Repo created

TMessagesProj/jni/voip/webrtc/call/rtp_transport_config.h

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+/*
+ *  Copyright (c) 2021 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_RTP_TRANSPORT_CONFIG_H_
+#define CALL_RTP_TRANSPORT_CONFIG_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/environment/environment.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/bitrate_settings.h"
+#include "api/transport/network_control.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+struct RtpTransportConfig {
+  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;
+
+  // 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;
+
+  // The burst interval of the pacer, see TaskQueuePacedSender constructor.
+  absl::optional<TimeDelta> pacer_burst_interval;
+};
+}  // namespace webrtc
+
+#endif  // CALL_RTP_TRANSPORT_CONFIG_H_