diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.cc | 50 |
1 files changed, 50 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.cc b/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.cc new file mode 100644 index 0000000000..4899d2d459 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.cc @@ -0,0 +1,50 @@ +/* + * 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 |