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/*
* Copyright 2004 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.
*/
#ifndef WEBRTC_PC_CHANNELMANAGER_H_
#define WEBRTC_PC_CHANNELMANAGER_H_
#include <memory>
#include <string>
#include <vector>
#include "webrtc/base/fileutils.h"
#include "webrtc/base/thread.h"
#include "webrtc/media/base/mediaengine.h"
#include "webrtc/pc/voicechannel.h"
namespace webrtc {
class MediaControllerInterface;
}
namespace cricket {
class VoiceChannel;
// ChannelManager allows the MediaEngine to run on a separate thread, and takes
// care of marshalling calls between threads. It also creates and keeps track of
// voice and video channels; by doing so, it can temporarily pause all the
// channels when a new audio or video device is chosen. The voice and video
// channels are stored in separate vectors, to easily allow operations on just
// voice or just video channels.
// ChannelManager also allows the application to discover what devices it has
// using device manager.
class ChannelManager {
public:
// For testing purposes. Allows the media engine and data media
// engine and dev manager to be mocks.
ChannelManager(std::unique_ptr<MediaEngineInterface> me,
std::unique_ptr<DataEngineInterface> dme,
rtc::Thread* worker_and_network);
// Same as above, but gives an easier default DataEngine.
ChannelManager(std::unique_ptr<MediaEngineInterface> me,
rtc::Thread* worker,
rtc::Thread* network);
~ChannelManager();
// Accessors for the worker thread, allowing it to be set after construction,
// but before Init. set_worker_thread will return false if called after Init.
rtc::Thread* worker_thread() const { return worker_thread_; }
bool set_worker_thread(rtc::Thread* thread) {
if (initialized_) {
return false;
}
worker_thread_ = thread;
return true;
}
rtc::Thread* network_thread() const { return network_thread_; }
bool set_network_thread(rtc::Thread* thread) {
if (initialized_) {
return false;
}
network_thread_ = thread;
return true;
}
MediaEngineInterface* media_engine() { return media_engine_.get(); }
// Retrieves the list of supported audio & video codec types.
// Can be called before starting the media engine.
void GetSupportedAudioSendCodecs(std::vector<AudioCodec>* codecs) const;
void GetSupportedAudioReceiveCodecs(std::vector<AudioCodec>* codecs) const;
void GetSupportedAudioRtpHeaderExtensions(RtpHeaderExtensions* ext) const;
void GetSupportedVideoCodecs(std::vector<VideoCodec>* codecs) const;
void GetSupportedVideoRtpHeaderExtensions(RtpHeaderExtensions* ext) const;
void GetSupportedDataCodecs(std::vector<DataCodec>* codecs) const;
// Indicates whether the media engine is started.
bool initialized() const { return initialized_; }
// Starts up the media engine.
bool Init();
// Shuts down the media engine.
void Terminate();
// The operations below all occur on the worker thread.
// Creates a voice channel, to be associated with the specified session.
VoiceChannel* CreateVoiceChannel(
webrtc::MediaControllerInterface* media_controller,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required,
const AudioOptions& options);
// Destroys a voice channel created with the Create API.
void DestroyVoiceChannel(VoiceChannel* voice_channel);
// Creates a video channel, synced with the specified voice channel, and
// associated with the specified session.
VideoChannel* CreateVideoChannel(
webrtc::MediaControllerInterface* media_controller,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required,
const VideoOptions& options);
// Destroys a video channel created with the Create API.
void DestroyVideoChannel(VideoChannel* video_channel);
RtpDataChannel* CreateRtpDataChannel(
webrtc::MediaControllerInterface* media_controller,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required);
// Destroys a data channel created with the Create API.
void DestroyRtpDataChannel(RtpDataChannel* data_channel);
// Indicates whether any channels exist.
bool has_channels() const {
return (!voice_channels_.empty() || !video_channels_.empty());
}
// RTX will be enabled/disabled in engines that support it. The supporting
// engines will start offering an RTX codec. Must be called before Init().
bool SetVideoRtxEnabled(bool enable);
// Define crypto options to set on newly created channels. Doesn't change
// options on already created channels.
bool SetCryptoOptions(const rtc::CryptoOptions& crypto_options);
// Starts/stops the local microphone and enables polling of the input level.
bool capturing() const { return capturing_; }
// The operations below occur on the main thread.
// Starts AEC dump using existing file, with a specified maximum file size in
// bytes. When the limit is reached, logging will stop and the file will be
// closed. If max_size_bytes is set to <= 0, no limit will be used.
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
// Stops recording AEC dump.
void StopAecDump();
private:
typedef std::vector<VoiceChannel*> VoiceChannels;
typedef std::vector<VideoChannel*> VideoChannels;
typedef std::vector<RtpDataChannel*> RtpDataChannels;
void Construct(std::unique_ptr<MediaEngineInterface> me,
std::unique_ptr<DataEngineInterface> dme,
rtc::Thread* worker_thread,
rtc::Thread* network_thread);
bool InitMediaEngine_w();
void DestructorDeletes_w();
void Terminate_w();
bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options);
VoiceChannel* CreateVoiceChannel_w(
webrtc::MediaControllerInterface* media_controller,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required,
const AudioOptions& options);
void DestroyVoiceChannel_w(VoiceChannel* voice_channel);
VideoChannel* CreateVideoChannel_w(
webrtc::MediaControllerInterface* media_controller,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required,
const VideoOptions& options);
void DestroyVideoChannel_w(VideoChannel* video_channel);
RtpDataChannel* CreateRtpDataChannel_w(
webrtc::MediaControllerInterface* media_controller,
DtlsTransportInternal* rtp_transport,
DtlsTransportInternal* rtcp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
const std::string* bundle_transport_name,
bool rtcp_mux_required,
bool srtp_required);
void DestroyRtpDataChannel_w(RtpDataChannel* data_channel);
std::unique_ptr<MediaEngineInterface> media_engine_;
std::unique_ptr<DataEngineInterface> data_media_engine_;
bool initialized_;
rtc::Thread* main_thread_;
rtc::Thread* worker_thread_;
rtc::Thread* network_thread_;
VoiceChannels voice_channels_;
VideoChannels video_channels_;
RtpDataChannels data_channels_;
bool enable_rtx_;
rtc::CryptoOptions crypto_options_;
bool capturing_;
};
} // namespace cricket
#endif // WEBRTC_PC_CHANNELMANAGER_H_