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