Index: content/renderer/media/webrtc_audio_processor_util.h |
diff --git a/content/renderer/media/webrtc_audio_processor_util.h b/content/renderer/media/webrtc_audio_processor_util.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..b2bb4e1be7b3813f98920acd0818c99d8993b64d |
--- /dev/null |
+++ b/content/renderer/media/webrtc_audio_processor_util.h |
@@ -0,0 +1,43 @@ |
+// Copyright 2013 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. |
+ |
+#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_UTIL_H_ |
+#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_UTIL_H_ |
+ |
+#include <string> |
+ |
+namespace webrtc { |
+ |
+class AudioFrame; |
+class AudioProcessing; |
+class MediaConstraintsInterface; |
+ |
+} |
+ |
+namespace content { |
+ |
+using webrtc::AudioProcessing; |
+using webrtc::MediaConstraintsInterface; |
+ |
+bool GetPropertyFromConstraints( |
+ const MediaConstraintsInterface* constraints, |
+ const std::string& key); |
+ |
+void EnableEchoCancellation(AudioProcessing* audio_processing); |
+ |
+void EnableNoiseSuppression(AudioProcessing* audio_processing); |
+ |
+void EnableHighPassFilter(AudioProcessing* audio_processing); |
+ |
+void EnableTypingDetection(AudioProcessing* audio_processing); |
+ |
+void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing); |
+ |
+void StartAecDump(AudioProcessing* audio_processin); |
+ |
+void StopAecDump(AudioProcessing* audio_processin); |
+ |
+} // namespace content |
+ |
+#endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_UTIL_H_ |