webrtc/api/mediastreaminterface.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.
 */

// 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 WEBRTC_API_MEDIASTREAMINTERFACE_H_
#define WEBRTC_API_MEDIASTREAMINTERFACE_H_

#include <string>
#include <vector>

#include "webrtc/base/basictypes.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.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 rtc::VideoSinkInterface<cricket::VideoFrame> {
 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;
  // Intended to replace RenderFrame.
  void OnFrame(const cricket::VideoFrame& frame) override {
    RenderFrame(&frame);
  }

 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 rtc::VideoSourceInterface<cricket::VideoFrame> {
 public:
  // Make an unqualified VideoSourceInterface resolve to
  // webrtc::VideoSourceInterface, not our base class
  // rtc::VideoSourceInterface<cricket::VideoFrame>.
  using VideoSourceInterface = webrtc::VideoSourceInterface;

  // AddRenderer and RemoveRenderer are for backwards compatibility
  // only. They are obsoleted by the methods of
  // rtc::VideoSourceInterface.
  virtual void AddRenderer(VideoRendererInterface* renderer) {
    AddOrUpdateSink(renderer, rtc::VideoSinkWants());
  }
  virtual void RemoveRenderer(VideoRendererInterface* renderer) {
    RemoveSink(renderer);
  }

  // Register a video sink for this track.
  void AddOrUpdateSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink,
                       const rtc::VideoSinkWants& wants) override{};
  void RemoveSink(
      rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override{};

  virtual VideoSourceInterface* GetSource() const = 0;

  // Return the track input sink. I.e., frames sent to this sink are
  // propagated to all renderers registered with the track. The
  // returned sink must not change between calls. Currently, this
  // method is used for remote tracks (VideoRtpReceiver); further
  // refactoring is planned for this path, it's unclear if this method
  // belongs here long term.

  // We do this instead of simply implementing the
  // VideoSourceInterface directly, because if we did the latter, we'd
  // need an OnFrame method in VideoTrackProxy, with a thread jump on
  // each call.

  // TODO(nisse): It has a default implementation so that mock
  // objects, in particular, chrome's MockWebRtcVideoTrack, doesn't
  // need to know about it. Consider removing the implementation (or
  // this comment) after refactoring dust settles.
  virtual rtc::VideoSinkInterface<cricket::VideoFrame>* GetSink() {
    return nullptr;
  };

 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; }

 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  // WEBRTC_API_MEDIASTREAMINTERFACE_H_