/* * libjingle * Copyright 2004 Google Inc. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions are met: * * 1. Redistributions of source code must retain the above copyright notice, * this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright notice, * this list of conditions and the following disclaimer in the documentation * and/or other materials provided with the distribution. * 3. The name of the author may not be used to endorse or promote products * derived from this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #ifndef TALK_SESSION_MEDIA_CHANNEL_H_ #define TALK_SESSION_MEDIA_CHANNEL_H_ #include #include #include "talk/base/asyncudpsocket.h" #include "talk/base/criticalsection.h" #include "talk/base/network.h" #include "talk/base/sigslot.h" #include "talk/base/window.h" #include "talk/media/base/mediachannel.h" #include "talk/media/base/mediaengine.h" #include "talk/media/base/screencastid.h" #include "talk/media/base/streamparams.h" #include "talk/media/base/videocapturer.h" #include "talk/p2p/base/session.h" #include "talk/p2p/client/socketmonitor.h" #include "talk/session/media/audiomonitor.h" #include "talk/session/media/bundlefilter.h" #include "talk/session/media/mediamonitor.h" #include "talk/session/media/mediasession.h" #include "talk/session/media/rtcpmuxfilter.h" #include "talk/session/media/srtpfilter.h" namespace cricket { struct CryptoParams; class MediaContentDescription; struct TypingMonitorOptions; class TypingMonitor; struct ViewRequest; enum SinkType { SINK_PRE_CRYPTO, // Sink packets before encryption or after decryption. SINK_POST_CRYPTO // Sink packets after encryption or before decryption. }; // BaseChannel contains logic common to voice and video, including // enable/mute, marshaling calls to a worker thread, and // connection and media monitors. // // 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 talk_base::MessageHandler, public sigslot::has_slots<>, public MediaChannel::NetworkInterface { public: BaseChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine, MediaChannel* channel, BaseSession* session, const std::string& content_name, bool rtcp); virtual ~BaseChannel(); bool Init(TransportChannel* transport_channel, TransportChannel* rtcp_transport_channel); // Deinit may be called multiple times and is simply ignored if it's alreay // done. void Deinit(); talk_base::Thread* worker_thread() const { return worker_thread_; } BaseSession* session() const { return session_; } const std::string& content_name() { return content_name_; } TransportChannel* transport_channel() const { return transport_channel_; } TransportChannel* rtcp_transport_channel() const { return rtcp_transport_channel_; } bool enabled() const { return enabled_; } // This function returns true if we are using SRTP. bool secure() const { return srtp_filter_.IsActive(); } // The following function returns true if we are using // DTLS-based keying. If you turned off SRTP later, however // you could have secure() == false and dtls_secure() == true. bool secure_dtls() const { return dtls_keyed_; } // This function returns true if we require secure channel for call setup. bool secure_required() const { return secure_required_; } bool writable() const { return writable_; } bool IsStreamMuted(uint32 ssrc); // Channel control bool SetLocalContent(const MediaContentDescription* content, ContentAction action, std::string* error_desc); bool SetRemoteContent(const MediaContentDescription* content, ContentAction action, std::string* error_desc); bool Enable(bool enable); // Mute sending media on the stream with SSRC |ssrc| // If there is only one sending stream SSRC 0 can be used. bool MuteStream(uint32 ssrc, bool mute); // Multiplexing bool AddRecvStream(const StreamParams& sp); bool RemoveRecvStream(uint32 ssrc); bool AddSendStream(const StreamParams& sp); bool RemoveSendStream(uint32 ssrc); // Monitoring void StartConnectionMonitor(int cms); void StopConnectionMonitor(); void set_srtp_signal_silent_time(uint32 silent_time) { srtp_filter_.set_signal_silent_time(silent_time); } void set_content_name(const std::string& content_name) { ASSERT(signaling_thread()->IsCurrent()); ASSERT(!writable_); if (session_->state() != BaseSession::STATE_INIT) { LOG(LS_ERROR) << "Content name for a channel can be changed only " << "when BaseSession is in STATE_INIT state."; return; } content_name_ = content_name; } template void RegisterSendSink(T* sink, void (T::*OnPacket)(const void*, size_t, bool), SinkType type) { talk_base::CritScope cs(&signal_send_packet_cs_); if (SINK_POST_CRYPTO == type) { SignalSendPacketPostCrypto.disconnect(sink); SignalSendPacketPostCrypto.connect(sink, OnPacket); } else { SignalSendPacketPreCrypto.disconnect(sink); SignalSendPacketPreCrypto.connect(sink, OnPacket); } } void UnregisterSendSink(sigslot::has_slots<>* sink, SinkType type) { talk_base::CritScope cs(&signal_send_packet_cs_); if (SINK_POST_CRYPTO == type) { SignalSendPacketPostCrypto.disconnect(sink); } else { SignalSendPacketPreCrypto.disconnect(sink); } } bool HasSendSinks(SinkType type) { talk_base::CritScope cs(&signal_send_packet_cs_); if (SINK_POST_CRYPTO == type) { return !SignalSendPacketPostCrypto.is_empty(); } else { return !SignalSendPacketPreCrypto.is_empty(); } } template void RegisterRecvSink(T* sink, void (T::*OnPacket)(const void*, size_t, bool), SinkType type) { talk_base::CritScope cs(&signal_recv_packet_cs_); if (SINK_POST_CRYPTO == type) { SignalRecvPacketPostCrypto.disconnect(sink); SignalRecvPacketPostCrypto.connect(sink, OnPacket); } else { SignalRecvPacketPreCrypto.disconnect(sink); SignalRecvPacketPreCrypto.connect(sink, OnPacket); } } void UnregisterRecvSink(sigslot::has_slots<>* sink, SinkType type) { talk_base::CritScope cs(&signal_recv_packet_cs_); if (SINK_POST_CRYPTO == type) { SignalRecvPacketPostCrypto.disconnect(sink); } else { SignalRecvPacketPreCrypto.disconnect(sink); } } bool HasRecvSinks(SinkType type) { talk_base::CritScope cs(&signal_recv_packet_cs_); if (SINK_POST_CRYPTO == type) { return !SignalRecvPacketPostCrypto.is_empty(); } else { return !SignalRecvPacketPreCrypto.is_empty(); } } BundleFilter* bundle_filter() { return &bundle_filter_; } const std::vector& local_streams() const { return local_streams_; } const std::vector& remote_streams() const { return remote_streams_; } // Used for latency measurements. sigslot::signal1 SignalFirstPacketReceived; // Used to alert UI when the muted status changes, perhaps autonomously. sigslot::repeater2 SignalAutoMuted; // Made public for easier testing. void SetReadyToSend(TransportChannel* channel, bool ready); protected: MediaEngineInterface* media_engine() const { return media_engine_; } virtual MediaChannel* media_channel() const { return media_channel_; } void set_rtcp_transport_channel(TransportChannel* transport); bool was_ever_writable() const { return was_ever_writable_; } void set_local_content_direction(MediaContentDirection direction) { local_content_direction_ = direction; } void set_remote_content_direction(MediaContentDirection direction) { remote_content_direction_ = direction; } bool IsReadyToReceive() const; bool IsReadyToSend() const; talk_base::Thread* signaling_thread() { return session_->signaling_thread(); } SrtpFilter* srtp_filter() { return &srtp_filter_; } bool rtcp() const { return rtcp_; } void FlushRtcpMessages(); // NetworkInterface implementation, called by MediaEngine virtual bool SendPacket(talk_base::Buffer* packet, talk_base::DiffServCodePoint dscp); virtual bool SendRtcp(talk_base::Buffer* packet, talk_base::DiffServCodePoint dscp); virtual int SetOption(SocketType type, talk_base::Socket::Option o, int val); // From TransportChannel void OnWritableState(TransportChannel* channel); virtual void OnChannelRead(TransportChannel* channel, const char* data, size_t len, const talk_base::PacketTime& packet_time, int flags); void OnReadyToSend(TransportChannel* channel); bool PacketIsRtcp(const TransportChannel* channel, const char* data, size_t len); bool SendPacket(bool rtcp, talk_base::Buffer* packet, talk_base::DiffServCodePoint dscp); virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet); void HandlePacket(bool rtcp, talk_base::Buffer* packet, const talk_base::PacketTime& packet_time); // Apply the new local/remote session description. void OnNewLocalDescription(BaseSession* session, ContentAction action); void OnNewRemoteDescription(BaseSession* session, ContentAction action); void EnableMedia_w(); void DisableMedia_w(); virtual bool MuteStream_w(uint32 ssrc, bool mute); bool IsStreamMuted_w(uint32 ssrc); void ChannelWritable_w(); void ChannelNotWritable_w(); bool AddRecvStream_w(const StreamParams& sp); bool RemoveRecvStream_w(uint32 ssrc); bool AddSendStream_w(const StreamParams& sp); bool RemoveSendStream_w(uint32 ssrc); virtual bool ShouldSetupDtlsSrtp() const; // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters. // |rtcp_channel| indicates whether to set up the RTP or RTCP filter. bool SetupDtlsSrtp(bool rtcp_channel); // Set the DTLS-SRTP cipher policy on this channel as appropriate. bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp); virtual void ChangeState() = 0; // Gets the content info appropriate to the channel (audio or video). virtual const ContentInfo* GetFirstContent( const SessionDescription* sdesc) = 0; bool UpdateLocalStreams_w(const std::vector& streams, ContentAction action, std::string* error_desc); bool UpdateRemoteStreams_w(const std::vector& streams, ContentAction action, std::string* error_desc); bool SetBaseLocalContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc); virtual bool SetLocalContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc) = 0; bool SetBaseRemoteContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc); virtual bool SetRemoteContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc) = 0; // Helper method to get RTP Absoulute SendTime extension header id if // present in remote supported extensions list. void MaybeCacheRtpAbsSendTimeHeaderExtension( const std::vector& extensions); bool CheckSrtpConfig(const std::vector& cryptos, bool* dtls, std::string* error_desc); bool SetSrtp_w(const std::vector& params, ContentAction action, ContentSource src, std::string* error_desc); bool SetRtcpMux_w(bool enable, ContentAction action, ContentSource src, std::string* error_desc); // From MessageHandler virtual void OnMessage(talk_base::Message* pmsg); // Handled in derived classes // Get the SRTP ciphers to use for RTP media virtual void GetSrtpCiphers(std::vector* ciphers) const = 0; virtual void OnConnectionMonitorUpdate(SocketMonitor* monitor, const std::vector& infos) = 0; // Helper function for invoking bool-returning methods on the worker thread. template bool InvokeOnWorker(const FunctorT& functor) { return worker_thread_->Invoke(functor); } private: sigslot::signal3 SignalSendPacketPreCrypto; sigslot::signal3 SignalSendPacketPostCrypto; sigslot::signal3 SignalRecvPacketPreCrypto; sigslot::signal3 SignalRecvPacketPostCrypto; talk_base::CriticalSection signal_send_packet_cs_; talk_base::CriticalSection signal_recv_packet_cs_; talk_base::Thread* worker_thread_; MediaEngineInterface* media_engine_; BaseSession* session_; MediaChannel* media_channel_; std::vector local_streams_; std::vector remote_streams_; std::string content_name_; bool rtcp_; TransportChannel* transport_channel_; TransportChannel* rtcp_transport_channel_; SrtpFilter srtp_filter_; RtcpMuxFilter rtcp_mux_filter_; BundleFilter bundle_filter_; talk_base::scoped_ptr socket_monitor_; bool enabled_; bool writable_; bool rtp_ready_to_send_; bool rtcp_ready_to_send_; bool was_ever_writable_; MediaContentDirection local_content_direction_; MediaContentDirection remote_content_direction_; std::set muted_streams_; bool has_received_packet_; bool dtls_keyed_; bool secure_required_; int rtp_abs_sendtime_extn_id_; }; // VoiceChannel is a specialization that adds support for early media, DTMF, // and input/output level monitoring. class VoiceChannel : public BaseChannel { public: VoiceChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine, VoiceMediaChannel* channel, BaseSession* session, const std::string& content_name, bool rtcp); ~VoiceChannel(); bool Init(); bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer); bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer); // downcasts a MediaChannel virtual VoiceMediaChannel* media_channel() const { return static_cast(BaseChannel::media_channel()); } bool SetRingbackTone(const void* buf, int len); void SetEarlyMedia(bool enable); // This signal is emitted when we have gone a period of time without // receiving early media. When received, a UI should start playing its // own ringing sound sigslot::signal1 SignalEarlyMediaTimeout; bool PlayRingbackTone(uint32 ssrc, bool play, bool loop); // TODO(ronghuawu): Replace PressDTMF with InsertDtmf. bool PressDTMF(int digit, bool playout); // Returns if the telephone-event has been negotiated. bool CanInsertDtmf(); // Send and/or play a DTMF |event| according to the |flags|. // The DTMF out-of-band signal will be used on sending. // The |ssrc| should be either 0 or a valid send stream ssrc. // The valid value for the |event| are 0 which corresponding to DTMF // event 0-9, *, #, A-D. bool InsertDtmf(uint32 ssrc, int event_code, int duration, int flags); bool SetOutputScaling(uint32 ssrc, double left, double right); // Get statistics about the current media session. bool GetStats(VoiceMediaInfo* stats); // Monitoring functions sigslot::signal2&> SignalConnectionMonitor; void StartMediaMonitor(int cms); void StopMediaMonitor(); sigslot::signal2 SignalMediaMonitor; void StartAudioMonitor(int cms); void StopAudioMonitor(); bool IsAudioMonitorRunning() const; sigslot::signal2 SignalAudioMonitor; void StartTypingMonitor(const TypingMonitorOptions& settings); void StopTypingMonitor(); bool IsTypingMonitorRunning() const; // Overrides BaseChannel::MuteStream_w. virtual bool MuteStream_w(uint32 ssrc, bool mute); int GetInputLevel_w(); int GetOutputLevel_w(); void GetActiveStreams_w(AudioInfo::StreamList* actives); // Signal errors from VoiceMediaChannel. Arguments are: // ssrc(uint32), and error(VoiceMediaChannel::Error). sigslot::signal3 SignalMediaError; // Configuration and setting. bool SetChannelOptions(const AudioOptions& options); private: // overrides from BaseChannel virtual void OnChannelRead(TransportChannel* channel, const char* data, size_t len, const talk_base::PacketTime& packet_time, int flags); virtual void ChangeState(); virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc); virtual bool SetLocalContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc); virtual bool SetRemoteContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc); bool SetRingbackTone_w(const void* buf, int len); bool PlayRingbackTone_w(uint32 ssrc, bool play, bool loop); void HandleEarlyMediaTimeout(); bool InsertDtmf_w(uint32 ssrc, int event, int duration, int flags); bool SetOutputScaling_w(uint32 ssrc, double left, double right); bool GetStats_w(VoiceMediaInfo* stats); virtual void OnMessage(talk_base::Message* pmsg); virtual void GetSrtpCiphers(std::vector* ciphers) const; virtual void OnConnectionMonitorUpdate( SocketMonitor* monitor, const std::vector& infos); virtual void OnMediaMonitorUpdate( VoiceMediaChannel* media_channel, const VoiceMediaInfo& info); void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info); void OnVoiceChannelError(uint32 ssrc, VoiceMediaChannel::Error error); void SendLastMediaError(); void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error); static const int kEarlyMediaTimeout = 1000; bool received_media_; talk_base::scoped_ptr media_monitor_; talk_base::scoped_ptr audio_monitor_; talk_base::scoped_ptr typing_monitor_; }; // VideoChannel is a specialization for video. class VideoChannel : public BaseChannel { public: // Make screen capturer virtual so that it can be overriden in testing. // E.g. used to test that window events are triggered correctly. class ScreenCapturerFactory { public: virtual VideoCapturer* CreateScreenCapturer(const ScreencastId& window) = 0; virtual ~ScreenCapturerFactory() {} }; VideoChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine, VideoMediaChannel* channel, BaseSession* session, const std::string& content_name, bool rtcp, VoiceChannel* voice_channel); ~VideoChannel(); bool Init(); bool SetRenderer(uint32 ssrc, VideoRenderer* renderer); bool ApplyViewRequest(const ViewRequest& request); // TODO(pthatcher): Refactor to use a "capture id" instead of an // ssrc here as the "key". VideoCapturer* AddScreencast(uint32 ssrc, const ScreencastId& id); bool SetCapturer(uint32 ssrc, VideoCapturer* capturer); bool RemoveScreencast(uint32 ssrc); // True if we've added a screencast. Doesn't matter if the capturer // has been started or not. bool IsScreencasting(); int GetScreencastFps(uint32 ssrc); int GetScreencastMaxPixels(uint32 ssrc); // Get statistics about the current media session. bool GetStats(const StatsOptions& options, VideoMediaInfo* stats); sigslot::signal2&> SignalConnectionMonitor; void StartMediaMonitor(int cms); void StopMediaMonitor(); sigslot::signal2 SignalMediaMonitor; sigslot::signal2 SignalScreencastWindowEvent; bool SendIntraFrame(); bool RequestIntraFrame(); sigslot::signal3 SignalMediaError; void SetScreenCaptureFactory( ScreenCapturerFactory* screencapture_factory); // Configuration and setting. bool SetChannelOptions(const VideoOptions& options); protected: // downcasts a MediaChannel virtual VideoMediaChannel* media_channel() const { return static_cast(BaseChannel::media_channel()); } private: typedef std::map ScreencastMap; struct ScreencastDetailsData; // overrides from BaseChannel virtual void ChangeState(); virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc); virtual bool SetLocalContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc); virtual bool SetRemoteContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc); bool ApplyViewRequest_w(const ViewRequest& request); VideoCapturer* AddScreencast_w(uint32 ssrc, const ScreencastId& id); bool RemoveScreencast_w(uint32 ssrc); void OnScreencastWindowEvent_s(uint32 ssrc, talk_base::WindowEvent we); bool IsScreencasting_w() const; void GetScreencastDetails_w(ScreencastDetailsData* d) const; void SetScreenCaptureFactory_w( ScreenCapturerFactory* screencapture_factory); bool GetStats_w(VideoMediaInfo* stats); virtual void OnMessage(talk_base::Message* pmsg); virtual void GetSrtpCiphers(std::vector* ciphers) const; virtual void OnConnectionMonitorUpdate( SocketMonitor* monitor, const std::vector& infos); virtual void OnMediaMonitorUpdate( VideoMediaChannel* media_channel, const VideoMediaInfo& info); virtual void OnScreencastWindowEvent(uint32 ssrc, talk_base::WindowEvent event); virtual void OnStateChange(VideoCapturer* capturer, CaptureState ev); bool GetLocalSsrc(const VideoCapturer* capturer, uint32* ssrc); void OnVideoChannelError(uint32 ssrc, VideoMediaChannel::Error error); void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error); VoiceChannel* voice_channel_; VideoRenderer* renderer_; talk_base::scoped_ptr screencapture_factory_; ScreencastMap screencast_capturers_; talk_base::scoped_ptr media_monitor_; talk_base::WindowEvent previous_we_; }; // DataChannel is a specialization for data. class DataChannel : public BaseChannel { public: DataChannel(talk_base::Thread* thread, DataMediaChannel* media_channel, BaseSession* session, const std::string& content_name, bool rtcp); ~DataChannel(); bool Init(); virtual bool SendData(const SendDataParams& params, const talk_base::Buffer& payload, SendDataResult* result); void StartMediaMonitor(int cms); void StopMediaMonitor(); // Should be called on the signaling thread only. bool ready_to_send_data() const { return ready_to_send_data_; } sigslot::signal2 SignalMediaMonitor; sigslot::signal2&> SignalConnectionMonitor; sigslot::signal3 SignalMediaError; sigslot::signal3 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 SignalReadyToSendData; protected: // downcasts a MediaChannel. virtual DataMediaChannel* media_channel() const { return static_cast(BaseChannel::media_channel()); } private: struct SendDataMessageData : public talk_base::MessageData { SendDataMessageData(const SendDataParams& params, const talk_base::Buffer* payload, SendDataResult* result) : params(params), payload(payload), result(result), succeeded(false) { } const SendDataParams& params; const talk_base::Buffer* payload; SendDataResult* result; bool succeeded; }; struct DataReceivedMessageData : public talk_base::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 talk_base::Buffer payload; }; typedef talk_base::TypedMessageData DataChannelReadyToSendMessageData; // overrides from BaseChannel virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc); // If data_channel_type_ is DCT_NONE, set it. Otherwise, check that // it's the same as what was set previously. Returns false if it's // set to one type one type and changed to another type later. bool SetDataChannelType(DataChannelType new_data_channel_type, std::string* error_desc); // Same as SetDataChannelType, but extracts the type from the // DataContentDescription. bool SetDataChannelTypeFromContent(const DataContentDescription* content, std::string* error_desc); virtual bool SetLocalContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc); virtual bool SetRemoteContent_w(const MediaContentDescription* content, ContentAction action, std::string* error_desc); virtual void ChangeState(); virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet); virtual void OnMessage(talk_base::Message* pmsg); virtual void GetSrtpCiphers(std::vector* ciphers) const; virtual void OnConnectionMonitorUpdate( SocketMonitor* monitor, const std::vector& infos); virtual void OnMediaMonitorUpdate( DataMediaChannel* media_channel, const DataMediaInfo& info); virtual bool ShouldSetupDtlsSrtp() const; void OnDataReceived( const ReceiveDataParams& params, const char* data, size_t len); void OnDataChannelError(uint32 ssrc, DataMediaChannel::Error error); void OnDataChannelReadyToSend(bool writable); void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error); talk_base::scoped_ptr media_monitor_; // TODO(pthatcher): Make a separate SctpDataChannel and // RtpDataChannel instead of using this. DataChannelType data_channel_type_; bool ready_to_send_data_; }; } // namespace cricket #endif // TALK_SESSION_MEDIA_CHANNEL_H_