| /* |
| * 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 API_MEDIASTREAMINTERFACE_H_ |
| #define API_MEDIASTREAMINTERFACE_H_ |
| |
| #include <stddef.h> |
| |
| #include <string> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/audio_options.h" |
| #include "api/video/video_frame.h" |
| #include "api/video/video_sink_interface.h" |
| #include "api/video/video_source_interface.h" |
| #include "modules/audio_processing/include/audio_processing_statistics.h" |
| #include "rtc_base/refcount.h" |
| #include "rtc_base/scoped_ref_ptr.h" |
| |
| 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 has an underlying source that |
| // provides media. A source can be shared by 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: |
| ~MediaSourceInterface() override = default; |
| }; |
| |
| // C++ version of MediaStreamTrack. |
| // See: https://www.w3.org/TR/mediacapture-streams/#mediastreamtrack |
| 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; |
| |
| // Track identifier. |
| virtual std::string id() const = 0; |
| |
| // A disabled track will produce silence (if audio) or black frames (if |
| // video). Can be disabled and re-enabled. |
| virtual bool enabled() const = 0; |
| virtual bool set_enabled(bool enable) = 0; |
| |
| // Live or ended. A track will never be live again after becoming ended. |
| virtual TrackState state() const = 0; |
| |
| protected: |
| ~MediaStreamTrackInterface() override = default; |
| }; |
| |
| // VideoTrackSourceInterface is a reference counted source used for |
| // VideoTracks. The same source can be used by multiple VideoTracks. |
| // VideoTrackSourceInterface is designed to be invoked on the signaling thread |
| // except for rtc::VideoSourceInterface<VideoFrame> methods that will be invoked |
| // on the worker thread via a VideoTrack. A custom implementation of a source |
| // can inherit AdaptedVideoTrackSource instead of directly implementing this |
| // interface. |
| class VideoTrackSourceInterface : public MediaSourceInterface, |
| public rtc::VideoSourceInterface<VideoFrame> { |
| public: |
| struct Stats { |
| // Original size of captured frame, before video adaptation. |
| int input_width; |
| int input_height; |
| }; |
| |
| // 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 don't 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 absl::optional<bool> 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. |
| // |
| // Implementation should avoid blocking. |
| virtual bool GetStats(Stats* stats) = 0; |
| |
| protected: |
| ~VideoTrackSourceInterface() override = default; |
| }; |
| |
| // VideoTrackInterface is designed to be invoked on the signaling thread except |
| // for rtc::VideoSourceInterface<VideoFrame> methods that must be invoked |
| // on the worker thread. |
| // PeerConnectionFactory::CreateVideoTrack can be used for creating a VideoTrack |
| // that ensures thread safety and that all methods are called on the right |
| // thread. |
| class VideoTrackInterface : public MediaStreamTrackInterface, |
| public rtc::VideoSourceInterface<VideoFrame> { |
| public: |
| // Video track content hint, used to override the source is_screencast |
| // property. |
| // See https://crbug.com/653531 and https://w3c.github.io/mst-content-hint. |
| enum class ContentHint { kNone, kFluid, kDetailed, kText }; |
| |
| // Register a video sink for this track. Used to connect the track to the |
| // underlying video engine. |
| void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink, |
| const rtc::VideoSinkWants& wants) override {} |
| void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {} |
| |
| virtual VideoTrackSourceInterface* GetSource() const = 0; |
| |
| virtual ContentHint content_hint() const; |
| virtual void set_content_hint(ContentHint hint) {} |
| |
| protected: |
| ~VideoTrackInterface() override = default; |
| }; |
| |
| // 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 by multiple AudioTracks. |
| class AudioSourceInterface : public MediaSourceInterface { |
| public: |
| class AudioObserver { |
| public: |
| virtual void OnSetVolume(double volume) = 0; |
| |
| protected: |
| virtual ~AudioObserver() {} |
| }; |
| |
| // TODO(deadbeef): Makes all the interfaces pure virtual after they're |
| // implemented in chromium. |
| |
| // Sets the volume of 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 observers 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) {} |
| |
| // Returns options for the AudioSource. |
| // (for some of the settings this approach is broken, e.g. setting |
| // audio network adaptation on the source is the wrong layer of abstraction). |
| virtual const cricket::AudioOptions options() const; |
| }; |
| |
| // Interface of the audio processor used by the audio track to collect |
| // statistics. |
| class AudioProcessorInterface : public rtc::RefCountInterface { |
| public: |
| struct AudioProcessorStatistics { |
| bool typing_noise_detected = false; |
| AudioProcessingStats apm_statistics; |
| }; |
| |
| // Get audio processor statistics. The |has_remote_tracks| argument should be |
| // set if there are active remote tracks (this would usually be true during |
| // a call). If there are no remote tracks some of the stats will not be set by |
| // the AudioProcessor, because they only make sense if there is at least one |
| // remote track. |
| virtual AudioProcessorStatistics GetStats(bool has_remote_tracks) = 0; |
| |
| protected: |
| ~AudioProcessorInterface() override = default; |
| }; |
| |
| class AudioTrackInterface : public MediaStreamTrackInterface { |
| public: |
| // TODO(deadbeef): 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(deadbeef): Change the interface to int GetSignalLevel() and pure |
| // virtual after it's implemented in chromium. |
| virtual bool GetSignalLevel(int* level); |
| |
| // Get the audio processor used by the audio track. Return null if the track |
| // does not have any processor. |
| // TODO(deadbeef): Make the interface pure virtual. |
| virtual rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor(); |
| |
| protected: |
| ~AudioTrackInterface() override = default; |
| }; |
| |
| typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> > AudioTrackVector; |
| typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> > VideoTrackVector; |
| |
| // C++ version of https://www.w3.org/TR/mediacapture-streams/#mediastream. |
| // |
| // A major difference is that remote audio/video tracks (received by a |
| // PeerConnection/RtpReceiver) are not synchronized simply by adding them to |
| // the same stream; a session description with the correct "a=msid" attributes |
| // must be pushed down. |
| // |
| // Thus, this interface acts as simply a container for tracks. |
| class MediaStreamInterface : public rtc::RefCountInterface, |
| public NotifierInterface { |
| public: |
| virtual std::string id() 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: |
| ~MediaStreamInterface() override = default; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_MEDIASTREAMINTERFACE_H_ |