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/*
* 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 "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/fec_controller.h"
#include "api/field_trials_view.h"
#include "api/metronome/metronome.h"
#include "api/neteq/neteq_factory.h"
#include "api/network_state_predictor.h"
#include "api/rtc_error.h"
#include "api/task_queue/task_queue_factory.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;
class RtcEventLog;
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);
// TODO(bugs.webrtc.org/15656): Deprecate and delete constructor below.
explicit CallConfig(RtcEventLog* event_log,
TaskQueueBase* network_task_queue = nullptr);
CallConfig(const CallConfig&);
~CallConfig();
RtpTransportConfig ExtractTransportConfig() const;
// TODO(bugs.webrtc.org/15656): Make non-optional when constructor that
// doesn't pass Environment is removed.
absl::optional<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;
// RtcEventLog to use for this call. Required.
// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
RtcEventLog* const event_log = nullptr;
// FecController to use for this call.
FecControllerFactoryInterface* fec_controller_factory = nullptr;
// Task Queue Factory to be used in this call. Required.
TaskQueueFactory* task_queue_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;
// Key-value mapping of internal configurations to apply,
// e.g. field trials.
const FieldTrialsView* trials = nullptr;
TaskQueueBase* const network_task_queue_ = nullptr;
// RtpTransportControllerSend to use for this call.
RtpTransportControllerSendFactoryInterface*
rtp_transport_controller_send_factory = nullptr;
Metronome* metronome = nullptr;
// Enables send packet batching from the egress RTP sender.
bool enable_send_packet_batching = false;
};
} // namespace webrtc
#endif // CALL_CALL_CONFIG_H_
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