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In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
40 lines
1.2 KiB
C++
40 lines
1.2 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 kAudioMinBitrateBps = 5000;
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constexpr int kMinBitrateBps = 10000;
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if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
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return kAudioMinBitrateBps;
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}
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return kMinBitrateBps;
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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 rtc::Optional<uint32_t>& incoming_bitrate,
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double noise_var)
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: bw_state(bw_state),
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incoming_bitrate(incoming_bitrate),
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noise_var(noise_var) {}
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RateControlInput::~RateControlInput() = default;
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} // namespace webrtc
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