| /* |
| * 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 <stddef.h> |
| #include <stdint.h> |
| |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/call/audio_sink.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/function_view.h" |
| #include "api/jsep.h" |
| #include "api/media_types.h" |
| #include "api/rtp_receiver_interface.h" |
| #include "api/rtp_transceiver_direction.h" |
| #include "api/scoped_refptr.h" |
| #include "api/sequence_checker.h" |
| #include "api/video/video_sink_interface.h" |
| #include "api/video/video_source_interface.h" |
| #include "call/rtp_demuxer.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 "modules/rtp_rtcp/source/rtp_packet_received.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/rtp_transport_internal.h" |
| #include "pc/session_description.h" |
| #include "pc/srtp_filter.h" |
| #include "pc/srtp_transport.h" |
| #include "rtc_base/async_packet_socket.h" |
| #include "rtc_base/async_udp_socket.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/message_handler.h" |
| #include "rtc_base/network.h" |
| #include "rtc_base/network/sent_packet.h" |
| #include "rtc_base/network_route.h" |
| #include "rtc_base/socket.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "rtc_base/task_utils/pending_task_safety_flag.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "rtc_base/thread_message.h" |
| #include "rtc_base/unique_id_generator.h" |
| |
| namespace webrtc { |
| class AudioSinkInterface; |
| } // 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::MessageHandlerAutoCleanup, |
| public sigslot::has_slots<>, |
| public MediaChannel::NetworkInterface, |
| public webrtc::RtpPacketSinkInterface { |
| 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); |
| |
| // 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 { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| return rtp_transport_ && rtp_transport_->IsSrtpActive(); |
| } |
| |
| // 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; |
| |
| webrtc::RtpTransportInternal* rtp_transport() const { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| return rtp_transport_; |
| } |
| |
| // 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; |
| // Controls whether this channel will receive packets on the basis of |
| // matching payload type alone. This is needed for legacy endpoints that |
| // don't signal SSRCs or use MID/RID, but doesn't make sense if there is |
| // more than channel of specific media type, As that creates an ambiguity. |
| // |
| // This method will also remove any existing streams that were bound to this |
| // channel on the basis of payload type, since one of these streams might |
| // actually belong to a new channel. See: crbug.com/webrtc/11477 |
| bool SetPayloadTypeDemuxingEnabled(bool enabled) 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_; |
| } |
| |
| // Used for latency measurements. |
| sigslot::signal1<ChannelInterface*>& SignalFirstPacketReceived() override; |
| |
| // Forward SignalSentPacket to worker thread. |
| sigslot::signal1<const rtc::SentPacket&>& SignalSentPacket(); |
| |
| // 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) |
| RTC_RUN_ON(network_thread()); |
| |
| // RtpPacketSinkInterface overrides. |
| void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override; |
| |
| MediaChannel* media_channel() const override { |
| return media_channel_.get(); |
| } |
| |
| protected: |
| bool was_ever_writable() const { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| return was_ever_writable_; |
| } |
| void set_local_content_direction(webrtc::RtpTransceiverDirection direction) { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| local_content_direction_ = direction; |
| } |
| void set_remote_content_direction(webrtc::RtpTransceiverDirection direction) { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| 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 RTC_RUN_ON(worker_thread()); |
| bool IsReadyToSendMedia_w() const RTC_RUN_ON(worker_thread()); |
| rtc::Thread* signaling_thread() const { return signaling_thread_; } |
| |
| void FlushRtcpMessages_n() RTC_RUN_ON(network_thread()); |
| |
| // 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 SendPacket(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options); |
| |
| void EnableMedia_w() RTC_RUN_ON(worker_thread()); |
| void DisableMedia_w() RTC_RUN_ON(worker_thread()); |
| |
| // 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() RTC_RUN_ON(network_thread()); |
| void ChannelWritable_n() RTC_RUN_ON(network_thread()); |
| void ChannelNotWritable_n() RTC_RUN_ON(network_thread()); |
| |
| bool AddRecvStream_w(const StreamParams& sp) RTC_RUN_ON(worker_thread()); |
| bool RemoveRecvStream_w(uint32_t ssrc) RTC_RUN_ON(worker_thread()); |
| void ResetUnsignaledRecvStream_w() RTC_RUN_ON(worker_thread()); |
| bool SetPayloadTypeDemuxingEnabled_w(bool enabled) |
| RTC_RUN_ON(worker_thread()); |
| bool AddSendStream_w(const StreamParams& sp) RTC_RUN_ON(worker_thread()); |
| bool RemoveSendStream_w(uint32_t ssrc) RTC_RUN_ON(worker_thread()); |
| |
| // Should be called whenever the conditions for |
| // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied). |
| // Updates the send/recv state of the media channel. |
| virtual void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) = 0; |
| |
| bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams, |
| webrtc::SdpType type, |
| std::string* error_desc) |
| RTC_RUN_ON(worker_thread()); |
| bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams, |
| webrtc::SdpType type, |
| std::string* error_desc) |
| RTC_RUN_ON(worker_thread()); |
| virtual bool SetLocalContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) |
| RTC_RUN_ON(worker_thread()) = 0; |
| virtual bool SetRemoteContent_w(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) |
| RTC_RUN_ON(worker_thread()) = 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> |
| T InvokeOnWorker(const rtc::Location& posted_from, |
| rtc::FunctionView<T()> functor) { |
| return worker_thread_->Invoke<T>(posted_from, functor); |
| } |
| |
| // Add |payload_type| to |demuxer_criteria_| if payload type demuxing is |
| // enabled. |
| void MaybeAddHandledPayloadType(int payload_type) RTC_RUN_ON(worker_thread()); |
| |
| void ClearHandledPayloadTypes() RTC_RUN_ON(worker_thread()); |
| |
| void UpdateRtpHeaderExtensionMap( |
| const RtpHeaderExtensions& header_extensions); |
| |
| bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread()); |
| |
| // Return description of media channel to facilitate logging |
| std::string ToString() const; |
| |
| void SetNegotiatedHeaderExtensions_w(const RtpHeaderExtensions& extensions) |
| RTC_RUN_ON(worker_thread()); |
| |
| // ChannelInterface overrides |
| RtpHeaderExtensions GetNegotiatedRtpHeaderExtensions() const override; |
| |
| private: |
| bool ConnectToRtpTransport() RTC_RUN_ON(network_thread()); |
| void DisconnectFromRtpTransport() RTC_RUN_ON(network_thread()); |
| void SignalSentPacket_n(const rtc::SentPacket& sent_packet) |
| RTC_RUN_ON(network_thread()); |
| |
| rtc::Thread* const worker_thread_; |
| rtc::Thread* const network_thread_; |
| rtc::Thread* const signaling_thread_; |
| rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> alive_; |
| sigslot::signal1<ChannelInterface*> SignalFirstPacketReceived_ |
| RTC_GUARDED_BY(signaling_thread_); |
| sigslot::signal1<const rtc::SentPacket&> SignalSentPacket_ |
| RTC_GUARDED_BY(worker_thread_); |
| |
| const std::string content_name_; |
| |
| bool has_received_packet_ = false; |
| |
| // Won't be set when using raw packet transports. SDP-specific thing. |
| // TODO(bugs.webrtc.org/12230): Written on network thread, read on |
| // worker thread (at least). |
| std::string transport_name_; |
| |
| webrtc::RtpTransportInternal* rtp_transport_ |
| RTC_GUARDED_BY(network_thread()) = nullptr; |
| |
| std::vector<std::pair<rtc::Socket::Option, int> > socket_options_ |
| RTC_GUARDED_BY(network_thread()); |
| std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_ |
| RTC_GUARDED_BY(network_thread()); |
| bool writable_ RTC_GUARDED_BY(network_thread()) = false; |
| bool was_ever_writable_n_ RTC_GUARDED_BY(network_thread()) = false; |
| bool was_ever_writable_ RTC_GUARDED_BY(worker_thread()) = false; |
| const bool srtp_required_ = true; |
| const webrtc::CryptoOptions crypto_options_; |
| |
| // MediaChannel related members that should be accessed from the worker |
| // thread. |
| const 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; |
| bool payload_type_demuxing_enabled_ RTC_GUARDED_BY(worker_thread()) = true; |
| std::vector<StreamParams> local_streams_ RTC_GUARDED_BY(worker_thread()); |
| std::vector<StreamParams> remote_streams_ RTC_GUARDED_BY(worker_thread()); |
| // TODO(bugs.webrtc.org/12230): local_content_direction and |
| // remote_content_direction are set on the worker thread, but accessed on the |
| // network thread. |
| webrtc::RtpTransceiverDirection local_content_direction_ = |
| webrtc::RtpTransceiverDirection::kInactive; |
| webrtc::RtpTransceiverDirection remote_content_direction_ = |
| webrtc::RtpTransceiverDirection::kInactive; |
| |
| // Cached list of payload types, used if payload type demuxing is re-enabled. |
| std::set<uint8_t> payload_types_ RTC_GUARDED_BY(worker_thread()); |
| // TODO(bugs.webrtc.org/12239): Modified on worker thread, accessed |
| // on network thread in RegisterRtpDemuxerSink_n (called from Init_w) |
| webrtc::RtpDemuxerCriteria demuxer_criteria_; |
| // Accessed on the worker thread, modified on the network thread from |
| // RegisterRtpDemuxerSink_w's Invoke. |
| webrtc::RtpDemuxerCriteria previous_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_; |
| |
| // |negotiated_header_extensions_| is read on the signaling thread, but |
| // written on the worker thread while being sync-invoked from the signal |
| // thread in SdpOfferAnswerHandler::PushdownMediaDescription(). Hence the lock |
| // isn't strictly needed, but it's anyway placed here for future safeness. |
| mutable webrtc::Mutex negotiated_header_extensions_lock_; |
| RtpHeaderExtensions negotiated_header_extensions_ |
| RTC_GUARDED_BY(negotiated_header_extensions_lock_); |
| }; |
| |
| // VoiceChannel is a specialization that adds support for early media, DTMF, |
| // and input/output level monitoring. |
| class VoiceChannel : public BaseChannel { |
| 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; |
| } |
| |
| 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 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) 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 MediaContentDescription* 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_ |