mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00

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}
69 lines
3.1 KiB
Text
69 lines
3.1 KiB
Text
/*
|
|
* Copyright 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.
|
|
*/
|
|
|
|
#import "WebRTC/RTCFieldTrials.h"
|
|
|
|
#include <memory>
|
|
|
|
#import "WebRTC/RTCLogging.h"
|
|
|
|
// Adding 'nogncheck' to disable the gn include headers check.
|
|
// We don't want to depend on 'system_wrappers:field_trial_default' because
|
|
// clients should be able to provide their own implementation.
|
|
#include "system_wrappers/include/field_trial_default.h" // nogncheck
|
|
|
|
NSString * const kRTCFieldTrialAudioSendSideBweKey = @"WebRTC-Audio-SendSideBwe";
|
|
NSString * const kRTCFieldTrialSendSideBweWithOverheadKey = @"WebRTC-SendSideBwe-WithOverhead";
|
|
NSString * const kRTCFieldTrialFlexFec03AdvertisedKey = @"WebRTC-FlexFEC-03-Advertised";
|
|
NSString * const kRTCFieldTrialFlexFec03Key = @"WebRTC-FlexFEC-03";
|
|
NSString * const kRTCFieldTrialImprovedBitrateEstimateKey = @"WebRTC-ImprovedBitrateEstimate";
|
|
NSString * const kRTCFieldTrialMedianSlopeFilterKey = @"WebRTC-BweMedianSlopeFilter";
|
|
NSString * const kRTCFieldTrialTrendlineFilterKey = @"WebRTC-BweTrendlineFilter";
|
|
NSString * const kRTCFieldTrialH264HighProfileKey = @"WebRTC-H264HighProfile";
|
|
NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey =
|
|
@"WebRTC-Audio-MinimizeResamplingOnMobile";
|
|
NSString * const kRTCFieldTrialEnabledValue = @"Enabled";
|
|
|
|
static std::unique_ptr<char[]> gFieldTrialInitString;
|
|
|
|
NSString *RTCFieldTrialMedianSlopeFilterValue(
|
|
size_t windowSize, double thresholdGain) {
|
|
NSString *format = @"Enabled-%zu,%lf";
|
|
return [NSString stringWithFormat:format, windowSize, thresholdGain];
|
|
}
|
|
|
|
NSString *RTCFieldTrialTrendlineFilterValue(
|
|
size_t windowSize, double smoothingCoeff, double thresholdGain) {
|
|
NSString *format = @"Enabled-%zu,%lf,%lf";
|
|
return [NSString stringWithFormat:format, windowSize, smoothingCoeff, thresholdGain];
|
|
}
|
|
|
|
void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials) {
|
|
if (!fieldTrials) {
|
|
RTCLogWarning(@"No fieldTrials provided.");
|
|
return;
|
|
}
|
|
// Assemble the keys and values into the field trial string.
|
|
// We don't perform any extra format checking. That should be done by the underlying WebRTC calls.
|
|
NSMutableString *fieldTrialInitString = [NSMutableString string];
|
|
for (NSString *key in fieldTrials) {
|
|
NSString *fieldTrialEntry = [NSString stringWithFormat:@"%@/%@/", key, fieldTrials[key]];
|
|
[fieldTrialInitString appendString:fieldTrialEntry];
|
|
}
|
|
size_t len = fieldTrialInitString.length + 1;
|
|
gFieldTrialInitString.reset(new char[len]);
|
|
if (![fieldTrialInitString getCString:gFieldTrialInitString.get()
|
|
maxLength:len
|
|
encoding:NSUTF8StringEncoding]) {
|
|
RTCLogError(@"Failed to convert field trial string.");
|
|
return;
|
|
}
|
|
webrtc::field_trial::InitFieldTrialsFromString(gFieldTrialInitString.get());
|
|
}
|