| /* |
| * 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. |
| */ |
| |
| #ifndef PC_CHANNEL_H_ |
| #define PC_CHANNEL_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/call/audio_sink.h" |
| #include "api/jsep.h" |
| #include "api/media_transport_config.h" |
| #include "api/rtp_receiver_interface.h" |
| #include "api/video/video_sink_interface.h" |
| #include "api/video/video_source_interface.h" |
| #include "call/rtp_packet_sink_interface.h" |
| #include "media/base/media_channel.h" |
| #include "media/base/media_engine.h" |
| #include "media/base/stream_params.h" |
| #include "p2p/base/dtls_transport_internal.h" |
| #include "p2p/base/packet_transport_internal.h" |
| #include "pc/channel_interface.h" |
| #include "pc/dtls_srtp_transport.h" |
| #include "pc/media_session.h" |
| #include "pc/rtp_transport.h" |
| #include "pc/srtp_filter.h" |
| #include "pc/srtp_transport.h" |
| #include "rtc_base/async_invoker.h" |
| #include "rtc_base/async_udp_socket.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/network.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| #include "rtc_base/unique_id_generator.h" |
| |
| namespace webrtc { |
| class AudioSinkInterface; |
| class MediaTransportInterface; |
| } // namespace webrtc |
| |
| namespace cricket { |
| |
| struct CryptoParams; |
| |
| // BaseChannel contains logic common to voice and video, including enable, |
| // marshaling calls to a worker and network threads, and connection and media |
| // monitors. |
| // |
| // BaseChannel assumes signaling and other threads are allowed to make |
| // synchronous calls to the worker thread, the worker thread makes synchronous |
| // calls only to the network thread, and the network thread can't be blocked by |
| // other threads. |
| // All methods with _n suffix must be called on network thread, |
| // methods with _w suffix on worker thread |
| // and methods with _s suffix on signaling thread. |
| // Network and worker threads may be the same thread. |
| // |
| // WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS! |
| // This is required to avoid a data race between the destructor modifying the |
| // vtable, and the media channel's thread using BaseChannel as the |
| // NetworkInterface. |
| |
| class BaseChannel : public ChannelInterface, |
| public rtc::MessageHandler, |
| public sigslot::has_slots<>, |
| public MediaChannel::NetworkInterface, |
| public webrtc::RtpPacketSinkInterface, |
| public webrtc::MediaTransportNetworkChangeCallback { |
| public: |
| // If |srtp_required| is true, the channel will not send or receive any |
| // RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP). |
| // The BaseChannel does not own the UniqueRandomIdGenerator so it is the |
| // responsibility of the user to ensure it outlives this object. |
| // TODO(zhihuang:) Create a BaseChannel::Config struct for the parameter lists |
| // which will make it easier to change the constructor. |
| 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, |
| rtc::UniqueRandomIdGenerator* ssrc_generator); |
| virtual ~BaseChannel(); |
| virtual void Init_w( |
| webrtc::RtpTransportInternal* rtp_transport, |
| const webrtc::MediaTransportConfig& media_transport_config); |
| |
| // Deinit may be called multiple times and is simply ignored if it's already |
| // done. |
| void Deinit(); |
| |
| rtc::Thread* worker_thread() const { return worker_thread_; } |
| rtc::Thread* network_thread() const { return network_thread_; } |
| const std::string& content_name() const override { return content_name_; } |
| // TODO(deadbeef): This is redundant; remove this. |
| const std::string& transport_name() const override { return transport_name_; } |
| bool enabled() const override { return enabled_; } |
| |
| // This function returns true if using SRTP (DTLS-based keying or SDES). |
| bool srtp_active() const { |
| return rtp_transport_ && rtp_transport_->IsSrtpActive(); |
| } |
| |
| bool writable() const { return writable_; } |
| |
| // Set an RTP level transport which could be an RtpTransport without |
| // encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP. |
| // This can be called from any thread and it hops to the network thread |
| // internally. It would replace the |SetTransports| and its variants. |
| bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) override; |
| |
| // Channel control |
| bool SetLocalContent(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) override; |
| bool SetRemoteContent(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) override; |
| |
| bool Enable(bool enable) override; |
| |
| const std::vector<StreamParams>& local_streams() const override { |
| return local_streams_; |
| } |
| const std::vector<StreamParams>& remote_streams() const override { |
| return remote_streams_; |
| } |
| |
| sigslot::signal2<BaseChannel*, bool> SignalDtlsSrtpSetupFailure; |
| void SignalDtlsSrtpSetupFailure_n(bool rtcp); |
| void SignalDtlsSrtpSetupFailure_s(bool rtcp); |
| |
| // Used for latency measurements. |
| sigslot::signal1<ChannelInterface*>& SignalFirstPacketReceived() override { |
| return SignalFirstPacketReceived_; |
| } |
| |
| // Forward SignalSentPacket to worker thread. |
| sigslot::signal1<const rtc::SentPacket&> SignalSentPacket; |
| |
| // Emitted whenever rtcp-mux is fully negotiated and the rtcp-transport can |
| // be destroyed. |
| // Fired on the network thread. |
| sigslot::signal1<const std::string&> SignalRtcpMuxFullyActive; |
| |
| rtc::PacketTransportInternal* rtp_packet_transport() { |
| if (rtp_transport_) { |
| return rtp_transport_->rtp_packet_transport(); |
| } |
| return nullptr; |
| } |
| |
| rtc::PacketTransportInternal* rtcp_packet_transport() { |
| if (rtp_transport_) { |
| return rtp_transport_->rtcp_packet_transport(); |
| } |
| return nullptr; |
| } |
| |
| // Returns media transport, can be null if media transport is not available. |
| webrtc::MediaTransportInterface* media_transport() { |
| return media_transport_config_.media_transport; |
| } |
| |
| // From RtpTransport - public for testing only |
| void OnTransportReadyToSend(bool ready); |
| |
| // Only public for unit tests. Otherwise, consider protected. |
| int SetOption(SocketType type, rtc::Socket::Option o, int val) override; |
| int SetOption_n(SocketType type, rtc::Socket::Option o, int val); |
| |
| // RtpPacketSinkInterface overrides. |
| void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override; |
| |
| // Used by the RTCStatsCollector tests to set the transport name without |
| // creating RtpTransports. |
| void set_transport_name_for_testing(const std::string& transport_name) { |
| transport_name_ = transport_name; |
| } |
| |
| MediaChannel* media_channel() const override { return media_channel_.get(); } |
| |
| protected: |
| bool was_ever_writable() const { return was_ever_writable_; } |
| void set_local_content_direction(webrtc::RtpTransceiverDirection direction) { |
| local_content_direction_ = direction; |
| } |
| void set_remote_content_direction(webrtc::RtpTransceiverDirection direction) { |
| remote_content_direction_ = direction; |
| } |
| // These methods verify that: |
| // * The required content description directions have been set. |
| // * The channel is enabled. |
| // * And for sending: |
| // - The SRTP filter is active if it's needed. |
| // - The transport has been writable before, meaning it should be at least |
| // possible to succeed in sending a packet. |
| // |
| // When any of these properties change, UpdateMediaSendRecvState_w should be |
| // called. |
| bool IsReadyToReceiveMedia_w() const; |
| bool IsReadyToSendMedia_w() const; |
| rtc::Thread* signaling_thread() { return signaling_thread_; } |
| |
| void FlushRtcpMessages_n(); |
| |
| // NetworkInterface implementation, called by MediaEngine |
| bool SendPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) override; |
| bool SendRtcp(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) override; |
| |
| // From RtpTransportInternal |
| void OnWritableState(bool writable); |
| |
| void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route); |
| |
| bool PacketIsRtcp(const rtc::PacketTransportInternal* transport, |
| const char* data, |
| size_t len); |
| bool SendPacket(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options); |
| |
| void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet, |
| int64_t packet_time_us); |
| |
| void OnPacketReceived(bool rtcp, |
| const rtc::CopyOnWriteBuffer& packet, |
| int64_t packet_time_us); |
| void ProcessPacket(bool rtcp, |
| const rtc::CopyOnWriteBuffer& packet, |
| int64_t packet_time_us); |
| |
| void EnableMedia_w(); |
| void DisableMedia_w(); |
| |
| // Performs actions if the RTP/RTCP writable state changed. This should |
| // be called whenever a channel's writable state changes or when RTCP muxing |
| // becomes active/inactive. |
| void UpdateWritableState_n(); |
| void ChannelWritable_n(); |
| void ChannelNotWritable_n(); |
| |
| bool AddRecvStream_w(const StreamParams& sp); |
| bool RemoveRecvStream_w(uint32_t ssrc); |
| bool AddSendStream_w(const StreamParams& sp); |
| bool RemoveSendStream_w(uint32_t ssrc); |
| |
| // Should be called whenever the conditions for |
| // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied). |
| // Updates the send/recv state of the media channel. |
| void UpdateMediaSendRecvState(); |
| virtual void UpdateMediaSendRecvState_w() = 0; |
| |
| bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams, |
| webrtc::SdpType type, |
| std::string* error_desc); |
| bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams, |
| webrtc::SdpType type, |
| std::string* error_desc); |
| virtual bool SetLocalContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) = 0; |
| virtual bool SetRemoteContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) = 0; |
| // Return a list of RTP header extensions with the non-encrypted extensions |
| // removed depending on the current crypto_options_ and only if both the |
| // non-encrypted and encrypted extension is present for the same URI. |
| RtpHeaderExtensions GetFilteredRtpHeaderExtensions( |
| const RtpHeaderExtensions& extensions); |
| |
| // From MessageHandler |
| void OnMessage(rtc::Message* pmsg) override; |
| |
| // Helper function template for invoking methods on the worker thread. |
| template <class T, class FunctorT> |
| T InvokeOnWorker(const rtc::Location& posted_from, const FunctorT& functor) { |
| return worker_thread_->Invoke<T>(posted_from, functor); |
| } |
| |
| void AddHandledPayloadType(int payload_type); |
| |
| void UpdateRtpHeaderExtensionMap( |
| const RtpHeaderExtensions& header_extensions); |
| |
| bool RegisterRtpDemuxerSink(); |
| |
| bool has_received_packet_ = false; |
| |
| private: |
| bool ConnectToRtpTransport(); |
| void DisconnectFromRtpTransport(); |
| void SignalSentPacket_n(const rtc::SentPacket& sent_packet); |
| void SignalSentPacket_w(const rtc::SentPacket& sent_packet); |
| bool IsReadyToSendMedia_n() const; |
| |
| // MediaTransportNetworkChangeCallback override. |
| void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route) override; |
| |
| rtc::Thread* const worker_thread_; |
| rtc::Thread* const network_thread_; |
| rtc::Thread* const signaling_thread_; |
| rtc::AsyncInvoker invoker_; |
| sigslot::signal1<ChannelInterface*> SignalFirstPacketReceived_; |
| |
| const std::string content_name_; |
| |
| // Won't be set when using raw packet transports. SDP-specific thing. |
| std::string transport_name_; |
| |
| webrtc::RtpTransportInternal* rtp_transport_ = nullptr; |
| |
| // Optional media transport configuration (experimental). |
| webrtc::MediaTransportConfig media_transport_config_; |
| |
| std::vector<std::pair<rtc::Socket::Option, int> > socket_options_; |
| std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_; |
| bool writable_ = false; |
| bool was_ever_writable_ = false; |
| const bool srtp_required_ = true; |
| webrtc::CryptoOptions crypto_options_; |
| |
| // MediaChannel related members that should be accessed from the worker |
| // thread. |
| std::unique_ptr<MediaChannel> media_channel_; |
| // Currently the |enabled_| flag is accessed from the signaling thread as |
| // well, but it can be changed only when signaling thread does a synchronous |
| // call to the worker thread, so it should be safe. |
| bool enabled_ = false; |
| std::vector<StreamParams> local_streams_; |
| std::vector<StreamParams> remote_streams_; |
| webrtc::RtpTransceiverDirection local_content_direction_ = |
| webrtc::RtpTransceiverDirection::kInactive; |
| webrtc::RtpTransceiverDirection remote_content_direction_ = |
| webrtc::RtpTransceiverDirection::kInactive; |
| |
| webrtc::RtpDemuxerCriteria demuxer_criteria_; |
| // This generator is used to generate SSRCs for local streams. |
| // This is needed in cases where SSRCs are not negotiated or set explicitly |
| // like in Simulcast. |
| // This object is not owned by the channel so it must outlive it. |
| rtc::UniqueRandomIdGenerator* const ssrc_generator_; |
| }; |
| |
| // VoiceChannel is a specialization that adds support for early media, DTMF, |
| // and input/output level monitoring. |
| class VoiceChannel : public BaseChannel, |
| public webrtc::AudioPacketReceivedObserver { |
| public: |
| VoiceChannel(rtc::Thread* worker_thread, |
| rtc::Thread* network_thread, |
| rtc::Thread* signaling_thread, |
| std::unique_ptr<VoiceMediaChannel> channel, |
| const std::string& content_name, |
| bool srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| rtc::UniqueRandomIdGenerator* ssrc_generator); |
| ~VoiceChannel(); |
| |
| // downcasts a MediaChannel |
| VoiceMediaChannel* media_channel() const override { |
| return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel()); |
| } |
| |
| cricket::MediaType media_type() const override { |
| return cricket::MEDIA_TYPE_AUDIO; |
| } |
| void Init_w( |
| webrtc::RtpTransportInternal* rtp_transport, |
| const webrtc::MediaTransportConfig& media_transport_config) override; |
| |
| private: |
| // overrides from BaseChannel |
| void UpdateMediaSendRecvState_w() override; |
| bool SetLocalContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) override; |
| bool SetRemoteContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) override; |
| |
| void OnFirstAudioPacketReceived(int64_t channel_id) override; |
| |
| // Last AudioSendParameters sent down to the media_channel() via |
| // SetSendParameters. |
| AudioSendParameters last_send_params_; |
| // Last AudioRecvParameters sent down to the media_channel() via |
| // SetRecvParameters. |
| AudioRecvParameters last_recv_params_; |
| }; |
| |
| // VideoChannel is a specialization for video. |
| class VideoChannel : public BaseChannel { |
| public: |
| 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, |
| rtc::UniqueRandomIdGenerator* ssrc_generator); |
| ~VideoChannel(); |
| |
| // downcasts a MediaChannel |
| VideoMediaChannel* media_channel() const override { |
| return static_cast<VideoMediaChannel*>(BaseChannel::media_channel()); |
| } |
| |
| void FillBitrateInfo(BandwidthEstimationInfo* bwe_info); |
| |
| cricket::MediaType media_type() const override { |
| return cricket::MEDIA_TYPE_VIDEO; |
| } |
| |
| private: |
| // overrides from BaseChannel |
| void UpdateMediaSendRecvState_w() override; |
| bool SetLocalContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) override; |
| bool SetRemoteContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) override; |
| |
| // Last VideoSendParameters sent down to the media_channel() via |
| // SetSendParameters. |
| VideoSendParameters last_send_params_; |
| // Last VideoRecvParameters sent down to the media_channel() via |
| // SetRecvParameters. |
| VideoRecvParameters last_recv_params_; |
| }; |
| |
| // RtpDataChannel is a specialization for data. |
| class RtpDataChannel : public BaseChannel { |
| public: |
| RtpDataChannel(rtc::Thread* worker_thread, |
| rtc::Thread* network_thread, |
| rtc::Thread* signaling_thread, |
| std::unique_ptr<DataMediaChannel> channel, |
| const std::string& content_name, |
| bool srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| rtc::UniqueRandomIdGenerator* ssrc_generator); |
| ~RtpDataChannel(); |
| // TODO(zhihuang): Remove this once the RtpTransport can be shared between |
| // BaseChannels. |
| void Init_w(DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport); |
| void Init_w( |
| webrtc::RtpTransportInternal* rtp_transport, |
| const webrtc::MediaTransportConfig& media_transport_config) override; |
| |
| virtual bool SendData(const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| SendDataResult* result); |
| |
| // Should be called on the signaling thread only. |
| bool ready_to_send_data() const { return ready_to_send_data_; } |
| |
| sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&> |
| SignalDataReceived; |
| // Signal for notifying when the channel becomes ready to send data. |
| // That occurs when the channel is enabled, the transport is writable, |
| // both local and remote descriptions are set, and the channel is unblocked. |
| sigslot::signal1<bool> SignalReadyToSendData; |
| cricket::MediaType media_type() const override { |
| return cricket::MEDIA_TYPE_DATA; |
| } |
| |
| protected: |
| // downcasts a MediaChannel. |
| DataMediaChannel* media_channel() const override { |
| return static_cast<DataMediaChannel*>(BaseChannel::media_channel()); |
| } |
| |
| private: |
| struct SendDataMessageData : public rtc::MessageData { |
| SendDataMessageData(const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer* payload, |
| SendDataResult* result) |
| : params(params), payload(payload), result(result), succeeded(false) {} |
| |
| const SendDataParams& params; |
| const rtc::CopyOnWriteBuffer* payload; |
| SendDataResult* result; |
| bool succeeded; |
| }; |
| |
| struct DataReceivedMessageData : public rtc::MessageData { |
| // We copy the data because the data will become invalid after we |
| // handle DataMediaChannel::SignalDataReceived but before we fire |
| // SignalDataReceived. |
| DataReceivedMessageData(const ReceiveDataParams& params, |
| const char* data, |
| size_t len) |
| : params(params), payload(data, len) {} |
| const ReceiveDataParams params; |
| const rtc::CopyOnWriteBuffer payload; |
| }; |
| |
| typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData; |
| |
| // overrides from BaseChannel |
| // Checks that data channel type is RTP. |
| bool CheckDataChannelTypeFromContent(const RtpDataContentDescription* content, |
| std::string* error_desc); |
| bool SetLocalContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) override; |
| bool SetRemoteContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) override; |
| void UpdateMediaSendRecvState_w() override; |
| |
| void OnMessage(rtc::Message* pmsg) override; |
| void OnDataReceived(const ReceiveDataParams& params, |
| const char* data, |
| size_t len); |
| void OnDataChannelReadyToSend(bool writable); |
| |
| bool ready_to_send_data_ = false; |
| |
| // Last DataSendParameters sent down to the media_channel() via |
| // SetSendParameters. |
| DataSendParameters last_send_params_; |
| // Last DataRecvParameters sent down to the media_channel() via |
| // SetRecvParameters. |
| DataRecvParameters last_recv_params_; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // PC_CHANNEL_H_ |