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cf2856b01c
BurstyPacer is currently controlled via field trials. In order for Chrome to be able to have burst without relying on a field trial, this parameter is added. When all burst experiments have concluded we may be able to have a hardcoded constant instead, but for now the parameter is added to RTCConfiguration. NOTRY=True Bug: chromium:1354491 Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38621}
88 lines
3 KiB
C++
88 lines
3 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_CALL_CONFIG_H_
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#define CALL_CALL_CONFIG_H_
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#include "api/fec_controller.h"
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#include "api/field_trials_view.h"
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#include "api/metronome/metronome.h"
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#include "api/neteq/neteq_factory.h"
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#include "api/network_state_predictor.h"
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#include "api/rtc_error.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "api/transport/bitrate_settings.h"
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#include "api/transport/network_control.h"
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#include "call/audio_state.h"
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#include "call/rtp_transport_config.h"
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#include "call/rtp_transport_controller_send_factory_interface.h"
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namespace webrtc {
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class AudioProcessing;
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class RtcEventLog;
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struct CallConfig {
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// If `network_task_queue` is set to nullptr, Call will assume that network
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// related callbacks will be made on the same TQ as the Call instance was
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// constructed on.
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explicit CallConfig(RtcEventLog* event_log,
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TaskQueueBase* network_task_queue = nullptr);
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CallConfig(const CallConfig&);
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RtpTransportConfig ExtractTransportConfig() const;
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~CallConfig();
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// Bitrate config used until valid bitrate estimates are calculated. Also
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// used to cap total bitrate used. This comes from the remote connection.
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BitrateConstraints bitrate_config;
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// AudioState which is possibly shared between multiple calls.
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rtc::scoped_refptr<AudioState> audio_state;
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// Audio Processing Module to be used in this call.
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AudioProcessing* audio_processing = nullptr;
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// RtcEventLog to use for this call. Required.
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// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
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RtcEventLog* const event_log = nullptr;
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// FecController to use for this call.
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FecControllerFactoryInterface* fec_controller_factory = nullptr;
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// Task Queue Factory to be used in this call. Required.
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TaskQueueFactory* task_queue_factory = nullptr;
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// NetworkStatePredictor to use for this call.
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NetworkStatePredictorFactoryInterface* network_state_predictor_factory =
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nullptr;
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// Network controller factory to use for this call.
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NetworkControllerFactoryInterface* network_controller_factory = nullptr;
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// NetEq factory to use for this call.
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NetEqFactory* neteq_factory = nullptr;
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// Key-value mapping of internal configurations to apply,
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// e.g. field trials.
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const FieldTrialsView* trials = nullptr;
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TaskQueueBase* const network_task_queue_ = nullptr;
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// RtpTransportControllerSend to use for this call.
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RtpTransportControllerSendFactoryInterface*
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rtp_transport_controller_send_factory = nullptr;
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Metronome* metronome = nullptr;
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// The burst interval of the pacer, see TaskQueuePacedSender constructor.
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absl::optional<TimeDelta> pacer_burst_interval;
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};
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} // namespace webrtc
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#endif // CALL_CALL_CONFIG_H_
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