Index: talk/app/webrtc/mediastreaminterface.h |
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h |
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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" |
-#include "webrtc/media/base/videosinkinterface.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: |
- // 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; |
- |
- // 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 // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_ |