This class collects the field trial based configuration of audio allocation and bandwidth in one place. This makes it easier overview and prepares for future cleanup of the trials. Bug: webrtc:9718 Change-Id: I34a441c0165b423f1e2ee63894337484684146ac Reviewed-on: https://webrtc-review.googlesource.com/c/118282 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26370}
38 lines
1.1 KiB
C++
38 lines
1.1 KiB
C++
/*
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* Copyright (c) 2016 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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#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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const char kBweTypeHistogram[] = "WebRTC.BWE.Types";
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namespace congestion_controller {
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int GetMinBitrateBps() {
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constexpr int kMinBitrateBps = 5000;
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return kMinBitrateBps;
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}
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DataRate GetMinBitrate() {
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return DataRate::bps(GetMinBitrateBps());
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}
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} // namespace congestion_controller
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RateControlInput::RateControlInput(
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BandwidthUsage bw_state,
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const absl::optional<DataRate>& estimated_throughput)
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: bw_state(bw_state), estimated_throughput(estimated_throughput) {}
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RateControlInput::~RateControlInput() = default;
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} // namespace webrtc
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