1 /* 2 * Copyright (c) 2015 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/common_audio/audio_ring_buffer.h" 12 13 #include "testing/gtest/include/gtest/gtest.h" 14 #include "webrtc/common_audio/channel_buffer.h" 15 16 namespace webrtc { 17 18 class AudioRingBufferTest : 19 public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > { 20 }; 21 22 void ReadAndWriteTest(const ChannelBuffer<float>& input, 23 size_t num_write_chunk_frames, 24 size_t num_read_chunk_frames, 25 size_t buffer_frames, 26 ChannelBuffer<float>* output) { 27 const size_t num_channels = input.num_channels(); 28 const size_t total_frames = input.num_frames(); 29 AudioRingBuffer buf(num_channels, buffer_frames); 30 rtc::scoped_ptr<float* []> slice(new float* [num_channels]); 31 32 size_t input_pos = 0; 33 size_t output_pos = 0; 34 while (input_pos + buf.WriteFramesAvailable() < total_frames) { 35 // Write until the buffer is as full as possible. 36 while (buf.WriteFramesAvailable() >= num_write_chunk_frames) { 37 buf.Write(input.Slice(slice.get(), input_pos), num_channels, 38 num_write_chunk_frames); 39 input_pos += num_write_chunk_frames; 40 } 41 // Read until the buffer is as empty as possible. 42 while (buf.ReadFramesAvailable() >= num_read_chunk_frames) { 43 EXPECT_LT(output_pos, total_frames); 44 buf.Read(output->Slice(slice.get(), output_pos), num_channels, 45 num_read_chunk_frames); 46 output_pos += num_read_chunk_frames; 47 } 48 } 49 50 // Write and read the last bit. 51 if (input_pos < total_frames) { 52 buf.Write(input.Slice(slice.get(), input_pos), num_channels, 53 total_frames - input_pos); 54 } 55 if (buf.ReadFramesAvailable()) { 56 buf.Read(output->Slice(slice.get(), output_pos), num_channels, 57 buf.ReadFramesAvailable()); 58 } 59 EXPECT_EQ(0u, buf.ReadFramesAvailable()); 60 } 61 62 TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) { 63 const size_t kFrames = 5000; 64 const size_t num_channels = ::testing::get<3>(GetParam()); 65 66 // Initialize the input data to an increasing sequence. 67 ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels)); 68 for (size_t i = 0; i < num_channels; ++i) 69 for (size_t j = 0; j < kFrames; ++j) 70 input.channels()[i][j] = (i + 1) * (j + 1); 71 72 ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels)); 73 ReadAndWriteTest(input, 74 ::testing::get<0>(GetParam()), 75 ::testing::get<1>(GetParam()), 76 ::testing::get<2>(GetParam()), 77 &output); 78 79 // Verify the read data matches the input. 80 for (size_t i = 0; i < num_channels; ++i) 81 for (size_t j = 0; j < kFrames; ++j) 82 EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]); 83 } 84 85 INSTANTIATE_TEST_CASE_P( 86 AudioRingBufferTest, AudioRingBufferTest, 87 ::testing::Combine(::testing::Values(10, 20, 42), // num_write_chunk_frames 88 ::testing::Values(1, 10, 17), // num_read_chunk_frames 89 ::testing::Values(100, 256), // buffer_frames 90 ::testing::Values(1, 4))); // num_channels 91 92 TEST_F(AudioRingBufferTest, MoveReadPosition) { 93 const size_t kNumChannels = 1; 94 const float kInputArray[] = {1, 2, 3, 4}; 95 const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray); 96 ChannelBuffer<float> input(kNumFrames, kNumChannels); 97 input.SetDataForTesting(kInputArray, kNumFrames); 98 AudioRingBuffer buf(kNumChannels, kNumFrames); 99 buf.Write(input.channels(), kNumChannels, kNumFrames); 100 101 buf.MoveReadPositionForward(3); 102 ChannelBuffer<float> output(1, kNumChannels); 103 buf.Read(output.channels(), kNumChannels, 1); 104 EXPECT_EQ(4, output.channels()[0][0]); 105 buf.MoveReadPositionBackward(3); 106 buf.Read(output.channels(), kNumChannels, 1); 107 EXPECT_EQ(2, output.channels()[0][0]); 108 } 109 110 } // namespace webrtc 111
