| /* |
| * Copyright (c) 2018 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 CALL_CALL_CONFIG_H_ |
| #define CALL_CALL_CONFIG_H_ |
| |
| #include <memory> |
| #include <optional> |
| |
| #include "api/environment/environment.h" |
| #include "api/fec_controller.h" |
| #include "api/metronome/metronome.h" |
| #include "api/neteq/neteq_factory.h" |
| #include "api/network_state_predictor.h" |
| #include "api/scoped_refptr.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/transport/bitrate_settings.h" |
| #include "api/transport/network_control.h" |
| #include "api/units/time_delta.h" |
| #include "call/audio_state.h" |
| #include "call/rtp_transport_config.h" |
| #include "call/rtp_transport_controller_send_factory_interface.h" |
| |
| namespace webrtc { |
| |
| class AudioProcessing; |
| |
| struct CallConfig { |
| // If `network_task_queue` is set to nullptr, Call will assume that network |
| // related callbacks will be made on the same TQ as the Call instance was |
| // constructed on. |
| explicit CallConfig(const Environment& env, |
| TaskQueueBase* network_task_queue = nullptr); |
| |
| // Move-only. |
| CallConfig(CallConfig&&) = default; |
| CallConfig& operator=(CallConfig&& other) = default; |
| |
| ~CallConfig(); |
| |
| RtpTransportConfig ExtractTransportConfig() const; |
| |
| Environment env; |
| |
| // Bitrate config used until valid bitrate estimates are calculated. Also |
| // used to cap total bitrate used. This comes from the remote connection. |
| BitrateConstraints bitrate_config; |
| |
| // AudioState which is possibly shared between multiple calls. |
| rtc::scoped_refptr<AudioState> audio_state; |
| |
| // Audio Processing Module to be used in this call. |
| AudioProcessing* audio_processing = nullptr; |
| |
| // FecController to use for this call. |
| FecControllerFactoryInterface* fec_controller_factory = nullptr; |
| |
| // NetworkStatePredictor to use for this call. |
| NetworkStatePredictorFactoryInterface* network_state_predictor_factory = |
| nullptr; |
| |
| // Call-specific Network controller factory to use. If this is set, it |
| // takes precedence over network_controller_factory. |
| std::unique_ptr<NetworkControllerFactoryInterface> |
| per_call_network_controller_factory; |
| // Network controller factory to use for this call if |
| // per_call_network_controller_factory is null. |
| NetworkControllerFactoryInterface* network_controller_factory = nullptr; |
| |
| // NetEq factory to use for this call. |
| NetEqFactory* neteq_factory = nullptr; |
| |
| TaskQueueBase* network_task_queue_ = nullptr; |
| // RtpTransportControllerSend to use for this call. |
| RtpTransportControllerSendFactoryInterface* |
| rtp_transport_controller_send_factory = nullptr; |
| |
| Metronome* decode_metronome = nullptr; |
| Metronome* encode_metronome = nullptr; |
| |
| // The burst interval of the pacer, see TaskQueuePacedSender constructor. |
| std::optional<TimeDelta> pacer_burst_interval; |
| |
| // Enables send packet batching from the egress RTP sender. |
| bool enable_send_packet_batching = false; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_CALL_CONFIG_H_ |