| /* |
| * 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/transport/media/media_transport_config.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" |
| |
| 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: |
| // Construct a ChannelManager with the specified media engine and data engine. |
| ChannelManager(std::unique_ptr<MediaEngineInterface> media_engine, |
| std::unique_ptr<DataEngineInterface> data_engine, |
| rtc::Thread* worker_thread, |
| rtc::Thread* network_thread); |
| ~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 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; |
| |
| // 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. |
| // 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 cricket::MediaConfig& media_config, |
| webrtc::RtpTransportInternal* rtp_transport, |
| const webrtc::MediaTransportConfig& media_transport_config, |
| 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 cricket::MediaConfig& media_config, |
| webrtc::RtpTransportInternal* rtp_transport, |
| const webrtc::MediaTransportConfig& media_transport_config, |
| 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 cricket::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); |
| |
| // Indicates whether any channels exist. |
| bool has_channels() const { |
| return (!voice_channels_.empty() || !video_channels_.empty() || |
| !data_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(webrtc::FileWrapper file, int64_t max_size_bytes); |
| |
| // Stops recording AEC dump. |
| void StopAecDump(); |
| |
| private: |
| std::unique_ptr<MediaEngineInterface> media_engine_; // Nullable. |
| std::unique_ptr<DataEngineInterface> data_engine_; // Non-null. |
| bool initialized_ = false; |
| rtc::Thread* main_thread_; |
| rtc::Thread* worker_thread_; |
| rtc::Thread* network_thread_; |
| |
| // Vector contents are non-null. |
| std::vector<std::unique_ptr<VoiceChannel>> voice_channels_; |
| std::vector<std::unique_ptr<VideoChannel>> video_channels_; |
| std::vector<std::unique_ptr<RtpDataChannel>> data_channels_; |
| |
| bool enable_rtx_ = false; |
| bool capturing_ = false; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // PC_CHANNEL_MANAGER_H_ |