| /* |
| * 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/media/base/mediaengine.h" |
| #include "webrtc/pc/voicechannel.h" |
| #include "webrtc/rtc_base/fileutils.h" |
| #include "webrtc/rtc_base/thread.h" |
| |
| 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::Call* call, |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_transport, |
| DtlsTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const AudioOptions& options); |
| // Version of the above that takes PacketTransportInternal. |
| VoiceChannel* CreateVoiceChannel( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| rtc::PacketTransportInternal* rtp_transport, |
| rtc::PacketTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| 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::Call* call, |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_transport, |
| DtlsTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const VideoOptions& options); |
| // Version of the above that takes PacketTransportInternal. |
| VideoChannel* CreateVideoChannel( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| rtc::PacketTransportInternal* rtp_transport, |
| rtc::PacketTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const VideoOptions& options); |
| // Destroys a video channel created with the Create API. |
| void DestroyVideoChannel(VideoChannel* video_channel); |
| RtpDataChannel* CreateRtpDataChannel( |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_transport, |
| DtlsTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| 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); |
| |
| // 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(); |
| VoiceChannel* CreateVoiceChannel_w( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const AudioOptions& options); |
| void DestroyVoiceChannel_w(VoiceChannel* voice_channel); |
| VideoChannel* CreateVideoChannel_w( |
| webrtc::Call* call, |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_dtls_transport, |
| DtlsTransportInternal* rtcp_dtls_transport, |
| rtc::PacketTransportInternal* rtp_packet_transport, |
| rtc::PacketTransportInternal* rtcp_packet_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| bool srtp_required, |
| const VideoOptions& options); |
| void DestroyVideoChannel_w(VideoChannel* video_channel); |
| RtpDataChannel* CreateRtpDataChannel_w( |
| const cricket::MediaConfig& media_config, |
| DtlsTransportInternal* rtp_transport, |
| DtlsTransportInternal* rtcp_transport, |
| rtc::Thread* signaling_thread, |
| const std::string& content_name, |
| 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_; |
| bool capturing_; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // WEBRTC_PC_CHANNELMANAGER_H_ |