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1 /* | 1 /* |
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2017 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 <cstdio> |
| 12 #include <memory> |
| 13 |
11 #include "webrtc/modules/audio_processing/test/conversational_speech/config.h" | 14 #include "webrtc/modules/audio_processing/test/conversational_speech/config.h" |
| 15 #include "webrtc/modules/audio_processing/test/conversational_speech/timing.h" |
12 #include "webrtc/test/gtest.h" | 16 #include "webrtc/test/gtest.h" |
| 17 #include "webrtc/test/testsupport/fileutils.h" |
13 | 18 |
14 namespace webrtc { | 19 namespace webrtc { |
15 namespace test { | 20 namespace test { |
16 namespace { | 21 namespace { |
17 | 22 |
| 23 using conversational_speech::LoadTiming; |
| 24 using conversational_speech::SaveTiming; |
| 25 using conversational_speech::Turn; |
| 26 |
18 const char* const audiotracks_path = "/path/to/audiotracks"; | 27 const char* const audiotracks_path = "/path/to/audiotracks"; |
19 const char* const timing_filepath = "/path/to/timing_file.txt"; | 28 const char* const timing_filepath = "/path/to/timing_file.txt"; |
20 const char* const output_path = "/path/to/output_dir"; | 29 const char* const output_path = "/path/to/output_dir"; |
21 | 30 |
| 31 const std::vector<Turn> expected_timing = { |
| 32 {"A", "a1", 0}, |
| 33 {"B", "b1", 0}, |
| 34 {"A", "a2", 100}, |
| 35 {"B", "b2", -200}, |
| 36 {"A", "a3", 0}, |
| 37 {"A", "a4", 0}, |
| 38 }; |
| 39 const std::size_t kNumberOfTurns = expected_timing.size(); |
| 40 |
22 } // namespace | 41 } // namespace |
23 | 42 |
24 TEST(ConversationalSpeechTest, Settings) { | 43 TEST(ConversationalSpeechTest, Settings) { |
25 conversational_speech::Config config( | 44 const conversational_speech::Config config( |
26 audiotracks_path, timing_filepath, output_path); | 45 audiotracks_path, timing_filepath, output_path); |
27 | 46 |
28 // Test getters. | 47 // Test getters. |
29 EXPECT_EQ(config.audiotracks_path(), audiotracks_path); | 48 EXPECT_EQ(audiotracks_path, config.audiotracks_path()); |
30 EXPECT_EQ(config.timing_filepath(), timing_filepath); | 49 EXPECT_EQ(timing_filepath, config.timing_filepath()); |
31 EXPECT_EQ(config.output_path(), output_path); | 50 EXPECT_EQ(output_path, config.output_path()); |
| 51 } |
| 52 |
| 53 TEST(ConversationalSpeechTest, ExpectedTimingSize) { |
| 54 // Check the expected timing size. |
| 55 EXPECT_EQ(kNumberOfTurns, 6u); |
| 56 } |
| 57 |
| 58 TEST(ConversationalSpeechTest, TimingSaveLoad) { |
| 59 // Save test timing. |
| 60 const std::string temporary_filepath = webrtc::test::TempFilename( |
| 61 webrtc::test::OutputPath(), "TempTimingTestFile"); |
| 62 SaveTiming(temporary_filepath, expected_timing); |
| 63 |
| 64 // Create a std::vector<Turn> instance by loading from file. |
| 65 std::vector<Turn> actual_timing = LoadTiming(temporary_filepath); |
| 66 std::remove(temporary_filepath.c_str()); |
| 67 |
| 68 // Check size. |
| 69 EXPECT_EQ(expected_timing.size(), actual_timing.size()); |
| 70 |
| 71 // Check Turn instances. |
| 72 for (size_t index = 0; index < expected_timing.size(); ++index) { |
| 73 EXPECT_EQ(expected_timing[index], actual_timing[index]) |
| 74 << "turn #" << index << " not matching"; |
| 75 } |
32 } | 76 } |
33 | 77 |
34 } // namespace test | 78 } // namespace test |
35 } // namespace webrtc | 79 } // namespace webrtc |
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