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/*
* 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 PC_RTPSENDER_H_
#define PC_RTPSENDER_H_
#include <memory>
#include <string>
#include <vector>
#include "api/mediastreaminterface.h"
#include "api/rtpsenderinterface.h"
#include "rtc_base/criticalsection.h"
#include "media/base/audiosource.h"
#include "media/base/mediachannel.h"
#include "pc/dtmfsender.h"
namespace webrtc {
class StatsCollector;
// Internal interface used by PeerConnection.
class RtpSenderInternal : public RtpSenderInterface {
public:
// Sets the underlying MediaEngine channel associated with this RtpSender.
// SetVoiceMediaChannel should be used for audio RtpSenders and
// SetVideoMediaChannel should be used for video RtpSenders. Must call the
// appropriate SetXxxMediaChannel(nullptr) before the media channel is
// destroyed.
virtual void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) = 0;
virtual void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_media_channel) = 0;
// 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;
virtual void set_stream_ids(const std::vector<std::string>& stream_ids) = 0;
virtual void Stop() = 0;
// Returns an ID that changes every time SetTrack() is called, but
// otherwise remains constant. Used to generate IDs for stats.
// The special value zero means that no track is attached.
virtual int AttachmentId() const = 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 DtmfProviderInterface,
public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
// StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
// at the appropriate times.
// Construct an AudioRtpSender with a null track, a single, randomly generated
// stream id, and a randomly generated ID.
AudioRtpSender(rtc::Thread* worker_thread, StatsCollector* stats);
// Construct an AudioRtpSender with the given track and stream ids. The
// sender ID will be set to the track's ID.
AudioRtpSender(rtc::Thread* worker_thread,
rtc::scoped_refptr<AudioTrackInterface> track,
const std::vector<std::string>& stream_ids,
StatsCollector* stats);
virtual ~AudioRtpSender();
// DtmfSenderProvider implementation.
bool CanInsertDtmf() override;
bool InsertDtmf(int code, int duration) override;
sigslot::signal0<>* GetOnDestroyedSignal() override;
// 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 { return stream_ids_; }
RtpParameters GetParameters() override;
RTCError SetParameters(const RtpParameters& parameters) override;
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
// RtpSenderInternal implementation.
void SetSsrc(uint32_t ssrc) override;
void set_stream_ids(const std::vector<std::string>& stream_ids) override {
stream_ids_ = stream_ids;
}
void Stop() override;
int AttachmentId() const override { return attachment_id_; }
void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) override {
media_channel_ = voice_media_channel;
}
void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_media_channel) override {
RTC_NOTREACHED();
}
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();
// Helper function to call SetAudioSend with "stop sending" parameters.
void ClearAudioSend();
sigslot::signal0<> SignalDestroyed;
rtc::Thread* const worker_thread_;
const std::string id_;
std::vector<std::string> stream_ids_;
cricket::VoiceMediaChannel* media_channel_ = nullptr;
StatsCollector* stats_;
rtc::scoped_refptr<AudioTrackInterface> track_;
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
rtc::Optional<std::string> last_transaction_id_;
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_;
int attachment_id_ = 0;
};
class VideoRtpSender : public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
// Construct a VideoRtpSender with a null track, a single, randomly generated
// stream id, and a randomly generated ID.
explicit VideoRtpSender(rtc::Thread* worker_thread);
// Construct a VideoRtpSender with the given track and stream ids. The
// sender ID will be set to the track's ID.
VideoRtpSender(rtc::Thread* worker_thread,
rtc::scoped_refptr<VideoTrackInterface> track,
const std::vector<std::string>& stream_ids);
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 { return stream_ids_; }
RtpParameters GetParameters() override;
RTCError SetParameters(const RtpParameters& parameters) override;
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
// RtpSenderInternal implementation.
void SetSsrc(uint32_t ssrc) override;
void set_stream_ids(const std::vector<std::string>& stream_ids) override {
stream_ids_ = stream_ids;
}
void Stop() override;
int AttachmentId() const override { return attachment_id_; }
void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) override {
RTC_NOTREACHED();
}
void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_media_channel) override {
media_channel_ = video_media_channel;
}
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();
rtc::Thread* worker_thread_;
const std::string id_;
std::vector<std::string> stream_ids_;
cricket::VideoMediaChannel* media_channel_ = nullptr;
rtc::scoped_refptr<VideoTrackInterface> track_;
rtc::Optional<std::string> last_transaction_id_;
uint32_t ssrc_ = 0;
VideoTrackInterface::ContentHint cached_track_content_hint_ =
VideoTrackInterface::ContentHint::kNone;
bool stopped_ = false;
int attachment_id_ = 0;
};
} // namespace webrtc
#endif // PC_RTPSENDER_H_