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TMessagesProj/jni/voip/webrtc/api/media_stream_interface.h

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+/*
+ *  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.
+
+#ifndef API_MEDIA_STREAM_INTERFACE_H_
+#define API_MEDIA_STREAM_INTERFACE_H_
+
+#include <stddef.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_options.h"
+#include "api/ref_count.h"
+#include "api/scoped_refptr.h"
+#include "api/video/recordable_encoded_frame.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+#include "api/video_track_source_constraints.h"
+#include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "rtc_base/system/rtc_export.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 RTC_EXPORT MediaSourceInterface : public webrtc::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 RTC_EXPORT MediaStreamTrackInterface : public webrtc::RefCountInterface,
+                                             public NotifierInterface {
+ public:
+  enum TrackState {
+    kLive,
+    kEnded,
+  };
+
+  static const char* const kAudioKind;
+  static const char* const 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;
+
+  // Returns true if encoded output can be enabled in the source.
+  virtual bool SupportsEncodedOutput() const = 0;
+
+  // Reliably cause a key frame to be generated in encoded output.
+  // TODO(bugs.webrtc.org/11115): find optimal naming.
+  virtual void GenerateKeyFrame() = 0;
+
+  // Add an encoded video sink to the source and additionally cause
+  // a key frame to be generated from the source. The sink will be
+  // invoked from a decoder queue.
+  virtual void AddEncodedSink(
+      rtc::VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;
+
+  // Removes an encoded video sink from the source.
+  virtual void RemoveEncodedSink(
+      rtc::VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;
+
+  // Notify about constraints set on the source. The information eventually gets
+  // routed to attached sinks via VideoSinkInterface<>::OnConstraintsChanged.
+  // The call is expected to happen on the network thread.
+  // TODO(crbug/1255737): make pure virtual once downstream project adapts.
+  virtual void ProcessConstraints(
+      const webrtc::VideoTrackSourceConstraints& constraints) {}
+
+ 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 RTC_EXPORT 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) {
+    RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
+  }
+
+  // In this method, `absolute_capture_timestamp_ms`, when available, is
+  // supposed to deliver the timestamp when this audio frame was originally
+  // captured. This timestamp MUST be based on the same clock as
+  // rtc::TimeMillis().
+  virtual void OnData(const void* audio_data,
+                      int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      absl::optional<int64_t> absolute_capture_timestamp_ms) {
+    // TODO(bugs.webrtc.org/10739): Deprecate the old OnData and make this one
+    // pure virtual.
+    return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
+                  number_of_frames);
+  }
+
+  // Returns the number of channels encoded by the sink. This can be less than
+  // the number_of_channels if down-mixing occur. A value of -1 means an unknown
+  // number.
+  virtual int NumPreferredChannels() const { return -1; }
+
+ protected:
+  virtual ~AudioTrackSinkInterface() {}
+};
+
+// AudioSourceInterface is a reference counted source used for AudioTracks.
+// The same source can be used by multiple AudioTracks.
+class RTC_EXPORT 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 webrtc::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 RTC_EXPORT 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 webrtc::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;
+
+  // Takes ownership of added tracks.
+  // Note: Default implementations are for avoiding link time errors in
+  // implementations that mock this API.
+  // TODO(bugs.webrtc.org/13980): Remove default implementations.
+  virtual bool AddTrack(rtc::scoped_refptr<AudioTrackInterface> track) {
+    RTC_CHECK_NOTREACHED();
+  }
+  virtual bool AddTrack(rtc::scoped_refptr<VideoTrackInterface> track) {
+    RTC_CHECK_NOTREACHED();
+  }
+  virtual bool RemoveTrack(rtc::scoped_refptr<AudioTrackInterface> track) {
+    RTC_CHECK_NOTREACHED();
+  }
+  virtual bool RemoveTrack(rtc::scoped_refptr<VideoTrackInterface> track) {
+    RTC_CHECK_NOTREACHED();
+  }
+  // Deprecated: Should use scoped_refptr versions rather than pointers.
+  [[deprecated("Pass a scoped_refptr")]] virtual bool AddTrack(
+      AudioTrackInterface* track) {
+    return AddTrack(rtc::scoped_refptr<AudioTrackInterface>(track));
+  }
+  [[deprecated("Pass a scoped_refptr")]] virtual bool AddTrack(
+      VideoTrackInterface* track) {
+    return AddTrack(rtc::scoped_refptr<VideoTrackInterface>(track));
+  }
+  [[deprecated("Pass a scoped_refptr")]] virtual bool RemoveTrack(
+      AudioTrackInterface* track) {
+    return RemoveTrack(rtc::scoped_refptr<AudioTrackInterface>(track));
+  }
+  [[deprecated("Pass a scoped_refptr")]] virtual bool RemoveTrack(
+      VideoTrackInterface* track) {
+    return RemoveTrack(rtc::scoped_refptr<VideoTrackInterface>(track));
+  }
+
+ protected:
+  ~MediaStreamInterface() override = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_MEDIA_STREAM_INTERFACE_H_