Home | History | Annotate | Download | only in media
      1 /*
      2  * libjingle
      3  * Copyright 2004 Google Inc.
      4  *
      5  * Redistribution and use in source and binary forms, with or without
      6  * modification, are permitted provided that the following conditions are met:
      7  *
      8  *  1. Redistributions of source code must retain the above copyright notice,
      9  *     this list of conditions and the following disclaimer.
     10  *  2. Redistributions in binary form must reproduce the above copyright notice,
     11  *     this list of conditions and the following disclaimer in the documentation
     12  *     and/or other materials provided with the distribution.
     13  *  3. The name of the author may not be used to endorse or promote products
     14  *     derived from this software without specific prior written permission.
     15  *
     16  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
     17  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
     18  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
     19  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
     20  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
     21  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
     22  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
     23  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
     24  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
     25  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
     26  */
     27 
     28 #ifndef TALK_SESSION_MEDIA_CHANNEL_H_
     29 #define TALK_SESSION_MEDIA_CHANNEL_H_
     30 
     31 #include <string>
     32 #include <vector>
     33 
     34 #include "talk/app/webrtc/datachannelinterface.h"
     35 #include "talk/base/asyncudpsocket.h"
     36 #include "talk/base/criticalsection.h"
     37 #include "talk/base/network.h"
     38 #include "talk/base/sigslot.h"
     39 #include "talk/base/window.h"
     40 #include "talk/media/base/mediachannel.h"
     41 #include "talk/media/base/mediaengine.h"
     42 #include "talk/media/base/screencastid.h"
     43 #include "talk/media/base/streamparams.h"
     44 #include "talk/media/base/videocapturer.h"
     45 #include "talk/p2p/base/session.h"
     46 #include "talk/p2p/client/socketmonitor.h"
     47 #include "talk/session/media/audiomonitor.h"
     48 #include "talk/session/media/mediamonitor.h"
     49 #include "talk/session/media/mediasession.h"
     50 #include "talk/session/media/rtcpmuxfilter.h"
     51 #include "talk/session/media/srtpfilter.h"
     52 #include "talk/session/media/ssrcmuxfilter.h"
     53 
     54 namespace cricket {
     55 
     56 struct CryptoParams;
     57 class MediaContentDescription;
     58 struct TypingMonitorOptions;
     59 class TypingMonitor;
     60 struct ViewRequest;
     61 
     62 enum SinkType {
     63   SINK_PRE_CRYPTO,  // Sink packets before encryption or after decryption.
     64   SINK_POST_CRYPTO  // Sink packets after encryption or before decryption.
     65 };
     66 
     67 // BaseChannel contains logic common to voice and video, including
     68 // enable/mute, marshaling calls to a worker thread, and
     69 // connection and media monitors.
     70 //
     71 // WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
     72 // This is required to avoid a data race between the destructor modifying the
     73 // vtable, and the media channel's thread using BaseChannel as the
     74 // NetworkInterface.
     75 
     76 class BaseChannel
     77     : public talk_base::MessageHandler, public sigslot::has_slots<>,
     78       public MediaChannel::NetworkInterface {
     79  public:
     80   BaseChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine,
     81               MediaChannel* channel, BaseSession* session,
     82               const std::string& content_name, bool rtcp);
     83   virtual ~BaseChannel();
     84   bool Init(TransportChannel* transport_channel,
     85             TransportChannel* rtcp_transport_channel);
     86   // Deinit may be called multiple times and is simply ignored if it's alreay
     87   // done.
     88   void Deinit();
     89 
     90   talk_base::Thread* worker_thread() const { return worker_thread_; }
     91   BaseSession* session() const { return session_; }
     92   const std::string& content_name() { return content_name_; }
     93   TransportChannel* transport_channel() const {
     94     return transport_channel_;
     95   }
     96   TransportChannel* rtcp_transport_channel() const {
     97     return rtcp_transport_channel_;
     98   }
     99   bool enabled() const { return enabled_; }
    100 
    101   // This function returns true if we are using SRTP.
    102   bool secure() const { return srtp_filter_.IsActive(); }
    103   // The following function returns true if we are using
    104   // DTLS-based keying. If you turned off SRTP later, however
    105   // you could have secure() == false and dtls_secure() == true.
    106   bool secure_dtls() const { return dtls_keyed_; }
    107   // This function returns true if we require secure channel for call setup.
    108   bool secure_required() const { return secure_required_; }
    109 
    110   bool writable() const { return writable_; }
    111   bool IsStreamMuted(uint32 ssrc);
    112 
    113   // Channel control
    114   bool SetLocalContent(const MediaContentDescription* content,
    115                        ContentAction action);
    116   bool SetRemoteContent(const MediaContentDescription* content,
    117                         ContentAction action);
    118   bool SetMaxSendBandwidth(int max_bandwidth);
    119 
    120   bool Enable(bool enable);
    121   // Mute sending media on the stream with SSRC |ssrc|
    122   // If there is only one sending stream SSRC 0 can be used.
    123   bool MuteStream(uint32 ssrc, bool mute);
    124 
    125   // Multiplexing
    126   bool AddRecvStream(const StreamParams& sp);
    127   bool RemoveRecvStream(uint32 ssrc);
    128   bool AddSendStream(const StreamParams& sp);
    129   bool RemoveSendStream(uint32 ssrc);
    130 
    131   // Monitoring
    132   void StartConnectionMonitor(int cms);
    133   void StopConnectionMonitor();
    134 
    135   void set_srtp_signal_silent_time(uint32 silent_time) {
    136     srtp_filter_.set_signal_silent_time(silent_time);
    137   }
    138 
    139   void set_content_name(const std::string& content_name) {
    140     ASSERT(signaling_thread()->IsCurrent());
    141     ASSERT(!writable_);
    142     if (session_->state() != BaseSession::STATE_INIT) {
    143       LOG(LS_ERROR) << "Content name for a channel can be changed only "
    144                     << "when BaseSession is in STATE_INIT state.";
    145       return;
    146     }
    147     content_name_ = content_name;
    148   }
    149 
    150   template <class T>
    151   void RegisterSendSink(T* sink,
    152                         void (T::*OnPacket)(const void*, size_t, bool),
    153                         SinkType type) {
    154     talk_base::CritScope cs(&signal_send_packet_cs_);
    155     if (SINK_POST_CRYPTO == type) {
    156       SignalSendPacketPostCrypto.disconnect(sink);
    157       SignalSendPacketPostCrypto.connect(sink, OnPacket);
    158     } else {
    159       SignalSendPacketPreCrypto.disconnect(sink);
    160       SignalSendPacketPreCrypto.connect(sink, OnPacket);
    161     }
    162   }
    163 
    164   void UnregisterSendSink(sigslot::has_slots<>* sink,
    165                           SinkType type) {
    166     talk_base::CritScope cs(&signal_send_packet_cs_);
    167     if (SINK_POST_CRYPTO == type) {
    168       SignalSendPacketPostCrypto.disconnect(sink);
    169     } else {
    170       SignalSendPacketPreCrypto.disconnect(sink);
    171     }
    172   }
    173 
    174   bool HasSendSinks(SinkType type) {
    175     talk_base::CritScope cs(&signal_send_packet_cs_);
    176     if (SINK_POST_CRYPTO == type) {
    177       return !SignalSendPacketPostCrypto.is_empty();
    178     } else {
    179       return !SignalSendPacketPreCrypto.is_empty();
    180     }
    181   }
    182 
    183   template <class T>
    184   void RegisterRecvSink(T* sink,
    185                         void (T::*OnPacket)(const void*, size_t, bool),
    186                         SinkType type) {
    187     talk_base::CritScope cs(&signal_recv_packet_cs_);
    188     if (SINK_POST_CRYPTO == type) {
    189       SignalRecvPacketPostCrypto.disconnect(sink);
    190       SignalRecvPacketPostCrypto.connect(sink, OnPacket);
    191     } else {
    192       SignalRecvPacketPreCrypto.disconnect(sink);
    193       SignalRecvPacketPreCrypto.connect(sink, OnPacket);
    194     }
    195   }
    196 
    197   void UnregisterRecvSink(sigslot::has_slots<>* sink,
    198                           SinkType type) {
    199     talk_base::CritScope cs(&signal_recv_packet_cs_);
    200     if (SINK_POST_CRYPTO == type) {
    201       SignalRecvPacketPostCrypto.disconnect(sink);
    202     } else {
    203       SignalRecvPacketPreCrypto.disconnect(sink);
    204     }
    205   }
    206 
    207   bool HasRecvSinks(SinkType type) {
    208     talk_base::CritScope cs(&signal_recv_packet_cs_);
    209     if (SINK_POST_CRYPTO == type) {
    210       return !SignalRecvPacketPostCrypto.is_empty();
    211     } else {
    212       return !SignalRecvPacketPreCrypto.is_empty();
    213     }
    214   }
    215 
    216   SsrcMuxFilter* ssrc_filter() { return &ssrc_filter_; }
    217 
    218   const std::vector<StreamParams>& local_streams() const {
    219     return local_streams_;
    220   }
    221   const std::vector<StreamParams>& remote_streams() const {
    222     return remote_streams_;
    223   }
    224 
    225   // Used for latency measurements.
    226   sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
    227 
    228   // Used to alert UI when the muted status changes, perhaps autonomously.
    229   sigslot::repeater2<BaseChannel*, bool> SignalAutoMuted;
    230 
    231   // Made public for easier testing.
    232   void SetReadyToSend(TransportChannel* channel, bool ready);
    233 
    234  protected:
    235   MediaEngineInterface* media_engine() const { return media_engine_; }
    236   virtual MediaChannel* media_channel() const { return media_channel_; }
    237   void set_rtcp_transport_channel(TransportChannel* transport);
    238   bool was_ever_writable() const { return was_ever_writable_; }
    239   void set_local_content_direction(MediaContentDirection direction) {
    240     local_content_direction_ = direction;
    241   }
    242   void set_remote_content_direction(MediaContentDirection direction) {
    243     remote_content_direction_ = direction;
    244   }
    245   bool IsReadyToReceive() const;
    246   bool IsReadyToSend() const;
    247   talk_base::Thread* signaling_thread() { return session_->signaling_thread(); }
    248   SrtpFilter* srtp_filter() { return &srtp_filter_; }
    249   bool rtcp() const { return rtcp_; }
    250 
    251   void Send(uint32 id, talk_base::MessageData* pdata = NULL);
    252   void Post(uint32 id, talk_base::MessageData* pdata = NULL);
    253   void PostDelayed(int cmsDelay, uint32 id = 0,
    254                    talk_base::MessageData* pdata = NULL);
    255   void Clear(uint32 id = talk_base::MQID_ANY,
    256              talk_base::MessageList* removed = NULL);
    257   void FlushRtcpMessages();
    258 
    259   // NetworkInterface implementation, called by MediaEngine
    260   virtual bool SendPacket(talk_base::Buffer* packet,
    261                           talk_base::DiffServCodePoint dscp);
    262   virtual bool SendRtcp(talk_base::Buffer* packet,
    263                         talk_base::DiffServCodePoint dscp);
    264   virtual int SetOption(SocketType type, talk_base::Socket::Option o, int val);
    265 
    266   // From TransportChannel
    267   void OnWritableState(TransportChannel* channel);
    268   virtual void OnChannelRead(TransportChannel* channel,
    269                              const char* data,
    270                              size_t len,
    271                              const talk_base::PacketTime& packet_time,
    272                              int flags);
    273   void OnReadyToSend(TransportChannel* channel);
    274 
    275   bool PacketIsRtcp(const TransportChannel* channel, const char* data,
    276                     size_t len);
    277   bool SendPacket(bool rtcp, talk_base::Buffer* packet,
    278                   talk_base::DiffServCodePoint dscp);
    279   virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet);
    280   void HandlePacket(bool rtcp, talk_base::Buffer* packet,
    281                     const talk_base::PacketTime& packet_time);
    282 
    283   // Apply the new local/remote session description.
    284   void OnNewLocalDescription(BaseSession* session, ContentAction action);
    285   void OnNewRemoteDescription(BaseSession* session, ContentAction action);
    286 
    287   void EnableMedia_w();
    288   void DisableMedia_w();
    289   virtual bool MuteStream_w(uint32 ssrc, bool mute);
    290   bool IsStreamMuted_w(uint32 ssrc);
    291   void ChannelWritable_w();
    292   void ChannelNotWritable_w();
    293   bool AddRecvStream_w(const StreamParams& sp);
    294   bool RemoveRecvStream_w(uint32 ssrc);
    295   bool AddSendStream_w(const StreamParams& sp);
    296   bool RemoveSendStream_w(uint32 ssrc);
    297   virtual bool ShouldSetupDtlsSrtp() const;
    298   // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
    299   // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
    300   bool SetupDtlsSrtp(bool rtcp_channel);
    301   // Set the DTLS-SRTP cipher policy on this channel as appropriate.
    302   bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
    303 
    304   virtual void ChangeState() = 0;
    305 
    306   // Gets the content info appropriate to the channel (audio or video).
    307   virtual const ContentInfo* GetFirstContent(
    308       const SessionDescription* sdesc) = 0;
    309   bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
    310                             ContentAction action);
    311   bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
    312                              ContentAction action);
    313   bool SetBaseLocalContent_w(const MediaContentDescription* content,
    314                              ContentAction action);
    315   virtual bool SetLocalContent_w(const MediaContentDescription* content,
    316                                  ContentAction action) = 0;
    317   bool SetBaseRemoteContent_w(const MediaContentDescription* content,
    318                               ContentAction action);
    319   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
    320                                   ContentAction action) = 0;
    321 
    322   bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos, bool* dtls);
    323   bool SetSrtp_w(const std::vector<CryptoParams>& params, ContentAction action,
    324                  ContentSource src);
    325   bool SetRtcpMux_w(bool enable, ContentAction action, ContentSource src);
    326 
    327   virtual bool SetMaxSendBandwidth_w(int max_bandwidth);
    328 
    329   // From MessageHandler
    330   virtual void OnMessage(talk_base::Message* pmsg);
    331 
    332   // Handled in derived classes
    333   // Get the SRTP ciphers to use for RTP media
    334   virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const = 0;
    335   virtual void OnConnectionMonitorUpdate(SocketMonitor* monitor,
    336       const std::vector<ConnectionInfo>& infos) = 0;
    337 
    338  private:
    339   sigslot::signal3<const void*, size_t, bool> SignalSendPacketPreCrypto;
    340   sigslot::signal3<const void*, size_t, bool> SignalSendPacketPostCrypto;
    341   sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPreCrypto;
    342   sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPostCrypto;
    343   talk_base::CriticalSection signal_send_packet_cs_;
    344   talk_base::CriticalSection signal_recv_packet_cs_;
    345 
    346   talk_base::Thread* worker_thread_;
    347   MediaEngineInterface* media_engine_;
    348   BaseSession* session_;
    349   MediaChannel* media_channel_;
    350   std::vector<StreamParams> local_streams_;
    351   std::vector<StreamParams> remote_streams_;
    352 
    353   std::string content_name_;
    354   bool rtcp_;
    355   TransportChannel* transport_channel_;
    356   TransportChannel* rtcp_transport_channel_;
    357   SrtpFilter srtp_filter_;
    358   RtcpMuxFilter rtcp_mux_filter_;
    359   SsrcMuxFilter ssrc_filter_;
    360   talk_base::scoped_ptr<SocketMonitor> socket_monitor_;
    361   bool enabled_;
    362   bool writable_;
    363   bool rtp_ready_to_send_;
    364   bool rtcp_ready_to_send_;
    365   bool was_ever_writable_;
    366   MediaContentDirection local_content_direction_;
    367   MediaContentDirection remote_content_direction_;
    368   std::set<uint32> muted_streams_;
    369   bool has_received_packet_;
    370   bool dtls_keyed_;
    371   bool secure_required_;
    372 };
    373 
    374 // VoiceChannel is a specialization that adds support for early media, DTMF,
    375 // and input/output level monitoring.
    376 class VoiceChannel : public BaseChannel {
    377  public:
    378   VoiceChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine,
    379                VoiceMediaChannel* channel, BaseSession* session,
    380                const std::string& content_name, bool rtcp);
    381   ~VoiceChannel();
    382   bool Init();
    383   bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer);
    384   bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
    385 
    386   // downcasts a MediaChannel
    387   virtual VoiceMediaChannel* media_channel() const {
    388     return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
    389   }
    390 
    391   bool SetRingbackTone(const void* buf, int len);
    392   void SetEarlyMedia(bool enable);
    393   // This signal is emitted when we have gone a period of time without
    394   // receiving early media. When received, a UI should start playing its
    395   // own ringing sound
    396   sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
    397 
    398   bool PlayRingbackTone(uint32 ssrc, bool play, bool loop);
    399   // TODO(ronghuawu): Replace PressDTMF with InsertDtmf.
    400   bool PressDTMF(int digit, bool playout);
    401   // Returns if the telephone-event has been negotiated.
    402   bool CanInsertDtmf();
    403   // Send and/or play a DTMF |event| according to the |flags|.
    404   // The DTMF out-of-band signal will be used on sending.
    405   // The |ssrc| should be either 0 or a valid send stream ssrc.
    406   // The valid value for the |event| are 0 which corresponding to DTMF
    407   // event 0-9, *, #, A-D.
    408   bool InsertDtmf(uint32 ssrc, int event_code, int duration, int flags);
    409   bool SetOutputScaling(uint32 ssrc, double left, double right);
    410   // Get statistics about the current media session.
    411   bool GetStats(VoiceMediaInfo* stats);
    412 
    413   // Monitoring functions
    414   sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
    415       SignalConnectionMonitor;
    416 
    417   void StartMediaMonitor(int cms);
    418   void StopMediaMonitor();
    419   sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
    420 
    421   void StartAudioMonitor(int cms);
    422   void StopAudioMonitor();
    423   bool IsAudioMonitorRunning() const;
    424   sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;
    425 
    426   void StartTypingMonitor(const TypingMonitorOptions& settings);
    427   void StopTypingMonitor();
    428   bool IsTypingMonitorRunning() const;
    429 
    430   // Overrides BaseChannel::MuteStream_w.
    431   virtual bool MuteStream_w(uint32 ssrc, bool mute);
    432 
    433   int GetInputLevel_w();
    434   int GetOutputLevel_w();
    435   void GetActiveStreams_w(AudioInfo::StreamList* actives);
    436 
    437   // Signal errors from VoiceMediaChannel.  Arguments are:
    438   //     ssrc(uint32), and error(VoiceMediaChannel::Error).
    439   sigslot::signal3<VoiceChannel*, uint32, VoiceMediaChannel::Error>
    440       SignalMediaError;
    441 
    442   // Configuration and setting.
    443   bool SetChannelOptions(const AudioOptions& options);
    444 
    445  private:
    446   // overrides from BaseChannel
    447   virtual void OnChannelRead(TransportChannel* channel,
    448                              const char* data, size_t len,
    449                              const talk_base::PacketTime& packet_time,
    450                              int flags);
    451   virtual void ChangeState();
    452   virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
    453   virtual bool SetLocalContent_w(const MediaContentDescription* content,
    454                                  ContentAction action);
    455   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
    456                                   ContentAction action);
    457   bool SetRingbackTone_w(const void* buf, int len);
    458   bool PlayRingbackTone_w(uint32 ssrc, bool play, bool loop);
    459   void HandleEarlyMediaTimeout();
    460   bool CanInsertDtmf_w();
    461   bool InsertDtmf_w(uint32 ssrc, int event, int duration, int flags);
    462   bool SetOutputScaling_w(uint32 ssrc, double left, double right);
    463   bool GetStats_w(VoiceMediaInfo* stats);
    464 
    465   virtual void OnMessage(talk_base::Message* pmsg);
    466   virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const;
    467   virtual void OnConnectionMonitorUpdate(
    468       SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos);
    469   virtual void OnMediaMonitorUpdate(
    470       VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
    471   void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
    472   void OnVoiceChannelError(uint32 ssrc, VoiceMediaChannel::Error error);
    473   void SendLastMediaError();
    474   void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
    475   // Configuration and setting.
    476   bool SetChannelOptions_w(const AudioOptions& options);
    477   bool SetRenderer_w(uint32 ssrc, AudioRenderer* renderer, bool is_local);
    478 
    479   static const int kEarlyMediaTimeout = 1000;
    480   bool received_media_;
    481   talk_base::scoped_ptr<VoiceMediaMonitor> media_monitor_;
    482   talk_base::scoped_ptr<AudioMonitor> audio_monitor_;
    483   talk_base::scoped_ptr<TypingMonitor> typing_monitor_;
    484 };
    485 
    486 // VideoChannel is a specialization for video.
    487 class VideoChannel : public BaseChannel {
    488  public:
    489   // Make screen capturer virtual so that it can be overriden in testing.
    490   // E.g. used to test that window events are triggered correctly.
    491   class ScreenCapturerFactory {
    492    public:
    493     virtual VideoCapturer* CreateScreenCapturer(const ScreencastId& window) = 0;
    494     virtual ~ScreenCapturerFactory() {}
    495   };
    496 
    497   VideoChannel(talk_base::Thread* thread, MediaEngineInterface* media_engine,
    498                VideoMediaChannel* channel, BaseSession* session,
    499                const std::string& content_name, bool rtcp,
    500                VoiceChannel* voice_channel);
    501   ~VideoChannel();
    502   bool Init();
    503 
    504   bool SetRenderer(uint32 ssrc, VideoRenderer* renderer);
    505   bool ApplyViewRequest(const ViewRequest& request);
    506 
    507   // TODO(pthatcher): Refactor to use a "capture id" instead of an
    508   // ssrc here as the "key".
    509   VideoCapturer* AddScreencast(uint32 ssrc, const ScreencastId& id);
    510   bool SetCapturer(uint32 ssrc, VideoCapturer* capturer);
    511   bool RemoveScreencast(uint32 ssrc);
    512   // True if we've added a screencast.  Doesn't matter if the capturer
    513   // has been started or not.
    514   bool IsScreencasting();
    515   int GetScreencastFps(uint32 ssrc);
    516   int GetScreencastMaxPixels(uint32 ssrc);
    517   // Get statistics about the current media session.
    518   bool GetStats(VideoMediaInfo* stats);
    519 
    520   sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&>
    521       SignalConnectionMonitor;
    522 
    523   void StartMediaMonitor(int cms);
    524   void StopMediaMonitor();
    525   sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
    526   sigslot::signal2<uint32, talk_base::WindowEvent> SignalScreencastWindowEvent;
    527 
    528   bool SendIntraFrame();
    529   bool RequestIntraFrame();
    530   sigslot::signal3<VideoChannel*, uint32, VideoMediaChannel::Error>
    531       SignalMediaError;
    532 
    533   void SetScreenCaptureFactory(
    534       ScreenCapturerFactory* screencapture_factory);
    535 
    536   // Configuration and setting.
    537   bool SetChannelOptions(const VideoOptions& options);
    538 
    539  protected:
    540   // downcasts a MediaChannel
    541   virtual VideoMediaChannel* media_channel() const {
    542     return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
    543   }
    544 
    545  private:
    546   typedef std::map<uint32, VideoCapturer*> ScreencastMap;
    547   struct ScreencastDetailsMessageData;
    548 
    549   // overrides from BaseChannel
    550   virtual void ChangeState();
    551   virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
    552   virtual bool SetLocalContent_w(const MediaContentDescription* content,
    553                                  ContentAction action);
    554   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
    555                                   ContentAction action);
    556   void SendIntraFrame_w() {
    557     media_channel()->SendIntraFrame();
    558   }
    559   void RequestIntraFrame_w() {
    560     media_channel()->RequestIntraFrame();
    561   }
    562 
    563   bool ApplyViewRequest_w(const ViewRequest& request);
    564   void SetRenderer_w(uint32 ssrc, VideoRenderer* renderer);
    565 
    566   VideoCapturer* AddScreencast_w(uint32 ssrc, const ScreencastId& id);
    567   bool SetCapturer_w(uint32 ssrc, VideoCapturer* capturer);
    568   bool RemoveScreencast_w(uint32 ssrc);
    569   void OnScreencastWindowEvent_s(uint32 ssrc, talk_base::WindowEvent we);
    570   bool IsScreencasting_w() const;
    571   void ScreencastDetails_w(ScreencastDetailsMessageData* d) const;
    572   void SetScreenCaptureFactory_w(
    573       ScreenCapturerFactory* screencapture_factory);
    574   bool GetStats_w(VideoMediaInfo* stats);
    575 
    576   virtual void OnMessage(talk_base::Message* pmsg);
    577   virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const;
    578   virtual void OnConnectionMonitorUpdate(
    579       SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos);
    580   virtual void OnMediaMonitorUpdate(
    581       VideoMediaChannel* media_channel, const VideoMediaInfo& info);
    582   virtual void OnScreencastWindowEvent(uint32 ssrc,
    583                                        talk_base::WindowEvent event);
    584   virtual void OnStateChange(VideoCapturer* capturer, CaptureState ev);
    585   bool GetLocalSsrc(const VideoCapturer* capturer, uint32* ssrc);
    586 
    587   void OnVideoChannelError(uint32 ssrc, VideoMediaChannel::Error error);
    588   void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
    589   // Configuration and setting.
    590   bool SetChannelOptions_w(const VideoOptions& options);
    591 
    592   VoiceChannel* voice_channel_;
    593   VideoRenderer* renderer_;
    594   talk_base::scoped_ptr<ScreenCapturerFactory> screencapture_factory_;
    595   ScreencastMap screencast_capturers_;
    596   talk_base::scoped_ptr<VideoMediaMonitor> media_monitor_;
    597 
    598   talk_base::WindowEvent previous_we_;
    599 };
    600 
    601 // DataChannel is a specialization for data.
    602 class DataChannel : public BaseChannel {
    603  public:
    604   DataChannel(talk_base::Thread* thread,
    605               DataMediaChannel* media_channel,
    606               BaseSession* session,
    607               const std::string& content_name,
    608               bool rtcp);
    609   ~DataChannel();
    610   bool Init();
    611 
    612   virtual bool SendData(const SendDataParams& params,
    613                         const talk_base::Buffer& payload,
    614                         SendDataResult* result);
    615 
    616   void StartMediaMonitor(int cms);
    617   void StopMediaMonitor();
    618 
    619   // Should be called on the signaling thread only.
    620   bool ready_to_send_data() const {
    621     return ready_to_send_data_;
    622   }
    623 
    624   sigslot::signal2<DataChannel*, const DataMediaInfo&> SignalMediaMonitor;
    625   sigslot::signal2<DataChannel*, const std::vector<ConnectionInfo>&>
    626       SignalConnectionMonitor;
    627   sigslot::signal3<DataChannel*, uint32, DataMediaChannel::Error>
    628       SignalMediaError;
    629   sigslot::signal3<DataChannel*,
    630                    const ReceiveDataParams&,
    631                    const talk_base::Buffer&>
    632       SignalDataReceived;
    633   // Signal for notifying when the channel becomes ready to send data.
    634   // That occurs when the channel is enabled, the transport is writable,
    635   // both local and remote descriptions are set, and the channel is unblocked.
    636   sigslot::signal1<bool> SignalReadyToSendData;
    637   // Signal for notifying when a new stream is added from the remote side. Used
    638   // for the in-band negotioation through the OPEN message for SCTP data
    639   // channel.
    640   sigslot::signal2<const std::string&, const webrtc::DataChannelInit&>
    641       SignalNewStreamReceived;
    642 
    643  protected:
    644   // downcasts a MediaChannel.
    645   virtual DataMediaChannel* media_channel() const {
    646     return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
    647   }
    648 
    649  private:
    650   struct SendDataMessageData : public talk_base::MessageData {
    651     SendDataMessageData(const SendDataParams& params,
    652                         const talk_base::Buffer* payload,
    653                         SendDataResult* result)
    654         : params(params),
    655           payload(payload),
    656           result(result),
    657           succeeded(false) {
    658     }
    659 
    660     const SendDataParams& params;
    661     const talk_base::Buffer* payload;
    662     SendDataResult* result;
    663     bool succeeded;
    664   };
    665 
    666   struct DataReceivedMessageData : public talk_base::MessageData {
    667     // We copy the data because the data will become invalid after we
    668     // handle DataMediaChannel::SignalDataReceived but before we fire
    669     // SignalDataReceived.
    670     DataReceivedMessageData(
    671         const ReceiveDataParams& params, const char* data, size_t len)
    672         : params(params),
    673           payload(data, len) {
    674     }
    675     const ReceiveDataParams params;
    676     const talk_base::Buffer payload;
    677   };
    678 
    679   typedef talk_base::TypedMessageData<bool> DataChannelReadyToSendMessageData;
    680 
    681   struct DataChannelNewStreamReceivedMessageData
    682       : public talk_base::MessageData {
    683     DataChannelNewStreamReceivedMessageData(
    684         const std::string& label, const webrtc::DataChannelInit& init)
    685         : label(label),
    686           init(init) {
    687     }
    688     const std::string label;
    689     const webrtc::DataChannelInit init;
    690   };
    691 
    692   // overrides from BaseChannel
    693   virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
    694   // If data_channel_type_ is DCT_NONE, set it.  Otherwise, check that
    695   // it's the same as what was set previously.  Returns false if it's
    696   // set to one type one type and changed to another type later.
    697   bool SetDataChannelType(DataChannelType new_data_channel_type);
    698   // Same as SetDataChannelType, but extracts the type from the
    699   // DataContentDescription.
    700   bool SetDataChannelTypeFromContent(const DataContentDescription* content);
    701   virtual bool SetMaxSendBandwidth_w(int max_bandwidth);
    702   virtual bool SetLocalContent_w(const MediaContentDescription* content,
    703                                  ContentAction action);
    704   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
    705                                   ContentAction action);
    706   virtual void ChangeState();
    707   virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet);
    708 
    709   virtual void OnMessage(talk_base::Message* pmsg);
    710   virtual void GetSrtpCiphers(std::vector<std::string>* ciphers) const;
    711   virtual void OnConnectionMonitorUpdate(
    712       SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos);
    713   virtual void OnMediaMonitorUpdate(
    714       DataMediaChannel* media_channel, const DataMediaInfo& info);
    715   virtual bool ShouldSetupDtlsSrtp() const;
    716   void OnDataReceived(
    717       const ReceiveDataParams& params, const char* data, size_t len);
    718   void OnDataChannelError(uint32 ssrc, DataMediaChannel::Error error);
    719   void OnDataChannelReadyToSend(bool writable);
    720   void OnDataChannelNewStreamReceived(const std::string& label,
    721                                       const webrtc::DataChannelInit& init);
    722   void OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode, SrtpFilter::Error error);
    723 
    724   talk_base::scoped_ptr<DataMediaMonitor> media_monitor_;
    725   // TODO(pthatcher): Make a separate SctpDataChannel and
    726   // RtpDataChannel instead of using this.
    727   DataChannelType data_channel_type_;
    728   bool ready_to_send_data_;
    729 };
    730 
    731 }  // namespace cricket
    732 
    733 #endif  // TALK_SESSION_MEDIA_CHANNEL_H_
    734