Move talk/app/webrtc to webrtc/api

talk/app/webrtc/mediastreamprovider.h

Index: talk/app/webrtc/mediastreamprovider.h
diff --git a/talk/app/webrtc/mediastreamprovider.h b/talk/app/webrtc/mediastreamprovider.h
deleted file mode 100644
index 103b3f36d4653b0e1990e14e00805a2831dc4715..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/mediastreamprovider.h
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- * 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,
- *     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_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
-#define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
-
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/media/base/videosinkinterface.h"
-
-namespace cricket {
-
-class AudioRenderer;
-class VideoCapturer;
-class VideoFrame;
-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,
-      rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 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_