Index: webrtc/modules/audio_processing/test/audio_buffer_tools.h |
diff --git a/webrtc/modules/audio_processing/test/audio_buffer_tools.h b/webrtc/modules/audio_processing/test/audio_buffer_tools.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..654691c893852818440e6025b800469b95a1ed02 |
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+++ b/webrtc/modules/audio_processing/test/audio_buffer_tools.h |
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+/* |
+ * Copyright (c) 2015 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. |
+ */ |
+ |
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_ |
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_ |
+ |
+#include <vector> |
+#include "webrtc/modules/audio_processing/audio_buffer.h" |
+#include "webrtc/modules/audio_processing/include/audio_processing.h" |
+ |
+namespace webrtc { |
+namespace test { |
+ |
+// Copies a vector into an audiobuffer. |
+void CopyVectorToAudioBuffer(const StreamConfig& stream_config, |
+ const std::vector<float>& source, |
+ AudioBuffer* destination); |
+ |
+// Extracts a vector from an audiobuffer. |
+std::vector<float> ExtractVectorFromAudioBuffer( |
+ const StreamConfig& stream_config, |
+ AudioBuffer* source); |
+ |
+} // namespace test |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_ |