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/*
* 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.
// TODO(perkj): Implement sources for local and remote audio tracks and
// remote video tracks.
class MediaSourceInterface : public rtc::RefCountInterface,
public NotifierInterface {
public:
enum SourceState {
kInitializing,
kLive,
kEnded,
kMuted
};
virtual SourceState state() 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.
};
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:
// TODO(guoweis): Remove this function. Obsolete. The implementation of
// VideoRendererInterface should be able to handle different frame size as
// well as pending rotation. If it can't apply the frame rotation by itself,
// it should call |frame|.GetCopyWithRotationApplied() to get a frame that has
// the rotation applied.
virtual void SetSize(int width, int height) {}
// |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() {}
};
// 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].
virtual void SetVolume(double volume) {}
// Registers/unregisters observer to the audio source.
virtual void RegisterAudioObserver(AudioObserver* observer) {}
virtual void UnregisterAudioObserver(AudioObserver* observer) {}
};
// 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,
int number_of_channels,
size_t number_of_frames) = 0;
protected:
virtual ~AudioTrackSinkInterface() {}
};
// 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_