| /* |
| * 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 "pc/channel.h" |
| |
| #include <iterator> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/memory/memory.h" |
| #include "api/call/audio_sink.h" |
| #include "media/base/media_constants.h" |
| #include "media/base/rtp_utils.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "p2p/base/packet_transport_internal.h" |
| #include "pc/channel_manager.h" |
| #include "pc/rtp_media_utils.h" |
| #include "rtc_base/bind.h" |
| #include "rtc_base/byte_order.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/dscp.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/network_route.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/synchronization/sequence_checker.h" |
| #include "rtc_base/trace_event.h" |
| |
| namespace cricket { |
| using rtc::Bind; |
| using rtc::UniqueRandomIdGenerator; |
| using webrtc::SdpType; |
| |
| namespace { |
| |
| struct SendPacketMessageData : public rtc::MessageData { |
| rtc::CopyOnWriteBuffer packet; |
| rtc::PacketOptions options; |
| }; |
| |
| // Finds a stream based on target's Primary SSRC or RIDs. |
| // This struct is used in BaseChannel::UpdateLocalStreams_w. |
| struct StreamFinder { |
| explicit StreamFinder(const StreamParams* target) : target_(target) { |
| RTC_DCHECK(target); |
| } |
| |
| bool operator()(const StreamParams& sp) const { |
| if (target_->has_ssrcs() && sp.has_ssrcs()) { |
| return sp.has_ssrc(target_->first_ssrc()); |
| } |
| |
| if (!target_->has_rids() && !sp.has_rids()) { |
| return false; |
| } |
| |
| const std::vector<RidDescription>& target_rids = target_->rids(); |
| const std::vector<RidDescription>& source_rids = sp.rids(); |
| if (source_rids.size() != target_rids.size()) { |
| return false; |
| } |
| |
| // Check that all RIDs match. |
| return std::equal(source_rids.begin(), source_rids.end(), |
| target_rids.begin(), |
| [](const RidDescription& lhs, const RidDescription& rhs) { |
| return lhs.rid == rhs.rid; |
| }); |
| } |
| |
| const StreamParams* target_; |
| }; |
| |
| } // namespace |
| |
| enum { |
| MSG_SEND_RTP_PACKET = 1, |
| MSG_SEND_RTCP_PACKET, |
| MSG_READYTOSENDDATA, |
| MSG_DATARECEIVED, |
| MSG_FIRSTPACKETRECEIVED, |
| }; |
| |
| static void SafeSetError(const std::string& message, std::string* error_desc) { |
| if (error_desc) { |
| *error_desc = message; |
| } |
| } |
| |
| template <class Codec> |
| void RtpParametersFromMediaDescription( |
| const MediaContentDescriptionImpl<Codec>* desc, |
| const RtpHeaderExtensions& extensions, |
| bool is_stream_active, |
| RtpParameters<Codec>* params) { |
| params->is_stream_active = is_stream_active; |
| params->codecs = desc->codecs(); |
| // TODO(bugs.webrtc.org/11513): 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(); |
| params->rtcp.remote_estimate = desc->remote_estimate(); |
| } |
| |
| template <class Codec> |
| void RtpSendParametersFromMediaDescription( |
| const MediaContentDescriptionImpl<Codec>* desc, |
| const RtpHeaderExtensions& extensions, |
| bool is_stream_active, |
| RtpSendParameters<Codec>* send_params) { |
| RtpParametersFromMediaDescription(desc, extensions, is_stream_active, |
| send_params); |
| send_params->max_bandwidth_bps = desc->bandwidth(); |
| send_params->extmap_allow_mixed = desc->extmap_allow_mixed(); |
| } |
| |
| 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 srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| UniqueRandomIdGenerator* ssrc_generator) |
| : worker_thread_(worker_thread), |
| network_thread_(network_thread), |
| signaling_thread_(signaling_thread), |
| content_name_(content_name), |
| srtp_required_(srtp_required), |
| crypto_options_(crypto_options), |
| media_channel_(std::move(media_channel)), |
| ssrc_generator_(ssrc_generator) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| RTC_DCHECK(ssrc_generator_); |
| demuxer_criteria_.mid = content_name; |
| RTC_LOG(LS_INFO) << "Created channel: " << ToString(); |
| } |
| |
| BaseChannel::~BaseChannel() { |
| TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel"); |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| |
| // Eats any outstanding messages or packets. |
| worker_thread_->Clear(&invoker_); |
| worker_thread_->Clear(this); |
| // The media channel is destroyed at the end of the destructor, since it |
| // is a std::unique_ptr. The transport channel (rtp_transport) must outlive |
| // the media channel. |
| } |
| |
| std::string BaseChannel::ToString() const { |
| rtc::StringBuilder sb; |
| sb << "{mid: " << content_name_; |
| if (media_channel_) { |
| sb << ", media_type: " << MediaTypeToString(media_channel_->media_type()); |
| } |
| sb << "}"; |
| return sb.Release(); |
| } |
| |
| bool BaseChannel::ConnectToRtpTransport() { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(rtp_transport_); |
| // TODO(bugs.webrtc.org/12230): This accesses demuxer_criteria_ on the |
| // networking thread. |
| if (!rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this)) { |
| RTC_LOG(LS_ERROR) << "Failed to set up demuxing for " << ToString(); |
| return false; |
| } |
| rtp_transport_->SignalReadyToSend.connect( |
| this, &BaseChannel::OnTransportReadyToSend); |
| rtp_transport_->SignalNetworkRouteChanged.connect( |
| this, &BaseChannel::OnNetworkRouteChanged); |
| rtp_transport_->SignalWritableState.connect(this, |
| &BaseChannel::OnWritableState); |
| rtp_transport_->SignalSentPacket.connect(this, |
| &BaseChannel::SignalSentPacket_n); |
| return true; |
| } |
| |
| void BaseChannel::DisconnectFromRtpTransport() { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(rtp_transport_); |
| rtp_transport_->UnregisterRtpDemuxerSink(this); |
| rtp_transport_->SignalReadyToSend.disconnect(this); |
| rtp_transport_->SignalNetworkRouteChanged.disconnect(this); |
| rtp_transport_->SignalWritableState.disconnect(this); |
| rtp_transport_->SignalSentPacket.disconnect(this); |
| } |
| |
| void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| |
| network_thread_->Invoke<void>( |
| RTC_FROM_HERE, [this, rtp_transport] { SetRtpTransport(rtp_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::Deinit() { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| media_channel_->SetInterface(/*iface=*/nullptr); |
| // 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, [&] { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| FlushRtcpMessages_n(); |
| |
| if (rtp_transport_) { |
| DisconnectFromRtpTransport(); |
| } |
| // Clear pending read packets/messages. |
| network_thread_->Clear(&invoker_); |
| network_thread_->Clear(this); |
| }); |
| } |
| |
| bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) { |
| if (!network_thread_->IsCurrent()) { |
| return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this, rtp_transport] { |
| return SetRtpTransport(rtp_transport); |
| }); |
| } |
| RTC_DCHECK_RUN_ON(network_thread()); |
| if (rtp_transport == rtp_transport_) { |
| return true; |
| } |
| |
| if (rtp_transport_) { |
| DisconnectFromRtpTransport(); |
| } |
| |
| rtp_transport_ = rtp_transport; |
| if (rtp_transport_) { |
| transport_name_ = rtp_transport_->transport_name(); |
| |
| if (!ConnectToRtpTransport()) { |
| RTC_LOG(LS_ERROR) << "Failed to connect to the new RtpTransport for " |
| << ToString() << "."; |
| return false; |
| } |
| OnTransportReadyToSend(rtp_transport_->IsReadyToSend()); |
| UpdateWritableState_n(); |
| |
| // Set the cached socket options. |
| for (const auto& pair : socket_options_) { |
| rtp_transport_->SetRtpOption(pair.first, pair.second); |
| } |
| if (!rtp_transport_->rtcp_mux_enabled()) { |
| for (const auto& pair : rtcp_socket_options_) { |
| rtp_transport_->SetRtcpOption(pair.first, pair.second); |
| } |
| } |
| } |
| return true; |
| } |
| |
| 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::SetLocalContent(const MediaContentDescription* content, |
| SdpType type, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent"); |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&BaseChannel::SetLocalContent_w, this, content, type, error_desc)); |
| } |
| |
| bool BaseChannel::SetRemoteContent(const MediaContentDescription* content, |
| SdpType type, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent"); |
| return InvokeOnWorker<bool>( |
| RTC_FROM_HERE, |
| Bind(&BaseChannel::SetRemoteContent_w, this, content, type, error_desc)); |
| } |
| |
| void BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) { |
| TRACE_EVENT0("webrtc", "BaseChannel::SetPayloadTypeDemuxingEnabled"); |
| InvokeOnWorker<void>( |
| RTC_FROM_HERE, |
| Bind(&BaseChannel::SetPayloadTypeDemuxingEnabled_w, this, enabled)); |
| } |
| |
| bool BaseChannel::UpdateRtpTransport(std::string* error_desc) { |
| return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this, error_desc] { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(rtp_transport_); |
| // TODO(bugs.webrtc.org/12230): This accesses demuxer_criteria_ on the |
| // networking thread. |
| if (!rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this)) { |
| RTC_LOG(LS_ERROR) << "Failed to set up demuxing for " << ToString(); |
| rtc::StringBuilder desc; |
| desc << "Failed to set up demuxing for m-section with mid='" |
| << content_name() << "'."; |
| SafeSetError(desc.str(), error_desc); |
| return false; |
| } |
| // NOTE: This doesn't take the BUNDLE case in account meaning the RTP header |
| // extension maps are not merged when BUNDLE is enabled. This is fine |
| // because the ID for MID should be consistent among all the RTP transports, |
| // and that's all RtpTransport uses this map for. |
| // |
| // TODO(deadbeef): Move this call to JsepTransport, there is no reason |
| // BaseChannel needs to be involved here. |
| if (media_type() != cricket::MEDIA_TYPE_DATA) { |
| rtp_transport_->UpdateRtpHeaderExtensionMap( |
| receive_rtp_header_extensions_); |
| } |
| return true; |
| }); |
| } |
| |
| bool BaseChannel::IsReadyToReceiveMedia_w() const { |
| // Receive data if we are enabled and have local content, |
| return enabled() && |
| webrtc::RtpTransceiverDirectionHasRecv(local_content_direction_); |
| } |
| |
| bool BaseChannel::IsReadyToSendMedia_w() const { |
| // Send outgoing data if we are enabled, have local and remote content, |
| // and we have had some form of connectivity. |
| return enabled() && |
| webrtc::RtpTransceiverDirectionHasRecv(remote_content_direction_) && |
| webrtc::RtpTransceiverDirectionHasSend(local_content_direction_) && |
| was_ever_writable(); |
| } |
| |
| 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(rtp_transport_); |
| switch (type) { |
| case ST_RTP: |
| socket_options_.push_back( |
| std::pair<rtc::Socket::Option, int>(opt, value)); |
| return rtp_transport_->SetRtpOption(opt, value); |
| case ST_RTCP: |
| rtcp_socket_options_.push_back( |
| std::pair<rtc::Socket::Option, int>(opt, value)); |
| return rtp_transport_->SetRtcpOption(opt, value); |
| } |
| return -1; |
| } |
| |
| void BaseChannel::OnWritableState(bool writable) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| if (writable) { |
| ChannelWritable_n(); |
| } else { |
| ChannelNotWritable_n(); |
| } |
| } |
| |
| void BaseChannel::OnNetworkRouteChanged( |
| absl::optional<rtc::NetworkRoute> network_route) { |
| RTC_LOG(LS_INFO) << "Network route for " << ToString() << " was changed."; |
| |
| RTC_DCHECK_RUN_ON(network_thread()); |
| 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_, [=] { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| media_channel_->OnNetworkRouteChanged(transport_name_, new_route); |
| }); |
| } |
| |
| sigslot::signal1<ChannelInterface*>& BaseChannel::SignalFirstPacketReceived() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| return SignalFirstPacketReceived_; |
| } |
| |
| sigslot::signal1<const rtc::SentPacket&>& BaseChannel::SignalSentPacket() { |
| // TODO(bugs.webrtc.org/11994): Uncomment this check once callers have been |
| // fixed to access this variable from the correct thread. |
| // RTC_DCHECK_RUN_ON(worker_thread_); |
| return SignalSentPacket_; |
| } |
| |
| void BaseChannel::OnTransportReadyToSend(bool ready) { |
| invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [=] { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| media_channel_->OnReadyToSend(ready); |
| }); |
| } |
| |
| bool BaseChannel::SendPacket(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) { |
| // Until all the code is migrated to use RtpPacketType instead of bool. |
| RtpPacketType packet_type = rtcp ? RtpPacketType::kRtcp : RtpPacketType::kRtp; |
| // 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; |
| } |
| RTC_DCHECK_RUN_ON(network_thread()); |
| |
| 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_ || !rtp_transport_->IsWritable(rtcp)) { |
| return false; |
| } |
| |
| // Protect ourselves against crazy data. |
| if (!IsValidRtpPacketSize(packet_type, packet->size())) { |
| RTC_LOG(LS_ERROR) << "Dropping outgoing " << ToString() << " " |
| << RtpPacketTypeToString(packet_type) |
| << " 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 for " << ToString() |
| << " when SRTP is inactive and crypto is required"; |
| RTC_NOTREACHED(); |
| return false; |
| } |
| |
| std::string packet_type = rtcp ? "RTCP" : "RTP"; |
| RTC_DLOG(LS_WARNING) << "Sending an " << packet_type |
| << " packet without encryption for " << ToString() |
| << "."; |
| } |
| |
| // Bon voyage. |
| return rtcp ? rtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS) |
| : rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS); |
| } |
| |
| void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) { |
| // Take packet time from the |parsed_packet|. |
| // RtpPacketReceived.arrival_time_ms = (timestamp_us + 500) / 1000; |
| int64_t packet_time_us = -1; |
| if (parsed_packet.arrival_time_ms() > 0) { |
| packet_time_us = parsed_packet.arrival_time_ms() * 1000; |
| } |
| |
| if (!has_received_packet_) { |
| 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 RTP packet when " |
| "SRTP is inactive and crypto is required " |
| << ToString(); |
| return; |
| } |
| |
| auto packet_buffer = parsed_packet.Buffer(); |
| |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, worker_thread_, [this, packet_buffer, packet_time_us] { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| media_channel_->OnPacketReceived(packet_buffer, packet_time_us); |
| }); |
| } |
| |
| void BaseChannel::EnableMedia_w() { |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| if (enabled_) |
| return; |
| |
| RTC_LOG(LS_INFO) << "Channel enabled: " << ToString(); |
| enabled_ = true; |
| UpdateMediaSendRecvState_w(); |
| } |
| |
| void BaseChannel::DisableMedia_w() { |
| RTC_DCHECK(worker_thread_ == rtc::Thread::Current()); |
| if (!enabled_) |
| return; |
| |
| RTC_LOG(LS_INFO) << "Channel disabled: " << ToString(); |
| enabled_ = false; |
| UpdateMediaSendRecvState_w(); |
| } |
| |
| void BaseChannel::UpdateWritableState_n() { |
| if (rtp_transport_->IsWritable(/*rtcp=*/true) && |
| rtp_transport_->IsWritable(/*rtcp=*/false)) { |
| ChannelWritable_n(); |
| } else { |
| ChannelNotWritable_n(); |
| } |
| } |
| |
| void BaseChannel::ChannelWritable_n() { |
| if (writable_) { |
| return; |
| } |
| writable_ = true; |
| |
| RTC_LOG(LS_INFO) << "Channel writable (" << ToString() << ")" |
| << (was_ever_writable_n_ ? "" : " for the first time"); |
| // We only have to do this AsyncInvoke once, when first transitioning to |
| // writable. |
| if (!was_ever_writable_n_) { |
| invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [this] { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| was_ever_writable_ = true; |
| UpdateMediaSendRecvState_w(); |
| }); |
| } |
| was_ever_writable_n_ = true; |
| } |
| |
| void BaseChannel::ChannelNotWritable_n() { |
| if (!writable_) { |
| return; |
| } |
| writable_ = false; |
| RTC_LOG(LS_INFO) << "Channel not writable (" << ToString() << ")"; |
| } |
| |
| bool BaseChannel::AddRecvStream_w(const StreamParams& sp) { |
| return media_channel()->AddRecvStream(sp); |
| } |
| |
| bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) { |
| return media_channel()->RemoveRecvStream(ssrc); |
| } |
| |
| void BaseChannel::ResetUnsignaledRecvStream_w() { |
| RTC_DCHECK(worker_thread() == rtc::Thread::Current()); |
| media_channel()->ResetUnsignaledRecvStream(); |
| } |
| |
| void BaseChannel::SetPayloadTypeDemuxingEnabled_w(bool enabled) { |
| if (enabled == payload_type_demuxing_enabled_) { |
| return; |
| } |
| payload_type_demuxing_enabled_ = enabled; |
| if (!enabled) { |
| // TODO(crbug.com/11477): This will remove *all* unsignaled streams (those |
| // without an explicitly signaled SSRC), which may include streams that |
| // were matched to this channel by MID or RID. Ideally we'd remove only the |
| // streams that were matched based on payload type alone, but currently |
| // there is no straightforward way to identify those streams. |
| media_channel()->ResetUnsignaledRecvStream(); |
| demuxer_criteria_.payload_types.clear(); |
| } else if (!payload_types_.empty()) { |
| demuxer_criteria_.payload_types.insert(payload_types_.begin(), |
| payload_types_.end()); |
| } |
| } |
| |
| bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams, |
| SdpType type, |
| std::string* error_desc) { |
| // In the case of RIDs (where SSRCs are not negotiated), this method will |
| // generate an SSRC for each layer in StreamParams. That representation will |
| // be stored internally in |local_streams_|. |
| // In subsequent offers, the same stream can appear in |streams| again |
| // (without the SSRCs), so it should be looked up using RIDs (if available) |
| // and then by primary SSRC. |
| // In both scenarios, it is safe to assume that the media channel will be |
| // created with a StreamParams object with SSRCs. However, it is not safe to |
| // assume that |local_streams_| will always have SSRCs as there are scenarios |
| // in which niether SSRCs or RIDs are negotiated. |
| |
| // Check for streams that have been removed. |
| bool ret = true; |
| for (const StreamParams& old_stream : local_streams_) { |
| if (!old_stream.has_ssrcs() || |
| GetStream(streams, StreamFinder(&old_stream))) { |
| continue; |
| } |
| if (!media_channel()->RemoveSendStream(old_stream.first_ssrc())) { |
| rtc::StringBuilder desc; |
| desc << "Failed to remove send stream with ssrc " |
| << old_stream.first_ssrc() << " from m-section with mid='" |
| << content_name() << "'."; |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| } |
| } |
| // Check for new streams. |
| std::vector<StreamParams> all_streams; |
| for (const StreamParams& stream : streams) { |
| StreamParams* existing = GetStream(local_streams_, StreamFinder(&stream)); |
| if (existing) { |
| // Parameters cannot change for an existing stream. |
| all_streams.push_back(*existing); |
| continue; |
| } |
| |
| all_streams.push_back(stream); |
| StreamParams& new_stream = all_streams.back(); |
| |
| if (!new_stream.has_ssrcs() && !new_stream.has_rids()) { |
| continue; |
| } |
| |
| RTC_DCHECK(new_stream.has_ssrcs() || new_stream.has_rids()); |
| if (new_stream.has_ssrcs() && new_stream.has_rids()) { |
| rtc::StringBuilder desc; |
| desc << "Failed to add send stream: " << new_stream.first_ssrc() |
| << " into m-section with mid='" << content_name() |
| << "'. Stream has both SSRCs and RIDs."; |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| continue; |
| } |
| |
| // At this point we use the legacy simulcast group in StreamParams to |
| // indicate that we want multiple layers to the media channel. |
| if (!new_stream.has_ssrcs()) { |
| // TODO(bugs.webrtc.org/10250): Indicate if flex is desired here. |
| new_stream.GenerateSsrcs(new_stream.rids().size(), /* rtx = */ true, |
| /* flex_fec = */ false, ssrc_generator_); |
| } |
| |
| if (media_channel()->AddSendStream(new_stream)) { |
| RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0] |
| << " into " << ToString(); |
| } else { |
| rtc::StringBuilder desc; |
| desc << "Failed to add send stream ssrc: " << new_stream.first_ssrc() |
| << " into m-section with mid='" << content_name() << "'"; |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| } |
| } |
| local_streams_ = all_streams; |
| return ret; |
| } |
| |
| bool BaseChannel::UpdateRemoteStreams_w( |
| const std::vector<StreamParams>& streams, |
| SdpType type, |
| std::string* error_desc) { |
| // Check for streams that have been removed. |
| bool ret = true; |
| for (const StreamParams& old_stream : remote_streams_) { |
| // If we no longer have an unsignaled stream, we would like to remove |
| // the unsignaled stream params that are cached. |
| if (!old_stream.has_ssrcs() && !HasStreamWithNoSsrcs(streams)) { |
| ResetUnsignaledRecvStream_w(); |
| RTC_LOG(LS_INFO) << "Reset unsignaled remote stream for " << ToString() |
| << "."; |
| } else if (old_stream.has_ssrcs() && |
| !GetStreamBySsrc(streams, old_stream.first_ssrc())) { |
| if (RemoveRecvStream_w(old_stream.first_ssrc())) { |
| RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc() |
| << " from " << ToString() << "."; |
| } else { |
| rtc::StringBuilder desc; |
| desc << "Failed to remove remote stream with ssrc " |
| << old_stream.first_ssrc() << " from m-section with mid='" |
| << content_name() << "'."; |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| } |
| } |
| } |
| demuxer_criteria_.ssrcs.clear(); |
| // Check for new streams. |
| for (const StreamParams& new_stream : streams) { |
| // We allow a StreamParams with an empty list of SSRCs, in which case the |
| // MediaChannel will cache the parameters and use them for any unsignaled |
| // stream received later. |
| if ((!new_stream.has_ssrcs() && !HasStreamWithNoSsrcs(remote_streams_)) || |
| !GetStreamBySsrc(remote_streams_, new_stream.first_ssrc())) { |
| if (AddRecvStream_w(new_stream)) { |
| RTC_LOG(LS_INFO) << "Add remote ssrc: " |
| << (new_stream.has_ssrcs() |
| ? std::to_string(new_stream.first_ssrc()) |
| : "unsignaled") |
| << " to " << ToString(); |
| } else { |
| rtc::StringBuilder desc; |
| desc << "Failed to add remote stream ssrc: " |
| << (new_stream.has_ssrcs() |
| ? std::to_string(new_stream.first_ssrc()) |
| : "unsignaled") |
| << " to " << ToString(); |
| SafeSetError(desc.str(), error_desc); |
| ret = false; |
| } |
| } |
| // Update the receiving SSRCs. |
| demuxer_criteria_.ssrcs.insert(new_stream.ssrcs.begin(), |
| new_stream.ssrcs.end()); |
| } |
| remote_streams_ = streams; |
| return ret; |
| } |
| |
| RtpHeaderExtensions BaseChannel::GetFilteredRtpHeaderExtensions( |
| const RtpHeaderExtensions& extensions) { |
| if (crypto_options_.srtp.enable_encrypted_rtp_header_extensions) { |
| RtpHeaderExtensions filtered; |
| absl::c_copy_if(extensions, std::back_inserter(filtered), |
| [](const webrtc::RtpExtension& extension) { |
| return !extension.encrypt; |
| }); |
| return filtered; |
| } |
| |
| return webrtc::RtpExtension::FilterDuplicateNonEncrypted(extensions); |
| } |
| |
| void BaseChannel::SetReceiveExtensions(const RtpHeaderExtensions& extensions) { |
| receive_rtp_header_extensions_ = extensions; |
| } |
| |
| 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_RUN_ON(network_thread()); |
| 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: { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| SignalFirstPacketReceived_(this); |
| break; |
| } |
| } |
| } |
| |
| void BaseChannel::MaybeAddHandledPayloadType(int payload_type) { |
| if (payload_type_demuxing_enabled_) { |
| demuxer_criteria_.payload_types.insert(static_cast<uint8_t>(payload_type)); |
| } |
| // Even if payload type demuxing is currently disabled, we need to remember |
| // the payload types in case it's re-enabled later. |
| payload_types_.insert(static_cast<uint8_t>(payload_type)); |
| } |
| |
| void BaseChannel::ClearHandledPayloadTypes() { |
| demuxer_criteria_.payload_types.clear(); |
| payload_types_.clear(); |
| } |
| |
| void BaseChannel::FlushRtcpMessages_n() { |
| // Flush all remaining RTCP messages. This should only be called in |
| // destructor. |
| 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(const rtc::SentPacket& sent_packet) { |
| invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, |
| [this, sent_packet] { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| SignalSentPacket()(sent_packet); |
| }); |
| } |
| |
| void BaseChannel::SetNegotiatedHeaderExtensions_w( |
| const RtpHeaderExtensions& extensions) { |
| TRACE_EVENT0("webrtc", __func__); |
| RtpHeaderExtensions extensions_copy = extensions; |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), |
| [this, extensions_copy = std::move(extensions_copy)] { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| negotiated_header_extensions_ = std::move(extensions_copy); |
| }); |
| } |
| |
| RtpHeaderExtensions BaseChannel::GetNegotiatedRtpHeaderExtensions() const { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return negotiated_header_extensions_; |
| } |
| |
| VoiceChannel::VoiceChannel(rtc::Thread* worker_thread, |
| rtc::Thread* network_thread, |
| rtc::Thread* signaling_thread, |
| std::unique_ptr<VoiceMediaChannel> media_channel, |
| const std::string& content_name, |
| bool srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| UniqueRandomIdGenerator* ssrc_generator) |
| : BaseChannel(worker_thread, |
| network_thread, |
| signaling_thread, |
| std::move(media_channel), |
| content_name, |
| srtp_required, |
| crypto_options, |
| ssrc_generator) {} |
| |
| VoiceChannel::~VoiceChannel() { |
| TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel"); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| Deinit(); |
| } |
| |
| void VoiceChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) { |
| BaseChannel::Init_w(rtp_transport); |
| } |
| |
| 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. |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| 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 |
| << " for " << ToString(); |
| } |
| |
| bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w"); |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| RTC_LOG(LS_INFO) << "Setting local voice description for " << ToString(); |
| |
| RTC_DCHECK(content); |
| if (!content) { |
| SafeSetError("Can't find audio content in local description.", error_desc); |
| return false; |
| } |
| |
| const AudioContentDescription* audio = content->as_audio(); |
| |
| if (type == SdpType::kAnswer) |
| SetNegotiatedHeaderExtensions_w(audio->rtp_header_extensions()); |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions()); |
| SetReceiveExtensions(rtp_header_extensions); |
| media_channel()->SetExtmapAllowMixed(audio->extmap_allow_mixed()); |
| |
| AudioRecvParameters recv_params = last_recv_params_; |
| RtpParametersFromMediaDescription( |
| audio, rtp_header_extensions, |
| webrtc::RtpTransceiverDirectionHasRecv(audio->direction()), &recv_params); |
| if (!media_channel()->SetRecvParameters(recv_params)) { |
| SafeSetError( |
| "Failed to set local audio description recv parameters for m-section " |
| "with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| |
| if (webrtc::RtpTransceiverDirectionHasRecv(audio->direction())) { |
| for (const AudioCodec& codec : audio->codecs()) { |
| MaybeAddHandledPayloadType(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(), type, error_desc)) { |
| SafeSetError( |
| "Failed to set local audio description streams for m-section with " |
| "mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| |
| set_local_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w"); |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| RTC_LOG(LS_INFO) << "Setting remote voice description for " << ToString(); |
| |
| RTC_DCHECK(content); |
| if (!content) { |
| SafeSetError("Can't find audio content in remote description.", error_desc); |
| return false; |
| } |
| |
| const AudioContentDescription* audio = content->as_audio(); |
| |
| if (type == SdpType::kAnswer) |
| SetNegotiatedHeaderExtensions_w(audio->rtp_header_extensions()); |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions()); |
| |
| AudioSendParameters send_params = last_send_params_; |
| RtpSendParametersFromMediaDescription( |
| audio, rtp_header_extensions, |
| webrtc::RtpTransceiverDirectionHasRecv(audio->direction()), &send_params); |
| send_params.mid = content_name(); |
| |
| bool parameters_applied = media_channel()->SetSendParameters(send_params); |
| if (!parameters_applied) { |
| SafeSetError( |
| "Failed to set remote audio description send parameters for m-section " |
| "with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| last_send_params_ = send_params; |
| |
| if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) { |
| RTC_DLOG(LS_VERBOSE) << "SetRemoteContent_w: remote side will not send - " |
| "disable payload type demuxing for " |
| << ToString(); |
| ClearHandledPayloadTypes(); |
| } |
| |
| // 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(), type, error_desc)) { |
| SafeSetError( |
| "Failed to set remote audio description streams for m-section with " |
| "mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| |
| set_remote_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| 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 srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| UniqueRandomIdGenerator* ssrc_generator) |
| : BaseChannel(worker_thread, |
| network_thread, |
| signaling_thread, |
| std::move(media_channel), |
| content_name, |
| srtp_required, |
| crypto_options, |
| ssrc_generator) {} |
| |
| VideoChannel::~VideoChannel() { |
| TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel"); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| Deinit(); |
| } |
| |
| 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. |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| bool send = IsReadyToSendMedia_w(); |
| if (!media_channel()->SetSend(send)) { |
| RTC_LOG(LS_ERROR) << "Failed to SetSend on video channel: " + ToString(); |
| // TODO(gangji): Report error back to server. |
| } |
| |
| RTC_LOG(LS_INFO) << "Changing video state, send=" << send << " for " |
| << ToString(); |
| } |
| |
| void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) { |
| InvokeOnWorker<void>(RTC_FROM_HERE, Bind(&VideoMediaChannel::FillBitrateInfo, |
| media_channel(), bwe_info)); |
| } |
| |
| bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w"); |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| RTC_LOG(LS_INFO) << "Setting local video description for " << ToString(); |
| |
| RTC_DCHECK(content); |
| if (!content) { |
| SafeSetError("Can't find video content in local description.", error_desc); |
| return false; |
| } |
| |
| const VideoContentDescription* video = content->as_video(); |
| |
| if (type == SdpType::kAnswer) |
| SetNegotiatedHeaderExtensions_w(video->rtp_header_extensions()); |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(video->rtp_header_extensions()); |
| SetReceiveExtensions(rtp_header_extensions); |
| media_channel()->SetExtmapAllowMixed(video->extmap_allow_mixed()); |
| |
| VideoRecvParameters recv_params = last_recv_params_; |
| RtpParametersFromMediaDescription( |
| video, rtp_header_extensions, |
| webrtc::RtpTransceiverDirectionHasRecv(video->direction()), &recv_params); |
| |
| VideoSendParameters send_params = last_send_params_; |
| |
| bool needs_send_params_update = false; |
| if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) { |
| for (auto& send_codec : send_params.codecs) { |
| auto* recv_codec = FindMatchingCodec(recv_params.codecs, send_codec); |
| if (recv_codec) { |
| if (!recv_codec->packetization && send_codec.packetization) { |
| send_codec.packetization.reset(); |
| needs_send_params_update = true; |
| } else if (recv_codec->packetization != send_codec.packetization) { |
| SafeSetError( |
| "Failed to set local answer due to invalid codec packetization " |
| "specified in m-section with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| } |
| } |
| } |
| |
| if (!media_channel()->SetRecvParameters(recv_params)) { |
| SafeSetError( |
| "Failed to set local video description recv parameters for m-section " |
| "with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| |
| if (webrtc::RtpTransceiverDirectionHasRecv(video->direction())) { |
| for (const VideoCodec& codec : video->codecs()) { |
| MaybeAddHandledPayloadType(codec.id); |
| } |
| } |
| |
| last_recv_params_ = recv_params; |
| |
| if (needs_send_params_update) { |
| if (!media_channel()->SetSendParameters(send_params)) { |
| SafeSetError("Failed to set send parameters for m-section with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| last_send_params_ = send_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(), type, error_desc)) { |
| SafeSetError( |
| "Failed to set local video description streams for m-section with " |
| "mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| |
| set_local_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w"); |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| RTC_LOG(LS_INFO) << "Setting remote video description for " << ToString(); |
| |
| RTC_DCHECK(content); |
| if (!content) { |
| SafeSetError("Can't find video content in remote description.", error_desc); |
| return false; |
| } |
| |
| const VideoContentDescription* video = content->as_video(); |
| |
| if (type == SdpType::kAnswer) |
| SetNegotiatedHeaderExtensions_w(video->rtp_header_extensions()); |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(video->rtp_header_extensions()); |
| |
| VideoSendParameters send_params = last_send_params_; |
| RtpSendParametersFromMediaDescription( |
| video, rtp_header_extensions, |
| webrtc::RtpTransceiverDirectionHasRecv(video->direction()), &send_params); |
| if (video->conference_mode()) { |
| send_params.conference_mode = true; |
| } |
| send_params.mid = content_name(); |
| |
| VideoRecvParameters recv_params = last_recv_params_; |
| |
| bool needs_recv_params_update = false; |
| if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) { |
| for (auto& recv_codec : recv_params.codecs) { |
| auto* send_codec = FindMatchingCodec(send_params.codecs, recv_codec); |
| if (send_codec) { |
| if (!send_codec->packetization && recv_codec.packetization) { |
| recv_codec.packetization.reset(); |
| needs_recv_params_update = true; |
| } else if (send_codec->packetization != recv_codec.packetization) { |
| SafeSetError( |
| "Failed to set remote answer due to invalid codec packetization " |
| "specifid in m-section with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| } |
| } |
| } |
| |
| if (!media_channel()->SetSendParameters(send_params)) { |
| SafeSetError( |
| "Failed to set remote video description send parameters for m-section " |
| "with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| last_send_params_ = send_params; |
| |
| if (needs_recv_params_update) { |
| if (!media_channel()->SetRecvParameters(recv_params)) { |
| SafeSetError("Failed to set recv parameters for m-section with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| last_recv_params_ = recv_params; |
| } |
| |
| if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) { |
| RTC_DLOG(LS_VERBOSE) << "SetRemoteContent_w: remote side will not send - " |
| "disable payload type demuxing for " |
| << ToString(); |
| ClearHandledPayloadTypes(); |
| } |
| |
| // 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(), type, error_desc)) { |
| SafeSetError( |
| "Failed to set remote video description streams for m-section with " |
| "mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| set_remote_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| 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 srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| UniqueRandomIdGenerator* ssrc_generator) |
| : BaseChannel(worker_thread, |
| network_thread, |
| signaling_thread, |
| std::move(media_channel), |
| content_name, |
| srtp_required, |
| crypto_options, |
| ssrc_generator) {} |
| |
| RtpDataChannel::~RtpDataChannel() { |
| TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel"); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| Deinit(); |
| } |
| |
| void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) { |
| BaseChannel::Init_w(rtp_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 MediaContentDescription* content, |
| std::string* error_desc) { |
| if (!content->as_rtp_data()) { |
| if (content->as_sctp()) { |
| SafeSetError("Data channel type mismatch. Expected RTP, got SCTP.", |
| error_desc); |
| } else { |
| SafeSetError("Data channel is not RTP or SCTP.", error_desc); |
| } |
| return false; |
| } |
| return true; |
| } |
| |
| bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "RtpDataChannel::SetLocalContent_w"); |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| RTC_LOG(LS_INFO) << "Setting local data description for " << ToString(); |
| |
| RTC_DCHECK(content); |
| if (!content) { |
| SafeSetError("Can't find data content in local description.", error_desc); |
| return false; |
| } |
| |
| if (!CheckDataChannelTypeFromContent(content, error_desc)) { |
| return false; |
| } |
| const RtpDataContentDescription* data = content->as_rtp_data(); |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(data->rtp_header_extensions()); |
| |
| DataRecvParameters recv_params = last_recv_params_; |
| RtpParametersFromMediaDescription( |
| data, rtp_header_extensions, |
| webrtc::RtpTransceiverDirectionHasRecv(data->direction()), &recv_params); |
| if (!media_channel()->SetRecvParameters(recv_params)) { |
| SafeSetError( |
| "Failed to set remote data description recv parameters for m-section " |
| "with mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| for (const DataCodec& codec : data->codecs()) { |
| MaybeAddHandledPayloadType(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(), type, error_desc)) { |
| SafeSetError( |
| "Failed to set local data description streams for m-section with " |
| "mid='" + |
| content_name() + "'.", |
| error_desc); |
| return false; |
| } |
| |
| set_local_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| return true; |
| } |
| |
| bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string* error_desc) { |
| TRACE_EVENT0("webrtc", "RtpDataChannel::SetRemoteContent_w"); |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| RTC_LOG(LS_INFO) << "Setting remote data description for " << ToString(); |
| |
| RTC_DCHECK(content); |
| if (!content) { |
| SafeSetError("Can't find data content in remote description.", error_desc); |
| return false; |
| } |
| |
| if (!CheckDataChannelTypeFromContent(content, error_desc)) { |
| return false; |
| } |
| |
| const RtpDataContentDescription* data = content->as_rtp_data(); |
| |
| // If the remote data doesn't have codecs, it must be empty, so ignore it. |
| if (!data->has_codecs()) { |
| return true; |
| } |
| |
| RtpHeaderExtensions rtp_header_extensions = |
| GetFilteredRtpHeaderExtensions(data->rtp_header_extensions()); |
| |
| RTC_LOG(LS_INFO) << "Setting remote data description for " << ToString(); |
| DataSendParameters send_params = last_send_params_; |
| RtpSendParametersFromMediaDescription<DataCodec>( |
| data, rtp_header_extensions, |
| webrtc::RtpTransceiverDirectionHasRecv(data->direction()), &send_params); |
| if (!media_channel()->SetSendParameters(send_params)) { |
| SafeSetError( |
| "Failed to set remote data description send parameters for m-section " |
| "with mid='" + |
| content_name() + "'.", |
| 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(), type, error_desc)) { |
| SafeSetError( |
| "Failed to set remote data description streams for m-section with " |
| "mid='" + |
| content_name() + "'.", |
| 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. |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| bool recv = IsReadyToReceiveMedia_w(); |
| if (!media_channel()->SetReceive(recv)) { |
| RTC_LOG(LS_ERROR) << "Failed to SetReceive on data channel: " << ToString(); |
| } |
| |
| // 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: " << ToString(); |
| } |
| |
| // Trigger SignalReadyToSendData asynchronously. |
| OnDataChannelReadyToSend(send); |
| |
| RTC_LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send |
| << " for " << ToString(); |
| } |
| |
| 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; |
| } |
| default: |
| BaseChannel::OnMessage(pmsg); |
| break; |
| } |
| } |
| |
| 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::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 |
