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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 <memory>
#include <optional>
#include "absl/base/nullability.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/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 "api/video/timing/video_jitter_timing_factory.h"
#include "call/audio_state.h"
#include "call/rtp_transport_config.h"
namespace webrtc {
class AudioProcessing;
struct CallConfig {
[[deprecated(
"Use CreateSingleThreaded or the multi-argument constructor instead.")]]
explicit CallConfig(const Environment& env,
TaskQueueBase* network_task_queue = nullptr);
CallConfig(const Environment& env,
TaskQueueBase* absl_nonnull worker_task_queue,
TaskQueueBase* absl_nonnull network_task_queue);
static CallConfig CreateWithJoinedWorkerAndNetworkQueue(
const Environment& env,
TaskQueueBase* absl_nonnull worker_and_network_queue);
// Creates a configuration for a single-threaded Call setup where signaling,
// worker, and network threads are all the same current thread.
// Note: This does not represent typical production configurations, where
// worker and network threads are usually separate background threads, but it
// is still a supported configuration (e.g. for testing or utility programs).
static CallConfig CreateSingleThreaded(const Environment& env);
// 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.
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;
VideoJitterTimingFactory* video_jitter_timing_factory = nullptr;
TaskQueueBase* network_task_queue_ = nullptr;
TaskQueueBase* worker_task_queue = 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_