Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/webrtcsession.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_API_WEBRTCSESSION_H_
 #define WEBRTC_API_WEBRTCSESSION_H_
 
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/datachannel.h"
 #include "webrtc/api/dtmfsender.h"
 #include "webrtc/api/mediacontroller.h"
-#include "webrtc/api/mediastreamprovider.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/pc/mediasession.h"
@@ skipping 75 matching lines @@
   TransportStatsMap transport_stats;
 };
 
 // A WebRtcSession manages general session state. This includes negotiation
 // of both the application-level and network-level protocols:  the former
 // defines what will be sent and the latter defines how it will be sent.  Each
 // network-level protocol is represented by a Transport object.  Each Transport
 // participates in the network-level negotiation.  The individual streams of
 // packets are represented by TransportChannels.  The application-level protocol
 // is represented by SessionDecription objects.
-class WebRtcSession : public AudioProviderInterface,
-                      public VideoProviderInterface,
-                      public DtmfProviderInterface,
-                      public DataChannelProviderInterface,
-                      public sigslot::has_slots<> {
+class WebRtcSession :
+
+    public DtmfProviderInterface,
+    public DataChannelProviderInterface,
+    public sigslot::has_slots<> {
  public:
   enum State {
     STATE_INIT = 0,
     STATE_SENTOFFER,         // Sent offer, waiting for answer.
     STATE_RECEIVEDOFFER,     // Received an offer. Need to send answer.
     STATE_SENTPRANSWER,      // Sent provisional answer. Need to send answer.
     STATE_RECEIVEDPRANSWER,  // Received provisional answer, waiting for answer.
     STATE_INPROGRESS,        // Offer/answer exchange completed.
     STATE_CLOSED,            // Close() was called.
   };
@@ skipping 94 matching lines @@
     return local_desc_.get();
   }
   const SessionDescriptionInterface* remote_description() const {
     return remote_desc_.get();
   }
 
   // Get the id used as a media stream track's "id" field from ssrc.
   virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
   virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
 
-  // AudioMediaProviderInterface implementation.
-  void SetAudioPlayout(uint32_t ssrc, bool enable) override;
-  void SetAudioSend(uint32_t ssrc,
-                    bool enable,
-                    const cricket::AudioOptions& options,
-                    cricket::AudioSource* source) override;
-  void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
-  void SetRawAudioSink(uint32_t ssrc,
-                       std::unique_ptr<AudioSinkInterface> sink) override;
-
-  RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const override;
-  bool SetAudioRtpSendParameters(uint32_t ssrc,
-                                 const RtpParameters& parameters) override;
-  RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const override;
-  bool SetAudioRtpReceiveParameters(uint32_t ssrc,
-                                    const RtpParameters& parameters) override;
-
-  // Implements VideoMediaProviderInterface.
-  void SetVideoPlayout(
-      uint32_t ssrc,
-      bool enable,
-      rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
-  void SetVideoSend(
-      uint32_t ssrc,
-      bool enable,
-      const cricket::VideoOptions* options,
-      rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
-
-  RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const override;
-  bool SetVideoRtpSendParameters(uint32_t ssrc,
-                                 const RtpParameters& parameters) override;
-  RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const override;
-  bool SetVideoRtpReceiveParameters(uint32_t ssrc,
-                                    const RtpParameters& parameters) override;
-
   // Implements DtmfProviderInterface.
   bool CanInsertDtmf(const std::string& track_id) override;
   bool InsertDtmf(const std::string& track_id,
                   int code, int duration) override;
   sigslot::signal0<>* GetOnDestroyedSignal() override;
 
   // Implements DataChannelProviderInterface.
   bool SendData(const cricket::SendDataParams& params,
                 const rtc::CopyOnWriteBuffer& payload,
                 cricket::SendDataResult* result) override;
@@ skipping 21 matching lines @@
 
   cricket::DataChannelType data_channel_type() const;
 
   bool IceRestartPending(const std::string& content_name) const;
 
   // Called when an RTCCertificate is generated or retrieved by
   // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
   void OnCertificateReady(
       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
   void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
-  // Called when the channel received the first packet.
-  void OnChannelFirstPacketReceived(cricket::BaseChannel*);
 
   // For unit test.
   bool waiting_for_certificate_for_testing() const;
   const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
 
   void set_metrics_observer(
       webrtc::MetricsObserverInterface* metrics_observer) {
     metrics_observer_ = metrics_observer;
   }
 
@@ skipping 205 matching lines @@
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   bool received_first_video_packet_ = false;
   bool received_first_audio_packet_ = false;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_WEBRTCSESSION_H_