webrtc/modules/audio_processing/test/runtime_setting_util.cc
Fredrik Hernqvist ca362855e1 Add PlayoutVolumeChange RuntimeSetting.
Add a PlayoutVolumeChange RuntimeSetting. Trigger an echo path change when the playout volume is changed.

Bug: webrtc:10608
Change-Id: I1e736b93c1865d08c7d2582f6fe00216c1e1f72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135746
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Fredrik Hernqvist <fhernqvist@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27913}
2019-05-10 14:12:23 +00:00

41 lines
1.6 KiB
C++

/*
* Copyright (c) 2018 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/audio_processing/test/runtime_setting_util.h"
#include "rtc_base/checks.h"
namespace webrtc {
void ReplayRuntimeSetting(AudioProcessing* apm,
const webrtc::audioproc::RuntimeSetting& setting) {
RTC_CHECK(apm);
// TODO(bugs.webrtc.org/9138): Add ability to handle different types
// of settings. Currently CapturePreGain, CaptureFixedPostGain and
// PlayoutVolumeChange are supported.
RTC_CHECK(setting.has_capture_pre_gain() ||
setting.has_capture_fixed_post_gain() ||
setting.has_playout_volume_change());
if (setting.has_capture_pre_gain()) {
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(
setting.capture_pre_gain()));
} else if (setting.has_capture_fixed_post_gain()) {
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain(
setting.capture_fixed_post_gain()));
} else if (setting.has_playout_volume_change()) {
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(
setting.playout_volume_change()));
}
}
} // namespace webrtc