| /* |
| * Copyright 2004 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 <algorithm> |
| #include <iterator> |
| #include <utility> |
| |
| #include "pc/channel.h" |
| |
| #include "api/call/audio_sink.h" |
| #include "media/base/mediaconstants.h" |
| #include "media/base/rtputils.h" |
| #include "rtc_base/bind.h" |
| #include "rtc_base/byteorder.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copyonwritebuffer.h" |
| #include "rtc_base/dscp.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/networkroute.h" |
| #include "rtc_base/ptr_util.h" |
| #include "rtc_base/trace_event.h" |
| // Adding 'nogncheck' to disable the gn include headers check to support modular |
| // WebRTC build targets. |
| #include "media/engine/webrtcvoiceengine.h" // nogncheck |
| #include "p2p/base/packettransportinternal.h" |
| #include "pc/channelmanager.h" |
| #include "pc/rtptransport.h" |
| #include "pc/srtptransport.h" |
| |
| namespace cricket { |
| using rtc::Bind; |
| |
| namespace { |
| // See comment below for why we need to use a pointer to a unique_ptr. |
| bool SetRawAudioSink_w(VoiceMediaChannel* channel, |
| uint32_t ssrc, |
| std::unique_ptr<webrtc::AudioSinkInterface>* sink) { |
| channel->SetRawAudioSink(ssrc, std::move(*sink)); |
| return true; |
| } |
| |
| struct SendPacketMessageData : public rtc::MessageData { |
| rtc::CopyOnWriteBuffer packet; |
| rtc::PacketOptions options; |
| }; |
| |
| } // namespace |
| |
| enum { |
| MSG_EARLYMEDIATIMEOUT = 1, |
| MSG_SEND_RTP_PACKET, |
| MSG_SEND_RTCP_PACKET, |
| MSG_CHANNEL_ERROR, |
| MSG_READYTOSENDDATA, |
| MSG_DATARECEIVED, |
| MSG_FIRSTPACKETRECEIVED, |
| }; |
| |
| // Value specified in RFC 5764. |
| static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp"; |
| |
| static const int kAgcMinus10db = -10; |
| |
| static void SafeSetError(const std::string& message, std::string* error_desc) { |
| if (error_desc) { |
| *error_desc = message; |
| } |
| } |
| |
| struct VoiceChannelErrorMessageData : public rtc::MessageData { |
| VoiceChannelErrorMessageData(uint32_t in_ssrc, |
| VoiceMediaChannel::Error in_error) |
| : ssrc(in_ssrc), error(in_error) {} |
| uint32_t ssrc; |
| VoiceMediaChannel::Error error; |
| }; |
| |
| struct VideoChannelErrorMessageData : public rtc::MessageData { |
| VideoChannelErrorMessageData(uint32_t in_ssrc, |
| VideoMediaChannel::Error in_error) |
| : ssrc(in_ssrc), error(in_error) {} |
| uint32_t ssrc; |
| VideoMediaChannel::Error error; |
| }; |
| |
| struct DataChannelErrorMessageData : public rtc::MessageData { |
| DataChannelErrorMessageData(uint32_t in_ssrc, |
| DataMediaChannel::Error in_error) |
| : ssrc(in_ssrc), error(in_error) {} |
| uint32_t ssrc; |
| DataMediaChannel::Error error; |
| }; |
| |
| static bool ValidPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) { |
| // Check the packet size. We could check the header too if needed. |
| return packet && IsValidRtpRtcpPacketSize(rtcp, packet->size()); |
| } |
| |
| static bool IsReceiveContentDirection(MediaContentDirection direction) { |
| return direction == MD_SENDRECV || direction == MD_RECVONLY; |
| } |
| |
| static bool IsSendContentDirection(MediaContentDirection direction) { |
| return direction == MD_SENDRECV || direction == MD_SENDONLY; |
| } |
| |
| template <class Codec> |
| void RtpParametersFromMediaDescription( |
| const MediaContentDescriptionImpl<Codec>* desc, |
| const RtpHeaderExtensions& extensions, |
| RtpParameters<Codec>* params) { |
| // TODO(pthatcher): Remove this once we're sure no one will give us |
| // a description without codecs (currently a CA_UPDATE with just |
| // streams can). |
| if (desc->has_codecs()) { |
| params->codecs = desc->codecs(); |
| } |
| // TODO(pthatcher): See if we really need |
| // rtp_header_extensions_set() and remove it if we don't. |
| if (desc->rtp_header_extensions_set()) { |
| params->extensions = extensions; |
| } |
| params->rtcp.reduced_size = desc->rtcp_reduced_size(); |
| } |
| |
| template <class Codec> |
| void RtpSendParametersFromMediaDescription( |
| const MediaContentDescriptionImpl<Codec>* desc, |
| const RtpHeaderExtensions& extensions, |
| RtpSendParameters<Codec>* send_params) { |
| RtpParametersFromMediaDescription(desc, extensions, send_params); |
| send_params->max_bandwidth_bps = desc->bandwidth(); |
| } |
| |
| BaseChannel::BaseChannel(rtc::Thread* worker_thread, |
| rtc::Thread* network_thread, |
| rtc::Thread* signaling_thread, |
| std::unique_ptr<MediaChannel> media_channel, |
| const std::string& content_name, |
| bool rtcp_mux_required, |
| bool srtp_required) |
| : worker_thread_(worker_thread), |
| network_thread_(network_thread), |
| signaling_thread_(signaling_thread), |
| content_name_(content_name), |
| rtcp_mux_required_(rtcp_mux_required), |
| srtp_required_(srtp_required), |
| media_channel_(std::move(media_channel)), |
| selected_candidate_pair_(nullptr) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| if (srtp_required) { |
| auto transport = |
| rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_required, content_name); |
| srtp_transport_ = transport.get(); |
| rtp_transport_ = std::move(transport); |
| #if defined(ENABLE_EXTERNAL_AUTH) |
| srtp_transport_->EnableExternalAuth(); |
| #endif |
| } else { |
| rtp_transport_ = rtc::MakeUnique<webrtc::RtpTransport>(rtcp_mux_required); |
| srtp_transport_ = nullptr; |
| } |
| rtp_transport_->SignalReadyToSend.connect( |
| this, &BaseChannel::OnTransportReadyToSend); |
| // TODO(zstein): RtpTransport::SignalPacketReceived will probably be replaced |
| // with a callback interface later so that the demuxer can select which |
| // channel to signal. |
| rtp_transport_->SignalPacketReceived.connect(this, |
| &BaseChannel::OnPacketReceived); |
| rtp_transport_->SignalNetworkRouteChanged.connect( |
| this, &BaseChannel::OnNetworkRouteChanged); |
| RTC_LOG(LS_INFO) << "Created channel for " << content_name; |
| } |
| |
| BaseChannel::~BaseChannel() { |
| TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel"); |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| Deinit(); |
| StopConnectionMonitor(); |
| // Eats any outstanding messages or packets. |
| worker_thread_->Clear(&invoker_); |
| worker_thread_->Clear(this); |
| // We must destroy the media channel before the transport channel, otherwise |
| // the media channel may try to send on the dead transport channel. NULLing |
| // is not an effective strategy since the sends will come on another thread. |
| media_channel_.reset(); |
| RTC_LOG(LS_INFO) << "Destroyed channel: " << content_name_; |
| } |
| |
| void BaseChannel::DisconnectTransportChannels_n() { |
| // Send any outstanding RTCP packets. |
| FlushRtcpMessages_n(); |
| |
| // Stop signals from transport channels, but keep them alive because |
| // media_channel may use them from a different thread. |
| if (rtp_dtls_transport_) { |
| DisconnectFromDtlsTransport(rtp_dtls_transport_); |
| } else if (rtp_transport_->rtp_packet_transport()) { |
| DisconnectFromPacketTransport(rtp_transport_->rtp_packet_transport()); |
| } |
| if (rtcp_dtls_transport_) { |
| DisconnectFromDtlsTransport(rtcp_dtls_transport_); |
| } else if (rtp_transport_->rtcp_packet_transport()) { |
| DisconnectFromPacketTransport(rtp_transport_->rtcp_packet_transport()); |
| } |
| |
| rtp_transport_->SetRtpPacketTransport(nullptr); |
| rtp_transport_->SetRtcpPacketTransport(nullptr); |
| |
| // Clear pending read packets/messages. |
| network_thread_->Clear(&invoker_); |
| network_thread_->Clear(this); |
| } |
| |
| void BaseChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| network_thread_->Invoke<void>(RTC_FROM_HERE, [&] { |
| return InitNetwork_n(rtp_dtls_transport, rtcp_dtls_transport, |
| rtp_packet_transport, rtcp_packet_transport); |
| }); |
| |
| // Both RTP and RTCP channels should be set, we can call SetInterface on |
| // the media channel and it can set network options. |
| media_channel_->SetInterface(this); |
| } |
| |
| void BaseChannel::InitNetwork_n( |
| DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| SetTransports_n(rtp_dtls_transport, rtcp_dtls_transport, rtp_packet_transport, |
| rtcp_packet_transport); |
| |
| if (rtcp_mux_required_) { |
| rtcp_mux_filter_.SetActive(); |
| } |
| } |
| |
| void BaseChannel::Deinit() { |
| RTC_DCHECK(worker_thread_->IsCurrent()); |
| media_channel_->SetInterface(NULL); |
| // Packets arrive on the network thread, processing packets calls virtual |
| // functions, so need to stop this process in Deinit that is called in |
| // derived classes destructor. |
| network_thread_->Invoke<void>( |
| RTC_FROM_HERE, Bind(&BaseChannel::DisconnectTransportChannels_n, this)); |
| } |
| |
| void BaseChannel::SetTransports(DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport) { |
| network_thread_->Invoke<void>( |
| RTC_FROM_HERE, |
| Bind(&BaseChannel::SetTransports_n, this, rtp_dtls_transport, |
| rtcp_dtls_transport, rtp_dtls_transport, rtcp_dtls_transport)); |
| } |
| |
| void BaseChannel::SetTransports( |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport) { |
| network_thread_->Invoke<void>( |
| RTC_FROM_HERE, Bind(&BaseChannel::SetTransports_n, this, nullptr, nullptr, |
| rtp_packet_transport, rtcp_packet_transport)); |
| } |
| |
| void BaseChannel::SetTransports_n( |
| DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| // Validate some assertions about the input. |
| RTC_DCHECK(rtp_packet_transport); |
| RTC_DCHECK_EQ(NeedsRtcpTransport(), rtcp_packet_transport != nullptr); |
| if (rtp_dtls_transport || rtcp_dtls_transport) { |
| // DTLS/non-DTLS pointers should be to the same object. |
| RTC_DCHECK(rtp_dtls_transport == rtp_packet_transport); |
| RTC_DCHECK(rtcp_dtls_transport == rtcp_packet_transport); |
| // Can't go from non-DTLS to DTLS. |
| RTC_DCHECK(!rtp_transport_->rtp_packet_transport() || rtp_dtls_transport_); |
| } else { |
| // Can't go from DTLS to non-DTLS. |
| RTC_DCHECK(!rtp_dtls_transport_); |
| } |
| // Transport names should be the same. |
| if (rtp_dtls_transport && rtcp_dtls_transport) { |
| RTC_DCHECK(rtp_dtls_transport->transport_name() == |
| rtcp_dtls_transport->transport_name()); |
| } |
| std::string debug_name; |
| if (rtp_dtls_transport) { |
| transport_name_ = rtp_dtls_transport->transport_name(); |
| debug_name = transport_name_; |
| } else { |
| debug_name = rtp_packet_transport->transport_name(); |
| } |
| if (rtp_packet_transport == rtp_transport_->rtp_packet_transport()) { |
| // Nothing to do if transport isn't changing. |
| return; |
| } |
| |
| // When using DTLS-SRTP, we must reset the SrtpTransport every time the |
| // DtlsTransport changes and wait until the DTLS handshake is complete to set |
| // the newly negotiated parameters. |
| if (ShouldSetupDtlsSrtp_n()) { |
| // Set |writable_| to false such that UpdateWritableState_w can set up |
| // DTLS-SRTP when |writable_| becomes true again. |
| writable_ = false; |
| dtls_active_ = false; |
| if (srtp_transport_) { |
| srtp_transport_->ResetParams(); |
| } |
| } |
| |
| // If this BaseChannel doesn't require RTCP mux and we haven't fully |
| // negotiated RTCP mux, we need an RTCP transport. |
| if (rtcp_packet_transport) { |
| RTC_LOG(LS_INFO) << "Setting RTCP Transport for " << content_name() |
| << " on " << debug_name << " transport " |
| << rtcp_packet_transport; |
| SetTransport_n(true, rtcp_dtls_transport, rtcp_packet_transport); |
| } |
| |
| RTC_LOG(LS_INFO) << "Setting RTP Transport for " << content_name() << " on " |
| << debug_name << " transport " << rtp_packet_transport; |
| SetTransport_n(false, rtp_dtls_transport, rtp_packet_transport); |
| |
| // Update aggregate writable/ready-to-send state between RTP and RTCP upon |
| // setting new transport channels. |
| UpdateWritableState_n(); |
| } |
| |
| void BaseChannel::SetTransport_n( |
| bool rtcp, |
| DtlsTransportInternal* new_dtls_transport, |
| rtc::PacketTransportInternal* new_packet_transport) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| if (new_dtls_transport) { |
| RTC_DCHECK(new_dtls_transport == new_packet_transport); |
| } |
| DtlsTransportInternal*& old_dtls_transport = |
| rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_; |
| rtc::PacketTransportInternal* old_packet_transport = |
| rtcp ? rtp_transport_->rtcp_packet_transport() |
| : rtp_transport_->rtp_packet_transport(); |
| |
| if (!old_packet_transport && !new_packet_transport) { |
| // Nothing to do. |
| return; |
| } |
| |
| RTC_DCHECK(old_packet_transport != new_packet_transport); |
| if (old_dtls_transport) { |
| DisconnectFromDtlsTransport(old_dtls_transport); |
| } else if (old_packet_transport) { |
| DisconnectFromPacketTransport(old_packet_transport); |
| } |
| |
| if (rtcp) { |
| rtp_transport_->SetRtcpPacketTransport(new_packet_transport); |
| } else { |
| rtp_transport_->SetRtpPacketTransport(new_packet_transport); |
| } |
| old_dtls_transport = new_dtls_transport; |
| |
| // If there's no new transport, we're done after disconnecting from old one. |
| if (!new_packet_transport) { |
| return; |
| } |
| |
| if (rtcp && new_dtls_transport) { |
| RTC_CHECK(!(ShouldSetupDtlsSrtp_n() && srtp_active())) |
| << "Setting RTCP for DTLS/SRTP after the DTLS is active " |
| << "should never happen."; |
| } |
| |
| if (new_dtls_transport) { |
| ConnectToDtlsTransport(new_dtls_transport); |
| } else { |
| ConnectToPacketTransport(new_packet_transport); |
| } |
| auto& socket_options = rtcp ? rtcp_socket_options_ : socket_options_; |
| for (const auto& pair : socket_options) { |
| new_packet_transport->SetOption(pair.first, pair.second); |
| } |
| } |
| |
| void BaseChannel::ConnectToDtlsTransport(DtlsTransportInternal* transport) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| |
| // TODO(zstein): de-dup with ConnectToPacketTransport |
| transport->SignalWritableState.connect(this, &BaseChannel::OnWritableState); |
| transport->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState); |
| transport->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n); |
| } |
| |
| void BaseChannel::DisconnectFromDtlsTransport( |
| DtlsTransportInternal* transport) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| transport->SignalWritableState.disconnect(this); |
| transport->SignalDtlsState.disconnect(this); |
| transport->SignalSentPacket.disconnect(this); |
| } |
| |
| void BaseChannel::ConnectToPacketTransport( |
| rtc::PacketTransportInternal* transport) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| transport->SignalWritableState.connect(this, &BaseChannel::OnWritableState); |
| transport->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n); |
| } |
| |
| void BaseChannel::DisconnectFromPacketTransport( |
| rtc::PacketTransportInternal* transport) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| transport->SignalWritableState.disconnect(this); |
| transport->SignalSentPacket.disconnect(this); |
| } |
| |
| bool BaseChannel::Enable(bool enable) { |
| worker_thread_->Invoke<void>( |
| RTC_FROM_HERE, |
| Bind(enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w, |
| this)); |
| return true; |
| } |
| |
| bool BaseChannel::AddRecvStream(const StreamParams& sp) { |
| return InvokeOnWorker<bool>(RTC_FROM_HERE, |
| Bind(&BaseChannel::AddRecvStream_w, this, sp)); |
| } |
| |
| bool BaseChannel::RemoveRecvStream(uint32_t ssrc) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc)); |
| } |
| |
| bool BaseChannel::AddSendStream(const StreamParams& sp) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, Bind(&MediaChannel::AddSendStream, media_channel(), sp)); |
| } |
| |
| bool BaseChannel::RemoveSendStream(uint32_t ssrc) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&MediaChannel::RemoveSendStream, media_channel(), ssrc)); |
| } |
| |
| bool BaseChannel::SetLocalContent(const MediaContentDescription* content, |
| ContentAction action, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent"); |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&BaseChannel::SetLocalContent_w, this, content, action, error_desc)); |
| } |
| |
| bool BaseChannel::SetRemoteContent(const MediaContentDescription* content, |
| ContentAction action, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent"); |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, Bind(&BaseChannel::SetRemoteContent_w, this, content, |
| action, error_desc)); |
| } |
| |
| void BaseChannel::StartConnectionMonitor(int cms) { |
| // We pass in the BaseChannel instead of the rtp_dtls_transport_ |
| // because if the rtp_dtls_transport_ changes, the ConnectionMonitor |
| // would be pointing to the wrong TransportChannel. |
| // We pass in the network thread because on that thread connection monitor |
| // will call BaseChannel::GetConnectionStats which must be called on the |
| // network thread. |
| connection_monitor_.reset( |
| new ConnectionMonitor(this, network_thread(), rtc::Thread::Current())); |
| connection_monitor_->SignalUpdate.connect( |
| this, &BaseChannel::OnConnectionMonitorUpdate); |
| connection_monitor_->Start(cms); |
| } |
| |
| void BaseChannel::StopConnectionMonitor() { |
| if (connection_monitor_) { |
| connection_monitor_->Stop(); |
| connection_monitor_.reset(); |
| } |
| } |
| |
| bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| if (!rtp_dtls_transport_) { |
| return false; |
| } |
| return rtp_dtls_transport_->ice_transport()->GetStats(infos); |
| } |
| |
| bool BaseChannel::NeedsRtcpTransport() { |
| // If this BaseChannel doesn't require RTCP mux and we haven't fully |
| // negotiated RTCP mux, we need an RTCP transport. |
| return !rtcp_mux_required_ && !rtcp_mux_filter_.IsFullyActive(); |
| } |
| |
| bool BaseChannel::IsReadyToReceiveMedia_w() const { |
| // Receive data if we are enabled and have local content, |
| return enabled() && IsReceiveContentDirection(local_content_direction_); |
| } |
| |
| bool BaseChannel::IsReadyToSendMedia_w() const { |
| // Need to access some state updated on the network thread. |
| return network_thread_->Invoke<bool>( |
| RTC_FROM_HERE, Bind(&BaseChannel::IsReadyToSendMedia_n, this)); |
| } |
| |
| bool BaseChannel::IsReadyToSendMedia_n() const { |
| // Send outgoing data if we are enabled, have local and remote content, |
| // and we have had some form of connectivity. |
| return enabled() && IsReceiveContentDirection(remote_content_direction_) && |
| IsSendContentDirection(local_content_direction_) && |
| was_ever_writable() && (srtp_active() || !ShouldSetupDtlsSrtp_n()); |
| } |
| |
| bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) { |
| return SendPacket(false, packet, options); |
| } |
| |
| bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) { |
| return SendPacket(true, packet, options); |
| } |
| |
| int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt, |
| int value) { |
| return network_thread_->Invoke<int>( |
| RTC_FROM_HERE, Bind(&BaseChannel::SetOption_n, this, type, opt, value)); |
| } |
| |
| int BaseChannel::SetOption_n(SocketType type, |
| rtc::Socket::Option opt, |
| int value) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| rtc::PacketTransportInternal* transport = nullptr; |
| switch (type) { |
| case ST_RTP: |
| transport = rtp_transport_->rtp_packet_transport(); |
| socket_options_.push_back( |
| std::pair<rtc::Socket::Option, int>(opt, value)); |
| break; |
| case ST_RTCP: |
| transport = rtp_transport_->rtcp_packet_transport(); |
| rtcp_socket_options_.push_back( |
| std::pair<rtc::Socket::Option, int>(opt, value)); |
| break; |
| } |
| return transport ? transport->SetOption(opt, value) : -1; |
| } |
| |
| void BaseChannel::OnWritableState(rtc::PacketTransportInternal* transport) { |
| RTC_DCHECK(transport == rtp_transport_->rtp_packet_transport() || |
| transport == rtp_transport_->rtcp_packet_transport()); |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| UpdateWritableState_n(); |
| } |
| |
| void BaseChannel::OnDtlsState(DtlsTransportInternal* transport, |
| DtlsTransportState state) { |
| if (!ShouldSetupDtlsSrtp_n()) { |
| return; |
| } |
| |
| // Reset the SrtpTransport if it's not the CONNECTED state. For the CONNECTED |
| // state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to |
| // cover other scenarios like the whole transport is writable (not just this |
| // TransportChannel) or when TransportChannel is attached after DTLS is |
| // negotiated. |
| if (state != DTLS_TRANSPORT_CONNECTED) { |
| dtls_active_ = false; |
| if (srtp_transport_) { |
| srtp_transport_->ResetParams(); |
| } |
| } |
| } |
| |
| void BaseChannel::OnNetworkRouteChanged( |
| rtc::Optional<rtc::NetworkRoute> network_route) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| rtc::NetworkRoute new_route; |
| if (network_route) { |
| new_route = *(network_route); |
| } |
| // Note: When the RTCP-muxing is not enabled, RTCP transport and RTP transport |
| // use the same transport name and MediaChannel::OnNetworkRouteChanged cannot |
| // work correctly. Intentionally leave it broken to simplify the code and |
| // encourage the users to stop using non-muxing RTCP. |
| invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [=] { |
| media_channel_->OnNetworkRouteChanged(transport_name_, new_route); |
| }); |
| } |
| |
| void BaseChannel::OnTransportReadyToSend(bool ready) { |
| invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, |
| [=] { media_channel_->OnReadyToSend(ready); }); |
| } |
| |
| bool BaseChannel::SendPacket(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) { |
| // SendPacket gets called from MediaEngine, on a pacer or an encoder thread. |
| // If the thread is not our network thread, we will post to our network |
| // so that the real work happens on our network. This avoids us having to |
| // synchronize access to all the pieces of the send path, including |
| // SRTP and the inner workings of the transport channels. |
| // The only downside is that we can't return a proper failure code if |
| // needed. Since UDP is unreliable anyway, this should be a non-issue. |
| if (!network_thread_->IsCurrent()) { |
| // Avoid a copy by transferring the ownership of the packet data. |
| int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET; |
| SendPacketMessageData* data = new SendPacketMessageData; |
| data->packet = std::move(*packet); |
| data->options = options; |
| network_thread_->Post(RTC_FROM_HERE, this, message_id, data); |
| return true; |
| } |
| TRACE_EVENT0("webrtc", "BaseChannel::SendPacket"); |
| |
| // Now that we are on the correct thread, ensure we have a place to send this |
| // packet before doing anything. (We might get RTCP packets that we don't |
| // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP |
| // transport. |
| if (!rtp_transport_->IsWritable(rtcp)) { |
| return false; |
| } |
| |
| // Protect ourselves against crazy data. |
| if (!ValidPacket(rtcp, packet)) { |
| RTC_LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " " |
| << RtpRtcpStringLiteral(rtcp) |
| << " packet: wrong size=" << packet->size(); |
| return false; |
| } |
| |
| if (!srtp_active()) { |
| if (srtp_required_) { |
| // The audio/video engines may attempt to send RTCP packets as soon as the |
| // streams are created, so don't treat this as an error for RTCP. |
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6809 |
| if (rtcp) { |
| return false; |
| } |
| // However, there shouldn't be any RTP packets sent before SRTP is set up |
| // (and SetSend(true) is called). |
| RTC_LOG(LS_ERROR) |
| << "Can't send outgoing RTP packet when SRTP is inactive" |
| << " and crypto is required"; |
| RTC_NOTREACHED(); |
| return false; |
| } |
| // Bon voyage. |
| return rtcp |
| ? rtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS) |
| : rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS); |
| } |
| RTC_DCHECK(srtp_transport_); |
| RTC_DCHECK(srtp_transport_->IsActive()); |
| // Bon voyage. |
| return rtcp ? srtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS) |
| : srtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS); |
| } |
| |
| bool BaseChannel::HandlesPayloadType(int packet_type) const { |
| return rtp_transport_->HandlesPayloadType(packet_type); |
| } |
| |
| void BaseChannel::OnPacketReceived(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) { |
| if (!has_received_packet_ && !rtcp) { |
| has_received_packet_ = true; |
| signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED); |
| } |
| |
| if (!srtp_active() && srtp_required_) { |
| // Our session description indicates that SRTP is required, but we got a |
| // packet before our SRTP filter is active. This means either that |
| // a) we got SRTP packets before we received the SDES keys, in which case |
| // we can't decrypt it anyway, or |
| // b) we got SRTP packets before DTLS completed on both the RTP and RTCP |
| // transports, so we haven't yet extracted keys, even if DTLS did |
| // complete on the transport that the packets are being sent on. It's |
| // really good practice to wait for both RTP and RTCP to be good to go |
| // before sending media, to prevent weird failure modes, so it's fine |
| // for us to just eat packets here. This is all sidestepped if RTCP mux |
| // is used anyway. |
| RTC_LOG(LS_WARNING) |
| << "Can't process incoming " << RtpRtcpStringLiteral(rtcp) |
| << " packet when SRTP is inactive and crypto is required"; |
| return; |
| } |
| |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, worker_thread_, |
| Bind(&BaseChannel::ProcessPacket, this, rtcp, *packet, packet_time)); |
| } |
| |
| void BaseChannel::ProcessPacket(bool rtcp, |
| const rtc::CopyOnWriteBuffer& packet, |
| const rtc::PacketTime& packet_time) { |
| RTC_DCHECK(worker_thread_->IsCurrent()); |
| |
| // Need to copy variable because OnRtcpReceived/OnPacketReceived |
| // requires non-const pointer to buffer. This doesn't memcpy the actual data. |
| rtc::CopyOnWriteBuffer data(packet); |
| if (rtcp) { |
| media_channel_->OnRtcpReceived(&data, packet_time); |
| } else { |
| media_channel_->OnPacketReceived(&data, packet_time); |
| } |
| } |
| |
| void BaseChannel::EnableMedia_w() { |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| if (enabled_) |
| return; |
| |
| RTC_LOG(LS_INFO) << "Channel enabled"; |
| enabled_ = true; |
| UpdateMediaSendRecvState_w(); |
| } |
| |
| void BaseChannel::DisableMedia_w() { |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| if (!enabled_) |
| return; |
| |
| RTC_LOG(LS_INFO) << "Channel disabled"; |
| enabled_ = false; |
| UpdateMediaSendRecvState_w(); |
| } |
| |
| void BaseChannel::UpdateWritableState_n() { |
| rtc::PacketTransportInternal* rtp_packet_transport = |
| rtp_transport_->rtp_packet_transport(); |
| rtc::PacketTransportInternal* rtcp_packet_transport = |
| rtp_transport_->rtcp_packet_transport(); |
| if (rtp_packet_transport && rtp_packet_transport->writable() && |
| (!rtcp_packet_transport || rtcp_packet_transport->writable())) { |
| ChannelWritable_n(); |
| } else { |
| ChannelNotWritable_n(); |
| } |
| } |
| |
| void BaseChannel::ChannelWritable_n() { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| if (writable_) { |
| return; |
| } |
| |
| RTC_LOG(LS_INFO) << "Channel writable (" << content_name_ << ")" |
| << (was_ever_writable_ ? "" : " for the first time"); |
| |
| if (selected_candidate_pair_) |
| RTC_LOG(LS_INFO) |
| << "Using " |
| << selected_candidate_pair_->local_candidate().ToSensitiveString() |
| << "->" |
| << selected_candidate_pair_->remote_candidate().ToSensitiveString(); |
| |
| was_ever_writable_ = true; |
| MaybeSetupDtlsSrtp_n(); |
| writable_ = true; |
| UpdateMediaSendRecvState(); |
| } |
| |
| void BaseChannel::SignalDtlsSrtpSetupFailure_n(bool rtcp) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), |
| Bind(&BaseChannel::SignalDtlsSrtpSetupFailure_s, this, rtcp)); |
| } |
| |
| void BaseChannel::SignalDtlsSrtpSetupFailure_s(bool rtcp) { |
| RTC_DCHECK(signaling_thread() == rtc::Thread::Current()); |
| SignalDtlsSrtpSetupFailure(this, rtcp); |
| } |
| |
| bool BaseChannel::ShouldSetupDtlsSrtp_n() const { |
| // Since DTLS is applied to all transports, checking RTP should be enough. |
| return rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive(); |
| } |
| |
| // This function returns true if either DTLS-SRTP is not in use |
| // *or* DTLS-SRTP is successfully set up. |
| bool BaseChannel::SetupDtlsSrtp_n(bool rtcp) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| bool ret = false; |
| |
| DtlsTransportInternal* transport = |
| rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_; |
| RTC_DCHECK(transport); |
| RTC_DCHECK(transport->IsDtlsActive()); |
| |
| int selected_crypto_suite; |
| |
| if (!transport->GetSrtpCryptoSuite(&selected_crypto_suite)) { |
| RTC_LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite"; |
| return false; |
| } |
| |
| RTC_LOG(LS_INFO) << "Installing keys from DTLS-SRTP on " << content_name() |
| << " " << RtpRtcpStringLiteral(rtcp); |
| |
| int key_len; |
| int salt_len; |
| if (!rtc::GetSrtpKeyAndSaltLengths(selected_crypto_suite, &key_len, |
| &salt_len)) { |
| RTC_LOG(LS_ERROR) << "Unknown DTLS-SRTP crypto suite" |
| << selected_crypto_suite; |
| return false; |
| } |
| |
| // OK, we're now doing DTLS (RFC 5764) |
| std::vector<unsigned char> dtls_buffer(key_len * 2 + salt_len * 2); |
| |
| // RFC 5705 exporter using the RFC 5764 parameters |
| if (!transport->ExportKeyingMaterial(kDtlsSrtpExporterLabel, NULL, 0, false, |
| &dtls_buffer[0], dtls_buffer.size())) { |
| RTC_LOG(LS_WARNING) << "DTLS-SRTP key export failed"; |
| RTC_NOTREACHED(); // This should never happen |
| return false; |
| } |
| |
| // Sync up the keys with the DTLS-SRTP interface |
| std::vector<unsigned char> client_write_key(key_len + salt_len); |
| std::vector<unsigned char> server_write_key(key_len + salt_len); |
| size_t offset = 0; |
| memcpy(&client_write_key[0], &dtls_buffer[offset], key_len); |
| offset += key_len; |
| memcpy(&server_write_key[0], &dtls_buffer[offset], key_len); |
| offset += key_len; |
| memcpy(&client_write_key[key_len], &dtls_buffer[offset], salt_len); |
| offset += salt_len; |
| memcpy(&server_write_key[key_len], &dtls_buffer[offset], salt_len); |
| |
| std::vector<unsigned char> *send_key, *recv_key; |
| rtc::SSLRole role; |
| if (!transport->GetSslRole(&role)) { |
| RTC_LOG(LS_WARNING) << "GetSslRole failed"; |
| return false; |
| } |
| |
| if (role == rtc::SSL_SERVER) { |
| send_key = &server_write_key; |
| recv_key = &client_write_key; |
| } else { |
| send_key = &client_write_key; |
| recv_key = &server_write_key; |
| } |
| |
| // Use an empty encrypted header extension ID vector if not set. This could |
| // happen when the DTLS handshake is completed before processing the |
| // Offer/Answer which contains the encrypted header extension IDs. |
| std::vector<int> send_extension_ids; |
| std::vector<int> recv_extension_ids; |
| if (catched_send_extension_ids_) { |
| send_extension_ids = *catched_send_extension_ids_; |
| } |
| if (catched_recv_extension_ids_) { |
| recv_extension_ids = *catched_recv_extension_ids_; |
| } |
| |
| if (rtcp) { |
| if (!dtls_active()) { |
| RTC_DCHECK(srtp_transport_); |
| ret = srtp_transport_->SetRtcpParams( |
| selected_crypto_suite, &(*send_key)[0], |
| static_cast<int>(send_key->size()), send_extension_ids, |
| selected_crypto_suite, &(*recv_key)[0], |
| static_cast<int>(recv_key->size()), recv_extension_ids); |
| } else { |
| // RTCP doesn't need to call SetRtpParam because it is only used |
| // to make the updated encrypted RTP header extension IDs take effect. |
| ret = true; |
| } |
| } else { |
| RTC_DCHECK(srtp_transport_); |
| ret = srtp_transport_->SetRtpParams( |
| selected_crypto_suite, &(*send_key)[0], |
| static_cast<int>(send_key->size()), send_extension_ids, |
| selected_crypto_suite, &(*recv_key)[0], |
| static_cast<int>(recv_key->size()), recv_extension_ids); |
| dtls_active_ = ret; |
| } |
| |
| if (!ret) { |
| RTC_LOG(LS_WARNING) << "DTLS-SRTP key installation failed"; |
| } |
| |
| return ret; |
| } |
| |
| void BaseChannel::MaybeSetupDtlsSrtp_n() { |
| if (dtls_active()) { |
| return; |
| } |
| |
| if (!ShouldSetupDtlsSrtp_n()) { |
| return; |
| } |
| |
| if (!srtp_transport_) { |
| EnableSrtpTransport_n(); |
| } |
| |
| if (!SetupDtlsSrtp_n(false)) { |
| SignalDtlsSrtpSetupFailure_n(false); |
| return; |
| } |
| |
| if (rtcp_dtls_transport_) { |
| if (!SetupDtlsSrtp_n(true)) { |
| SignalDtlsSrtpSetupFailure_n(true); |
| return; |
| } |
| } |
| } |
| |
| void BaseChannel::ChannelNotWritable_n() { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| if (!writable_) |
| return; |
| |
| RTC_LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")"; |
| writable_ = false; |
| UpdateMediaSendRecvState(); |
| } |
| |
| bool BaseChannel::SetRtpTransportParameters( |
| const MediaContentDescription* content, |
| ContentAction action, |
| ContentSource src, |
| const RtpHeaderExtensions& extensions, |
| std::string* error_desc) { |
| if (action == CA_UPDATE) { |
| // These parameters never get changed by a CA_UDPATE. |
| return true; |
| } |
| |
| std::vector<int> encrypted_extension_ids; |
| for (const webrtc::RtpExtension& extension : extensions) { |
| if (extension.encrypt) { |
| RTC_LOG(LS_INFO) << "Using " << (src == CS_LOCAL ? "local" : "remote") |
| << " encrypted extension: " << extension.ToString(); |
| encrypted_extension_ids.push_back(extension.id); |
| } |
| } |
| |
| // Cache srtp_required_ for belt and suspenders check on SendPacket |
| return network_thread_->Invoke<bool>( |
| RTC_FROM_HERE, Bind(&BaseChannel::SetRtpTransportParameters_n, this, |
| content, action, src, encrypted_extension_ids, |
| error_desc)); |
| } |
| |
| bool BaseChannel::SetRtpTransportParameters_n( |
| const MediaContentDescription* content, |
| ContentAction action, |
| ContentSource src, |
| const std::vector<int>& encrypted_extension_ids, |
| std::string* error_desc) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| |
| if (!SetSrtp_n(content->cryptos(), action, src, encrypted_extension_ids, |
| error_desc)) { |
| return false; |
| } |
| |
| if (!SetRtcpMux_n(content->rtcp_mux(), action, src, error_desc)) { |
| return false; |
| } |
| |
| return true; |
| } |
| |
| // |dtls| will be set to true if DTLS is active for transport and crypto is |
| // empty. |
| bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos, |
| bool* dtls, |
| std::string* error_desc) { |
| *dtls = rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive(); |
| if (*dtls && !cryptos.empty()) { |
| SafeSetError("Cryptos must be empty when DTLS is active.", error_desc); |
| return false; |
| } |
| return true; |
| } |
| |
| void BaseChannel::EnableSrtpTransport_n() { |
| if (srtp_transport_ == nullptr) { |
| rtp_transport_->SignalReadyToSend.disconnect(this); |
| rtp_transport_->SignalPacketReceived.disconnect(this); |
| rtp_transport_->SignalNetworkRouteChanged.disconnect(this); |
| |
| auto transport = rtc::MakeUnique<webrtc::SrtpTransport>( |
| std::move(rtp_transport_), content_name_); |
| srtp_transport_ = transport.get(); |
| rtp_transport_ = std::move(transport); |
| |
| rtp_transport_->SignalReadyToSend.connect( |
| this, &BaseChannel::OnTransportReadyToSend); |
| rtp_transport_->SignalPacketReceived.connect( |
| this, &BaseChannel::OnPacketReceived); |
| rtp_transport_->SignalNetworkRouteChanged.connect( |
| this, &BaseChannel::OnNetworkRouteChanged); |
| RTC_LOG(LS_INFO) << "Wrapping RtpTransport in SrtpTransport."; |
| } |
| } |
| |
| bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos, |
| ContentAction action, |
| ContentSource src, |
| const std::vector<int>& encrypted_extension_ids, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "BaseChannel::SetSrtp_w"); |
| if (action == CA_UPDATE) { |
| // no crypto params. |
| return true; |
| } |
| bool ret = false; |
| bool dtls = false; |
| ret = CheckSrtpConfig_n(cryptos, &dtls, error_desc); |
| if (!ret) { |
| return false; |
| } |
| |
| // If SRTP was not required, but we're setting a description that uses SDES, |
| // we need to upgrade to an SrtpTransport. |
| if (!srtp_transport_ && !dtls && !cryptos.empty()) { |
| EnableSrtpTransport_n(); |
| } |
| |
| bool encrypted_header_extensions_id_changed = |
| EncryptedHeaderExtensionIdsChanged(src, encrypted_extension_ids); |
| CacheEncryptedHeaderExtensionIds(src, encrypted_extension_ids); |
| |
| switch (action) { |
| case CA_OFFER: |
| // If DTLS is already active on the channel, we could be renegotiating |
| // here. We don't update the srtp filter. |
| if (!dtls) { |
| ret = sdes_negotiator_.SetOffer(cryptos, src); |
| } |
| break; |
| case CA_PRANSWER: |
| // If we're doing DTLS-SRTP, we don't want to update the filter |
| // with an answer, because we already have SRTP parameters. |
| if (!dtls) { |
| ret = sdes_negotiator_.SetProvisionalAnswer(cryptos, src); |
| } |
| break; |
| case CA_ANSWER: |
| // If we're doing DTLS-SRTP, we don't want to update the filter |
| // with an answer, because we already have SRTP parameters. |
| if (!dtls) { |
| ret = sdes_negotiator_.SetAnswer(cryptos, src); |
| } |
| break; |
| default: |
| break; |
| } |
| |
| // If setting an SDES answer succeeded, apply the negotiated parameters |
| // to the SRTP transport. |
| if ((action == CA_PRANSWER || action == CA_ANSWER) && !dtls && ret) { |
| if (sdes_negotiator_.send_cipher_suite() && |
| sdes_negotiator_.recv_cipher_suite()) { |
| RTC_DCHECK(catched_send_extension_ids_); |
| RTC_DCHECK(catched_recv_extension_ids_); |
| ret = srtp_transport_->SetRtpParams( |
| *(sdes_negotiator_.send_cipher_suite()), |
| sdes_negotiator_.send_key().data(), |
| static_cast<int>(sdes_negotiator_.send_key().size()), |
| *(catched_send_extension_ids_), |
| *(sdes_negotiator_.recv_cipher_suite()), |
| sdes_negotiator_.recv_key().data(), |
| static_cast<int>(sdes_negotiator_.recv_key().size()), |
| *(catched_recv_extension_ids_)); |
| } else { |
| RTC_LOG(LS_INFO) << "No crypto keys are provided for SDES."; |
| if (action == CA_ANSWER && srtp_transport_) { |
| // Explicitly reset the |srtp_transport_| if no crypto param is |
| // provided in the answer. No need to call |ResetParams()| for |
| // |sdes_negotiator_| because it resets the params inside |SetAnswer|. |
| srtp_transport_->ResetParams(); |
| } |
| } |
| } |
| |
| // Only update SRTP transport if using DTLS. SDES is handled internally |
| // by the SRTP filter. |
| if (ret && dtls_active() && rtp_dtls_transport_ && |
| rtp_dtls_transport_->dtls_state() == DTLS_TRANSPORT_CONNECTED && |
| encrypted_header_extensions_id_changed) { |
| ret = SetupDtlsSrtp_n(/*rtcp=*/false); |
| } |
| |
| if (!ret) { |
| SafeSetError("Failed to setup SRTP.", error_desc); |
| return false; |
| } |
| return true; |
| } |
| |
| bool BaseChannel::SetRtcpMux_n(bool enable, |
| ContentAction action, |
| ContentSource src, |
| std::string* error_desc) { |
| // Provide a more specific error message for the RTCP mux "require" policy |
| // case. |
| if (rtcp_mux_required_ && !enable) { |
| SafeSetError( |
| "rtcpMuxPolicy is 'require', but media description does not " |
| "contain 'a=rtcp-mux'.", |
| error_desc); |
| return false; |
| } |
| bool ret = false; |
| switch (action) { |
| case CA_OFFER: |
| ret = rtcp_mux_filter_.SetOffer(enable, src); |
| break; |
| case CA_PRANSWER: |
| // This may activate RTCP muxing, but we don't yet destroy the transport |
| // because the final answer may deactivate it. |
| ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src); |
| break; |
| case CA_ANSWER: |
| ret = rtcp_mux_filter_.SetAnswer(enable, src); |
| if (ret && rtcp_mux_filter_.IsActive()) { |
| // We permanently activated RTCP muxing; signal that we no longer need |
| // the RTCP transport. |
| std::string debug_name = |
| transport_name_.empty() |
| ? rtp_transport_->rtp_packet_transport()->transport_name() |
| : transport_name_; |
| RTC_LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name() |
| << "; no longer need RTCP transport for " |
| << debug_name; |
| if (rtp_transport_->rtcp_packet_transport()) { |
| SetTransport_n(true, nullptr, nullptr); |
| SignalRtcpMuxFullyActive(transport_name_); |
| } |
| UpdateWritableState_n(); |
| } |
| break; |
| case CA_UPDATE: |
| // No RTCP mux info. |
| ret = true; |
| break; |
| default: |
| break; |
| } |
| if (!ret) { |
| SafeSetError("Failed to setup RTCP mux filter.", error_desc); |
| return false; |
| } |
| rtp_transport_->SetRtcpMuxEnabled(rtcp_mux_filter_.IsActive()); |
| // |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or |
| // CA_ANSWER, but we only want to tear down the RTCP transport if we received |
| // a final answer. |
| if (rtcp_mux_filter_.IsActive()) { |
| // If the RTP transport is already writable, then so are we. |
| if (rtp_transport_->rtp_packet_transport()->writable()) { |
| ChannelWritable_n(); |
| } |
| } |
| |
| return true; |
| } |
| |
| bool BaseChannel::AddRecvStream_w(const StreamParams& sp) { |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| return media_channel()->AddRecvStream(sp); |
| } |
| |
| bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) { |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| return media_channel()->RemoveRecvStream(ssrc); |
| } |
| |
| bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams, |
| ContentAction action, |
| std::string* error_desc) { |
| if (!(action == CA_OFFER || action == CA_ANSWER || |
| action == CA_PRANSWER || action == CA_UPDATE)) |
| return false; |
| |
| // If this is an update, streams only contain streams that have changed. |
| if (action == CA_UPDATE) { |
| for (StreamParamsVec::const_iterator it = streams.begin(); |
| it != streams.end(); ++it) { |
| const StreamParams* existing_stream = |
| GetStreamByIds(local_streams_, it->groupid, it->id); |
| if (!existing_stream && it->has_ssrcs()) { |
| if (media_channel()->AddSendStream(*it)) { |
| local_streams_.push_back(*it); |
| RTC_LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc(); |
| } else { |
| std::ostringstream desc; |
| desc << "Failed to add send stream ssrc: " << it->first_ssrc(); |
| SafeSetError(desc.str(), error_desc); |
| return false; |
| } |
| } else if (existing_stream && !it->has_ssrcs()) { |
| if (!media_channel()->RemoveSendStream(existing_stream->first_ssrc())) { |
| std::ostringstream desc; |
| desc << "Failed to remove send stream with ssrc " |
| << it->first_ssrc() << "."; |
| SafeSetError(desc.str(), error_desc); |
| return false; |
| } |
| RemoveStreamBySsrc(&local_streams_, existing_stream->first_ssrc()); |
| } else { |
| RTC_LOG(LS_WARNING) << "Ignore unsupported stream update"; |
| } |
| } |
| return true; |
| } |
| // Else streams are all the streams we want to send. |
| |
| // Check for streams that have been removed. |
| bool ret = true; |
| for (StreamParamsVec::const_iterator it = local_streams_.begin(); |
| it != local_streams_.end(); ++it) { |
| if (!GetStreamBySsrc(streams, it->first_ssrc())) { |
| if (!media_channel()->RemoveSendStream(it->first_ssrc())) { |
| std::ostringstream desc; |
| desc << "Failed to remove send stream with ssrc " |
| << it->first_ssrc() << "."; |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| } |
| } |
| } |
| // Check for new streams. |
| for (StreamParamsVec::const_iterator it = streams.begin(); |
| it != streams.end(); ++it) { |
| if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) { |
| if (media_channel()->AddSendStream(*it)) { |
| RTC_LOG(LS_INFO) << "Add send stream ssrc: " << it->ssrcs[0]; |
| } else { |
| std::ostringstream desc; |
| desc << "Failed to add send stream ssrc: " << it->first_ssrc(); |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| } |
| } |
| } |
| local_streams_ = streams; |
| return ret; |
| } |
| |
| bool BaseChannel::UpdateRemoteStreams_w( |
| const std::vector<StreamParams>& streams, |
| ContentAction action, |
| std::string* error_desc) { |
| if (!(action == CA_OFFER || action == CA_ANSWER || |
| action == CA_PRANSWER || action == CA_UPDATE)) |
| return false; |
| |
| // If this is an update, streams only contain streams that have changed. |
| if (action == CA_UPDATE) { |
| for (StreamParamsVec::const_iterator it = streams.begin(); |
| it != streams.end(); ++it) { |
| const StreamParams* existing_stream = |
| GetStreamByIds(remote_streams_, it->groupid, it->id); |
| if (!existing_stream && it->has_ssrcs()) { |
| if (AddRecvStream_w(*it)) { |
| remote_streams_.push_back(*it); |
| RTC_LOG(LS_INFO) << "Add remote stream ssrc: " << it->first_ssrc(); |
| } else { |
| std::ostringstream desc; |
| desc << "Failed to add remote stream ssrc: " << it->first_ssrc(); |
| SafeSetError(desc.str(), error_desc); |
| return false; |
| } |
| } else if (existing_stream && !it->has_ssrcs()) { |
| if (!RemoveRecvStream_w(existing_stream->first_ssrc())) { |
| std::ostringstream desc; |
| desc << "Failed to remove remote stream with ssrc " |
| << it->first_ssrc() << "."; |
| SafeSetError(desc.str(), error_desc); |
| return false; |
| } |
| RemoveStreamBySsrc(&remote_streams_, existing_stream->first_ssrc()); |
| } else { |
| RTC_LOG(LS_WARNING) |
| << "Ignore unsupported stream update." |
| << " Stream exists? " << (existing_stream != nullptr) |
| << " new stream = " << it->ToString(); |
| } |
| } |
| return true; |
| } |
| // Else streams are all the streams we want to receive. |
| |
| // Check for streams that have been removed. |
| bool ret = true; |
| for (StreamParamsVec::const_iterator it = remote_streams_.begin(); |
| it != remote_streams_.end(); ++it) { |
| if (!GetStreamBySsrc(streams, it->first_ssrc())) { |
| if (!RemoveRecvStream_w(it->first_ssrc())) { |
| std::ostringstream desc; |
| desc << "Failed to remove remote stream with ssrc " |
| << it->first_ssrc() << "."; |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| } |
| } |
| } |
| // Check for new streams. |
| for (StreamParamsVec::const_iterator it = streams.begin(); |
| it != streams.end(); ++it) { |
| if (!GetStreamBySsrc(remote_streams_, it->first_ssrc())) { |
| if (AddRecvStream_w(*it)) { |
| RTC_LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0]; |
| } else { |
| std::ostringstream desc; |
| desc << "Failed to add remote stream ssrc: " << it->first_ssrc(); |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| } |
| } |
| } |
| remote_streams_ = streams; |
| return ret; |
| } |
| |
| RtpHeaderExtensions BaseChannel::GetFilteredRtpHeaderExtensions( |
| const RtpHeaderExtensions& extensions) { |
| if (!rtp_dtls_transport_ || |
| !rtp_dtls_transport_->crypto_options() |
| .enable_encrypted_rtp_header_extensions) { |
| RtpHeaderExtensions filtered; |
| auto pred = [](const webrtc::RtpExtension& extension) { |
| return !extension.encrypt; |
| }; |
| std::copy_if(extensions.begin(), extensions.end(), |
| std::back_inserter(filtered), pred); |
| return filtered; |
| } |
| |
| return webrtc::RtpExtension::FilterDuplicateNonEncrypted(extensions); |
| } |
| |
| void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w( |
| const std::vector<webrtc::RtpExtension>& extensions) { |
| // Absolute Send Time extension id is used only with external auth, |
| // so do not bother searching for it and making asyncronious call to set |
| // something that is not used. |
| #if defined(ENABLE_EXTERNAL_AUTH) |
| const webrtc::RtpExtension* send_time_extension = |
| webrtc::RtpExtension::FindHeaderExtensionByUri( |
| extensions, webrtc::RtpExtension::kAbsSendTimeUri); |
| int rtp_abs_sendtime_extn_id = |
| send_time_extension ? send_time_extension->id : -1; |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, network_thread_, |
| Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n, this, |
| rtp_abs_sendtime_extn_id)); |
| #endif |
| } |
| |
| void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n( |
| int rtp_abs_sendtime_extn_id) { |
| if (srtp_transport_) { |
| srtp_transport_->CacheRtpAbsSendTimeHeaderExtension( |
| rtp_abs_sendtime_extn_id); |
| } else { |
| RTC_LOG(LS_WARNING) |
| << "Trying to cache the Absolute Send Time extension id " |
| "but the SRTP is not active."; |
| } |
| } |
| |
| void BaseChannel::OnMessage(rtc::Message *pmsg) { |
| TRACE_EVENT0("webrtc", "BaseChannel::OnMessage"); |
| switch (pmsg->message_id) { |
| case MSG_SEND_RTP_PACKET: |
| case MSG_SEND_RTCP_PACKET: { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| SendPacketMessageData* data = |
| static_cast<SendPacketMessageData*>(pmsg->pdata); |
| bool rtcp = pmsg->message_id == MSG_SEND_RTCP_PACKET; |
| SendPacket(rtcp, &data->packet, data->options); |
| delete data; |
| break; |
| } |
| case MSG_FIRSTPACKETRECEIVED: { |
| SignalFirstPacketReceived(this); |
| break; |
| } |
| } |
| } |
| |
| void BaseChannel::AddHandledPayloadType(int payload_type) { |
| rtp_transport_->AddHandledPayloadType(payload_type); |
| } |
| |
| void BaseChannel::FlushRtcpMessages_n() { |
| // Flush all remaining RTCP messages. This should only be called in |
| // destructor. |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| rtc::MessageList rtcp_messages; |
| network_thread_->Clear(this, MSG_SEND_RTCP_PACKET, &rtcp_messages); |
| for (const auto& message : rtcp_messages) { |
| network_thread_->Send(RTC_FROM_HERE, this, MSG_SEND_RTCP_PACKET, |
| message.pdata); |
| } |
| } |
| |
| void BaseChannel::SignalSentPacket_n( |
| rtc::PacketTransportInternal* /* transport */, |
| const rtc::SentPacket& sent_packet) { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, worker_thread_, |
| rtc::Bind(&BaseChannel::SignalSentPacket_w, this, sent_packet)); |
| } |
| |
| void BaseChannel::SignalSentPacket_w(const rtc::SentPacket& sent_packet) { |
| RTC_DCHECK(worker_thread_->IsCurrent()); |
| SignalSentPacket(sent_packet); |
| } |
| |
| void BaseChannel::CacheEncryptedHeaderExtensionIds( |
| cricket::ContentSource source, |
| const std::vector<int>& extension_ids) { |
| source == ContentSource::CS_LOCAL |
| ? catched_recv_extension_ids_.emplace(extension_ids) |
| : catched_send_extension_ids_.emplace(extension_ids); |
| } |
| |
| bool BaseChannel::EncryptedHeaderExtensionIdsChanged( |
| cricket::ContentSource source, |
| const std::vector<int>& new_extension_ids) { |
| if (source == ContentSource::CS_LOCAL) { |
| return !catched_recv_extension_ids_ || |
| (*catched_recv_extension_ids_) != new_extension_ids; |
| } else { |
| return !catched_send_extension_ids_ || |
| (*catched_send_extension_ids_) != new_extension_ids; |
| } |
| } |
| |
| VoiceChannel::VoiceChannel(rtc::Thread* worker_thread, |
| rtc::Thread* network_thread, |
| rtc::Thread* signaling_thread, |
| MediaEngineInterface* media_engine, |
| std::unique_ptr<VoiceMediaChannel> media_channel, |
| const std::string& content_name, |
| bool rtcp_mux_required, |
| bool srtp_required) |
| : BaseChannel(worker_thread, |
| network_thread, |
| signaling_thread, |
| std::move(media_channel), |
| content_name, |
| rtcp_mux_required, |
| srtp_required), |
| media_engine_(media_engine) {} |
| |
| VoiceChannel::~VoiceChannel() { |
| TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel"); |
| StopAudioMonitor(); |
| StopMediaMonitor(); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| Deinit(); |
| } |
| |
| bool VoiceChannel::SetAudioSend(uint32_t ssrc, |
| bool enable, |
| const AudioOptions* options, |
| AudioSource* source) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, Bind(&VoiceMediaChannel::SetAudioSend, media_channel(), |
| ssrc, enable, options, source)); |
| } |
| |
| // TODO(juberti): Handle early media the right way. We should get an explicit |
| // ringing message telling us to start playing local ringback, which we cancel |
| // if any early media actually arrives. For now, we do the opposite, which is |
| // to wait 1 second for early media, and start playing local ringback if none |
| // arrives. |
| void VoiceChannel::SetEarlyMedia(bool enable) { |
| if (enable) { |
| // Start the early media timeout |
| worker_thread()->PostDelayed(RTC_FROM_HERE, kEarlyMediaTimeout, this, |
| MSG_EARLYMEDIATIMEOUT); |
| } else { |
| // Stop the timeout if currently going. |
| worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT); |
| } |
| } |
| |
| bool VoiceChannel::CanInsertDtmf() { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, Bind(&VoiceMediaChannel::CanInsertDtmf, media_channel())); |
| } |
| |
| bool VoiceChannel::InsertDtmf(uint32_t ssrc, |
| int event_code, |
| int duration) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&VoiceChannel::InsertDtmf_w, this, ssrc, event_code, duration)); |
| } |
| |
| bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&VoiceMediaChannel::SetOutputVolume, media_channel(), ssrc, volume)); |
| } |
| |
| void VoiceChannel::SetRawAudioSink( |
| uint32_t ssrc, |
| std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
| // We need to work around Bind's lack of support for unique_ptr and ownership |
| // passing. So we invoke to our own little routine that gets a pointer to |
| // our local variable. This is OK since we're synchronously invoking. |
| InvokeOnWorker<bool>(RTC_FROM_HERE, |
| Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink)); |
| } |
| |
| webrtc::RtpParameters VoiceChannel::GetRtpSendParameters(uint32_t ssrc) const { |
| return worker_thread()->Invoke<webrtc::RtpParameters>( |
| RTC_FROM_HERE, Bind(&VoiceChannel::GetRtpSendParameters_w, this, ssrc)); |
| } |
| |
| webrtc::RtpParameters VoiceChannel::GetRtpSendParameters_w( |
| uint32_t ssrc) const { |
| return media_channel()->GetRtpSendParameters(ssrc); |
| } |
| |
| bool VoiceChannel::SetRtpSendParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&VoiceChannel::SetRtpSendParameters_w, this, ssrc, parameters)); |
| } |
| |
| bool VoiceChannel::SetRtpSendParameters_w(uint32_t ssrc, |
| webrtc::RtpParameters parameters) { |
| return media_channel()->SetRtpSendParameters(ssrc, parameters); |
| } |
| |
| webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters( |
| uint32_t ssrc) const { |
| return worker_thread()->Invoke<webrtc::RtpParameters>( |
| RTC_FROM_HERE, |
| Bind(&VoiceChannel::GetRtpReceiveParameters_w, this, ssrc)); |
| } |
| |
| webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters_w( |
| uint32_t ssrc) const { |
| return media_channel()->GetRtpReceiveParameters(ssrc); |
| } |
| |
| bool VoiceChannel::SetRtpReceiveParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&VoiceChannel::SetRtpReceiveParameters_w, this, ssrc, parameters)); |
| } |
| |
| bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc, |
| webrtc::RtpParameters parameters) { |
| return media_channel()->SetRtpReceiveParameters(ssrc, parameters); |
| } |
| |
| bool VoiceChannel::GetStats(VoiceMediaInfo* stats) { |
| return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats, |
| media_channel(), stats)); |
| } |
| |
| std::vector<webrtc::RtpSource> VoiceChannel::GetSources(uint32_t ssrc) const { |
| return worker_thread()->Invoke<std::vector<webrtc::RtpSource>>( |
| RTC_FROM_HERE, Bind(&VoiceChannel::GetSources_w, this, ssrc)); |
| } |
| |
| std::vector<webrtc::RtpSource> VoiceChannel::GetSources_w(uint32_t ssrc) const { |
| RTC_DCHECK(worker_thread()->IsCurrent()); |
| return media_channel()->GetSources(ssrc); |
| } |
| |
| void VoiceChannel::StartMediaMonitor(int cms) { |
| media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), |
| rtc::Thread::Current())); |
| media_monitor_->SignalUpdate.connect( |
| this, &VoiceChannel::OnMediaMonitorUpdate); |
| media_monitor_->Start(cms); |
| } |
| |
| void VoiceChannel::StopMediaMonitor() { |
| if (media_monitor_) { |
| media_monitor_->Stop(); |
| media_monitor_->SignalUpdate.disconnect(this); |
| media_monitor_.reset(); |
| } |
| } |
| |
| void VoiceChannel::StartAudioMonitor(int cms) { |
| audio_monitor_.reset(new AudioMonitor(this, rtc::Thread::Current())); |
| audio_monitor_ |
| ->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate); |
| audio_monitor_->Start(cms); |
| } |
| |
| void VoiceChannel::StopAudioMonitor() { |
| if (audio_monitor_) { |
| audio_monitor_->Stop(); |
| audio_monitor_.reset(); |
| } |
| } |
| |
| bool VoiceChannel::IsAudioMonitorRunning() const { |
| return (audio_monitor_.get() != NULL); |
| } |
| |
| int VoiceChannel::GetInputLevel_w() { |
| return media_engine_->GetInputLevel(); |
| } |
| |
| int VoiceChannel::GetOutputLevel_w() { |
| return media_channel()->GetOutputLevel(); |
| } |
| |
| void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) { |
| media_channel()->GetActiveStreams(actives); |
| } |
| |
| void VoiceChannel::OnPacketReceived(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) { |
| BaseChannel::OnPacketReceived(rtcp, packet, packet_time); |
| // Set a flag when we've received an RTP packet. If we're waiting for early |
| // media, this will disable the timeout. |
| if (!received_media_ && !rtcp) { |
| received_media_ = true; |
| } |
| } |
| |
| void BaseChannel::UpdateMediaSendRecvState() { |
| RTC_DCHECK(network_thread_->IsCurrent()); |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, worker_thread_, |
| Bind(&BaseChannel::UpdateMediaSendRecvState_w, this)); |
| } |
| |
| void VoiceChannel::UpdateMediaSendRecvState_w() { |
| // Render incoming data if we're the active call, and we have the local |
| // content. We receive data on the default channel and multiplexed streams. |
| bool recv = IsReadyToReceiveMedia_w(); |
| media_channel()->SetPlayout(recv); |
| |
| // Send outgoing data if we're the active call, we have the remote content, |
| // and we have had some form of connectivity. |
| bool send = IsReadyToSendMedia_w(); |
| media_channel()->SetSend(send); |
| |
| RTC_LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send; |
| } |
| |
| bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, |
| ContentAction action, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w"); |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| RTC_LOG(LS_INFO) << "Setting local voice description"; |
| |
| const AudioContentDescription* audio = |
| static_cast<const AudioContentDescription*>(content); |
| RTC_DCHECK(audio != NULL); |
| if (!audio) { |
| SafeSetError("Can't find audio content in local description.", error_desc); |
| return false; |
| } |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions()); |
| |
| if (!SetRtpTransportParameters(content, action, CS_LOCAL, |
| rtp_header_extensions, error_desc)) { |
| return false; |
| } |
| |
| AudioRecvParameters recv_params = last_recv_params_; |
| RtpParametersFromMediaDescription(audio, rtp_header_extensions, &recv_params); |
| if (!media_channel()->SetRecvParameters(recv_params)) { |
| SafeSetError("Failed to set local audio description recv parameters.", |
| error_desc); |
| return false; |
| } |
| for (const AudioCodec& codec : audio->codecs()) { |
| AddHandledPayloadType(codec.id); |
| } |
| last_recv_params_ = recv_params; |
| |
| // TODO(pthatcher): Move local streams into AudioSendParameters, and |
| // only give it to the media channel once we have a remote |
| // description too (without a remote description, we won't be able |
| // to send them anyway). |
| if (!UpdateLocalStreams_w(audio->streams(), action, error_desc)) { |
| SafeSetError("Failed to set local audio description streams.", error_desc); |
| return false; |
| } |
| |
| set_local_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| ContentAction action, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w"); |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| RTC_LOG(LS_INFO) << "Setting remote voice description"; |
| |
| const AudioContentDescription* audio = |
| static_cast<const AudioContentDescription*>(content); |
| RTC_DCHECK(audio != NULL); |
| if (!audio) { |
| SafeSetError("Can't find audio content in remote description.", error_desc); |
| return false; |
| } |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions()); |
| |
| if (!SetRtpTransportParameters(content, action, CS_REMOTE, |
| rtp_header_extensions, error_desc)) { |
| return false; |
| } |
| |
| AudioSendParameters send_params = last_send_params_; |
| RtpSendParametersFromMediaDescription(audio, rtp_header_extensions, |
| &send_params); |
| if (audio->agc_minus_10db()) { |
| send_params.options.adjust_agc_delta = rtc::Optional<int>(kAgcMinus10db); |
| } |
| |
| bool parameters_applied = media_channel()->SetSendParameters(send_params); |
| if (!parameters_applied) { |
| SafeSetError("Failed to set remote audio description send parameters.", |
| error_desc); |
| return false; |
| } |
| last_send_params_ = send_params; |
| |
| // TODO(pthatcher): Move remote streams into AudioRecvParameters, |
| // and only give it to the media channel once we have a local |
| // description too (without a local description, we won't be able to |
| // recv them anyway). |
| if (!UpdateRemoteStreams_w(audio->streams(), action, error_desc)) { |
| SafeSetError("Failed to set remote audio description streams.", error_desc); |
| return false; |
| } |
| |
| if (audio->rtp_header_extensions_set()) { |
| MaybeCacheRtpAbsSendTimeHeaderExtension_w(rtp_header_extensions); |
| } |
| |
| set_remote_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| void VoiceChannel::HandleEarlyMediaTimeout() { |
| // This occurs on the main thread, not the worker thread. |
| if (!received_media_) { |
| RTC_LOG(LS_INFO) << "No early media received before timeout"; |
| SignalEarlyMediaTimeout(this); |
| } |
| } |
| |
| bool VoiceChannel::InsertDtmf_w(uint32_t ssrc, |
| int event, |
| int duration) { |
| if (!enabled()) { |
| return false; |
| } |
| return media_channel()->InsertDtmf(ssrc, event, duration); |
| } |
| |
| void VoiceChannel::OnMessage(rtc::Message *pmsg) { |
| switch (pmsg->message_id) { |
| case MSG_EARLYMEDIATIMEOUT: |
| HandleEarlyMediaTimeout(); |
| break; |
| case MSG_CHANNEL_ERROR: { |
| VoiceChannelErrorMessageData* data = |
| static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata); |
| delete data; |
| break; |
| } |
| default: |
| BaseChannel::OnMessage(pmsg); |
| break; |
| } |
| } |
| |
| void VoiceChannel::OnConnectionMonitorUpdate( |
| ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) { |
| SignalConnectionMonitor(this, infos); |
| } |
| |
| void VoiceChannel::OnMediaMonitorUpdate( |
| VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) { |
| RTC_DCHECK(media_channel == this->media_channel()); |
| SignalMediaMonitor(this, info); |
| } |
| |
| void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor, |
| const AudioInfo& info) { |
| SignalAudioMonitor(this, info); |
| } |
| |
| VideoChannel::VideoChannel(rtc::Thread* worker_thread, |
| rtc::Thread* network_thread, |
| rtc::Thread* signaling_thread, |
| std::unique_ptr<VideoMediaChannel> media_channel, |
| const std::string& content_name, |
| bool rtcp_mux_required, |
| bool srtp_required) |
| : BaseChannel(worker_thread, |
| network_thread, |
| signaling_thread, |
| std::move(media_channel), |
| content_name, |
| rtcp_mux_required, |
| srtp_required) {} |
| |
| VideoChannel::~VideoChannel() { |
| TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel"); |
| StopMediaMonitor(); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| |
| Deinit(); |
| } |
| |
| bool VideoChannel::SetSink(uint32_t ssrc, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| worker_thread()->Invoke<void>( |
| RTC_FROM_HERE, |
| Bind(&VideoMediaChannel::SetSink, media_channel(), ssrc, sink)); |
| return true; |
| } |
| |
| bool VideoChannel::SetVideoSend( |
| uint32_t ssrc, |
| bool mute, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, Bind(&VideoMediaChannel::SetVideoSend, media_channel(), |
| ssrc, mute, options, source)); |
| } |
| |
| webrtc::RtpParameters VideoChannel::GetRtpSendParameters(uint32_t ssrc) const { |
| return worker_thread()->Invoke<webrtc::RtpParameters>( |
| RTC_FROM_HERE, Bind(&VideoChannel::GetRtpSendParameters_w, this, ssrc)); |
| } |
| |
| webrtc::RtpParameters VideoChannel::GetRtpSendParameters_w( |
| uint32_t ssrc) const { |
| return media_channel()->GetRtpSendParameters(ssrc); |
| } |
| |
| bool VideoChannel::SetRtpSendParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&VideoChannel::SetRtpSendParameters_w, this, ssrc, parameters)); |
| } |
| |
| bool VideoChannel::SetRtpSendParameters_w(uint32_t ssrc, |
| webrtc::RtpParameters parameters) { |
| return media_channel()->SetRtpSendParameters(ssrc, parameters); |
| } |
| |
| webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters( |
| uint32_t ssrc) const { |
| return worker_thread()->Invoke<webrtc::RtpParameters>( |
| RTC_FROM_HERE, |
| Bind(&VideoChannel::GetRtpReceiveParameters_w, this, ssrc)); |
| } |
| |
| webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters_w( |
| uint32_t ssrc) const { |
| return media_channel()->GetRtpReceiveParameters(ssrc); |
| } |
| |
| bool VideoChannel::SetRtpReceiveParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&VideoChannel::SetRtpReceiveParameters_w, this, ssrc, parameters)); |
| } |
| |
| bool VideoChannel::SetRtpReceiveParameters_w(uint32_t ssrc, |
| webrtc::RtpParameters parameters) { |
| return media_channel()->SetRtpReceiveParameters(ssrc, parameters); |
| } |
| |
| void VideoChannel::UpdateMediaSendRecvState_w() { |
| // Send outgoing data if we're the active call, we have the remote content, |
| // and we have had some form of connectivity. |
| bool send = IsReadyToSendMedia_w(); |
| if (!media_channel()->SetSend(send)) { |
| RTC_LOG(LS_ERROR) << "Failed to SetSend on video channel"; |
| // TODO(gangji): Report error back to server. |
| } |
| |
| RTC_LOG(LS_INFO) << "Changing video state, send=" << send; |
| } |
| |
| void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) { |
| InvokeOnWorker<void>(RTC_FROM_HERE, Bind(&VideoMediaChannel::FillBitrateInfo, |
| media_channel(), bwe_info)); |
| } |
| |
| bool VideoChannel::GetStats(VideoMediaInfo* stats) { |
| return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VideoMediaChannel::GetStats, |
| media_channel(), stats)); |
| } |
| |
| void VideoChannel::StartMediaMonitor(int cms) { |
| media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(), |
| rtc::Thread::Current())); |
| media_monitor_->SignalUpdate.connect( |
| this, &VideoChannel::OnMediaMonitorUpdate); |
| media_monitor_->Start(cms); |
| } |
| |
| void VideoChannel::StopMediaMonitor() { |
| if (media_monitor_) { |
| media_monitor_->Stop(); |
| media_monitor_.reset(); |
| } |
| } |
| |
| bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, |
| ContentAction action, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w"); |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| RTC_LOG(LS_INFO) << "Setting local video description"; |
| |
| const VideoContentDescription* video = |
| static_cast<const VideoContentDescription*>(content); |
| RTC_DCHECK(video != NULL); |
| if (!video) { |
| SafeSetError("Can't find video content in local description.", error_desc); |
| return false; |
| } |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(video->rtp_header_extensions()); |
| |
| if (!SetRtpTransportParameters(content, action, CS_LOCAL, |
| rtp_header_extensions, error_desc)) { |
| return false; |
| } |
| |
| VideoRecvParameters recv_params = last_recv_params_; |
| RtpParametersFromMediaDescription(video, rtp_header_extensions, &recv_params); |
| if (!media_channel()->SetRecvParameters(recv_params)) { |
| SafeSetError("Failed to set local video description recv parameters.", |
| error_desc); |
| return false; |
| } |
| for (const VideoCodec& codec : video->codecs()) { |
| AddHandledPayloadType(codec.id); |
| } |
| last_recv_params_ = recv_params; |
| |
| // TODO(pthatcher): Move local streams into VideoSendParameters, and |
| // only give it to the media channel once we have a remote |
| // description too (without a remote description, we won't be able |
| // to send them anyway). |
| if (!UpdateLocalStreams_w(video->streams(), action, error_desc)) { |
| SafeSetError("Failed to set local video description streams.", error_desc); |
| return false; |
| } |
| |
| set_local_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| ContentAction action, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w"); |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| RTC_LOG(LS_INFO) << "Setting remote video description"; |
| |
| const VideoContentDescription* video = |
| static_cast<const VideoContentDescription*>(content); |
| RTC_DCHECK(video != NULL); |
| if (!video) { |
| SafeSetError("Can't find video content in remote description.", error_desc); |
| return false; |
| } |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(video->rtp_header_extensions()); |
| |
| if (!SetRtpTransportParameters(content, action, CS_REMOTE, |
| rtp_header_extensions, error_desc)) { |
| return false; |
| } |
| |
| VideoSendParameters send_params = last_send_params_; |
| RtpSendParametersFromMediaDescription(video, rtp_header_extensions, |
| &send_params); |
| if (video->conference_mode()) { |
| send_params.conference_mode = true; |
| } |
| |
| bool parameters_applied = media_channel()->SetSendParameters(send_params); |
| |
| if (!parameters_applied) { |
| SafeSetError("Failed to set remote video description send parameters.", |
| error_desc); |
| return false; |
| } |
| last_send_params_ = send_params; |
| |
| // TODO(pthatcher): Move remote streams into VideoRecvParameters, |
| // and only give it to the media channel once we have a local |
| // description too (without a local description, we won't be able to |
| // recv them anyway). |
| if (!UpdateRemoteStreams_w(video->streams(), action, error_desc)) { |
| SafeSetError("Failed to set remote video description streams.", error_desc); |
| return false; |
| } |
| |
| if (video->rtp_header_extensions_set()) { |
| MaybeCacheRtpAbsSendTimeHeaderExtension_w(rtp_header_extensions); |
| } |
| |
| set_remote_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| void VideoChannel::OnMessage(rtc::Message *pmsg) { |
| switch (pmsg->message_id) { |
| case MSG_CHANNEL_ERROR: { |
| const VideoChannelErrorMessageData* data = |
| static_cast<VideoChannelErrorMessageData*>(pmsg->pdata); |
| delete data; |
| break; |
| } |
| default: |
| BaseChannel::OnMessage(pmsg); |
| break; |
| } |
| } |
| |
| void VideoChannel::OnConnectionMonitorUpdate( |
| ConnectionMonitor* monitor, const std::vector<ConnectionInfo> &infos) { |
| SignalConnectionMonitor(this, infos); |
| } |
| |
| // TODO(pthatcher): Look into removing duplicate code between |
| // audio, video, and data, perhaps by using templates. |
| void VideoChannel::OnMediaMonitorUpdate( |
| VideoMediaChannel* media_channel, const VideoMediaInfo &info) { |
| RTC_DCHECK(media_channel == this->media_channel()); |
| SignalMediaMonitor(this, info); |
| } |
| |
| RtpDataChannel::RtpDataChannel(rtc::Thread* worker_thread, |
| rtc::Thread* network_thread, |
| rtc::Thread* signaling_thread, |
| std::unique_ptr<DataMediaChannel> media_channel, |
| const std::string& content_name, |
| bool rtcp_mux_required, |
| bool srtp_required) |
| : BaseChannel(worker_thread, |
| network_thread, |
| signaling_thread, |
| std::move(media_channel), |
| content_name, |
| rtcp_mux_required, |
| srtp_required) {} |
| |
| RtpDataChannel::~RtpDataChannel() { |
| TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel"); |
| StopMediaMonitor(); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| |
| Deinit(); |
| } |
| |
| void RtpDataChannel::Init_w( |
| DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport) { |
| BaseChannel::Init_w(rtp_dtls_transport, rtcp_dtls_transport, |
| rtp_packet_transport, rtcp_packet_transport); |
| |
| media_channel()->SignalDataReceived.connect(this, |
| &RtpDataChannel::OnDataReceived); |
| media_channel()->SignalReadyToSend.connect( |
| this, &RtpDataChannel::OnDataChannelReadyToSend); |
| } |
| |
| bool RtpDataChannel::SendData(const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| SendDataResult* result) { |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, Bind(&DataMediaChannel::SendData, media_channel(), params, |
| payload, result)); |
| } |
| |
| bool RtpDataChannel::CheckDataChannelTypeFromContent( |
| const DataContentDescription* content, |
| std::string* error_desc) { |
| bool is_sctp = ((content->protocol() == kMediaProtocolSctp) || |
| (content->protocol() == kMediaProtocolDtlsSctp)); |
| // It's been set before, but doesn't match. That's bad. |
| if (is_sctp) { |
| SafeSetError("Data channel type mismatch. Expected RTP, got SCTP.", |
| error_desc); |
| return false; |
| } |
| return true; |
| } |
| |
| bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content, |
| ContentAction action, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "RtpDataChannel::SetLocalContent_w"); |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| RTC_LOG(LS_INFO) << "Setting local data description"; |
| |
| const DataContentDescription* data = |
| static_cast<const DataContentDescription*>(content); |
| RTC_DCHECK(data != NULL); |
| if (!data) { |
| SafeSetError("Can't find data content in local description.", error_desc); |
| return false; |
| } |
| |
| if (!CheckDataChannelTypeFromContent(data, error_desc)) { |
| return false; |
| } |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(data->rtp_header_extensions()); |
| |
| if (!SetRtpTransportParameters(content, action, CS_LOCAL, |
| rtp_header_extensions, error_desc)) { |
| return false; |
| } |
| |
| DataRecvParameters recv_params = last_recv_params_; |
| RtpParametersFromMediaDescription(data, rtp_header_extensions, &recv_params); |
| if (!media_channel()->SetRecvParameters(recv_params)) { |
| SafeSetError("Failed to set remote data description recv parameters.", |
| error_desc); |
| return false; |
| } |
| for (const DataCodec& codec : data->codecs()) { |
| AddHandledPayloadType(codec.id); |
| } |
| last_recv_params_ = recv_params; |
| |
| // TODO(pthatcher): Move local streams into DataSendParameters, and |
| // only give it to the media channel once we have a remote |
| // description too (without a remote description, we won't be able |
| // to send them anyway). |
| if (!UpdateLocalStreams_w(data->streams(), action, error_desc)) { |
| SafeSetError("Failed to set local data description streams.", error_desc); |
| return false; |
| } |
| |
| set_local_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| ContentAction action, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "RtpDataChannel::SetRemoteContent_w"); |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| |
| const DataContentDescription* data = |
| static_cast<const DataContentDescription*>(content); |
| RTC_DCHECK(data != NULL); |
| if (!data) { |
| SafeSetError("Can't find data content in remote description.", error_desc); |
| return false; |
| } |
| |
| // If the remote data doesn't have codecs and isn't an update, it |
| // must be empty, so ignore it. |
| if (!data->has_codecs() && action != CA_UPDATE) { |
| return true; |
| } |
| |
| if (!CheckDataChannelTypeFromContent(data, error_desc)) { |
| return false; |
| } |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(data->rtp_header_extensions()); |
| |
| RTC_LOG(LS_INFO) << "Setting remote data description"; |
| if (!SetRtpTransportParameters(content, action, CS_REMOTE, |
| rtp_header_extensions, error_desc)) { |
| return false; |
| } |
| |
| DataSendParameters send_params = last_send_params_; |
| RtpSendParametersFromMediaDescription<DataCodec>(data, rtp_header_extensions, |
| &send_params); |
| if (!media_channel()->SetSendParameters(send_params)) { |
| SafeSetError("Failed to set remote data description send parameters.", |
| error_desc); |
| return false; |
| } |
| last_send_params_ = send_params; |
| |
| // TODO(pthatcher): Move remote streams into DataRecvParameters, |
| // and only give it to the media channel once we have a local |
| // description too (without a local description, we won't be able to |
| // recv them anyway). |
| if (!UpdateRemoteStreams_w(data->streams(), action, error_desc)) { |
| SafeSetError("Failed to set remote data description streams.", |
| error_desc); |
| return false; |
| } |
| |
| set_remote_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| void RtpDataChannel::UpdateMediaSendRecvState_w() { |
| // Render incoming data if we're the active call, and we have the local |
| // content. We receive data on the default channel and multiplexed streams. |
| bool recv = IsReadyToReceiveMedia_w(); |
| if (!media_channel()->SetReceive(recv)) { |
| RTC_LOG(LS_ERROR) << "Failed to SetReceive on data channel"; |
| } |
| |
| // Send outgoing data if we're the active call, we have the remote content, |
| // and we have had some form of connectivity. |
| bool send = IsReadyToSendMedia_w(); |
| if (!media_channel()->SetSend(send)) { |
| RTC_LOG(LS_ERROR) << "Failed to SetSend on data channel"; |
| } |
| |
| // Trigger SignalReadyToSendData asynchronously. |
| OnDataChannelReadyToSend(send); |
| |
| RTC_LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send; |
| } |
| |
| void RtpDataChannel::OnMessage(rtc::Message* pmsg) { |
| switch (pmsg->message_id) { |
| case MSG_READYTOSENDDATA: { |
| DataChannelReadyToSendMessageData* data = |
| static_cast<DataChannelReadyToSendMessageData*>(pmsg->pdata); |
| ready_to_send_data_ = data->data(); |
| SignalReadyToSendData(ready_to_send_data_); |
| delete data; |
| break; |
| } |
| case MSG_DATARECEIVED: { |
| DataReceivedMessageData* data = |
| static_cast<DataReceivedMessageData*>(pmsg->pdata); |
| SignalDataReceived(data->params, data->payload); |
| delete data; |
| break; |
| } |
| case MSG_CHANNEL_ERROR: { |
| const DataChannelErrorMessageData* data = |
| static_cast<DataChannelErrorMessageData*>(pmsg->pdata); |
| delete data; |
| break; |
| } |
| default: |
| BaseChannel::OnMessage(pmsg); |
| break; |
| } |
| } |
| |
| void RtpDataChannel::OnConnectionMonitorUpdate( |
| ConnectionMonitor* monitor, |
| const std::vector<ConnectionInfo>& infos) { |
| SignalConnectionMonitor(this, infos); |
| } |
| |
| void RtpDataChannel::StartMediaMonitor(int cms) { |
| media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(), |
| rtc::Thread::Current())); |
| media_monitor_->SignalUpdate.connect(this, |
| &RtpDataChannel::OnMediaMonitorUpdate); |
| media_monitor_->Start(cms); |
| } |
| |
| void RtpDataChannel::StopMediaMonitor() { |
| if (media_monitor_) { |
| media_monitor_->Stop(); |
| media_monitor_->SignalUpdate.disconnect(this); |
| media_monitor_.reset(); |
| } |
| } |
| |
| void RtpDataChannel::OnMediaMonitorUpdate(DataMediaChannel* media_channel, |
| const DataMediaInfo& info) { |
| RTC_DCHECK(media_channel == this->media_channel()); |
| SignalMediaMonitor(this, info); |
| } |
| |
| void RtpDataChannel::OnDataReceived(const ReceiveDataParams& params, |
| const char* data, |
| size_t len) { |
| DataReceivedMessageData* msg = new DataReceivedMessageData( |
| params, data, len); |
| signaling_thread()->Post(RTC_FROM_HERE, this, MSG_DATARECEIVED, msg); |
| } |
| |
| void RtpDataChannel::OnDataChannelError(uint32_t ssrc, |
| DataMediaChannel::Error err) { |
| DataChannelErrorMessageData* data = new DataChannelErrorMessageData( |
| ssrc, err); |
| signaling_thread()->Post(RTC_FROM_HERE, this, MSG_CHANNEL_ERROR, data); |
| } |
| |
| void RtpDataChannel::OnDataChannelReadyToSend(bool writable) { |
| // This is usded for congestion control to indicate that the stream is ready |
| // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates |
| // that the transport channel is ready. |
| signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA, |
| new DataChannelReadyToSendMessageData(writable)); |
| } |
| |
| } // namespace cricket |