From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 21:33:14 +0200 Subject: Adding upstream version 115.7.0esr. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/call/call_config.h | 88 ++++++++++++++++++++++++++++++++ 1 file changed, 88 insertions(+) create mode 100644 third_party/libwebrtc/call/call_config.h (limited to 'third_party/libwebrtc/call/call_config.h') diff --git a/third_party/libwebrtc/call/call_config.h b/third_party/libwebrtc/call/call_config.h new file mode 100644 index 0000000000..6df4ab7ed4 --- /dev/null +++ b/third_party/libwebrtc/call/call_config.h @@ -0,0 +1,88 @@ +/* + * 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/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(RtcEventLog* event_log, + TaskQueueBase* network_task_queue = nullptr); + CallConfig(const CallConfig&); + RtpTransportConfig ExtractTransportConfig() const; + ~CallConfig(); + + // 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 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; + + // The burst interval of the pacer, see TaskQueuePacedSender constructor. + absl::optional pacer_burst_interval; +}; + +} // namespace webrtc + +#endif // CALL_CALL_CONFIG_H_ -- cgit v1.2.3