Support for remote audio into tracks

talk/app/webrtc/mediastreamprovider.h

 /*
  * libjingle
  * Copyright 2012 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,
@@ skipping 11 matching lines @@
  * 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_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace cricket {
 
 class AudioRenderer;
 class VideoCapturer;
 class VideoRenderer;
 struct AudioOptions;
 struct VideoOptions;
 
 }  // namespace cricket
 
 namespace webrtc {
 
+class AudioSinkInterface;
+
 // TODO(deadbeef): Change the key from an ssrc to a "sender_id" or
 // "receiver_id" string, which will be the MSID in the short term and MID in
 // the long term.
 
 // TODO(deadbeef): These interfaces are effectively just a way for the
 // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
 // refactored away eventually, as the classes converge.
 
 // This interface is called by AudioRtpSender/Receivers to change the settings
 // of an audio track connected to certain PeerConnection.
 class AudioProviderInterface {
  public:
   // Enable/disable the audio playout of a remote audio track with |ssrc|.
   virtual void SetAudioPlayout(uint32_t ssrc, bool enable) = 0;
   // Enable/disable sending audio on the local audio track with |ssrc|.
   // When |enable| is true |options| should be applied to the audio track.
   virtual void SetAudioSend(uint32_t ssrc,
                             bool enable,
                             const cricket::AudioOptions& options,
                             cricket::AudioRenderer* renderer) = 0;
 
   // Sets the audio playout volume of a remote audio track with |ssrc|.
   // |volume| is in the range of [0, 10].
   virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0;
 
+  // Allows for setting a direct audio sink for an incoming audio source.
+  // Only one audio sink is supported per ssrc and ownership of the sink is
+  // passed to the provider.
+  virtual void SetRawAudioSink(
+      uint32_t ssrc,
+      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
+
  protected:
   virtual ~AudioProviderInterface() {}
 };
 
 // This interface is called by VideoRtpSender/Receivers to change the settings
 // of a video track connected to a certain PeerConnection.
 class VideoProviderInterface {
  public:
   virtual bool SetCaptureDevice(uint32_t ssrc,
                                 cricket::VideoCapturer* camera) = 0;
   // Enable/disable the video playout of a remote video track with |ssrc|.
   virtual void SetVideoPlayout(uint32_t ssrc,
                                bool enable,
                                cricket::VideoRenderer* renderer) = 0;
   // Enable sending video on the local video track with |ssrc|.
   virtual void SetVideoSend(uint32_t ssrc,
                             bool enable,
                             const cricket::VideoOptions* options) = 0;
 
  protected:
   virtual ~VideoProviderInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_