tor-browser

media_stream_interface.h (13636B)

---

/*
*  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 <stdint.h>

#include <optional>
#include <string>
#include <vector>

#include "api/audio/audio_processing_statistics.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 "rtc_base/checks.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 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 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 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 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 std::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(
     VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;

 // Removes an encoded video sink from the source.
 virtual void RemoveEncodedSink(
     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 VideoTrackSourceConstraints& /* constraints */) {}

protected:
 ~VideoTrackSourceInterface() override = default;
};

// VideoTrackInterface is designed to be invoked on the signaling thread except
// for 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 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(VideoSinkInterface<VideoFrame>* /* sink */,
                      const VideoSinkWants& /* wants */) override {}
 void RemoveSink(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
 // 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,
     std::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 AudioOptions options() const;
};

// Interface of the audio processor used by the audio track to collect
// statistics.
class AudioProcessorInterface : public 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 scoped_refptr<AudioProcessorInterface> GetAudioProcessor();

protected:
 ~AudioTrackInterface() override = default;
};

typedef std::vector<scoped_refptr<AudioTrackInterface> > AudioTrackVector;
typedef std::vector<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 RefCountInterface,
                            public NotifierInterface {
public:
 virtual std::string id() const = 0;

 virtual AudioTrackVector GetAudioTracks() = 0;
 virtual VideoTrackVector GetVideoTracks() = 0;
 virtual scoped_refptr<AudioTrackInterface> FindAudioTrack(
     const std::string& track_id) = 0;
 virtual 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(scoped_refptr<AudioTrackInterface> /* track */) {
   RTC_CHECK_NOTREACHED();
 }
 virtual bool AddTrack(scoped_refptr<VideoTrackInterface> /* track */) {
   RTC_CHECK_NOTREACHED();
 }
 virtual bool RemoveTrack(scoped_refptr<AudioTrackInterface> /* track */) {
   RTC_CHECK_NOTREACHED();
 }
 virtual bool RemoveTrack(scoped_refptr<VideoTrackInterface> /* track */) {
   RTC_CHECK_NOTREACHED();
 }

protected:
 ~MediaStreamInterface() override = default;
};

}  // namespace webrtc

#endif  // API_MEDIA_STREAM_INTERFACE_H_