Sebastian Jansson 470a5eae93 Introduces common AudioAllocationSettings class.
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}
2019-01-23 12:13:29 +00:00

38 lines
1.1 KiB
C++

/*
* Copyright (c) 2016 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.
*/
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
const char kBweTypeHistogram[] = "WebRTC.BWE.Types";
namespace congestion_controller {
int GetMinBitrateBps() {
constexpr int kMinBitrateBps = 5000;
return kMinBitrateBps;
}
DataRate GetMinBitrate() {
return DataRate::bps(GetMinBitrateBps());
}
} // namespace congestion_controller
RateControlInput::RateControlInput(
BandwidthUsage bw_state,
const absl::optional<DataRate>& estimated_throughput)
: bw_state(bw_state), estimated_throughput(estimated_throughput) {}
RateControlInput::~RateControlInput() = default;
} // namespace webrtc