap() const {
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auto params = SenderParameters::ToStringMap();
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params["options"] = options.ToString();
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return params;
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}
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VideoSenderParameters::VideoSenderParameters() = default;
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VideoSenderParameters::~VideoSenderParameters() = default;
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std::map<std::string, std::string> VideoSenderParameters::ToStringMap() const {
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auto params = SenderParameters::ToStringMap();
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params["conference_mode"] = (conference_mode ? "yes" : "no");
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return params;
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}
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// --------------------- MediaChannelUtil::TransportForMediaChannels -----
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MediaChannelUtil::TransportForMediaChannels::TransportForMediaChannels(
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webrtc::TaskQueueBase* network_thread,
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bool enable_dscp)
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: network_safety_(webrtc::PendingTaskSafetyFlag::CreateDetachedInactive()),
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network_thread_(network_thread),
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enable_dscp_(enable_dscp) {}
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MediaChannelUtil::TransportForMediaChannels::~TransportForMediaChannels() {
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RTC_DCHECK(!network_interface_);
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}
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bool MediaChannelUtil::TransportForMediaChannels::SendRtcp(
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rtc::ArrayView<const uint8_t> packet) {
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auto send = [this, packet = rtc::CopyOnWriteBuffer(
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packet, kMaxRtpPacketLen)]() mutable {
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rtc::PacketOptions rtc_options;
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if (DscpEnabled()) {
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rtc_options.dscp = PreferredDscp();
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}
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DoSendPacket(&packet, true, rtc_options);
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};
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if (network_thread_->IsCurrent()) {
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send();
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} else {
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network_thread_->PostTask(SafeTask(network_safety_, std::move(send)));
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}
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return true;
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}
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bool MediaChannelUtil::TransportForMediaChannels::SendRtp(
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rtc::ArrayView<const uint8_t> packet,
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const webrtc::PacketOptions& options) {
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auto send =
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[this, packet_id = options.packet_id,
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included_in_feedback = options.included_in_feedback,
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included_in_allocation = options.included_in_allocation,
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batchable = options.batchable,
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last_packet_in_batch = options.last_packet_in_batch,
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is_media = options.is_media, ect_1 = options.send_as_ect1,
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packet = rtc::CopyOnWriteBuffer(packet, kMaxRtpPacketLen)]() mutable {
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rtc::PacketOptions rtc_options;
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rtc_options.packet_id = packet_id;
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if (DscpEnabled()) {
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rtc_options.dscp = PreferredDscp();
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}
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rtc_options.info_signaled_after_sent.included_in_feedback =
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included_in_feedback;
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rtc_options.info_signaled_after_sent.included_in_allocation =
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included_in_allocation;
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rtc_options.info_signaled_after_sent.is_media = is_media;
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rtc_options.ecn_1 = ect_1;
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rtc_options.batchable = batchable;
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rtc_options.last_packet_in_batch = last_packet_in_batch;
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DoSendPacket(&packet, false, rtc_options);
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};
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// TODO(bugs.webrtc.org/11993): ModuleRtpRtcpImpl2 and related classes (e.g.
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// RTCPSender) aren't aware of the network thread and may trigger calls to
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// this function from different threads. Update those classes to keep
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// network traffic on the network thread.
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if (network_thread_->IsCurrent()) {
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send();
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} else {
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network_thread_->PostTask(SafeTask(network_safety_, std::move(send)));
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}
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return true;
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}
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void MediaChannelUtil::TransportForMediaChannels::SetInterface(
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MediaChannelNetworkInterface* iface) {
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RTC_DCHECK_RUN_ON(network_thread_);
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iface ? network_safety_->SetAlive() : network_safety_->SetNotAlive();
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network_interface_ = iface;
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UpdateDscp();
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}
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void MediaChannelUtil::TransportForMediaChannels::UpdateDscp() {
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rtc::DiffServCodePoint value =
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enable_dscp_ ? preferred_dscp_ : rtc::DSCP_DEFAULT;
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int ret = SetOptionLocked(MediaChannelNetworkInterface::ST_RTP,
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rtc::Socket::OPT_DSCP, value);
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if (ret == 0)
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SetOptionLocked(MediaChannelNetworkInterface::ST_RTCP,
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rtc::Socket::OPT_DSCP, value);
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}
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bool MediaChannelUtil::TransportForMediaChannels::DoSendPacket(
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rtc::CopyOnWriteBuffer* packet,
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bool rtcp,
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const rtc::PacketOptions& options) {
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RTC_DCHECK_RUN_ON(network_thread_);
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if (!network_interface_)
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return false;
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return (!rtcp) ? network_interface_->SendPacket(packet, options)
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: network_interface_->SendRtcp(packet, options);
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}
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int MediaChannelUtil::TransportForMediaChannels::SetOption(
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MediaChannelNetworkInterface::SocketType type,
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rtc::Socket::Option opt,
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int option) {
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RTC_DCHECK_RUN_ON(network_thread_);
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return SetOptionLocked(type, opt, option);
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}
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int MediaChannelUtil::TransportForMediaChannels::SetOptionLocked(
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MediaChannelNetworkInterface::SocketType type,
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rtc::Socket::Option opt,
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int option) {
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if (!network_interface_)
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return -1;
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return network_interface_->SetOption(type, opt, option);
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}
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void MediaChannelUtil::TransportForMediaChannels::SetPreferredDscp(
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rtc::DiffServCodePoint new_dscp) {
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if (!network_thread_->IsCurrent()) {
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// This is currently the common path as the derived channel classes
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// get called on the worker thread. There are still some tests though
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// that call directly on the network thread.
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network_thread_->PostTask(SafeTask(
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network_safety_, [this, new_dscp]() { SetPreferredDscp(new_dscp); }));
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return;
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}
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RTC_DCHECK_RUN_ON(network_thread_);
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if (new_dscp == preferred_dscp_)
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return;
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preferred_dscp_ = new_dscp;
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UpdateDscp();
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}
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} // namespace cricket
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