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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. PeerConnectionManager
// interface provides the factory methods to create MediaStream and MediaTracks.
#include <string>
#include <vector>
#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/videoframe.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.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 have an underlying source that
// provide media. A source can be shared with multiple tracks.
class MediaSourceInterface : public rtc::RefCountInterface,
public NotifierInterface {
enum SourceState {
virtual SourceState state() const = 0;
virtual bool remote() const = 0;
virtual ~MediaSourceInterface() {}
// Information about a track.
class MediaStreamTrackInterface : public rtc::RefCountInterface,
public NotifierInterface {
enum TrackState {
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;
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<cricket::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 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<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.
// Should avoid blocking.
virtual bool GetStats(Stats* stats) = 0;
virtual ~VideoTrackSourceInterface() {}
class VideoTrackInterface
: public MediaStreamTrackInterface,
public rtc::VideoSourceInterface<cricket::VideoFrame> {
// Register a video sink for this track.
void AddOrUpdateSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override{};
void RemoveSink(
rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override{};
virtual VideoTrackSourceInterface* GetSource() const = 0;
virtual ~VideoTrackInterface() {}
// 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) = 0;
virtual ~AudioTrackSinkInterface() {}
// AudioSourceInterface is a reference counted source used for AudioTracks.
// The same source can be used in multiple AudioTracks.
class AudioSourceInterface : public MediaSourceInterface {
class AudioObserver {
virtual void OnSetVolume(double volume) = 0;
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 {
struct AudioProcessorStats {
AudioProcessorStats() : typing_noise_detected(false),
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;
virtual ~AudioProcessorInterface() {}
class AudioTrackInterface : public MediaStreamTrackInterface {
// 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<AudioProcessorInterface>
GetAudioProcessor() { return NULL; }
virtual ~AudioTrackInterface() {}
typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
class MediaStreamInterface : public rtc::RefCountInterface,
public NotifierInterface {
virtual std::string label() 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;
virtual ~MediaStreamInterface() {}
} // namespace webrtc