337 lines
12 KiB
C
337 lines
12 KiB
C
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/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file contains interfaces for MediaStream, MediaTrack and MediaSource.
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// These interfaces are used for implementing MediaStream and MediaTrack as
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// defined in http://dev.w3.org/2011/webrtc/editor/webrtc.html#stream-api. These
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// interfaces must be used only with PeerConnection.
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#ifndef API_MEDIA_STREAM_INTERFACE_H_
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#define API_MEDIA_STREAM_INTERFACE_H_
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#include <stddef.h>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/audio_options.h"
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#include "api/scoped_refptr.h"
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#include "api/video/recordable_encoded_frame.h"
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#include "api/video/video_frame.h"
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#include "api/video/video_sink_interface.h"
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#include "api/video/video_source_interface.h"
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#include "modules/audio_processing/include/audio_processing_statistics.h"
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#include "rtc_base/ref_count.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// Generic observer interface.
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class ObserverInterface {
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public:
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virtual void OnChanged() = 0;
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protected:
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virtual ~ObserverInterface() {}
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};
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class NotifierInterface {
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public:
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virtual void RegisterObserver(ObserverInterface* observer) = 0;
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virtual void UnregisterObserver(ObserverInterface* observer) = 0;
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virtual ~NotifierInterface() {}
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};
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// Base class for sources. A MediaStreamTrack has an underlying source that
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// provides media. A source can be shared by multiple tracks.
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class RTC_EXPORT MediaSourceInterface : public rtc::RefCountInterface,
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public NotifierInterface {
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public:
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enum SourceState { kInitializing, kLive, kEnded, kMuted };
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virtual SourceState state() const = 0;
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virtual bool remote() const = 0;
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protected:
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~MediaSourceInterface() override = default;
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};
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// C++ version of MediaStreamTrack.
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// See: https://www.w3.org/TR/mediacapture-streams/#mediastreamtrack
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class RTC_EXPORT MediaStreamTrackInterface : public rtc::RefCountInterface,
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public NotifierInterface {
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public:
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enum TrackState {
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kLive,
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kEnded,
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};
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static const char* const kAudioKind;
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static const char* const kVideoKind;
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// The kind() method must return kAudioKind only if the object is a
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// subclass of AudioTrackInterface, and kVideoKind only if the
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// object is a subclass of VideoTrackInterface. It is typically used
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// to protect a static_cast<> to the corresponding subclass.
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virtual std::string kind() const = 0;
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// Track identifier.
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virtual std::string id() const = 0;
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// A disabled track will produce silence (if audio) or black frames (if
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// video). Can be disabled and re-enabled.
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virtual bool enabled() const = 0;
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virtual bool set_enabled(bool enable) = 0;
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// Live or ended. A track will never be live again after becoming ended.
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virtual TrackState state() const = 0;
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protected:
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~MediaStreamTrackInterface() override = default;
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};
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// VideoTrackSourceInterface is a reference counted source used for
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// VideoTracks. The same source can be used by multiple VideoTracks.
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// VideoTrackSourceInterface is designed to be invoked on the signaling thread
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// except for rtc::VideoSourceInterface<VideoFrame> methods that will be invoked
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// on the worker thread via a VideoTrack. A custom implementation of a source
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// can inherit AdaptedVideoTrackSource instead of directly implementing this
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// interface.
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class VideoTrackSourceInterface : public MediaSourceInterface,
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public rtc::VideoSourceInterface<VideoFrame> {
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public:
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struct Stats {
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// Original size of captured frame, before video adaptation.
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int input_width;
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int input_height;
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};
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// Indicates that parameters suitable for screencasts should be automatically
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// applied to RtpSenders.
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// TODO(perkj): Remove these once all known applications have moved to
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// explicitly setting suitable parameters for screencasts and don't need this
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// implicit behavior.
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virtual bool is_screencast() const = 0;
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// Indicates that the encoder should denoise video before encoding it.
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// If it is not set, the default configuration is used which is different
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// depending on video codec.
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// TODO(perkj): Remove this once denoising is done by the source, and not by
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// the encoder.
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virtual absl::optional<bool> needs_denoising() const = 0;
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// Returns false if no stats are available, e.g, for a remote source, or a
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// source which has not seen its first frame yet.
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//
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// Implementation should avoid blocking.
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virtual bool GetStats(Stats* stats) = 0;
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// Returns true if encoded output can be enabled in the source.
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virtual bool SupportsEncodedOutput() const = 0;
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// Reliably cause a key frame to be generated in encoded output.
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// TODO(bugs.webrtc.org/11115): find optimal naming.
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virtual void GenerateKeyFrame() = 0;
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// Add an encoded video sink to the source and additionally cause
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// a key frame to be generated from the source. The sink will be
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// invoked from a decoder queue.
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// TODO(bugs.webrtc.org/11114): make pure virtual once downstream project
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// adapts.
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virtual void AddEncodedSink(
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rtc::VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;
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// Removes an encoded video sink from the source.
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virtual void RemoveEncodedSink(
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rtc::VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;
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protected:
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~VideoTrackSourceInterface() override = default;
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};
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// VideoTrackInterface is designed to be invoked on the signaling thread except
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// for rtc::VideoSourceInterface<VideoFrame> methods that must be invoked
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// on the worker thread.
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// PeerConnectionFactory::CreateVideoTrack can be used for creating a VideoTrack
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// that ensures thread safety and that all methods are called on the right
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// thread.
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class RTC_EXPORT VideoTrackInterface
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: public MediaStreamTrackInterface,
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public rtc::VideoSourceInterface<VideoFrame> {
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public:
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// Video track content hint, used to override the source is_screencast
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// property.
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// See https://crbug.com/653531 and https://w3c.github.io/mst-content-hint.
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enum class ContentHint { kNone, kFluid, kDetailed, kText };
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// Register a video sink for this track. Used to connect the track to the
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// underlying video engine.
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void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
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const rtc::VideoSinkWants& wants) override {}
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void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
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virtual VideoTrackSourceInterface* GetSource() const = 0;
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virtual ContentHint content_hint() const;
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virtual void set_content_hint(ContentHint hint) {}
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protected:
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~VideoTrackInterface() override = default;
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};
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// Interface for receiving audio data from a AudioTrack.
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class AudioTrackSinkInterface {
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public:
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virtual void OnData(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) {
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RTC_NOTREACHED() << "This method must be overridden, or not used.";
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}
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// In this method, |absolute_capture_timestamp_ms|, when available, is
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// supposed to deliver the timestamp when this audio frame was originally
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// captured. This timestamp MUST be based on the same clock as
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// rtc::TimeMillis().
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virtual void OnData(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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absl::optional<int64_t> absolute_capture_timestamp_ms) {
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// TODO(bugs.webrtc.org/10739): Deprecate the old OnData and make this one
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// pure virtual.
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return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
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number_of_frames);
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}
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protected:
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virtual ~AudioTrackSinkInterface() {}
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};
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// AudioSourceInterface is a reference counted source used for AudioTracks.
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// The same source can be used by multiple AudioTracks.
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class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
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public:
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class AudioObserver {
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public:
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virtual void OnSetVolume(double volume) = 0;
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protected:
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virtual ~AudioObserver() {}
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};
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// TODO(deadbeef): Makes all the interfaces pure virtual after they're
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// implemented in chromium.
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// Sets the volume of the source. |volume| is in the range of [0, 10].
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// TODO(tommi): This method should be on the track and ideally volume should
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// be applied in the track in a way that does not affect clones of the track.
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virtual void SetVolume(double volume) {}
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// Registers/unregisters observers to the audio source.
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virtual void RegisterAudioObserver(AudioObserver* observer) {}
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virtual void UnregisterAudioObserver(AudioObserver* observer) {}
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// TODO(tommi): Make pure virtual.
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virtual void AddSink(AudioTrackSinkInterface* sink) {}
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virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
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// Returns options for the AudioSource.
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// (for some of the settings this approach is broken, e.g. setting
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// audio network adaptation on the source is the wrong layer of abstraction).
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virtual const cricket::AudioOptions options() const;
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};
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// Interface of the audio processor used by the audio track to collect
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// statistics.
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class AudioProcessorInterface : public rtc::RefCountInterface {
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public:
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struct AudioProcessorStatistics {
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bool typing_noise_detected = false;
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AudioProcessingStats apm_statistics;
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};
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// Get audio processor statistics. The |has_remote_tracks| argument should be
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// set if there are active remote tracks (this would usually be true during
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// a call). If there are no remote tracks some of the stats will not be set by
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// the AudioProcessor, because they only make sense if there is at least one
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// remote track.
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virtual AudioProcessorStatistics GetStats(bool has_remote_tracks) = 0;
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protected:
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~AudioProcessorInterface() override = default;
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};
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class RTC_EXPORT AudioTrackInterface : public MediaStreamTrackInterface {
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public:
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// TODO(deadbeef): Figure out if the following interface should be const or
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// not.
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virtual AudioSourceInterface* GetSource() const = 0;
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// Add/Remove a sink that will receive the audio data from the track.
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virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
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virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
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// Get the signal level from the audio track.
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// Return true on success, otherwise false.
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// TODO(deadbeef): Change the interface to int GetSignalLevel() and pure
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// virtual after it's implemented in chromium.
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virtual bool GetSignalLevel(int* level);
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// Get the audio processor used by the audio track. Return null if the track
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// does not have any processor.
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// TODO(deadbeef): Make the interface pure virtual.
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virtual rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor();
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protected:
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~AudioTrackInterface() override = default;
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};
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typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> > AudioTrackVector;
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typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> > VideoTrackVector;
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// C++ version of https://www.w3.org/TR/mediacapture-streams/#mediastream.
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//
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// A major difference is that remote audio/video tracks (received by a
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// PeerConnection/RtpReceiver) are not synchronized simply by adding them to
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// the same stream; a session description with the correct "a=msid" attributes
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// must be pushed down.
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//
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// Thus, this interface acts as simply a container for tracks.
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class MediaStreamInterface : public rtc::RefCountInterface,
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public NotifierInterface {
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public:
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virtual std::string id() const = 0;
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virtual AudioTrackVector GetAudioTracks() = 0;
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virtual VideoTrackVector GetVideoTracks() = 0;
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virtual rtc::scoped_refptr<AudioTrackInterface> FindAudioTrack(
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const std::string& track_id) = 0;
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virtual rtc::scoped_refptr<VideoTrackInterface> FindVideoTrack(
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const std::string& track_id) = 0;
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virtual bool AddTrack(AudioTrackInterface* track) = 0;
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virtual bool AddTrack(VideoTrackInterface* track) = 0;
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virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
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virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
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protected:
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~MediaStreamInterface() override = default;
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};
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} // namespace webrtc
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#endif // API_MEDIA_STREAM_INTERFACE_H_
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