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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
58 lines
2 KiB
C++
58 lines
2 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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#include "modules/audio_processing/agc2/agc2_common.h"
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#include <stdio.h>
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#include <string>
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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float GetInitialSaturationMarginDb() {
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constexpr char kForceInitialSaturationMarginFieldTrial[] =
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"WebRTC-Audio-Agc2ForceInitialSaturationMargin";
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const bool use_forced_initial_saturation_margin =
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webrtc::field_trial::IsEnabled(kForceInitialSaturationMarginFieldTrial);
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if (use_forced_initial_saturation_margin) {
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const std::string field_trial_string = webrtc::field_trial::FindFullName(
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kForceInitialSaturationMarginFieldTrial);
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float margin_db = -1;
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if (sscanf(field_trial_string.c_str(), "Enabled-%f", &margin_db) == 1 &&
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margin_db >= 12.f && margin_db <= 25.f) {
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return margin_db;
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}
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}
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constexpr float kDefaultInitialSaturationMarginDb = 20.f;
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return kDefaultInitialSaturationMarginDb;
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}
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float GetExtraSaturationMarginOffsetDb() {
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constexpr char kForceExtraSaturationMarginFieldTrial[] =
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"WebRTC-Audio-Agc2ForceExtraSaturationMargin";
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const bool use_forced_extra_saturation_margin =
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webrtc::field_trial::IsEnabled(kForceExtraSaturationMarginFieldTrial);
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if (use_forced_extra_saturation_margin) {
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const std::string field_trial_string = webrtc::field_trial::FindFullName(
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kForceExtraSaturationMarginFieldTrial);
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float margin_db = -1;
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if (sscanf(field_trial_string.c_str(), "Enabled-%f", &margin_db) == 1 &&
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margin_db >= 0.f && margin_db <= 10.f) {
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return margin_db;
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}
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}
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constexpr float kDefaultExtraSaturationMarginDb = 2.f;
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return kDefaultExtraSaturationMarginDb;
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}
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} // namespace webrtc
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