| /* | 
 |  *  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 "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/transport/bitrate_settings.h" | 
 | #include "api/transport/network_control.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); | 
 |  | 
 |   CallConfig(const CallConfig&); | 
 |  | 
 |   ~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; | 
 |  | 
 |   // Network controller factory to use for this call. | 
 |   NetworkControllerFactoryInterface* network_controller_factory = nullptr; | 
 |  | 
 |   // NetEq factory to use for this call. | 
 |   NetEqFactory* neteq_factory = nullptr; | 
 |  | 
 |   TaskQueueBase* const 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. | 
 |   absl::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_ |