rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/api/rtpsender.h

217 lines
6.7 KiB
C++

/*
* Copyright 2015 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 classes that implement RtpSenderInterface.
// An RtpSender associates a MediaStreamTrackInterface with an underlying
// transport (provided by AudioProviderInterface/VideoProviderInterface)
#ifndef WEBRTC_API_RTPSENDER_H_
#define WEBRTC_API_RTPSENDER_H_
#include <memory>
#include <string>
#include "webrtc/api/mediastreamprovider.h"
#include "webrtc/api/rtpsenderinterface.h"
#include "webrtc/api/statscollector.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/media/base/audiosource.h"
namespace webrtc {
// Internal interface used by PeerConnection.
class RtpSenderInternal : public RtpSenderInterface {
public:
// Used to set the SSRC of the sender, once a local description has been set.
// If |ssrc| is 0, this indiates that the sender should disconnect from the
// underlying transport (this occurs if the sender isn't seen in a local
// description).
virtual void SetSsrc(uint32_t ssrc) = 0;
// TODO(deadbeef): Support one sender having multiple stream ids.
virtual void set_stream_id(const std::string& stream_id) = 0;
virtual std::string stream_id() const = 0;
virtual void Stop() = 0;
};
// LocalAudioSinkAdapter receives data callback as a sink to the local
// AudioTrack, and passes the data to the sink of AudioSource.
class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
public cricket::AudioSource {
public:
LocalAudioSinkAdapter();
virtual ~LocalAudioSinkAdapter();
private:
// AudioSinkInterface implementation.
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override;
// cricket::AudioSource implementation.
void SetSink(cricket::AudioSource::Sink* sink) override;
cricket::AudioSource::Sink* sink_;
// Critical section protecting |sink_|.
rtc::CriticalSection lock_;
};
class AudioRtpSender : public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
// StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
// at the appropriate times.
AudioRtpSender(AudioTrackInterface* track,
const std::string& stream_id,
AudioProviderInterface* provider,
StatsCollector* stats);
// Randomly generates stream_id.
AudioRtpSender(AudioTrackInterface* track,
AudioProviderInterface* provider,
StatsCollector* stats);
// Randomly generates id and stream_id.
AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
virtual ~AudioRtpSender();
// ObserverInterface implementation
void OnChanged() override;
// RtpSenderInterface implementation
bool SetTrack(MediaStreamTrackInterface* track) override;
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_;
}
uint32_t ssrc() const override { return ssrc_; }
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
std::string id() const override { return id_; }
std::vector<std::string> stream_ids() const override {
std::vector<std::string> ret = {stream_id_};
return ret;
}
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
// RtpSenderInternal implementation.
void SetSsrc(uint32_t ssrc) override;
void set_stream_id(const std::string& stream_id) override {
stream_id_ = stream_id;
}
std::string stream_id() const override { return stream_id_; }
void Stop() override;
private:
// TODO(nisse): Since SSRC == 0 is technically valid, figure out
// some other way to test if we have a valid SSRC.
bool can_send_track() const { return track_ && ssrc_; }
// Helper function to construct options for
// AudioProviderInterface::SetAudioSend.
void SetAudioSend();
std::string id_;
std::string stream_id_;
AudioProviderInterface* provider_;
StatsCollector* stats_;
rtc::scoped_refptr<AudioTrackInterface> track_;
uint32_t ssrc_ = 0;
bool cached_track_enabled_ = false;
bool stopped_ = false;
// Used to pass the data callback from the |track_| to the other end of
// cricket::AudioSource.
std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
};
class VideoRtpSender : public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
VideoRtpSender(VideoTrackInterface* track,
const std::string& stream_id,
VideoProviderInterface* provider);
// Randomly generates stream_id.
VideoRtpSender(VideoTrackInterface* track, VideoProviderInterface* provider);
// Randomly generates id and stream_id.
explicit VideoRtpSender(VideoProviderInterface* provider);
virtual ~VideoRtpSender();
// ObserverInterface implementation
void OnChanged() override;
// RtpSenderInterface implementation
bool SetTrack(MediaStreamTrackInterface* track) override;
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_;
}
uint32_t ssrc() const override { return ssrc_; }
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
std::string id() const override { return id_; }
std::vector<std::string> stream_ids() const override {
std::vector<std::string> ret = {stream_id_};
return ret;
}
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
// RtpSenderInternal implementation.
void SetSsrc(uint32_t ssrc) override;
void set_stream_id(const std::string& stream_id) override {
stream_id_ = stream_id;
}
std::string stream_id() const override { return stream_id_; }
void Stop() override;
private:
bool can_send_track() const { return track_ && ssrc_; }
// Helper function to construct options for
// VideoProviderInterface::SetVideoSend.
void SetVideoSend();
// Helper function to call SetVideoSend with "stop sending" parameters.
void ClearVideoSend();
std::string id_;
std::string stream_id_;
VideoProviderInterface* provider_;
rtc::scoped_refptr<VideoTrackInterface> track_;
uint32_t ssrc_ = 0;
bool cached_track_enabled_ = false;
bool stopped_ = false;
};
} // namespace webrtc
#endif // WEBRTC_API_RTPSENDER_H_