1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
12
13 #include <limits.h>
14 #include <memory>
15
16 #include "testing/gmock/include/gmock/gmock.h"
17 #include "testing/gtest/include/gtest/gtest.h"
18 #include "webrtc/base/scoped_ptr.h"
19 #include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
20
21 using ::testing::_;
22 using ::testing::AtLeast;
23 using ::testing::InSequence;
24 using ::testing::Return;
25
26 namespace webrtc {
27
28 // The fake audio data is 0,1,..SCHAR_MAX-1,0,1,... This is to make it easy
29 // to detect errors. This function verifies that the buffers contain such data.
30 // E.g. if there are two buffers of size 3, buffer 1 would contain 0,1,2 and
31 // buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around
32 // will happen.
33 // |buffer| is the audio buffer to verify.
VerifyBuffer(const int8_t * buffer,int buffer_number,int size)34 bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
35 int start_value = (buffer_number * size) % SCHAR_MAX;
36 for (int i = 0; i < size; ++i) {
37 if (buffer[i] != (i + start_value) % SCHAR_MAX) {
38 return false;
39 }
40 }
41 return true;
42 }
43
44 // This function replaces the real AudioDeviceBuffer::GetPlayoutData when it's
45 // called (which is done implicitly when calling GetBufferData). It writes the
46 // sequence 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a
47 // buffer of different size than the one VerifyBuffer verifies.
48 // |iteration| is the number of calls made to UpdateBuffer prior to this call.
49 // |samples_per_10_ms| is the number of samples that should be written to the
50 // buffer (|arg0|).
ACTION_P2(UpdateBuffer,iteration,samples_per_10_ms)51 ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) {
52 int8_t* buffer = static_cast<int8_t*>(arg0);
53 int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
54 int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
55 for (int i = 0; i < bytes_per_10_ms; ++i) {
56 buffer[i] = (i + start_value) % SCHAR_MAX;
57 }
58 return samples_per_10_ms;
59 }
60
61 // Writes a periodic ramp pattern to the supplied |buffer|. See UpdateBuffer()
62 // for details.
UpdateInputBuffer(int8_t * buffer,int iteration,int size)63 void UpdateInputBuffer(int8_t* buffer, int iteration, int size) {
64 int start_value = (iteration * size) % SCHAR_MAX;
65 for (int i = 0; i < size; ++i) {
66 buffer[i] = (i + start_value) % SCHAR_MAX;
67 }
68 }
69
70 // Action macro which verifies that the recorded 10ms chunk of audio data
71 // (in |arg0|) contains the correct reference values even if they have been
72 // supplied using a buffer size that is smaller or larger than 10ms.
73 // See VerifyBuffer() for details.
ACTION_P2(VerifyInputBuffer,iteration,samples_per_10_ms)74 ACTION_P2(VerifyInputBuffer, iteration, samples_per_10_ms) {
75 const int8_t* buffer = static_cast<const int8_t*>(arg0);
76 int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
77 int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
78 for (int i = 0; i < bytes_per_10_ms; ++i) {
79 EXPECT_EQ(buffer[i], (i + start_value) % SCHAR_MAX);
80 }
81 return 0;
82 }
83
RunFineBufferTest(int sample_rate,int frame_size_in_samples)84 void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
85 const int kSamplesPer10Ms = sample_rate * 10 / 1000;
86 const int kFrameSizeBytes =
87 frame_size_in_samples * static_cast<int>(sizeof(int16_t));
88 const int kNumberOfFrames = 5;
89 // Ceiling of integer division: 1 + ((x - 1) / y)
90 const int kNumberOfUpdateBufferCalls =
91 1 + ((kNumberOfFrames * frame_size_in_samples - 1) / kSamplesPer10Ms);
92
93 MockAudioDeviceBuffer audio_device_buffer;
94 EXPECT_CALL(audio_device_buffer, RequestPlayoutData(_))
95 .WillRepeatedly(Return(kSamplesPer10Ms));
96 {
97 InSequence s;
98 for (int i = 0; i < kNumberOfUpdateBufferCalls; ++i) {
99 EXPECT_CALL(audio_device_buffer, GetPlayoutData(_))
100 .WillOnce(UpdateBuffer(i, kSamplesPer10Ms))
101 .RetiresOnSaturation();
102 }
103 }
104 {
105 InSequence s;
106 for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
107 EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
108 .WillOnce(VerifyInputBuffer(j, kSamplesPer10Ms))
109 .RetiresOnSaturation();
110 }
111 }
112 EXPECT_CALL(audio_device_buffer, SetVQEData(_, _, _))
113 .Times(kNumberOfUpdateBufferCalls - 1);
114 EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
115 .Times(kNumberOfUpdateBufferCalls - 1)
116 .WillRepeatedly(Return(kSamplesPer10Ms));
117
118 FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
119 sample_rate);
120
121 rtc::scoped_ptr<int8_t[]> out_buffer;
122 out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);
123 rtc::scoped_ptr<int8_t[]> in_buffer;
124 in_buffer.reset(new int8_t[kFrameSizeBytes]);
125 for (int i = 0; i < kNumberOfFrames; ++i) {
126 fine_buffer.GetPlayoutData(out_buffer.get());
127 EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
128 UpdateInputBuffer(in_buffer.get(), i, kFrameSizeBytes);
129 fine_buffer.DeliverRecordedData(in_buffer.get(), kFrameSizeBytes, 0, 0);
130 }
131 }
132
TEST(FineBufferTest,BufferLessThan10ms)133 TEST(FineBufferTest, BufferLessThan10ms) {
134 const int kSampleRate = 44100;
135 const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
136 const int kFrameSizeSamples = kSamplesPer10Ms - 50;
137 RunFineBufferTest(kSampleRate, kFrameSizeSamples);
138 }
139
TEST(FineBufferTest,GreaterThan10ms)140 TEST(FineBufferTest, GreaterThan10ms) {
141 const int kSampleRate = 44100;
142 const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
143 const int kFrameSizeSamples = kSamplesPer10Ms + 50;
144 RunFineBufferTest(kSampleRate, kFrameSizeSamples);
145 }
146
147 } // namespace webrtc
148