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Side by Side Diff: webrtc/voice_engine/test/auto_test/voe_output_test.cc

Issue 1457023002: Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include "testing/gtest/include/gtest/gtest.h" 11 #include "testing/gtest/include/gtest/gtest.h"
12 #include "webrtc/base/random.h"
12 #include "webrtc/base/scoped_ptr.h" 13 #include "webrtc/base/scoped_ptr.h"
13 #include "webrtc/base/timeutils.h" 14 #include "webrtc/base/timeutils.h"
14 #include "webrtc/system_wrappers/include/sleep.h" 15 #include "webrtc/system_wrappers/include/sleep.h"
15 #include "webrtc/test/channel_transport/channel_transport.h" 16 #include "webrtc/test/channel_transport/channel_transport.h"
16 #include "webrtc/test/random.h"
17 #include "webrtc/test/testsupport/fileutils.h" 17 #include "webrtc/test/testsupport/fileutils.h"
18 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h" 18 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
19 19
20 namespace { 20 namespace {
21 21
22 const char kIp[] = "127.0.0.1"; 22 const char kIp[] = "127.0.0.1";
23 const int kPort = 1234; 23 const int kPort = 1234;
24 const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000}; 24 const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
25 25
26 } // namespace 26 } // namespace
27 27
28 namespace voetest { 28 namespace voetest {
29 29
30 using webrtc::test::Random; 30 using webrtc::Random;
31 using webrtc::test::VoiceChannelTransport; 31 using webrtc::test::VoiceChannelTransport;
32 32
33 // This test allows a check on the output signal in an end-to-end call. 33 // This test allows a check on the output signal in an end-to-end call.
34 class OutputTest { 34 class OutputTest {
35 public: 35 public:
36 OutputTest(int16_t lower_bound, int16_t upper_bound); 36 OutputTest(int16_t lower_bound, int16_t upper_bound);
37 ~OutputTest(); 37 ~OutputTest();
38 38
39 void Start(); 39 void Start();
40 40
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194 test.EnableOutputCheck(); 194 test.EnableOutputCheck();
195 webrtc::SleepMs(random.Rand(kCheckTimeMs - kCheckTimeMs / 10, 195 webrtc::SleepMs(random.Rand(kCheckTimeMs - kCheckTimeMs / 10,
196 kCheckTimeMs + kCheckTimeMs / 10)); 196 kCheckTimeMs + kCheckTimeMs / 10));
197 test.DisableOutputCheck(); 197 test.DisableOutputCheck();
198 test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate)); 198 test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate));
199 test.Unmute(); 199 test.Unmute();
200 } 200 }
201 } 201 }
202 202
203 } // namespace voetest 203 } // namespace voetest
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