/* * 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 #include #include "webrtc/base/basictypes.h" #include "webrtc/base/refcount.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/base/optional.h" #include "webrtc/media/base/mediachannel.h" #include "webrtc/media/base/videosinkinterface.h" #include "webrtc/media/base/videosourceinterface.h" namespace cricket { class AudioRenderer; 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 { kLive, kEnded, }; static const char kAudioKind[]; static const char kVideoKind[]; // The kind() method must return kAudioKind only if the object is a // subclass of AudioTrackInterface, and kVideoKind only if the // object is a subclass of VideoTrackInterface. It is typically used // to protect a static_cast<> to the corresponding subclass. 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; protected: virtual ~MediaStreamTrackInterface() {} }; // VideoTrackSourceInterface is a reference counted source used for VideoTracks. // The same source can be used in multiple VideoTracks. class VideoTrackSourceInterface : public MediaSourceInterface, public rtc::VideoSourceInterface { public: struct Stats { // Original size of captured frame, before video adaptation. int input_width; int input_height; }; virtual void Stop() = 0; virtual void Restart() = 0; // Indicates that parameters suitable for screencasts should be automatically // applied to RtpSenders. // TODO(perkj): Remove these once all known applications have moved to // explicitly setting suitable parameters for screencasts and dont' need this // implicit behavior. virtual bool is_screencast() const = 0; // Indicates that the encoder should denoise video before encoding it. // If it is not set, the default configuration is used which is different // depending on video codec. // TODO(perkj): Remove this once denoising is done by the source, and not by // the encoder. virtual rtc::Optional needs_denoising() const = 0; // Returns false if no stats are available, e.g, for a remote // source, or a source which has not seen its first frame yet. // Should avoid blocking. virtual bool GetStats(Stats* stats) = 0; protected: virtual ~VideoTrackSourceInterface() {} }; class VideoTrackInterface : public MediaStreamTrackInterface, public rtc::VideoSourceInterface { public: // Register a video sink for this track. void AddOrUpdateSink(rtc::VideoSinkInterface* sink, const rtc::VideoSinkWants& wants) override{}; void RemoveSink( rtc::VideoSinkInterface* sink) override{}; virtual VideoTrackSourceInterface* 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), aec_divergent_filter_fraction(0.0) {} ~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; float aec_divergent_filter_fraction; }; // 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 GetAudioProcessor() { return NULL; } protected: virtual ~AudioTrackInterface() {} }; typedef std::vector > AudioTrackVector; typedef std::vector > 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 FindAudioTrack(const std::string& track_id) = 0; virtual rtc::scoped_refptr 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_