blob: 051228d75194487b155432e5d286f26831965569 [file] [log] [blame]
// Copyright 2018 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "media/webrtc/webrtc_switches.h"
#include "base/command_line.h"
#include "build/build_config.h"
namespace switches {
// Override the default minimum starting volume of the Automatic Gain Control
// algorithm in WebRTC used with audio tracks from getUserMedia.
// The valid range is 12-255. Values outside that range will be clamped
// to the lowest or highest valid value inside WebRTC.
// TODO(tommi): Remove this switch when is fixed.
const char kAgcStartupMinVolume[] = "agc-startup-min-volume";
} // namespace switches
namespace features {
// When enabled we will tell WebRTC that we want to use the
// Windows.Graphics.Capture API based DesktopCapturer, if it is available.
const base::Feature kWebRtcAllowWgcDesktopCapturer{
"AllowWgcDesktopCapturer", base::FEATURE_DISABLED_BY_DEFAULT};
// Enables multichannel capture audio to be processed without downmixing in the
// WebRTC audio processing module.
const base::Feature kWebRtcEnableCaptureMultiChannelApm{
"WebRtcEnableCaptureMultiChannelApm", base::FEATURE_ENABLED_BY_DEFAULT};
// Kill-switch allowing deactivation of the support for 48 kHz internal
// processing in the WebRTC audio processing module when running on an ARM
// platform.
const base::Feature kWebRtcAllow48kHzProcessingOnArm{
"WebRtcAllow48kHzProcessingOnArm", base::FEATURE_ENABLED_BY_DEFAULT};
// Enables the WebRTC Agc2 digital adaptation with WebRTC Agc1 analog
// adaptation. Feature for Is sent to WebRTC.
const base::Feature kWebRtcHybridAgc{"WebRtcHybridAgc",
// Enables and configures the clipping control in the WebRTC analog AGC.
const base::Feature kWebRtcAnalogAgcClippingControl{
"WebRtcAnalogAgcClippingControl", base::FEATURE_DISABLED_BY_DEFAULT};
} // namespace features