Move talk/app/webrtc to webrtc/api

talk/app/webrtc/mediastreaminterface.h

Index: talk/app/webrtc/mediastreaminterface.h
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
deleted file mode 100644
index 9b137d9f761006bc45477b455fc58b91236a7069..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/mediastreaminterface.h
+++ /dev/null
@@ -1,276 +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.
- */
-
-// This file contains interfaces for MediaStream, MediaTrack and MediaSource.
-// These interfaces are used for implementing MediaStream and MediaTrack as
-// defined in http://dev.w3.org/2011/webrtc/editor/webrtc.html#stream-api. These
-// interfaces must be used only with PeerConnection. PeerConnectionManager
-// interface provides the factory methods to create MediaStream and MediaTracks.
-
-#ifndef TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_
-#define TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_
-
-#include <string>
-#include <vector>
-
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-
-namespace cricket {
-
-class AudioRenderer;
-class VideoCapturer;
-class VideoRenderer;
-class VideoFrame;
-
-}  // namespace cricket
-
-namespace webrtc {
-
-// Generic observer interface.
-class ObserverInterface {
- public:
-  virtual void OnChanged() = 0;
-
- protected:
-  virtual ~ObserverInterface() {}
-};
-
-class NotifierInterface {
- public:
-  virtual void RegisterObserver(ObserverInterface* observer) = 0;
-  virtual void UnregisterObserver(ObserverInterface* observer) = 0;
-
-  virtual ~NotifierInterface() {}
-};
-
-// Base class for sources. A MediaStreamTrack have an underlying source that
-// provide media. A source can be shared with multiple tracks.
-class MediaSourceInterface : public rtc::RefCountInterface,
-                             public NotifierInterface {
- public:
-  enum SourceState {
-    kInitializing,
-    kLive,
-    kEnded,
-    kMuted
-  };
-
-  virtual SourceState state() const = 0;
-
-  virtual bool remote() const = 0;
-
- protected:
-  virtual ~MediaSourceInterface() {}
-};
-
-// Information about a track.
-class MediaStreamTrackInterface : public rtc::RefCountInterface,
-                                  public NotifierInterface {
- public:
-  enum TrackState {
-    kInitializing,  // Track is beeing negotiated.
-    kLive = 1,  // Track alive
-    kEnded = 2,  // Track have ended
-    kFailed = 3,  // Track negotiation failed.
-  };
-
-  static const char kAudioKind[];
-  static const char kVideoKind[];
-
-  virtual std::string kind() const = 0;
-  virtual std::string id() const = 0;
-  virtual bool enabled() const = 0;
-  virtual TrackState state() const = 0;
-  virtual bool set_enabled(bool enable) = 0;
-  // These methods should be called by implementation only.
-  virtual bool set_state(TrackState new_state) = 0;
-
- protected:
-  virtual ~MediaStreamTrackInterface() {}
-};
-
-// Interface for rendering VideoFrames from a VideoTrack
-class VideoRendererInterface {
- public:
-  // |frame| may have pending rotation. For clients which can't apply rotation,
-  // |frame|->GetCopyWithRotationApplied() will return a frame that has the
-  // rotation applied.
-  virtual void RenderFrame(const cricket::VideoFrame* frame) = 0;
-
- protected:
-  // The destructor is protected to prevent deletion via the interface.
-  // This is so that we allow reference counted classes, where the destructor
-  // should never be public, to implement the interface.
-  virtual ~VideoRendererInterface() {}
-};
-
-class VideoSourceInterface;
-
-class VideoTrackInterface : public MediaStreamTrackInterface {
- public:
-  // Register a renderer that will render all frames received on this track.
-  virtual void AddRenderer(VideoRendererInterface* renderer) = 0;
-  // Deregister a renderer.
-  virtual void RemoveRenderer(VideoRendererInterface* renderer) = 0;
-
-  virtual VideoSourceInterface* GetSource() const = 0;
-
- protected:
-  virtual ~VideoTrackInterface() {}
-};
-
-// Interface for receiving audio data from a AudioTrack.
-class AudioTrackSinkInterface {
- public:
-  virtual void OnData(const void* audio_data,
-                      int bits_per_sample,
-                      int sample_rate,
-                      size_t number_of_channels,
-                      size_t number_of_frames) = 0;
-
- protected:
-  virtual ~AudioTrackSinkInterface() {}
-};
-
-// AudioSourceInterface is a reference counted source used for AudioTracks.
-// The same source can be used in multiple AudioTracks.
-class AudioSourceInterface : public MediaSourceInterface {
- public:
-  class AudioObserver {
-   public:
-    virtual void OnSetVolume(double volume) = 0;
-
-   protected:
-    virtual ~AudioObserver() {}
-  };
-
-  // TODO(xians): Makes all the interface pure virtual after Chrome has their
-  // implementations.
-  // Sets the volume to the source. |volume| is in  the range of [0, 10].
-  // TODO(tommi): This method should be on the track and ideally volume should
-  // be applied in the track in a way that does not affect clones of the track.
-  virtual void SetVolume(double volume) {}
-
-  // Registers/unregisters observer to the audio source.
-  virtual void RegisterAudioObserver(AudioObserver* observer) {}
-  virtual void UnregisterAudioObserver(AudioObserver* observer) {}
-
-  // TODO(tommi): Make pure virtual.
-  virtual void AddSink(AudioTrackSinkInterface* sink) {}
-  virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
-};
-
-// Interface of the audio processor used by the audio track to collect
-// statistics.
-class AudioProcessorInterface : public rtc::RefCountInterface {
- public:
-  struct AudioProcessorStats {
-    AudioProcessorStats() : typing_noise_detected(false),
-                            echo_return_loss(0),
-                            echo_return_loss_enhancement(0),
-                            echo_delay_median_ms(0),
-                            aec_quality_min(0.0),
-                            echo_delay_std_ms(0) {}
-    ~AudioProcessorStats() {}
-
-    bool typing_noise_detected;
-    int echo_return_loss;
-    int echo_return_loss_enhancement;
-    int echo_delay_median_ms;
-    float aec_quality_min;
-    int echo_delay_std_ms;
-  };
-
-  // Get audio processor statistics.
-  virtual void GetStats(AudioProcessorStats* stats) = 0;
-
- protected:
-  virtual ~AudioProcessorInterface() {}
-};
-
-class AudioTrackInterface : public MediaStreamTrackInterface {
- public:
-  // TODO(xians): Figure out if the following interface should be const or not.
-  virtual AudioSourceInterface* GetSource() const =  0;
-
-  // Add/Remove a sink that will receive the audio data from the track.
-  virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
-  virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
-
-  // Get the signal level from the audio track.
-  // Return true on success, otherwise false.
-  // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
-  // after Chrome has the correct implementation of the interface.
-  virtual bool GetSignalLevel(int* level) { return false; }
-
-  // Get the audio processor used by the audio track. Return NULL if the track
-  // does not have any processor.
-  // TODO(xians): Make the interface pure virtual.
-  virtual rtc::scoped_refptr<AudioProcessorInterface>
-      GetAudioProcessor() { return NULL; }
-
-  // Get a pointer to the audio renderer of this AudioTrack.
-  // The pointer is valid for the lifetime of this AudioTrack.
-  // TODO(xians): Remove the following interface after Chrome switches to
-  // AddSink() and RemoveSink() interfaces.
-  virtual cricket::AudioRenderer* GetRenderer() { return NULL; }
-
- protected:
-  virtual ~AudioTrackInterface() {}
-};
-
-typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
-    AudioTrackVector;
-typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
-    VideoTrackVector;
-
-class MediaStreamInterface : public rtc::RefCountInterface,
-                             public NotifierInterface {
- public:
-  virtual std::string label() const = 0;
-
-  virtual AudioTrackVector GetAudioTracks() = 0;
-  virtual VideoTrackVector GetVideoTracks() = 0;
-  virtual rtc::scoped_refptr<AudioTrackInterface>
-      FindAudioTrack(const std::string& track_id) = 0;
-  virtual rtc::scoped_refptr<VideoTrackInterface>
-      FindVideoTrack(const std::string& track_id) = 0;
-
-  virtual bool AddTrack(AudioTrackInterface* track) = 0;
-  virtual bool AddTrack(VideoTrackInterface* track) = 0;
-  virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
-  virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
-
- protected:
-  virtual ~MediaStreamInterface() {}
-};
-
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_