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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/vad/vad_audio_proc.h" | 15 #include "webrtc/modules/audio_processing/agc/agc_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 <string> | 20 #include "gtest/gtest.h" |
21 | 21 #include "webrtc/modules/audio_processing/agc/common.h" |
22 #include "testing/gtest/include/gtest/gtest.h" | |
23 #include "webrtc/modules/audio_processing/vad/common.h" | |
24 #include "webrtc/modules/interface/module_common_types.h" | 22 #include "webrtc/modules/interface/module_common_types.h" |
25 #include "webrtc/test/testsupport/fileutils.h" | 23 #include "webrtc/test/testsupport/fileutils.h" |
26 | 24 |
27 namespace webrtc { | 25 namespace webrtc { |
28 | 26 |
29 TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) { | 27 TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) { |
30 VadAudioProc audioproc; | 28 AgcAudioProc audioproc; |
31 | 29 |
32 std::string peak_file_name = | 30 std::string peak_file_name = |
33 test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat"); | 31 test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat"); |
34 FILE* peak_file = fopen(peak_file_name.c_str(), "rb"); | 32 FILE* peak_file = fopen(peak_file_name.c_str(), "rb"); |
35 ASSERT_TRUE(peak_file != NULL); | 33 ASSERT_TRUE(peak_file != NULL); |
36 | 34 |
37 std::string pcm_file_name = | 35 std::string pcm_file_name = |
38 test::ResourcePath("audio_processing/agc/agc_audio", "pcm"); | 36 test::ResourcePath("audio_processing/agc/agc_audio", "pcm"); |
39 FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb"); | 37 FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb"); |
40 ASSERT_TRUE(pcm_file != NULL); | 38 ASSERT_TRUE(pcm_file != NULL); |
41 | 39 |
42 // Read 10 ms audio in each iteration. | 40 // Read 10 ms audio in each iteration. |
43 const size_t kDataLength = kLength10Ms; | 41 const size_t kDataLength = kLength10Ms; |
44 int16_t data[kDataLength] = {0}; | 42 int16_t data[kDataLength] = { 0 }; |
45 AudioFeatures features; | 43 AudioFeatures features; |
46 double sp[kMaxNumFrames]; | 44 double sp[kMaxNumFrames]; |
47 while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) { | 45 while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) { |
48 audioproc.ExtractFeatures(data, kDataLength, &features); | 46 audioproc.ExtractFeatures(data, kDataLength, &features); |
49 if (features.num_frames > 0) { | 47 if (features.num_frames > 0) { |
50 ASSERT_LT(features.num_frames, kMaxNumFrames); | 48 ASSERT_LT(features.num_frames, kMaxNumFrames); |
51 // Read reference values. | 49 // Read reference values. |
52 const size_t num_frames = features.num_frames; | 50 const size_t num_frames = features.num_frames; |
53 ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file)); | 51 ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file)); |
54 for (int n = 0; n < features.num_frames; n++) | 52 for (int n = 0; n < features.num_frames; n++) |
55 EXPECT_NEAR(features.spectral_peak[n], sp[n], 3); | 53 EXPECT_NEAR(features.spectral_peak[n], sp[n], 3); |
56 } | 54 } |
57 } | 55 } |
58 | 56 |
59 fclose(peak_file); | 57 fclose(peak_file); |
60 fclose(pcm_file); | 58 fclose(pcm_file); |
61 } | 59 } |
62 | 60 |
63 } // namespace webrtc | 61 } // namespace webrtc |
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