Chromium Code Reviews| Index: webrtc/pc/channel.cc |
| diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc |
| index 4d47e878b8e2b8c7d7c858c439d822f9d95a4720..9348b37b058f7f43dc42d5d81fedd42950cfece1 100644 |
| --- a/webrtc/pc/channel.cc |
| +++ b/webrtc/pc/channel.cc |
| @@ -37,12 +37,18 @@ bool SetRawAudioSink_w(VoiceMediaChannel* channel, |
| 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_RTPPACKET, |
| - MSG_RTCPPACKET, |
| + MSG_SEND_RTP_PACKET, |
| + MSG_SEND_RTCP_PACKET, |
| MSG_CHANNEL_ERROR, |
| MSG_READYTOSENDDATA, |
| MSG_DATARECEIVED, |
| @@ -61,11 +67,6 @@ static void SafeSetError(const std::string& message, std::string* error_desc) { |
| } |
| } |
| -struct PacketMessageData : public rtc::MessageData { |
| - rtc::CopyOnWriteBuffer packet; |
| - rtc::PacketOptions options; |
| -}; |
| - |
| struct VoiceChannelErrorMessageData : public rtc::MessageData { |
| VoiceChannelErrorMessageData(uint32_t in_ssrc, |
| VoiceMediaChannel::Error in_error) |
| @@ -142,12 +143,14 @@ void RtpSendParametersFromMediaDescription( |
| send_params->max_bandwidth_bps = desc->bandwidth(); |
| } |
| -BaseChannel::BaseChannel(rtc::Thread* thread, |
| +BaseChannel::BaseChannel(rtc::Thread* worker_thread, |
| + rtc::Thread* network_thread, |
| MediaChannel* media_channel, |
| TransportController* transport_controller, |
| const std::string& content_name, |
| bool rtcp) |
| - : worker_thread_(thread), |
| + : worker_thread_(worker_thread), |
| + network_thread_(network_thread), |
| transport_controller_(transport_controller), |
| media_channel_(media_channel), |
| content_name_(content_name), |
| @@ -166,6 +169,9 @@ BaseChannel::BaseChannel(rtc::Thread* thread, |
| secure_required_(false), |
| rtp_abs_sendtime_extn_id_(-1) { |
| ASSERT(worker_thread_ == rtc::Thread::Current()); |
| + if (transport_controller) { |
| + RTC_DCHECK_EQ(network_thread, transport_controller->worker_thread()); |
| + } |
| LOG(LS_INFO) << "Created channel for " << content_name; |
| } |
| @@ -174,14 +180,22 @@ BaseChannel::~BaseChannel() { |
| ASSERT(worker_thread_ == rtc::Thread::Current()); |
| Deinit(); |
| StopConnectionMonitor(); |
| - FlushRtcpMessages(); // Send any outstanding RTCP packets. |
| - worker_thread_->Clear(this); // eats any outstanding messages or packets |
| + // Send any outstanding RTCP packets. |
| + network_thread_->Invoke<void>(Bind(&BaseChannel::FlushRtcpMessages_n, this)); |
| + // 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. |
| delete media_channel_; |
| // Note that we don't just call set_transport_channel(nullptr) because that |
| // would call a pure virtual method which we can't do from a destructor. |
| + network_thread_->Invoke<void>(Bind(&BaseChannel::Destruct_n, this)); |
| + LOG(LS_INFO) << "Destroyed channel"; |
| +} |
| + |
| +void BaseChannel::Destruct_n() { |
|
pthatcher1
2016/05/11 04:50:01
I think we could call this something more specific
danilchap
2016/05/11 12:19:16
Done.
|
| if (transport_channel_) { |
| DisconnectFromTransportChannel(transport_channel_); |
| transport_controller_->DestroyTransportChannel_w( |
| @@ -192,39 +206,49 @@ BaseChannel::~BaseChannel() { |
| transport_controller_->DestroyTransportChannel_w( |
| transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTCP); |
| } |
| - LOG(LS_INFO) << "Destroyed channel"; |
| + network_thread_->Clear(this); |
| } |
| bool BaseChannel::Init() { |
| - if (!SetTransport(content_name())) { |
| + if (!network_thread_->Invoke<bool>(Bind(&BaseChannel::Init_n, this))) { |
| + return false; |
| + } |
| + |
| + // Both RTP and RTCP channels are set, we can call SetInterface on |
| + // media channel and it can set network options. |
| + RTC_DCHECK(worker_thread_->IsCurrent()); |
| + media_channel_->SetInterface(this); |
| + return true; |
| +} |
| + |
| +bool BaseChannel::Init_n() { |
|
pthatcher1
2016/05/11 04:50:01
And this InitNetwork_n.
danilchap
2016/05/11 12:19:16
Done.
|
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| + if (!SetTransport_n(content_name())) { |
| return false; |
| } |
| - if (!SetDtlsSrtpCryptoSuites(transport_channel(), false)) { |
| + if (!SetDtlsSrtpCryptoSuites(transport_channel_, false)) { |
| return false; |
| } |
| if (rtcp_transport_enabled() && |
| - !SetDtlsSrtpCryptoSuites(rtcp_transport_channel(), true)) { |
| + !SetDtlsSrtpCryptoSuites(rtcp_transport_channel_, true)) { |
| return false; |
| } |
| - |
| - // Both RTP and RTCP channels are set, we can call SetInterface on |
| - // media channel and it can set network options. |
| - media_channel_->SetInterface(this); |
| return true; |
| } |
| void BaseChannel::Deinit() { |
| + RTC_DCHECK(worker_thread_->IsCurrent()); |
| media_channel_->SetInterface(NULL); |
| } |
| bool BaseChannel::SetTransport(const std::string& transport_name) { |
| - return worker_thread_->Invoke<bool>( |
| - Bind(&BaseChannel::SetTransport_w, this, transport_name)); |
| + return network_thread_->Invoke<bool>( |
| + Bind(&BaseChannel::SetTransport_n, this, transport_name)); |
| } |
| -bool BaseChannel::SetTransport_w(const std::string& transport_name) { |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| +bool BaseChannel::SetTransport_n(const std::string& transport_name) { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| if (transport_name == transport_name_) { |
| // Nothing to do if transport name isn't changing |
| @@ -234,7 +258,7 @@ bool BaseChannel::SetTransport_w(const std::string& transport_name) { |
| // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport |
| // changes and wait until the DTLS handshake is complete to set the newly |
| // negotiated parameters. |
| - if (ShouldSetupDtlsSrtp()) { |
| + if (ShouldSetupDtlsSrtp_n()) { |
| // Set |writable_| to false such that UpdateWritableState_w can set up |
| // DTLS-SRTP when the writable_ becomes true again. |
| writable_ = false; |
| @@ -249,7 +273,7 @@ bool BaseChannel::SetTransport_w(const std::string& transport_name) { |
| transport_controller_->CreateTransportChannel_w( |
| transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP), |
| false /* update_writablity */); |
| - if (!rtcp_transport_channel()) { |
| + if (!rtcp_transport_channel_) { |
| return false; |
| } |
| } |
| @@ -257,7 +281,7 @@ bool BaseChannel::SetTransport_w(const std::string& transport_name) { |
| // We're not updating the writablity during the transition state. |
| set_transport_channel(transport_controller_->CreateTransportChannel_w( |
| transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP)); |
| - if (!transport_channel()) { |
| + if (!transport_channel_) { |
| return false; |
| } |
| @@ -266,14 +290,14 @@ bool BaseChannel::SetTransport_w(const std::string& transport_name) { |
| // We can only update the RTCP ready to send after set_transport_channel has |
| // handled channel writability. |
| SetReadyToSend( |
| - true, rtcp_transport_channel() && rtcp_transport_channel()->writable()); |
| + true, rtcp_transport_channel_ && rtcp_transport_channel_->writable()); |
| } |
| transport_name_ = transport_name; |
| return true; |
| } |
| void BaseChannel::set_transport_channel(TransportChannel* new_tc) { |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| TransportChannel* old_tc = transport_channel_; |
| if (!old_tc && !new_tc) { |
| @@ -299,13 +323,13 @@ void BaseChannel::set_transport_channel(TransportChannel* new_tc) { |
| // Update aggregate writable/ready-to-send state between RTP and RTCP upon |
| // setting new channel |
| - UpdateWritableState_w(); |
| + UpdateWritableState_n(); |
| SetReadyToSend(false, new_tc && new_tc->writable()); |
| } |
| void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc, |
| bool update_writablity) { |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| TransportChannel* old_tc = rtcp_transport_channel_; |
| if (!old_tc && !new_tc) { |
| @@ -323,7 +347,7 @@ void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc, |
| rtcp_transport_channel_ = new_tc; |
| if (new_tc) { |
| - RTC_CHECK(!(ShouldSetupDtlsSrtp() && srtp_filter_.IsActive())) |
| + RTC_CHECK(!(ShouldSetupDtlsSrtp_n() && srtp_filter_.IsActive())) |
| << "Setting RTCP for DTLS/SRTP after SrtpFilter is active " |
| << "should never happen."; |
| ConnectToTransportChannel(new_tc); |
| @@ -335,13 +359,13 @@ void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc, |
| if (update_writablity) { |
| // Update aggregate writable/ready-to-send state between RTP and RTCP upon |
| // setting new channel |
| - UpdateWritableState_w(); |
| + UpdateWritableState_n(); |
| SetReadyToSend(true, new_tc && new_tc->writable()); |
| } |
| } |
| void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) { |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState); |
| tc->SignalReadPacket.connect(this, &BaseChannel::OnChannelRead); |
| @@ -349,15 +373,18 @@ void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) { |
| tc->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState); |
| tc->SignalSelectedCandidatePairChanged.connect( |
| this, &BaseChannel::OnSelectedCandidatePairChanged); |
| + tc->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n); |
| } |
| void BaseChannel::DisconnectFromTransportChannel(TransportChannel* tc) { |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| tc->SignalWritableState.disconnect(this); |
| tc->SignalReadPacket.disconnect(this); |
| tc->SignalReadyToSend.disconnect(this); |
| tc->SignalDtlsState.disconnect(this); |
| + tc->SignalSelectedCandidatePairChanged.disconnect(this); |
| + tc->SignalSentPacket.disconnect(this); |
| } |
| bool BaseChannel::Enable(bool enable) { |
| @@ -405,8 +432,10 @@ void BaseChannel::StartConnectionMonitor(int cms) { |
| // We pass in the BaseChannel instead of the transport_channel_ |
| // because if the transport_channel_ changes, the ConnectionMonitor |
| // would be pointing to the wrong TransportChannel. |
| - connection_monitor_.reset(new ConnectionMonitor( |
| - this, worker_thread(), rtc::Thread::Current())); |
| + // We pass in the network thread because on that thread connection monitor |
| + // would pull stats with BaseChannel::GetConnectionStats. |
|
pthatcher1
2016/05/11 04:50:01
Maybe change "would pull stats with BaseChannel::G
danilchap
2016/05/11 12:19:16
Done.
|
| + connection_monitor_.reset( |
| + new ConnectionMonitor(this, network_thread(), rtc::Thread::Current())); |
| connection_monitor_->SignalUpdate.connect( |
| this, &BaseChannel::OnConnectionMonitorUpdate); |
| connection_monitor_->Start(cms); |
| @@ -420,7 +449,7 @@ void BaseChannel::StopConnectionMonitor() { |
| } |
| bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) { |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| return transport_channel_->GetStats(infos); |
| } |
| @@ -435,7 +464,7 @@ bool BaseChannel::IsReadyToSend() const { |
| return enabled() && IsReceiveContentDirection(remote_content_direction_) && |
| IsSendContentDirection(local_content_direction_) && |
| was_ever_writable() && |
| - (srtp_filter_.IsActive() || !ShouldSetupDtlsSrtp()); |
| + (srtp_filter_.IsActive() || !ShouldSetupDtlsSrtp_n()); |
| } |
| bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet, |
| @@ -450,7 +479,15 @@ bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet, |
| int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt, |
| int value) { |
| - TransportChannel* channel = NULL; |
| + return network_thread_->Invoke<int>( |
| + 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()); |
| + TransportChannel* channel = nullptr; |
| switch (type) { |
| case ST_RTP: |
| channel = transport_channel_; |
| @@ -467,8 +504,10 @@ int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt, |
| } |
| void BaseChannel::OnWritableState(TransportChannel* channel) { |
| - ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_); |
| - UpdateWritableState_w(); |
| + RTC_DCHECK(channel == transport_channel_ || |
| + channel == rtcp_transport_channel_); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| + UpdateWritableState_n(); |
| } |
| void BaseChannel::OnChannelRead(TransportChannel* channel, |
| @@ -477,7 +516,7 @@ void BaseChannel::OnChannelRead(TransportChannel* channel, |
| int flags) { |
| TRACE_EVENT0("webrtc", "BaseChannel::OnChannelRead"); |
| // OnChannelRead gets called from P2PSocket; now pass data to MediaEngine |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| // When using RTCP multiplexing we might get RTCP packets on the RTP |
| // transport. We feed RTP traffic into the demuxer to determine if it is RTCP. |
| @@ -493,7 +532,7 @@ void BaseChannel::OnReadyToSend(TransportChannel* channel) { |
| void BaseChannel::OnDtlsState(TransportChannel* channel, |
| DtlsTransportState state) { |
| - if (!ShouldSetupDtlsSrtp()) { |
| + if (!ShouldSetupDtlsSrtp_n()) { |
| return; |
| } |
| @@ -512,6 +551,8 @@ void BaseChannel::OnSelectedCandidatePairChanged( |
| CandidatePairInterface* selected_candidate_pair, |
| int last_sent_packet_id) { |
| ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| + std::string transport_name = channel->transport_name(); |
| rtc::NetworkRoute network_route; |
| if (selected_candidate_pair) { |
| network_route = rtc::NetworkRoute( |
| @@ -519,26 +560,27 @@ void BaseChannel::OnSelectedCandidatePairChanged( |
| selected_candidate_pair->remote_candidate().network_id(), |
| last_sent_packet_id); |
| } |
| - media_channel()->OnNetworkRouteChanged(channel->transport_name(), |
| - network_route); |
| + invoker_.AsyncInvoke<void>( |
| + worker_thread_, Bind(&MediaChannel::OnNetworkRouteChanged, media_channel_, |
| + transport_name, network_route)); |
| } |
| void BaseChannel::SetReadyToSend(bool rtcp, bool ready) { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| if (rtcp) { |
| rtcp_ready_to_send_ = ready; |
| } else { |
| rtp_ready_to_send_ = ready; |
| } |
| - if (rtp_ready_to_send_ && |
| - // In the case of rtcp mux |rtcp_transport_channel_| will be null. |
| - (rtcp_ready_to_send_ || !rtcp_transport_channel_)) { |
| - // Notify the MediaChannel when both rtp and rtcp channel can send. |
| - media_channel_->OnReadyToSend(true); |
| - } else { |
| - // Notify the MediaChannel when either rtp or rtcp channel can't send. |
| - media_channel_->OnReadyToSend(false); |
| - } |
| + bool ready_to_send = |
| + (rtp_ready_to_send_ && |
| + // In the case of rtcp mux |rtcp_transport_channel_| will be null. |
| + (rtcp_ready_to_send_ || !rtcp_transport_channel_)); |
| + |
| + invoker_.AsyncInvoke<void>( |
| + worker_thread_, |
| + Bind(&MediaChannel::OnReadyToSend, media_channel_, ready_to_send)); |
| } |
| bool BaseChannel::PacketIsRtcp(const TransportChannel* channel, |
| @@ -550,22 +592,23 @@ bool BaseChannel::PacketIsRtcp(const TransportChannel* channel, |
| bool BaseChannel::SendPacket(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) { |
| - // SendPacket gets called from MediaEngine, typically on an encoder thread. |
| - // If the thread is not our worker thread, we will post to our worker |
| - // so that the real work happens on our worker. This avoids us having to |
| + // 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 (rtc::Thread::Current() != worker_thread_) { |
| + if (!network_thread_->IsCurrent()) { |
| // Avoid a copy by transferring the ownership of the packet data. |
| - int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET; |
| - PacketMessageData* data = new PacketMessageData; |
| + int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET; |
| + SendPacketMessageData* data = new SendPacketMessageData; |
| data->packet = std::move(*packet); |
| data->options = options; |
| - worker_thread_->Post(this, message_id, data); |
| + network_thread_->Post(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 |
| @@ -589,6 +632,7 @@ bool BaseChannel::SendPacket(bool rtcp, |
| updated_options = options; |
| // Protect if needed. |
| if (srtp_filter_.IsActive()) { |
| + TRACE_EVENT0("webrtc", "SRTP Encode"); |
| bool res; |
| uint8_t* data = packet->data(); |
| int len = static_cast<int>(packet->size()); |
| @@ -656,9 +700,9 @@ bool BaseChannel::SendPacket(bool rtcp, |
| } |
| // Bon voyage. |
| - int ret = |
| - channel->SendPacket(packet->data<char>(), packet->size(), updated_options, |
| - (secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0); |
| + int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL; |
| + int ret = channel->SendPacket(packet->data<char>(), packet->size(), |
| + updated_options, flags); |
| if (ret != static_cast<int>(packet->size())) { |
| if (channel->GetError() == EWOULDBLOCK) { |
| LOG(LS_WARNING) << "Got EWOULDBLOCK from socket."; |
| @@ -687,6 +731,7 @@ bool BaseChannel::WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) { |
| void BaseChannel::HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| if (!WantsPacket(rtcp, packet)) { |
| return; |
| } |
| @@ -700,6 +745,7 @@ void BaseChannel::HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet, |
| // Unprotect the packet, if needed. |
| if (srtp_filter_.IsActive()) { |
| + TRACE_EVENT0("webrtc", "SRTP Decode"); |
| char* data = packet->data<char>(); |
| int len = static_cast<int>(packet->size()); |
| bool res; |
| @@ -743,11 +789,22 @@ void BaseChannel::HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet, |
| return; |
| } |
| - // Push it down to the media channel. |
| - if (!rtcp) { |
| - media_channel_->OnPacketReceived(packet, packet_time); |
| + invoker_.AsyncInvoke<void>( |
| + worker_thread_, |
| + Bind(&BaseChannel::OnPacketReceived, this, rtcp, *packet, packet_time)); |
| +} |
| + |
| +void BaseChannel::OnPacketReceived(bool rtcp, |
| + const rtc::CopyOnWriteBuffer& packet, |
| + const rtc::PacketTime& packet_time) { |
| + RTC_DCHECK(worker_thread_->IsCurrent()); |
| + // Need to copy variable because OnRtcpReceived/OnPacketReceived |
| + // require non-const pointer to buffer. This doesn't memcpy the actual data. |
|
pthatcher1
2016/05/11 04:50:01
require => requires
danilchap
2016/05/11 12:19:16
Done.
|
| + rtc::CopyOnWriteBuffer data(packet); |
| + if (rtcp) { |
| + media_channel_->OnRtcpReceived(&data, packet_time); |
| } else { |
| - media_channel_->OnRtcpReceived(packet, packet_time); |
| + media_channel_->OnPacketReceived(&data, packet_time); |
| } |
| } |
| @@ -786,7 +843,7 @@ void BaseChannel::EnableMedia_w() { |
| LOG(LS_INFO) << "Channel enabled"; |
| enabled_ = true; |
| - ChangeState(); |
| + ChangeState_w(); |
| } |
| void BaseChannel::DisableMedia_w() { |
| @@ -796,20 +853,20 @@ void BaseChannel::DisableMedia_w() { |
| LOG(LS_INFO) << "Channel disabled"; |
| enabled_ = false; |
| - ChangeState(); |
| + ChangeState_w(); |
| } |
| -void BaseChannel::UpdateWritableState_w() { |
| +void BaseChannel::UpdateWritableState_n() { |
| if (transport_channel_ && transport_channel_->writable() && |
| (!rtcp_transport_channel_ || rtcp_transport_channel_->writable())) { |
| - ChannelWritable_w(); |
| + ChannelWritable_n(); |
| } else { |
| - ChannelNotWritable_w(); |
| + ChannelNotWritable_n(); |
| } |
| } |
| -void BaseChannel::ChannelWritable_w() { |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| +void BaseChannel::ChannelWritable_n() { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| if (writable_) { |
| return; |
| } |
| @@ -829,15 +886,16 @@ void BaseChannel::ChannelWritable_w() { |
| } |
| was_ever_writable_ = true; |
| - MaybeSetupDtlsSrtp_w(); |
| + MaybeSetupDtlsSrtp_n(); |
| writable_ = true; |
| ChangeState(); |
| } |
| -void BaseChannel::SignalDtlsSetupFailure_w(bool rtcp) { |
| - ASSERT(worker_thread() == rtc::Thread::Current()); |
| - signaling_thread()->Invoke<void>(Bind( |
| - &BaseChannel::SignalDtlsSetupFailure_s, this, rtcp)); |
| +void BaseChannel::SignalDtlsSetupFailure_n(bool rtcp) { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| + invoker_.AsyncInvoke<void>( |
| + signaling_thread(), |
| + Bind(&BaseChannel::SignalDtlsSetupFailure_s, this, rtcp)); |
| } |
| void BaseChannel::SignalDtlsSetupFailure_s(bool rtcp) { |
| @@ -857,14 +915,15 @@ bool BaseChannel::SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp) { |
| return tc->SetSrtpCryptoSuites(crypto_suites); |
| } |
| -bool BaseChannel::ShouldSetupDtlsSrtp() const { |
| +bool BaseChannel::ShouldSetupDtlsSrtp_n() const { |
| // Since DTLS is applied to all channels, checking RTP should be enough. |
| return transport_channel_ && transport_channel_->IsDtlsActive(); |
| } |
| // This function returns true if either DTLS-SRTP is not in use |
| // *or* DTLS-SRTP is successfully set up. |
| -bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) { |
| +bool BaseChannel::SetupDtlsSrtp_n(bool rtcp_channel) { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| bool ret = false; |
| TransportChannel* channel = |
| @@ -950,30 +1009,30 @@ bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) { |
| return ret; |
| } |
| -void BaseChannel::MaybeSetupDtlsSrtp_w() { |
| +void BaseChannel::MaybeSetupDtlsSrtp_n() { |
| if (srtp_filter_.IsActive()) { |
| return; |
| } |
| - if (!ShouldSetupDtlsSrtp()) { |
| + if (!ShouldSetupDtlsSrtp_n()) { |
| return; |
| } |
| - if (!SetupDtlsSrtp(false)) { |
| - SignalDtlsSetupFailure_w(false); |
| + if (!SetupDtlsSrtp_n(false)) { |
| + SignalDtlsSetupFailure_n(false); |
| return; |
| } |
| if (rtcp_transport_channel_) { |
| - if (!SetupDtlsSrtp(true)) { |
| - SignalDtlsSetupFailure_w(true); |
| + if (!SetupDtlsSrtp_n(true)) { |
| + SignalDtlsSetupFailure_n(true); |
| return; |
| } |
| } |
| } |
| -void BaseChannel::ChannelNotWritable_w() { |
| - ASSERT(worker_thread_ == rtc::Thread::Current()); |
| +void BaseChannel::ChannelNotWritable_n() { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| if (!writable_) |
| return; |
| @@ -982,7 +1041,7 @@ void BaseChannel::ChannelNotWritable_w() { |
| ChangeState(); |
| } |
| -bool BaseChannel::SetRtpTransportParameters_w( |
| +bool BaseChannel::SetRtpTransportParameters( |
| const MediaContentDescription* content, |
| ContentAction action, |
| ContentSource src, |
| @@ -993,15 +1052,27 @@ bool BaseChannel::SetRtpTransportParameters_w( |
| } |
| // Cache secure_required_ for belt and suspenders check on SendPacket |
| + return network_thread_->Invoke<bool>( |
| + Bind(&BaseChannel::SetRtpTransportParameters_n, this, content, action, |
| + src, error_desc)); |
| +} |
| + |
| +bool BaseChannel::SetRtpTransportParameters_n( |
| + const MediaContentDescription* content, |
| + ContentAction action, |
| + ContentSource src, |
| + std::string* error_desc) { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| + |
| if (src == CS_LOCAL) { |
| set_secure_required(content->crypto_required() != CT_NONE); |
| } |
| - if (!SetSrtp_w(content->cryptos(), action, src, error_desc)) { |
| + if (!SetSrtp_n(content->cryptos(), action, src, error_desc)) { |
| return false; |
| } |
| - if (!SetRtcpMux_w(content->rtcp_mux(), action, src, error_desc)) { |
| + if (!SetRtcpMux_n(content->rtcp_mux(), action, src, error_desc)) { |
| return false; |
| } |
| @@ -1022,7 +1093,7 @@ bool BaseChannel::CheckSrtpConfig(const std::vector<CryptoParams>& cryptos, |
| return true; |
| } |
| -bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos, |
| +bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos, |
| ContentAction action, |
| ContentSource src, |
| std::string* error_desc) { |
| @@ -1070,11 +1141,10 @@ bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos, |
| } |
| void BaseChannel::ActivateRtcpMux() { |
| - worker_thread_->Invoke<void>(Bind( |
| - &BaseChannel::ActivateRtcpMux_w, this)); |
| + network_thread_->Invoke<void>(Bind(&BaseChannel::ActivateRtcpMux_n, this)); |
| } |
| -void BaseChannel::ActivateRtcpMux_w() { |
| +void BaseChannel::ActivateRtcpMux_n() { |
| if (!rtcp_mux_filter_.IsActive()) { |
| rtcp_mux_filter_.SetActive(); |
| set_rtcp_transport_channel(nullptr, true); |
| @@ -1082,7 +1152,8 @@ void BaseChannel::ActivateRtcpMux_w() { |
| } |
| } |
| -bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action, |
| +bool BaseChannel::SetRtcpMux_n(bool enable, |
| + ContentAction action, |
| ContentSource src, |
| std::string* error_desc) { |
| bool ret = false; |
| @@ -1121,7 +1192,7 @@ bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action, |
| if (rtcp_mux_filter_.IsActive()) { |
| // If the RTP transport is already writable, then so are we. |
| if (transport_channel_->writable()) { |
| - ChannelWritable_w(); |
| + ChannelWritable_n(); |
| } |
| } |
| @@ -1296,12 +1367,14 @@ void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension( |
| void BaseChannel::OnMessage(rtc::Message *pmsg) { |
| TRACE_EVENT0("webrtc", "BaseChannel::OnMessage"); |
| switch (pmsg->message_id) { |
| - case MSG_RTPPACKET: |
| - case MSG_RTCPPACKET: { |
| - PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata); |
| - SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet, |
| - data->options); |
| - delete data; // because it is Posted |
| + 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: { |
| @@ -1311,25 +1384,39 @@ void BaseChannel::OnMessage(rtc::Message *pmsg) { |
| } |
| } |
| -void BaseChannel::FlushRtcpMessages() { |
| +void BaseChannel::FlushRtcpMessages_n() { |
| // Flush all remaining RTCP messages. This should only be called in |
| // destructor. |
| - ASSERT(rtc::Thread::Current() == worker_thread_); |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| rtc::MessageList rtcp_messages; |
| - worker_thread_->Clear(this, MSG_RTCPPACKET, &rtcp_messages); |
| - for (rtc::MessageList::iterator it = rtcp_messages.begin(); |
| - it != rtcp_messages.end(); ++it) { |
| - worker_thread_->Send(this, MSG_RTCPPACKET, it->pdata); |
| + network_thread_->Clear(this, MSG_SEND_RTCP_PACKET, &rtcp_messages); |
| + for (const auto& message : rtcp_messages) { |
| + network_thread_->Send(this, MSG_SEND_RTCP_PACKET, message.pdata); |
| } |
| } |
| -VoiceChannel::VoiceChannel(rtc::Thread* thread, |
| +void BaseChannel::SignalSentPacket_n(TransportChannel* /* channel */, |
| + const rtc::SentPacket& sent_packet) { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| + invoker_.AsyncInvoke<void>( |
| + 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); |
| +} |
| + |
| +VoiceChannel::VoiceChannel(rtc::Thread* worker_thread, |
| + rtc::Thread* network_thread, |
| MediaEngineInterface* media_engine, |
| VoiceMediaChannel* media_channel, |
| TransportController* transport_controller, |
| const std::string& content_name, |
| bool rtcp) |
| - : BaseChannel(thread, |
| + : BaseChannel(worker_thread, |
| + network_thread, |
| media_channel, |
| transport_controller, |
| content_name, |
| @@ -1487,7 +1574,13 @@ void VoiceChannel::OnChannelRead(TransportChannel* channel, |
| } |
| } |
| -void VoiceChannel::ChangeState() { |
| +void BaseChannel::ChangeState() { |
| + RTC_DCHECK(network_thread_->IsCurrent()); |
| + invoker_.AsyncInvoke<void>(worker_thread_, |
| + Bind(&BaseChannel::ChangeState_w, this)); |
| +} |
| + |
| +void VoiceChannel::ChangeState_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 = IsReadyToReceive(); |
| @@ -1521,7 +1614,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, |
| return false; |
| } |
| - if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) { |
| + if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) { |
| return false; |
| } |
| @@ -1547,7 +1640,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, |
| } |
| set_local_content_direction(content->direction()); |
| - ChangeState(); |
| + ChangeState_w(); |
| return true; |
| } |
| @@ -1566,7 +1659,7 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| return false; |
| } |
| - if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) { |
| + if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) { |
| return false; |
| } |
| @@ -1598,7 +1691,7 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| } |
| set_remote_content_direction(content->direction()); |
| - ChangeState(); |
| + ChangeState_w(); |
| return true; |
| } |
| @@ -1656,12 +1749,14 @@ void VoiceChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const { |
| GetSupportedAudioCryptoSuites(crypto_suites); |
| } |
| -VideoChannel::VideoChannel(rtc::Thread* thread, |
| +VideoChannel::VideoChannel(rtc::Thread* worker_thread, |
| + rtc::Thread* network_thread, |
| VideoMediaChannel* media_channel, |
| TransportController* transport_controller, |
| const std::string& content_name, |
| bool rtcp) |
| - : BaseChannel(thread, |
| + : BaseChannel(worker_thread, |
| + network_thread, |
| media_channel, |
| transport_controller, |
| content_name, |
| @@ -1723,7 +1818,8 @@ bool VideoChannel::SetRtpParameters_w(uint32_t ssrc, |
| webrtc::RtpParameters parameters) { |
| return media_channel()->SetRtpParameters(ssrc, parameters); |
| } |
| -void VideoChannel::ChangeState() { |
| + |
| +void VideoChannel::ChangeState_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 = IsReadyToSend(); |
| @@ -1775,7 +1871,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, |
| return false; |
| } |
| - if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) { |
| + if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) { |
| return false; |
| } |
| @@ -1801,7 +1897,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, |
| } |
| set_local_content_direction(content->direction()); |
| - ChangeState(); |
| + ChangeState_w(); |
| return true; |
| } |
| @@ -1820,8 +1916,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| return false; |
| } |
| - |
| - if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) { |
| + if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) { |
| return false; |
| } |
| @@ -1854,7 +1949,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| } |
| set_remote_content_direction(content->direction()); |
| - ChangeState(); |
| + ChangeState_w(); |
| return true; |
| } |
| @@ -1889,12 +1984,14 @@ void VideoChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const { |
| GetSupportedVideoCryptoSuites(crypto_suites); |
| } |
| -DataChannel::DataChannel(rtc::Thread* thread, |
| +DataChannel::DataChannel(rtc::Thread* worker_thread, |
| + rtc::Thread* network_thread, |
| DataMediaChannel* media_channel, |
| TransportController* transport_controller, |
| const std::string& content_name, |
| bool rtcp) |
| - : BaseChannel(thread, |
| + : BaseChannel(worker_thread, |
| + network_thread, |
| media_channel, |
| transport_controller, |
| content_name, |
| @@ -1998,7 +2095,7 @@ bool DataChannel::SetLocalContent_w(const MediaContentDescription* content, |
| } |
| if (data_channel_type_ == DCT_RTP) { |
| - if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) { |
| + if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) { |
| return false; |
| } |
| } |
| @@ -2030,7 +2127,7 @@ bool DataChannel::SetLocalContent_w(const MediaContentDescription* content, |
| } |
| set_local_content_direction(content->direction()); |
| - ChangeState(); |
| + ChangeState_w(); |
| return true; |
| } |
| @@ -2060,7 +2157,7 @@ bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| LOG(LS_INFO) << "Setting remote data description"; |
| if (data_channel_type_ == DCT_RTP && |
| - !SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) { |
| + !SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) { |
| return false; |
| } |
| @@ -2085,11 +2182,11 @@ bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| } |
| set_remote_content_direction(content->direction()); |
| - ChangeState(); |
| + ChangeState_w(); |
| return true; |
| } |
| -void DataChannel::ChangeState() { |
| +void DataChannel::ChangeState_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 = IsReadyToReceive(); |
| @@ -2199,8 +2296,8 @@ void DataChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const { |
| GetSupportedDataCryptoSuites(crypto_suites); |
| } |
| -bool DataChannel::ShouldSetupDtlsSrtp() const { |
| - return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp(); |
| +bool DataChannel::ShouldSetupDtlsSrtp_n() const { |
| + return data_channel_type_ == DCT_RTP && BaseChannel::ShouldSetupDtlsSrtp_n(); |
| } |
| void DataChannel::OnStreamClosedRemotely(uint32_t sid) { |