1 files changed, 32 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
new file mode 100644
index 00000000000..d094db0f9b3
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2014 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_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqPerformanceTest {
+ public:
+  // Runs a performance test with parameters as follows:
+  //   |runtime_ms|: the simulation time, i.e., the duration of the audio data.
+  //   |lossrate|: drop one out of |lossrate| packets, e.g., one out of 10.
+  //   |drift_factor|: clock drift in [0, 1].
+  // Returns the runtime in ms.
+  static int64_t Run(int runtime_ms, int lossrate, double drift_factor);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_