Temp hook in WebRtcSession to VideoChannel.

Review URL: http://webrtc-codereview.appspot.com/195001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@689 4adac7df-926f-26a2-2b94-8c16560cd09d

third_party_mods/libjingle/libjingle.gyp

@@ -522,6 +522,7 @@
         '<(libjingle_orig)/source/talk/p2p/client/basicportallocator.h',
         '<(libjingle_orig)/source/talk/p2p/client/httpportallocator.cc',
         '<(libjingle_orig)/source/talk/p2p/client/httpportallocator.h',
+        '<(libjingle_mods)/source/talk/p2p/client/fakeportallocator.h',
         '<(libjingle_orig)/source/talk/p2p/client/sessionmanagertask.h',
         '<(libjingle_orig)/source/talk/p2p/client/sessionsendtask.h',
         '<(libjingle_orig)/source/talk/p2p/client/socketmonitor.cc',
@@ -530,8 +531,8 @@
         '<(libjingle_orig)/source/talk/session/phone/audiomonitor.h',
         '<(libjingle_orig)/source/talk/session/phone/call.cc',
         '<(libjingle_orig)/source/talk/session/phone/call.h',
-        '<(libjingle_orig)/source/talk/session/phone/channel.cc',
-        '<(libjingle_orig)/source/talk/session/phone/channel.h',
+        '<(libjingle_mods)/source/talk/session/phone/channel.cc',
+        '<(libjingle_mods)/source/talk/session/phone/channel.h',
         '<(libjingle_orig)/source/talk/session/phone/channelmanager.cc',
         '<(libjingle_orig)/source/talk/session/phone/channelmanager.h',
         '<(libjingle_orig)/source/talk/session/phone/codec.cc',
@@ -757,7 +758,7 @@
             '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionmanager_unittest.cc',
             '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionmessage_unittest.cc',
             '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionsignaling_unittest.cc',
-            #'<(libjingle_mods)/source/talk/app/webrtc_dev/webrtcsession_unittest.cc',
+            '<(libjingle_mods)/source/talk/app/webrtc_dev/webrtcsession_unittest.cc',
           ],
         } , {
           'type': 'none',

third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.cc

@@ -216,8 +216,12 @@ void WebRtcSession::OnTransportCandidatesReady(
   if (local_candidates_.size() == kAllowedCandidates)
     return;
   InsertTransportCandidates(candidates);
-  if (local_candidates_.size() == kAllowedCandidates)
-    pc_signaling_->Initialize(candidates);
+  if (local_candidates_.size() == kAllowedCandidates) {
+    pc_signaling_->Initialize(local_candidates_);
+    // TODO(mallinath) - Remove signal when a new interface added for
+    // PC signaling.
+    SignalCandidatesReady(this, local_candidates_);
+  }
 }
 
 void WebRtcSession::OnTransportChannelGone(cricket::Transport* transport) {
@@ -303,12 +307,31 @@ bool WebRtcSession::GetVideoSourceParamInfo(
 
 void WebRtcSession::ProcessLocalMediaChanges(
     const cricket::SessionDescription* sdesc) {
-  //TODO(mallinath) - Handling of local media stream changes in active session
+  // TODO(mallinath) - Handling of local media stream changes in active session
 }
 
 void WebRtcSession::ProcessRemoteMediaChanges(
     const cricket::SessionDescription* sdesc) {
-  //TODO(mallinath) - Handling of remote media stream changes in active session
+  // TODO(mallinath) - Handling of remote media stream changes in active session
+}
+
+void WebRtcSession::SetCaptureDevice(uint32 ssrc,
+                                     VideoCaptureModule* camera) {
+  // should be called from a signaling thread
+  ASSERT(signaling_thread()->IsCurrent());
+  video_channel_->SetCaptureDevice(ssrc, camera);
+}
+
+void WebRtcSession::SetLocalRenderer(uint32 ssrc,
+                                     cricket::VideoRenderer* renderer) {
+  ASSERT(signaling_thread()->IsCurrent());
+  video_channel_->SetLocalRenderer(ssrc, renderer);
+}
+
+void WebRtcSession::SetRemoteRenderer(uint32 ssrc,
+                                      cricket::VideoRenderer* renderer) {
+  ASSERT(signaling_thread()->IsCurrent());
+  video_channel_->SetRenderer(ssrc, renderer);
 }
 
 }  // namespace webrtc

third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession.h

@@ -72,18 +72,21 @@ class WebRtcSession : public cricket::BaseSession,
   // Generic error message callback from WebRtcSession.
   // TODO(mallinath) - It may be necessary to supply error code as well.
   sigslot::signal0<> SignalError;
+  // This signal added for testing. Shouldn't be registered by other
+  // objects.
+  sigslot::signal2<WebRtcSession*,
+                   cricket::Candidates&> SignalCandidatesReady;
 
   void ProcessSessionUpdate(const cricket::SessionDescription* local_desc,
                             const cricket::SessionDescription* remote_desc);
 
  private:
   // Implements MediaProviderInterface.
-  // TODO(mallinath): Add proper implementation.
-  virtual void SetCaptureDevice(uint32 ssrc, VideoCaptureModule* camera) {};
+  virtual void SetCaptureDevice(uint32 ssrc, VideoCaptureModule* camera);
   virtual void SetLocalRenderer(uint32 ssrc,
-                                cricket::VideoRenderer* renderer) {};
+                                cricket::VideoRenderer* renderer);
   virtual void SetRemoteRenderer(uint32 ssrc,
-                                 cricket::VideoRenderer* renderer) {};
+                                 cricket::VideoRenderer* renderer);
 
   // Callback handling from PeerConnectionSignaling
   void OnSignalUpdateSessionDescription(
@@ -95,8 +98,9 @@ class WebRtcSession : public cricket::BaseSession,
   virtual void OnTransportRequestSignaling(cricket::Transport* transport);
   virtual void OnTransportConnecting(cricket::Transport* transport);
   virtual void OnTransportWritable(cricket::Transport* transport);
-  virtual void OnTransportCandidatesReady(cricket::Transport* transport,
-                                  const cricket::Candidates& candidates);
+  virtual void OnTransportCandidatesReady(
+      cricket::Transport* transport,
+      const cricket::Candidates& candidates);
   virtual void OnTransportChannelGone(cricket::Transport* transport);
 
   // Creates channels for voice and video.

third_party_mods/libjingle/source/talk/app/webrtc_dev/webrtcsession_unittest.cc

@@ -32,12 +32,10 @@
 #include "talk/session/phone/channelmanager.h"
 #include "talk/p2p/client/fakeportallocator.h"
 
-class MockPeerConnectionSignaling {
-
-};
-
-class WebRtcSessionTest : public testing::Test {
+class WebRtcSessionTest : public testing::Test,
+                          public sigslot::has_slots<> {
  public:
+  cricket::MediaSessionDescriptionFactory* media_factory_;
   WebRtcSessionTest() {
   }
 
@@ -53,16 +51,26 @@ class WebRtcSessionTest : public testing::Test {
     pc_signaling_.reset(
         new webrtc::PeerConnectionSignaling(channel_manager_.get(),
                                             signaling_thread_));
+    media_factory_ =
+        new cricket::MediaSessionDescriptionFactory(channel_manager_.get());
   }
 
   bool InitializeSession() {
     return session_.get()->Initialize();
   }
+
   bool CheckChannels() {
     return (session_->voice_channel() != NULL &&
             session_->video_channel() != NULL);
   }
 
+  bool CheckTransportChannels() {
+    EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, "rtp") != NULL);
+    EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, "rtcp") != NULL);
+    EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, "video_rtp") != NULL);
+    EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, "video_rtcp") != NULL);
+  }
+
   void Init() {
     ASSERT_TRUE(channel_manager_.get() != NULL);
     ASSERT_TRUE(session_.get() == NULL);
@@ -70,11 +78,38 @@ class WebRtcSessionTest : public testing::Test {
     session_.reset(new webrtc::WebRtcSession(
         channel_manager_.get(), worker_thread_, signaling_thread_,
         port_allocator_.get(), pc_signaling_.get()));
+    session_->SignalCandidatesReady.connect(
+        this, &WebRtcSessionTest::OnCandidatesReady);
     EXPECT_TRUE(InitializeSession());
-    EXPECT_TRUE(CheckChannels());
+  }
+  void OnCandidatesReady(webrtc::WebRtcSession* session,
+                         cricket::Candidates& candidates) {
+    for (cricket::Candidates::iterator iter = candidates.begin();
+         iter != candidates.end(); ++iter) {
+      local_candidates_.push_back(*iter);
+    }
+  }
+  cricket::Candidates& local_candidates() {
+    return local_candidates_;
+  }
+  cricket::SessionDescription* CreateOffer(bool video) {
+    cricket::MediaSessionOptions options;
+    options.is_video = true;
+    // Source params not set
+    cricket::SessionDescription* sdp = media_factory_->CreateOffer(options);
+    return sdp;
+  }
+  cricket::SessionDescription* CreateAnswer(
+      cricket::SessionDescription* offer, bool video) {
+    cricket::MediaSessionOptions options;
+    options.is_video = video;
+    cricket::SessionDescription* sdp =
+        media_factory_->CreateAnswer(offer, options);
   }
 
  private:
+  cricket::Candidates local_candidates_;
+  cricket::Candidates remote_candidates_;
   talk_base::Thread* signaling_thread_;
   talk_base::Thread* worker_thread_;
   talk_base::scoped_ptr<cricket::PortAllocator> port_allocator_;
@@ -85,4 +120,9 @@ class WebRtcSessionTest : public testing::Test {
 
 TEST_F(WebRtcSessionTest, TestInitialize) {
   WebRtcSessionTest::Init();
+  EXPECT_TRUE(CheckChannels());
+  CheckTransportChannels();
+  talk_base::Thread::Current()->ProcessMessages(1000);
+  EXPECT_EQ(4u, local_candidates().size());
 }
+

third_party_mods/libjingle/source/talk/session/phone/channel.h

@@ -0,0 +1,510 @@
+/*
+ * libjingle
+ * Copyright 2004--2007, 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_PHONE_CHANNEL_H_
+#define TALK_SESSION_PHONE_CHANNEL_H_
+
+#include <string>
+#include <vector>
+
+#include "talk/base/asyncudpsocket.h"
+#include "talk/base/criticalsection.h"
+#include "talk/base/network.h"
+#include "talk/base/sigslot.h"
+#include "talk/p2p/client/socketmonitor.h"
+#include "talk/p2p/base/session.h"
+#include "talk/session/phone/audiomonitor.h"
+#include "talk/session/phone/mediachannel.h"
+#include "talk/session/phone/mediaengine.h"
+#include "talk/session/phone/mediamonitor.h"
+#include "talk/session/phone/rtcpmuxfilter.h"
+#include "talk/session/phone/srtpfilter.h"
+
+namespace webrtc {
+class VideoCaptureModule;
+}
+
+namespace cricket {
+
+class MediaContentDescription;
+struct CryptoParams;
+
+enum {
+  MSG_ENABLE = 1,
+  MSG_DISABLE = 2,
+  MSG_MUTE = 3,
+  MSG_UNMUTE = 4,
+  MSG_SETREMOTECONTENT = 5,
+  MSG_SETLOCALCONTENT = 6,
+  MSG_EARLYMEDIATIMEOUT = 8,
+  MSG_PRESSDTMF = 9,
+  MSG_SETRENDERER = 10,
+  MSG_ADDSTREAM = 11,
+  MSG_REMOVESTREAM = 12,
+  MSG_SETRINGBACKTONE = 13,
+  MSG_PLAYRINGBACKTONE = 14,
+  MSG_SETMAXSENDBANDWIDTH = 15,
+  MSG_SETRTCPCNAME = 18,
+  MSG_SENDINTRAFRAME = 19,
+  MSG_REQUESTINTRAFRAME = 20,
+  MSG_RTPPACKET = 22,
+  MSG_RTCPPACKET = 23,
+  MSG_CHANNEL_ERROR = 24,
+  MSG_ENABLECPUADAPTATION = 25,
+  MSG_DISABLECPUADAPTATION = 26,
+  MSG_SCALEVOLUME = 27
+};
+
+// BaseChannel contains logic common to voice and video, including
+// enable/mute, marshaling calls to a worker thread, and
+// connection and media monitors.
+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,
+              TransportChannel* transport_channel);
+  virtual ~BaseChannel();
+
+  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_; }
+  bool secure() const { return srtp_filter_.IsActive(); }
+
+  // Channel control
+  bool SetRtcpCName(const std::string& cname);
+  bool SetLocalContent(const MediaContentDescription* content,
+                       ContentAction action);
+  bool SetRemoteContent(const MediaContentDescription* content,
+                        ContentAction action);
+  bool SetMaxSendBandwidth(int max_bandwidth);
+
+  bool Enable(bool enable);
+  bool Mute(bool mute);
+
+  // Multiplexing
+  bool RemoveStream(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);
+  }
+
+  template <class T>
+  void RegisterSendSink(T* sink,
+                        void (T::*OnPacket)(const void*, size_t, bool)) {
+    talk_base::CritScope cs(&signal_send_packet_cs_);
+    SignalSendPacket.disconnect(sink);
+    SignalSendPacket.connect(sink, OnPacket);
+  }
+
+  void UnregisterSendSink(sigslot::has_slots<>* sink) {
+    talk_base::CritScope cs(&signal_send_packet_cs_);
+    SignalSendPacket.disconnect(sink);
+  }
+
+  bool HasSendSinks() {
+    talk_base::CritScope cs(&signal_send_packet_cs_);
+    return !SignalSendPacket.is_empty();
+  }
+
+  template <class T>
+  void RegisterRecvSink(T* sink,
+                        void (T::*OnPacket)(const void*, size_t, bool)) {
+    talk_base::CritScope cs(&signal_recv_packet_cs_);
+    SignalRecvPacket.disconnect(sink);
+    SignalRecvPacket.connect(sink, OnPacket);
+  }
+
+  void UnregisterRecvSink(sigslot::has_slots<>* sink) {
+    talk_base::CritScope cs(&signal_recv_packet_cs_);
+    SignalRecvPacket.disconnect(sink);
+  }
+
+  bool HasRecvSinks() {
+    talk_base::CritScope cs(&signal_recv_packet_cs_);
+    return !SignalRecvPacket.is_empty();
+  }
+
+ protected:
+  MediaEngineInterface* media_engine() const { return media_engine_; }
+  virtual MediaChannel* media_channel() const { return media_channel_; }
+  void set_rtcp_transport_channel(TransportChannel* transport);
+  bool writable() const { return writable_; }
+  bool was_ever_writable() const { return was_ever_writable_; }
+  bool has_local_content() const { return has_local_content_; }
+  bool has_remote_content() const { return has_remote_content_; }
+  void set_has_local_content(bool has) { has_local_content_ = has; }
+  void set_has_remote_content(bool has) { has_remote_content_ = has; }
+  bool muted() const { return muted_; }
+  talk_base::Thread* signaling_thread() { return session_->signaling_thread(); }
+  SrtpFilter* srtp_filter() { return &srtp_filter_; }
+
+  void Send(uint32 id, talk_base::MessageData *pdata = NULL);
+  void Post(uint32 id, talk_base::MessageData *pdata = NULL);
+  void PostDelayed(int cmsDelay, uint32 id = 0,
+                   talk_base::MessageData *pdata = NULL);
+  void Clear(uint32 id = talk_base::MQID_ANY,
+             talk_base::MessageList* removed = NULL);
+  void FlushRtcpMessages();
+
+  // NetworkInterface implementation, called by MediaEngine
+  virtual bool SendPacket(talk_base::Buffer* packet);
+  virtual bool SendRtcp(talk_base::Buffer* packet);
+  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);
+
+  bool PacketIsRtcp(const TransportChannel* channel, const char* data,
+                    size_t len);
+  bool SendPacket(bool rtcp, talk_base::Buffer* packet);
+  void HandlePacket(bool rtcp, talk_base::Buffer* packet);
+
+  // Setting the send codec based on the remote description.
+  void OnSessionState(BaseSession* session, BaseSession::State state);
+  void OnRemoteDescriptionUpdate(BaseSession* session);
+
+  void EnableMedia_w();
+  void DisableMedia_w();
+  void MuteMedia_w();
+  void UnmuteMedia_w();
+  void ChannelWritable_w();
+  void ChannelNotWritable_w();
+
+  struct StreamMessageData : public talk_base::MessageData {
+    StreamMessageData(uint32 s1, uint32 s2) : ssrc1(s1), ssrc2(s2) {}
+    uint32 ssrc1;
+    uint32 ssrc2;
+  };
+  virtual void RemoveStream_w(uint32 ssrc) = 0;
+
+  virtual void ChangeState() = 0;
+
+  struct SetRtcpCNameData : public talk_base::MessageData {
+    explicit SetRtcpCNameData(const std::string& cname)
+        : cname(cname), result(false) {}
+    std::string cname;
+    bool result;
+  };
+  bool SetRtcpCName_w(const std::string& cname);
+
+  struct SetContentData : public talk_base::MessageData {
+    SetContentData(const MediaContentDescription* content,
+                   ContentAction action)
+        : content(content), action(action), result(false) {}
+    const MediaContentDescription* content;
+    ContentAction action;
+    bool result;
+  };
+
+  // Gets the content appropriate to the channel (audio or video).
+  virtual const MediaContentDescription* GetFirstContent(
+      const SessionDescription* sdesc) = 0;
+  virtual bool SetLocalContent_w(const MediaContentDescription* content,
+                                 ContentAction action) = 0;
+  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
+                                  ContentAction action) = 0;
+
+  bool SetSrtp_w(const std::vector<CryptoParams>& params, ContentAction action,
+                 ContentSource src);
+  bool SetRtcpMux_w(bool enable, ContentAction action, ContentSource src);
+
+  struct SetBandwidthData : public talk_base::MessageData {
+    explicit SetBandwidthData(int value) : value(value), result(false) {}
+    int value;
+    bool result;
+  };
+  bool SetMaxSendBandwidth_w(int max_bandwidth);
+
+  // From MessageHandler
+  virtual void OnMessage(talk_base::Message *pmsg);
+
+  // Handled in derived classes
+  virtual void OnConnectionMonitorUpdate(SocketMonitor *monitor,
+      const std::vector<ConnectionInfo> &infos) = 0;
+
+ private:
+  sigslot::signal3<const void*, size_t, bool> SignalSendPacket;
+  sigslot::signal3<const void*, size_t, bool> SignalRecvPacket;
+  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::string content_name_;
+  TransportChannel *transport_channel_;
+  TransportChannel *rtcp_transport_channel_;
+  SrtpFilter srtp_filter_;
+  RtcpMuxFilter rtcp_mux_filter_;
+  talk_base::scoped_ptr<SocketMonitor> socket_monitor_;
+  bool enabled_;
+  bool writable_;
+  bool was_ever_writable_;
+  bool has_local_content_;
+  bool has_remote_content_;
+  bool muted_;
+};
+
+// 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();
+
+  // downcasts a MediaChannel
+  virtual VoiceMediaChannel* media_channel() const {
+    return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
+  }
+
+  // Add an incoming stream with the specified SSRC.
+  bool AddStream(uint32 ssrc);
+
+  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);
+  bool PressDTMF(int digit, bool playout);
+  bool SetOutputScaling(uint32 ssrc, double left, double right);
+
+  // 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;
+
+  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;
+
+ private:
+  struct SetRingbackToneMessageData : public talk_base::MessageData {
+    SetRingbackToneMessageData(const void* b, int l)
+        : buf(b),
+          len(l),
+          result(false) {
+    }
+    const void* buf;
+    int len;
+    bool result;
+  };
+  struct PlayRingbackToneMessageData : public talk_base::MessageData {
+    PlayRingbackToneMessageData(uint32 s, bool p, bool l)
+        : ssrc(s),
+          play(p),
+          loop(l),
+          result(false) {
+    }
+    uint32 ssrc;
+    bool play;
+    bool loop;
+    bool result;
+  };
+  struct DtmfMessageData : public talk_base::MessageData {
+    DtmfMessageData(int d, bool p)
+        : digit(d),
+          playout(p),
+          result(false) {
+    }
+    int digit;
+    bool playout;
+    bool result;
+  };
+  struct ScaleVolumeMessageData : public talk_base::MessageData {
+    ScaleVolumeMessageData(uint32 s, double l, double r)
+        : ssrc(s),
+          left(l),
+          right(r),
+          result(false) {
+    }
+    uint32 ssrc;
+    double left;
+    double right;
+    bool result;
+  };
+
+  // overrides from BaseChannel
+  virtual void OnChannelRead(TransportChannel* channel,
+                             const char *data, size_t len);
+  virtual void ChangeState();
+  virtual const MediaContentDescription* GetFirstContent(
+      const SessionDescription* sdesc);
+  virtual bool SetLocalContent_w(const MediaContentDescription* content,
+                                 ContentAction action);
+  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
+                                  ContentAction action);
+
+  void AddStream_w(uint32 ssrc);
+  void RemoveStream_w(uint32 ssrc);
+
+  bool SetRingbackTone_w(const void* buf, int len);
+  bool PlayRingbackTone_w(uint32 ssrc, bool play, bool loop);
+  void HandleEarlyMediaTimeout();
+  bool PressDTMF_w(int digit, bool playout);
+  bool SetOutputScaling_w(uint32 ssrc, double left, double right);
+
+  virtual void OnMessage(talk_base::Message *pmsg);
+  virtual void OnConnectionMonitorUpdate(
+      SocketMonitor *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_;
+  talk_base::scoped_ptr<VoiceMediaMonitor> media_monitor_;
+  talk_base::scoped_ptr<AudioMonitor> audio_monitor_;
+};
+
+// VideoChannel is a specialization for video.
+class VideoChannel : public BaseChannel {
+ public:
+  VideoChannel(talk_base::Thread *thread, MediaEngineInterface *media_engine,
+               VideoMediaChannel *channel, BaseSession *session,
+               const std::string& content_name, bool rtcp,
+               VoiceChannel *voice_channel);
+  ~VideoChannel();
+
+  // downcasts a MediaChannel
+  virtual VideoMediaChannel* media_channel() const {
+    return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
+  }
+
+  // Add an incoming stream with the specified SSRC.
+  bool AddStream(uint32 ssrc, uint32 voice_ssrc);
+
+  bool SetRenderer(uint32 ssrc, VideoRenderer* renderer);
+
+
+  sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo> &>
+      SignalConnectionMonitor;
+
+  void StartMediaMonitor(int cms);
+  void StopMediaMonitor();
+  sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
+
+  bool SendIntraFrame();
+  bool RequestIntraFrame();
+  void EnableCpuAdaptation(bool enable);
+
+  sigslot::signal3<VideoChannel*, uint32, VideoMediaChannel::Error>
+      SignalMediaError;
+
+  void SetCaptureDevice(uint32 ssrc, webrtc::VideoCaptureModule* camera);
+  void SetLocalRenderer(uint32 ssrc, VideoRenderer* renderer);
+
+ private:
+  // overrides from BaseChannel
+  virtual void ChangeState();
+  virtual const MediaContentDescription* GetFirstContent(
+      const SessionDescription* sdesc);
+  virtual bool SetLocalContent_w(const MediaContentDescription* content,
+                                 ContentAction action);
+  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
+                                  ContentAction action);
+
+  void AddStream_w(uint32 ssrc, uint32 voice_ssrc);
+  void RemoveStream_w(uint32 ssrc);
+
+  void SendIntraFrame_w() {
+    media_channel()->SendIntraFrame();
+  }
+  void RequestIntraFrame_w() {
+    media_channel()->RequestIntraFrame();
+  }
+  void EnableCpuAdaptation_w(bool enable) {
+    // TODO: The following call will clear all other options, which is
+    // OK now since SetOptions is not used in video media channel. In the
+    // future, add GetOptions() method and change the options.
+    media_channel()->SetOptions(enable ? OPT_CPU_ADAPTATION : 0);
+  }
+
+  struct RenderMessageData : public talk_base::MessageData {
+    RenderMessageData(uint32 s, VideoRenderer* r) : ssrc(s), renderer(r) {}
+    uint32 ssrc;
+    VideoRenderer* renderer;
+  };
+
+
+  void SetRenderer_w(uint32 ssrc, VideoRenderer* renderer);
+
+
+  virtual void OnMessage(talk_base::Message *pmsg);
+  virtual void OnConnectionMonitorUpdate(
+      SocketMonitor *monitor, const std::vector<ConnectionInfo> &infos);
+  virtual void OnMediaMonitorUpdate(
+      VideoMediaChannel *media_channel, const VideoMediaInfo& info);
+  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<VideoMediaMonitor> media_monitor_;
+};
+
+}  // namespace cricket
+
+#endif  // TALK_SESSION_PHONE_CHANNEL_H_