This CL removes direct access to SendSideCongestionController (SSCC) via the RtpTransportControllerSend interface and replaces all usages with calls on RtpTransportControllerSend which will in turn calls SSCC. This prepares for later refactor of RtpTransportControllerSend. Bug: webrtc:8415 Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5 Reviewed-on: https://webrtc-review.googlesource.com/53860 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22044}
35 lines
1.4 KiB
C++
35 lines
1.4 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 MODULES_CONGESTION_CONTROLLER_INCLUDE_NETWORK_CHANGED_OBSERVER_H_
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#define MODULES_CONGESTION_CONTROLLER_INCLUDE_NETWORK_CHANGED_OBSERVER_H_
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#include <stdint.h>
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namespace webrtc {
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// Observer class for bitrate changes announced due to change in bandwidth
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// estimate or due to that the send pacer is full. Fraction loss and rtt is
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// also part of this callback to allow the observer to optimize its settings
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// for different types of network environments. The bitrate does not include
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// packet headers and is measured in bits per second.
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class NetworkChangedObserver {
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public:
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virtual void OnNetworkChanged(uint32_t bitrate_bps,
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uint8_t fraction_loss, // 0 - 255.
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int64_t rtt_ms,
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int64_t probing_interval_ms) = 0;
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protected:
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virtual ~NetworkChangedObserver() {}
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};
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} // namespace webrtc
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#endif // MODULES_CONGESTION_CONTROLLER_INCLUDE_NETWORK_CHANGED_OBSERVER_H_
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