webrtc_m130/webrtc/pc/rtpsender.h
zhihuang 38ede13042 Support building WebRTC without audio and video.
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).

The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.

The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.

Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)

BUG=webrtc:7613

Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 19:52:32 +00:00

254 lines
8.2 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_PC_RTPSENDER_H_
#define WEBRTC_PC_RTPSENDER_H_
#include <memory>
#include <string>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/rtpsenderinterface.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/criticalsection.h"
// Adding 'nogncheck' to disable the gn include headers check to support modular
// WebRTC build targets.
#include "webrtc/media/base/audiosource.h" // nogncheck
#include "webrtc/pc/channel.h"
#include "webrtc/pc/dtmfsender.h"
#include "webrtc/pc/statscollector.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 DtmfProviderInterface,
public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInternal> {
public:
// StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
// at the appropriate times.
// |channel| can be null if one does not exist yet.
AudioRtpSender(AudioTrackInterface* track,
const std::string& stream_id,
cricket::VoiceChannel* channel,
StatsCollector* stats);
// Randomly generates stream_id.
// |channel| can be null if one does not exist yet.
AudioRtpSender(AudioTrackInterface* track,
cricket::VoiceChannel* channel,
StatsCollector* stats);
// Randomly generates id and stream_id.
// |channel| can be null if one does not exist yet.
AudioRtpSender(cricket::VoiceChannel* channel, 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 {
std::vector<std::string> ret = {stream_id_};
return ret;
}
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const 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;
// Does not take ownership.
// Should call SetChannel(nullptr) before |channel| is destroyed.
void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; }
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();
void CreateDtmfSender();
sigslot::signal0<> SignalDestroyed;
std::string id_;
std::string stream_id_;
cricket::VoiceChannel* channel_ = nullptr;
StatsCollector* stats_;
rtc::scoped_refptr<AudioTrackInterface> track_;
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
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:
// |channel| can be null if one does not exist yet.
VideoRtpSender(VideoTrackInterface* track,
const std::string& stream_id,
cricket::VideoChannel* channel);
// Randomly generates stream_id.
// |channel| can be null if one does not exist yet.
VideoRtpSender(VideoTrackInterface* track, cricket::VideoChannel* channel);
// Randomly generates id and stream_id.
// |channel| can be null if one does not exist yet.
explicit VideoRtpSender(cricket::VideoChannel* channel);
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;
rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const 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;
// Does not take ownership.
// Should call SetChannel(nullptr) before |channel| is destroyed.
void SetChannel(cricket::VideoChannel* channel) { channel_ = 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();
std::string id_;
std::string stream_id_;
cricket::VideoChannel* channel_ = nullptr;
rtc::scoped_refptr<VideoTrackInterface> track_;
uint32_t ssrc_ = 0;
bool cached_track_enabled_ = false;
VideoTrackInterface::ContentHint cached_track_content_hint_ =
VideoTrackInterface::ContentHint::kNone;
bool stopped_ = false;
};
} // namespace webrtc
#endif // WEBRTC_PC_RTPSENDER_H_