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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 PC_CHANNEL_MANAGER_H_
#define PC_CHANNEL_MANAGER_H_
#include <stdint.h>
#include <memory>
#include <string>
#include <vector>
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_parameters.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "pc/channel.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/system/file_wrapper.h"
#include "rtc_base/thread.h"
#include "rtc_base/unique_id_generator.h"
namespace cricket {
// 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 final {
public:
// Returns an initialized instance of ChannelManager.
// If media_engine is non-nullptr, then the returned ChannelManager instance
// will own that reference and media engine initialization
static std::unique_ptr<ChannelManager> Create(
std::unique_ptr<MediaEngineInterface> media_engine,
std::unique_ptr<DataEngineInterface> data_engine,
bool enable_rtx,
rtc::Thread* worker_thread,
rtc::Thread* network_thread);
ChannelManager() = delete;
~ChannelManager();
rtc::Thread* worker_thread() const { return worker_thread_; }
rtc::Thread* network_thread() const { return network_thread_; }
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 GetSupportedVideoSendCodecs(std::vector<VideoCodec>* codecs) const;
void GetSupportedVideoReceiveCodecs(std::vector<VideoCodec>* codecs) const;
void GetSupportedDataCodecs(std::vector<DataCodec>* codecs) const;
RtpHeaderExtensions GetDefaultEnabledAudioRtpHeaderExtensions() const;
std::vector<webrtc::RtpHeaderExtensionCapability>
GetSupportedAudioRtpHeaderExtensions() const;
RtpHeaderExtensions GetDefaultEnabledVideoRtpHeaderExtensions() const;
std::vector<webrtc::RtpHeaderExtensionCapability>
GetSupportedVideoRtpHeaderExtensions() const;
// The operations below all occur on the worker thread.
// ChannelManager retains ownership of the created channels, so clients should
// call the appropriate Destroy*Channel method when done.
// Creates a voice channel, to be associated with the specified session.
VoiceChannel* CreateVoiceChannel(webrtc::Call* call,
const MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
const AudioOptions& options);
// Destroys a voice channel created by CreateVoiceChannel.
void DestroyVoiceChannel(VoiceChannel* voice_channel);
// Creates a video channel, synced with the specified voice channel, and
// associated with the specified session.
// Version of the above that takes PacketTransportInternal.
VideoChannel* CreateVideoChannel(
webrtc::Call* call,
const MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
const VideoOptions& options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory);
// Destroys a video channel created by CreateVideoChannel.
void DestroyVideoChannel(VideoChannel* video_channel);
RtpDataChannel* CreateRtpDataChannel(
const MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
const webrtc::CryptoOptions& crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
// Destroys a data channel created by CreateRtpDataChannel.
void DestroyRtpDataChannel(RtpDataChannel* data_channel);
// 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(webrtc::FileWrapper file, int64_t max_size_bytes);
// Stops recording AEC dump.
void StopAecDump();
private:
ChannelManager(std::unique_ptr<MediaEngineInterface> media_engine,
std::unique_ptr<DataEngineInterface> data_engine,
bool enable_rtx,
rtc::Thread* worker_thread,
rtc::Thread* network_thread);
const std::unique_ptr<MediaEngineInterface> media_engine_; // Nullable.
const std::unique_ptr<DataEngineInterface> data_engine_; // Non-null.
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
// Vector contents are non-null.
std::vector<std::unique_ptr<VoiceChannel>> voice_channels_
RTC_GUARDED_BY(worker_thread_);
std::vector<std::unique_ptr<VideoChannel>> video_channels_
RTC_GUARDED_BY(worker_thread_);
std::vector<std::unique_ptr<RtpDataChannel>> data_channels_
RTC_GUARDED_BY(worker_thread_);
const bool enable_rtx_;
};
} // namespace cricket
#endif // PC_CHANNEL_MANAGER_H_