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/android/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/modules/audio_device/mock_audio_device_buffer.h" 19 #include "webrtc/system_wrappers/interface/scoped_ptr.h" 20 21 using ::testing::_; 22 using ::testing::InSequence; 23 using ::testing::Return; 24 25 namespace webrtc { 26 27 // The fake audio data is 0,1,..SCHAR_MAX-1,0,1,... This is to make it easy 28 // to detect errors. This function verifies that the buffers contain such data. 29 // E.g. if there are two buffers of size 3, buffer 1 would contain 0,1,2 and 30 // buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around 31 // will happen. 32 // |buffer| is the audio buffer to verify. 33 bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) { 34 int start_value = (buffer_number * size) % SCHAR_MAX; 35 for (int i = 0; i < size; ++i) { 36 if (buffer[i] != (i + start_value) % SCHAR_MAX) { 37 return false; 38 } 39 } 40 return true; 41 } 42 43 // This function replaces GetPlayoutData when it's called (which is done 44 // implicitly when calling GetBufferData). It writes the sequence 45 // 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a buffer of 46 // different size than the one VerifyBuffer verifies. 47 // |iteration| is the number of calls made to UpdateBuffer prior to this call. 48 // |samples_per_10_ms| is the number of samples that should be written to the 49 // buffer (|arg0|). 50 ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) { 51 int8_t* buffer = static_cast<int8_t*>(arg0); 52 int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t)); 53 int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX; 54 for (int i = 0; i < bytes_per_10_ms; ++i) { 55 buffer[i] = (i + start_value) % SCHAR_MAX; 56 } 57 return samples_per_10_ms; 58 } 59 60 void RunFineBufferTest(int sample_rate, int frame_size_in_samples) { 61 const int kSamplesPer10Ms = sample_rate * 10 / 1000; 62 const int kFrameSizeBytes = frame_size_in_samples * 63 static_cast<int>(sizeof(int16_t)); 64 const int kNumberOfFrames = 5; 65 // Ceiling of integer division: 1 + ((x - 1) / y) 66 const int kNumberOfUpdateBufferCalls = 67 1 + ((kNumberOfFrames * frame_size_in_samples - 1) / kSamplesPer10Ms); 68 69 MockAudioDeviceBuffer audio_device_buffer; 70 EXPECT_CALL(audio_device_buffer, RequestPlayoutData(_)) 71 .WillRepeatedly(Return(kSamplesPer10Ms)); 72 { 73 InSequence s; 74 for (int i = 0; i < kNumberOfUpdateBufferCalls; ++i) { 75 EXPECT_CALL(audio_device_buffer, GetPlayoutData(_)) 76 .WillOnce(UpdateBuffer(i, kSamplesPer10Ms)) 77 .RetiresOnSaturation(); 78 } 79 } 80 FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes, 81 sample_rate); 82 83 scoped_ptr<int8_t[]> out_buffer; 84 out_buffer.reset( 85 new int8_t[fine_buffer.RequiredBufferSizeBytes()]); 86 for (int i = 0; i < kNumberOfFrames; ++i) { 87 fine_buffer.GetBufferData(out_buffer.get()); 88 EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes)); 89 } 90 } 91 92 TEST(FineBufferTest, BufferLessThan10ms) { 93 const int kSampleRate = 44100; 94 const int kSamplesPer10Ms = kSampleRate * 10 / 1000; 95 const int kFrameSizeSamples = kSamplesPer10Ms - 50; 96 RunFineBufferTest(kSampleRate, kFrameSizeSamples); 97 } 98 99 TEST(FineBufferTest, GreaterThan10ms) { 100 const int kSampleRate = 44100; 101 const int kSamplesPer10Ms = kSampleRate * 10 / 1000; 102 const int kFrameSizeSamples = kSamplesPer10Ms + 50; 103 RunFineBufferTest(kSampleRate, kFrameSizeSamples); 104 } 105 106 } // namespace webrtc 107
