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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 // We don't test the value of pitch gain and lags as they are created by iSAC | 11 // We don't test the value of pitch gain and lags as they are created by iSAC |
12 // routines. However, interpolation of pitch-gain and lags is in a separate | 12 // routines. However, interpolation of pitch-gain and lags is in a separate |
13 // class and has its own unit-test. | 13 // class and has its own unit-test. |
14 | 14 |
15 #include "webrtc/modules/audio_processing/agc/agc_audio_proc.h" | 15 #include "webrtc/modules/audio_processing/vad/vad_audio_proc.h" |
16 | 16 |
17 #include <math.h> | 17 #include <math.h> |
18 #include <stdio.h> | 18 #include <stdio.h> |
19 | 19 |
20 #include "gtest/gtest.h" | 20 #include <string> |
21 #include "webrtc/modules/audio_processing/agc/common.h" | 21 |
| 22 #include "testing/gtest/include/gtest/gtest.h" |
| 23 #include "webrtc/modules/audio_processing/vad/common.h" |
22 #include "webrtc/modules/interface/module_common_types.h" | 24 #include "webrtc/modules/interface/module_common_types.h" |
23 #include "webrtc/test/testsupport/fileutils.h" | 25 #include "webrtc/test/testsupport/fileutils.h" |
24 | 26 |
25 namespace webrtc { | 27 namespace webrtc { |
26 | 28 |
27 TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) { | 29 TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) { |
28 AgcAudioProc audioproc; | 30 VadAudioProc audioproc; |
29 | 31 |
30 std::string peak_file_name = | 32 std::string peak_file_name = |
31 test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat"); | 33 test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat"); |
32 FILE* peak_file = fopen(peak_file_name.c_str(), "rb"); | 34 FILE* peak_file = fopen(peak_file_name.c_str(), "rb"); |
33 ASSERT_TRUE(peak_file != NULL); | 35 ASSERT_TRUE(peak_file != NULL); |
34 | 36 |
35 std::string pcm_file_name = | 37 std::string pcm_file_name = |
36 test::ResourcePath("audio_processing/agc/agc_audio", "pcm"); | 38 test::ResourcePath("audio_processing/agc/agc_audio", "pcm"); |
37 FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb"); | 39 FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb"); |
38 ASSERT_TRUE(pcm_file != NULL); | 40 ASSERT_TRUE(pcm_file != NULL); |
39 | 41 |
40 // Read 10 ms audio in each iteration. | 42 // Read 10 ms audio in each iteration. |
41 const size_t kDataLength = kLength10Ms; | 43 const size_t kDataLength = kLength10Ms; |
42 int16_t data[kDataLength] = { 0 }; | 44 int16_t data[kDataLength] = {0}; |
43 AudioFeatures features; | 45 AudioFeatures features; |
44 double sp[kMaxNumFrames]; | 46 double sp[kMaxNumFrames]; |
45 while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) { | 47 while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) { |
46 audioproc.ExtractFeatures(data, kDataLength, &features); | 48 audioproc.ExtractFeatures(data, kDataLength, &features); |
47 if (features.num_frames > 0) { | 49 if (features.num_frames > 0) { |
48 ASSERT_LT(features.num_frames, kMaxNumFrames); | 50 ASSERT_LT(features.num_frames, kMaxNumFrames); |
49 // Read reference values. | 51 // Read reference values. |
50 const size_t num_frames = features.num_frames; | 52 const size_t num_frames = features.num_frames; |
51 ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file)); | 53 ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file)); |
52 for (int n = 0; n < features.num_frames; n++) | 54 for (int n = 0; n < features.num_frames; n++) |
53 EXPECT_NEAR(features.spectral_peak[n], sp[n], 3); | 55 EXPECT_NEAR(features.spectral_peak[n], sp[n], 3); |
54 } | 56 } |
55 } | 57 } |
56 | 58 |
57 fclose(peak_file); | 59 fclose(peak_file); |
58 fclose(pcm_file); | 60 fclose(pcm_file); |
59 } | 61 } |
60 | 62 |
61 } // namespace webrtc | 63 } // namespace webrtc |
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