| /* |
| * 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 <string> |
| #include <vector> |
| |
| #include "talk/media/base/mediachannel.h" |
| #include "talk/media/base/mediaengine.h" |
| #include "talk/media/base/streamparams.h" |
| #include "talk/media/base/videocapturer.h" |
| #include "webrtc/p2p/base/session.h" |
| #include "webrtc/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" |
| #include "webrtc/base/asyncudpsocket.h" |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/network.h" |
| #include "webrtc/base/sigslot.h" |
| #include "webrtc/base/window.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 rtc::MessageHandler, public sigslot::has_slots<>, |
| public MediaChannel::NetworkInterface, |
| public ConnectionStatsGetter { |
| public: |
| BaseChannel(rtc::Thread* thread, MediaEngineInterface* media_engine, |
| MediaChannel* channel, BaseSession* session, |
| const std::string& content_name, bool rtcp); |
| virtual ~BaseChannel(); |
| bool Init(); |
| // Deinit may be called multiple times and is simply ignored if it's alreay |
| // done. |
| void Deinit(); |
| |
| rtc::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); |
| |
| bool PushdownLocalDescription(const SessionDescription* local_desc, |
| ContentAction action, |
| std::string* error_desc); |
| bool PushdownRemoteDescription(const SessionDescription* remote_desc, |
| ContentAction action, |
| std::string* error_desc); |
| // 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(); |
| // For ConnectionStatsGetter, used by ConnectionMonitor |
| virtual bool GetConnectionStats(ConnectionInfos* infos) override; |
| |
| void set_srtp_signal_silent_time(uint32 silent_time) { |
| srtp_filter_.set_signal_silent_time(silent_time); |
| } |
| |
| template <class T> |
| void RegisterSendSink(T* sink, |
| void (T::*OnPacket)(const void*, size_t, bool), |
| SinkType type) { |
| rtc::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) { |
| rtc::CritScope cs(&signal_send_packet_cs_); |
| if (SINK_POST_CRYPTO == type) { |
| SignalSendPacketPostCrypto.disconnect(sink); |
| } else { |
| SignalSendPacketPreCrypto.disconnect(sink); |
| } |
| } |
| |
| bool HasSendSinks(SinkType type) { |
| rtc::CritScope cs(&signal_send_packet_cs_); |
| if (SINK_POST_CRYPTO == type) { |
| return !SignalSendPacketPostCrypto.is_empty(); |
| } else { |
| return !SignalSendPacketPreCrypto.is_empty(); |
| } |
| } |
| |
| template <class T> |
| void RegisterRecvSink(T* sink, |
| void (T::*OnPacket)(const void*, size_t, bool), |
| SinkType type) { |
| rtc::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) { |
| rtc::CritScope cs(&signal_recv_packet_cs_); |
| if (SINK_POST_CRYPTO == type) { |
| SignalRecvPacketPostCrypto.disconnect(sink); |
| } else { |
| SignalRecvPacketPreCrypto.disconnect(sink); |
| } |
| } |
| |
| bool HasRecvSinks(SinkType type) { |
| rtc::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<StreamParams>& local_streams() const { |
| return local_streams_; |
| } |
| const std::vector<StreamParams>& remote_streams() const { |
| return remote_streams_; |
| } |
| |
| sigslot::signal2<BaseChannel*, bool> SignalDtlsSetupFailure; |
| void SignalDtlsSetupFailure_w(bool rtcp); |
| void SignalDtlsSetupFailure_s(bool rtcp); |
| |
| // Used for latency measurements. |
| sigslot::signal1<BaseChannel*> SignalFirstPacketReceived; |
| |
| // Used to alert UI when the muted status changes, perhaps autonomously. |
| sigslot::repeater2<BaseChannel*, bool> SignalAutoMuted; |
| |
| // Made public for easier testing. |
| void SetReadyToSend(TransportChannel* channel, bool ready); |
| |
| // Only public for unit tests. Otherwise, consider protected. |
| virtual int SetOption(SocketType type, rtc::Socket::Option o, int val); |
| |
| protected: |
| MediaEngineInterface* media_engine() const { return media_engine_; } |
| virtual MediaChannel* media_channel() const { return media_channel_; } |
| // Sets the transport_channel_ and rtcp_transport_channel_. If |
| // |rtcp| is false, set rtcp_transport_channel_ is set to NULL. Get |
| // the transport channels from |session|. |
| // TODO(pthatcher): Pass in a Transport instead of a BaseSession. |
| bool SetTransportChannels(BaseSession* session, bool rtcp); |
| bool SetTransportChannels_w(BaseSession* session, bool rtcp); |
| void set_transport_channel(TransportChannel* transport); |
| 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; |
| rtc::Thread* signaling_thread() { return session_->signaling_thread(); } |
| SrtpFilter* srtp_filter() { return &srtp_filter_; } |
| bool rtcp() const { return rtcp_; } |
| |
| void ConnectToTransportChannel(TransportChannel* tc); |
| void DisconnectFromTransportChannel(TransportChannel* tc); |
| |
| void FlushRtcpMessages(); |
| |
| // NetworkInterface implementation, called by MediaEngine |
| virtual bool SendPacket(rtc::Buffer* packet, |
| rtc::DiffServCodePoint dscp); |
| virtual bool SendRtcp(rtc::Buffer* packet, |
| rtc::DiffServCodePoint dscp); |
| |
| // From TransportChannel |
| void OnWritableState(TransportChannel* channel); |
| virtual void OnChannelRead(TransportChannel* channel, |
| const char* data, |
| size_t len, |
| const rtc::PacketTime& packet_time, |
| int flags); |
| void OnReadyToSend(TransportChannel* channel); |
| |
| bool PacketIsRtcp(const TransportChannel* channel, const char* data, |
| size_t len); |
| bool SendPacket(bool rtcp, rtc::Buffer* packet, |
| rtc::DiffServCodePoint dscp); |
| virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet); |
| void HandlePacket(bool rtcp, rtc::Buffer* packet, |
| const rtc::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<StreamParams>& streams, |
| ContentAction action, |
| std::string* error_desc); |
| bool UpdateRemoteStreams_w(const std::vector<StreamParams>& 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<RtpHeaderExtension>& extensions); |
| |
| bool SetRecvRtpHeaderExtensions_w(const MediaContentDescription* content, |
| MediaChannel* media_channel, |
| std::string* error_desc); |
| bool SetSendRtpHeaderExtensions_w(const MediaContentDescription* content, |
| MediaChannel* media_channel, |
| std::string* error_desc); |
| |
| bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos, |
| bool* dtls, |
| std::string* error_desc); |
| bool SetSrtp_w(const std::vector<CryptoParams>& 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(rtc::Message* pmsg); |
| |
| // Handled in derived classes |
| // Get the SRTP ciphers to use for RTP media |
| virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const = 0; |
| virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor, |
| const std::vector<ConnectionInfo>& infos) = 0; |
| |
| // Helper function for invoking bool-returning methods on the worker thread. |
| template <class FunctorT> |
| bool InvokeOnWorker(const FunctorT& functor) { |
| return worker_thread_->Invoke<bool>(functor); |
| } |
| |
| private: |
| sigslot::signal3<const void*, size_t, bool> SignalSendPacketPreCrypto; |
| sigslot::signal3<const void*, size_t, bool> SignalSendPacketPostCrypto; |
| sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPreCrypto; |
| sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPostCrypto; |
| rtc::CriticalSection signal_send_packet_cs_; |
| rtc::CriticalSection signal_recv_packet_cs_; |
| |
| rtc::Thread* worker_thread_; |
| MediaEngineInterface* media_engine_; |
| BaseSession* session_; |
| MediaChannel* media_channel_; |
| std::vector<StreamParams> local_streams_; |
| std::vector<StreamParams> remote_streams_; |
| |
| const std::string content_name_; |
| bool rtcp_; |
| TransportChannel* transport_channel_; |
| TransportChannel* rtcp_transport_channel_; |
| SrtpFilter srtp_filter_; |
| RtcpMuxFilter rtcp_mux_filter_; |
| BundleFilter bundle_filter_; |
| rtc::scoped_ptr<ConnectionMonitor> connection_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<uint32> 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(rtc::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<VoiceMediaChannel*>(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<VoiceChannel*> 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<VoiceChannel*, const std::vector<ConnectionInfo>&> |
| SignalConnectionMonitor; |
| |
| void StartMediaMonitor(int cms); |
| void StopMediaMonitor(); |
| sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor; |
| |
| void StartAudioMonitor(int cms); |
| void StopAudioMonitor(); |
| bool IsAudioMonitorRunning() const; |
| sigslot::signal2<VoiceChannel*, const AudioInfo&> 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<VoiceChannel*, uint32, VoiceMediaChannel::Error> |
| 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 rtc::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(rtc::Message* pmsg); |
| virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const; |
| virtual void OnConnectionMonitorUpdate( |
| ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& 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_; |
| rtc::scoped_ptr<VoiceMediaMonitor> media_monitor_; |
| rtc::scoped_ptr<AudioMonitor> audio_monitor_; |
| rtc::scoped_ptr<TypingMonitor> typing_monitor_; |
| }; |
| |
| // VideoChannel is a specialization for video. |
| class VideoChannel : public BaseChannel { |
| public: |
| VideoChannel(rtc::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". |
| // Passes ownership of the capturer to the channel. |
| bool AddScreencast(uint32 ssrc, VideoCapturer* capturer); |
| 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(VideoMediaInfo* stats); |
| |
| sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&> |
| SignalConnectionMonitor; |
| |
| void StartMediaMonitor(int cms); |
| void StopMediaMonitor(); |
| sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor; |
| sigslot::signal2<uint32, rtc::WindowEvent> SignalScreencastWindowEvent; |
| |
| bool SendIntraFrame(); |
| bool RequestIntraFrame(); |
| sigslot::signal3<VideoChannel*, uint32, VideoMediaChannel::Error> |
| SignalMediaError; |
| |
| // Configuration and setting. |
| bool SetChannelOptions(const VideoOptions& options); |
| |
| protected: |
| // downcasts a MediaChannel |
| virtual VideoMediaChannel* media_channel() const { |
| return static_cast<VideoMediaChannel*>(BaseChannel::media_channel()); |
| } |
| |
| private: |
| typedef std::map<uint32, VideoCapturer*> 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); |
| |
| bool AddScreencast_w(uint32 ssrc, VideoCapturer* capturer); |
| bool RemoveScreencast_w(uint32 ssrc); |
| void OnScreencastWindowEvent_s(uint32 ssrc, rtc::WindowEvent we); |
| bool IsScreencasting_w() const; |
| void GetScreencastDetails_w(ScreencastDetailsData* d) const; |
| bool GetStats_w(VideoMediaInfo* stats); |
| |
| virtual void OnMessage(rtc::Message* pmsg); |
| virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const; |
| virtual void OnConnectionMonitorUpdate( |
| ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos); |
| virtual void OnMediaMonitorUpdate( |
| VideoMediaChannel* media_channel, const VideoMediaInfo& info); |
| virtual void OnScreencastWindowEvent(uint32 ssrc, |
| rtc::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_; |
| ScreencastMap screencast_capturers_; |
| rtc::scoped_ptr<VideoMediaMonitor> media_monitor_; |
| |
| rtc::WindowEvent previous_we_; |
| }; |
| |
| // DataChannel is a specialization for data. |
| class DataChannel : public BaseChannel { |
| public: |
| DataChannel(rtc::Thread* thread, |
| DataMediaChannel* media_channel, |
| BaseSession* session, |
| const std::string& content_name, |
| bool rtcp); |
| ~DataChannel(); |
| bool Init(); |
| |
| virtual bool SendData(const SendDataParams& params, |
| const rtc::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<DataChannel*, const DataMediaInfo&> SignalMediaMonitor; |
| sigslot::signal2<DataChannel*, const std::vector<ConnectionInfo>&> |
| SignalConnectionMonitor; |
| sigslot::signal3<DataChannel*, uint32, DataMediaChannel::Error> |
| SignalMediaError; |
| sigslot::signal3<DataChannel*, |
| const ReceiveDataParams&, |
| const rtc::Buffer&> |
| 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; |
| // Signal for notifying that the remote side has closed the DataChannel. |
| sigslot::signal1<uint32> SignalStreamClosedRemotely; |
| |
| protected: |
| // downcasts a MediaChannel. |
| virtual DataMediaChannel* media_channel() const { |
| return static_cast<DataMediaChannel*>(BaseChannel::media_channel()); |
| } |
| |
| private: |
| struct SendDataMessageData : public rtc::MessageData { |
| SendDataMessageData(const SendDataParams& params, |
| const rtc::Buffer* payload, |
| SendDataResult* result) |
| : params(params), |
| payload(payload), |
| result(result), |
| succeeded(false) { |
| } |
| |
| const SendDataParams& params; |
| const rtc::Buffer* 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::Buffer payload; |
| }; |
| |
| typedef rtc::TypedMessageData<bool> 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, rtc::Buffer* packet); |
| |
| virtual void OnMessage(rtc::Message* pmsg); |
| virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const; |
| virtual void OnConnectionMonitorUpdate( |
| ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& 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); |
| void OnStreamClosedRemotely(uint32 sid); |
| |
| rtc::scoped_ptr<DataMediaMonitor> 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_ |
