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diff --git a/third_party/libwebrtc/media/base/media_channel_impl.cc b/third_party/libwebrtc/media/base/media_channel_impl.cc
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+++ b/third_party/libwebrtc/media/base/media_channel_impl.cc
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+/*
+ * Copyright (c) 2018 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 "media/base/media_channel_impl.h"
+
+#include <map>
+#include <string>
+#include <type_traits>
+#include <utility>
+
+#include "absl/functional/any_invocable.h"
+#include "api/audio_options.h"
+#include "api/media_stream_interface.h"
+#include "api/rtc_error.h"
+#include "api/rtp_sender_interface.h"
+#include "api/units/time_delta.h"
+#include "api/video/video_timing.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "common_video/include/quality_limitation_reason.h"
+#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/rtp_utils.h"
+#include "media/base/stream_params.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "rtc_base/checks.h"
+
+namespace cricket {
+using webrtc::FrameDecryptorInterface;
+using webrtc::FrameEncryptorInterface;
+using webrtc::FrameTransformerInterface;
+using webrtc::PendingTaskSafetyFlag;
+using webrtc::SafeTask;
+using webrtc::TaskQueueBase;
+using webrtc::VideoTrackInterface;
+
+VideoOptions::VideoOptions()
+ : content_hint(VideoTrackInterface::ContentHint::kNone) {}
+VideoOptions::~VideoOptions() = default;
+
+MediaChannel::MediaChannel(TaskQueueBase* network_thread, bool enable_dscp)
+ : enable_dscp_(enable_dscp),
+ network_safety_(PendingTaskSafetyFlag::CreateDetachedInactive()),
+ network_thread_(network_thread) {}
+
+MediaChannel::~MediaChannel() {
+ RTC_DCHECK(!network_interface_);
+}
+
+void MediaChannel::SetInterface(MediaChannelNetworkInterface* iface) {
+ RTC_DCHECK_RUN_ON(network_thread_);
+ iface ? network_safety_->SetAlive() : network_safety_->SetNotAlive();
+ network_interface_ = iface;
+ UpdateDscp();
+}
+
+int MediaChannel::GetRtpSendTimeExtnId() const {
+ return -1;
+}
+
+void MediaChannel::SetFrameEncryptor(
+ uint32_t ssrc,
+ rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
+ // Placeholder should be pure virtual once internal supports it.
+}
+
+void MediaChannel::SetFrameDecryptor(
+ uint32_t ssrc,
+ rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
+ // Placeholder should be pure virtual once internal supports it.
+}
+
+bool MediaChannel::SendPacket(rtc::CopyOnWriteBuffer* packet,
+ const rtc::PacketOptions& options) {
+ return DoSendPacket(packet, false, options);
+}
+
+bool MediaChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet,
+ const rtc::PacketOptions& options) {
+ return DoSendPacket(packet, true, options);
+}
+
+int MediaChannel::SetOption(MediaChannelNetworkInterface::SocketType type,
+ rtc::Socket::Option opt,
+ int option) {
+ RTC_DCHECK_RUN_ON(network_thread_);
+ return SetOptionLocked(type, opt, option);
+}
+
+// Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
+// Set to true if it's allowed to mix one- and two-byte RTP header extensions
+// in the same stream. The setter and getter must only be called from
+// worker_thread.
+void MediaChannel::SetExtmapAllowMixed(bool extmap_allow_mixed) {
+ extmap_allow_mixed_ = extmap_allow_mixed;
+}
+
+bool MediaChannel::ExtmapAllowMixed() const {
+ return extmap_allow_mixed_;
+}
+
+bool MediaChannel::HasNetworkInterface() const {
+ RTC_DCHECK_RUN_ON(network_thread_);
+ return network_interface_ != nullptr;
+}
+
+void MediaChannel::SetEncoderToPacketizerFrameTransformer(
+ uint32_t ssrc,
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {}
+
+void MediaChannel::SetDepacketizerToDecoderFrameTransformer(
+ uint32_t ssrc,
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {}
+
+int MediaChannel::SetOptionLocked(MediaChannelNetworkInterface::SocketType type,
+ rtc::Socket::Option opt,
+ int option) {
+ if (!network_interface_)
+ return -1;
+ return network_interface_->SetOption(type, opt, option);
+}
+
+bool MediaChannel::DscpEnabled() const {
+ return enable_dscp_;
+}
+
+// This is the DSCP value used for both RTP and RTCP channels if DSCP is
+// enabled. It can be changed at any time via `SetPreferredDscp`.
+rtc::DiffServCodePoint MediaChannel::PreferredDscp() const {
+ RTC_DCHECK_RUN_ON(network_thread_);
+ return preferred_dscp_;
+}
+
+void MediaChannel::SetPreferredDscp(rtc::DiffServCodePoint new_dscp) {
+ if (!network_thread_->IsCurrent()) {
+ // This is currently the common path as the derived channel classes
+ // get called on the worker thread. There are still some tests though
+ // that call directly on the network thread.
+ network_thread_->PostTask(SafeTask(
+ network_safety_, [this, new_dscp]() { SetPreferredDscp(new_dscp); }));
+ return;
+ }
+
+ RTC_DCHECK_RUN_ON(network_thread_);
+ if (new_dscp == preferred_dscp_)
+ return;
+
+ preferred_dscp_ = new_dscp;
+ UpdateDscp();
+}
+
+rtc::scoped_refptr<PendingTaskSafetyFlag> MediaChannel::network_safety() {
+ return network_safety_;
+}
+
+void MediaChannel::UpdateDscp() {
+ rtc::DiffServCodePoint value =
+ enable_dscp_ ? preferred_dscp_ : rtc::DSCP_DEFAULT;
+ int ret = SetOptionLocked(MediaChannelNetworkInterface::ST_RTP,
+ rtc::Socket::OPT_DSCP, value);
+ if (ret == 0)
+ SetOptionLocked(MediaChannelNetworkInterface::ST_RTCP,
+ rtc::Socket::OPT_DSCP, value);
+}
+
+bool MediaChannel::DoSendPacket(rtc::CopyOnWriteBuffer* packet,
+ bool rtcp,
+ const rtc::PacketOptions& options) {
+ RTC_DCHECK_RUN_ON(network_thread_);
+ if (!network_interface_)
+ return false;
+
+ return (!rtcp) ? network_interface_->SendPacket(packet, options)
+ : network_interface_->SendRtcp(packet, options);
+}
+
+void MediaChannel::SendRtp(const uint8_t* data,
+ size_t len,
+ const webrtc::PacketOptions& options) {
+ auto send =
+ [this, packet_id = options.packet_id,
+ included_in_feedback = options.included_in_feedback,
+ included_in_allocation = options.included_in_allocation,
+ packet = rtc::CopyOnWriteBuffer(data, len, kMaxRtpPacketLen)]() mutable {
+ rtc::PacketOptions rtc_options;
+ rtc_options.packet_id = packet_id;
+ if (DscpEnabled()) {
+ rtc_options.dscp = PreferredDscp();
+ }
+ rtc_options.info_signaled_after_sent.included_in_feedback =
+ included_in_feedback;
+ rtc_options.info_signaled_after_sent.included_in_allocation =
+ included_in_allocation;
+ SendPacket(&packet, rtc_options);
+ };
+
+ // TODO(bugs.webrtc.org/11993): ModuleRtpRtcpImpl2 and related classes (e.g.
+ // RTCPSender) aren't aware of the network thread and may trigger calls to
+ // this function from different threads. Update those classes to keep
+ // network traffic on the network thread.
+ if (network_thread_->IsCurrent()) {
+ send();
+ } else {
+ network_thread_->PostTask(SafeTask(network_safety_, std::move(send)));
+ }
+}
+
+void MediaChannel::SendRtcp(const uint8_t* data, size_t len) {
+ auto send = [this, packet = rtc::CopyOnWriteBuffer(
+ data, len, kMaxRtpPacketLen)]() mutable {
+ rtc::PacketOptions rtc_options;
+ if (DscpEnabled()) {
+ rtc_options.dscp = PreferredDscp();
+ }
+ SendRtcp(&packet, rtc_options);
+ };
+
+ if (network_thread_->IsCurrent()) {
+ send();
+ } else {
+ network_thread_->PostTask(SafeTask(network_safety_, std::move(send)));
+ }
+}
+
+MediaSenderInfo::MediaSenderInfo() = default;
+MediaSenderInfo::~MediaSenderInfo() = default;
+
+MediaReceiverInfo::MediaReceiverInfo() = default;
+MediaReceiverInfo::~MediaReceiverInfo() = default;
+
+VoiceSenderInfo::VoiceSenderInfo() = default;
+VoiceSenderInfo::~VoiceSenderInfo() = default;
+
+VoiceReceiverInfo::VoiceReceiverInfo() = default;
+VoiceReceiverInfo::~VoiceReceiverInfo() = default;
+
+VideoSenderInfo::VideoSenderInfo() = default;
+VideoSenderInfo::~VideoSenderInfo() = default;
+
+VideoReceiverInfo::VideoReceiverInfo() = default;
+VideoReceiverInfo::~VideoReceiverInfo() = default;
+
+VoiceMediaInfo::VoiceMediaInfo() = default;
+VoiceMediaInfo::~VoiceMediaInfo() = default;
+
+VideoMediaInfo::VideoMediaInfo() = default;
+VideoMediaInfo::~VideoMediaInfo() = default;
+
+VideoMediaSendInfo::VideoMediaSendInfo() = default;
+VideoMediaSendInfo::~VideoMediaSendInfo() = default;
+
+VoiceMediaSendInfo::VoiceMediaSendInfo() = default;
+VoiceMediaSendInfo::~VoiceMediaSendInfo() = default;
+
+VideoMediaReceiveInfo::VideoMediaReceiveInfo() = default;
+VideoMediaReceiveInfo::~VideoMediaReceiveInfo() = default;
+
+VoiceMediaReceiveInfo::VoiceMediaReceiveInfo() = default;
+VoiceMediaReceiveInfo::~VoiceMediaReceiveInfo() = default;
+
+AudioSendParameters::AudioSendParameters() = default;
+AudioSendParameters::~AudioSendParameters() = default;
+
+std::map<std::string, std::string> AudioSendParameters::ToStringMap() const {
+ auto params = RtpSendParameters<AudioCodec>::ToStringMap();
+ params["options"] = options.ToString();
+ return params;
+}
+
+cricket::MediaType VoiceMediaChannel::media_type() const {
+ return cricket::MediaType::MEDIA_TYPE_AUDIO;
+}
+
+VideoSendParameters::VideoSendParameters() = default;
+VideoSendParameters::~VideoSendParameters() = default;
+
+std::map<std::string, std::string> VideoSendParameters::ToStringMap() const {
+ auto params = RtpSendParameters<VideoCodec>::ToStringMap();
+ params["conference_mode"] = (conference_mode ? "yes" : "no");
+ return params;
+}
+
+cricket::MediaType VideoMediaChannel::media_type() const {
+ return cricket::MediaType::MEDIA_TYPE_VIDEO;
+}
+
+void VideoMediaChannel::SetVideoCodecSwitchingEnabled(bool enabled) {}
+
+} // namespace cricket