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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 These
// interfaces must be used only with PeerConnection.
#include <stddef.h>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_options.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 "modules/audio_processing/include/audio_processing_statistics.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// Generic observer interface.
class ObserverInterface {
virtual void OnChanged() = 0;
virtual ~ObserverInterface() {}
class NotifierInterface {
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 rtc::RefCountInterface,
public NotifierInterface {
enum SourceState { kInitializing, kLive, kEnded, kMuted };
virtual SourceState state() const = 0;
virtual bool remote() const = 0;
~MediaSourceInterface() override = default;
// C++ version of MediaStreamTrack.
// See:
class RTC_EXPORT MediaStreamTrackInterface : public rtc::RefCountInterface,
public NotifierInterface {
enum TrackState {
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;
~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> {
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( 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.
// TODO( make pure virtual once downstream project
// adapts.
virtual void AddEncodedSink(
rtc::VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;
// Removes an encoded video sink from the source.
virtual void RemoveEncodedSink(
rtc::VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;
~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> {
// Video track content hint, used to override the source is_screencast
// property.
// See and
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) {}
~VideoTrackInterface() override = default;
// Interface for receiving audio data from a AudioTrack.
class AudioTrackSinkInterface {
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_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( Deprecate the old OnData and make this one
// pure virtual.
return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
// 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; }
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 {
class AudioObserver {
virtual void OnSetVolume(double volume) = 0;
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 {
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;
~AudioProcessorInterface() override = default;
class RTC_EXPORT AudioTrackInterface : public MediaStreamTrackInterface {
// 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();
~AudioTrackInterface() override = default;
typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> > AudioTrackVector;
typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> > VideoTrackVector;
// C++ version of
// 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 {
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;
~MediaStreamInterface() override = default;
} // namespace webrtc