1 files changed, 37 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
new file mode 100644
index 00000000000..d9210668dde
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2012 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_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+
+#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
+
+#include "gmock/gmock.h"
+
+namespace webrtc {
+
+class MockBufferLevelFilter : public BufferLevelFilter {
+ public:
+  virtual ~MockBufferLevelFilter() { Die(); }
+  MOCK_METHOD0(Die,
+      void());
+  MOCK_METHOD0(Reset,
+      void());
+  MOCK_METHOD3(Update,
+      void(int buffer_size_packets, int time_stretched_samples,
+           int packet_len_samples));
+  MOCK_METHOD1(SetTargetBufferLevel,
+      void(int target_buffer_level));
+  MOCK_CONST_METHOD0(filtered_current_level,
+      int());
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_