/* * 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_ #include 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 // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_