modules/audio_processing/test/runtime_setting_util.cc

/*
 *  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()));
  } else if (setting.has_playout_audio_device_change()) {
    apm->SetRuntimeSetting(
        AudioProcessing::RuntimeSetting::CreatePlayoutAudioDeviceChange(
            {setting.playout_audio_device_change().id(),
             setting.playout_audio_device_change().max_volume()}));
  } else if (setting.has_capture_output_used()) {
    apm->SetRuntimeSetting(
        AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
            setting.capture_output_used()));
  }
}
}  // namespace webrtc