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Bug: webrtc:12338 Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34621}
32 lines
1.1 KiB
C++
32 lines
1.1 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
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#include <stdint.h>
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namespace webrtc {
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namespace test {
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class NetEqPerformanceTest {
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public:
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// Runs a performance test with parameters as follows:
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// `runtime_ms`: the simulation time, i.e., the duration of the audio data.
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// `lossrate`: drop one out of `lossrate` packets, e.g., one out of 10.
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// `drift_factor`: clock drift in [0, 1].
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// Returns the runtime in ms.
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static int64_t Run(int runtime_ms, int lossrate, double drift_factor);
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
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