andrew@webrtc.org | 041035b | 2015-01-26 21:23:53 +0000 | [diff] [blame] | 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/modules/audio_processing/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.samples_per_channel(); |
| 29 | AudioRingBuffer buf(num_channels, buffer_frames); |
| 30 | 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(), static_cast<int>(input_pos)), |
| 38 | num_channels, 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(), static_cast<int>(output_pos)), |
| 45 | num_channels, 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(), static_cast<int>(input_pos)), |
| 53 | num_channels, total_frames - input_pos); |
| 54 | if (buf.ReadFramesAvailable()) |
| 55 | buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)), |
| 56 | num_channels, buf.ReadFramesAvailable()); |
| 57 | EXPECT_EQ(0u, buf.ReadFramesAvailable()); |
| 58 | } |
| 59 | |
| 60 | TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) { |
| 61 | const size_t kFrames = 5000; |
| 62 | const size_t num_channels = ::testing::get<3>(GetParam()); |
| 63 | |
| 64 | // Initialize the input data to an increasing sequence. |
| 65 | ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels)); |
| 66 | for (size_t i = 0; i < num_channels; ++i) |
| 67 | for (size_t j = 0; j < kFrames; ++j) |
andrew@webrtc.org | 922cfcd | 2015-01-27 21:59:33 +0000 | [diff] [blame^] | 68 | input.channels()[i][j] = (i + 1) * (j + 1); |
andrew@webrtc.org | 041035b | 2015-01-26 21:23:53 +0000 | [diff] [blame] | 69 | |
| 70 | ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels)); |
| 71 | ReadAndWriteTest(input, |
| 72 | ::testing::get<0>(GetParam()), |
| 73 | ::testing::get<1>(GetParam()), |
| 74 | ::testing::get<2>(GetParam()), |
| 75 | &output); |
| 76 | |
| 77 | // Verify the read data matches the input. |
| 78 | for (size_t i = 0; i < num_channels; ++i) |
| 79 | for (size_t j = 0; j < kFrames; ++j) |
| 80 | EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]); |
| 81 | } |
| 82 | |
| 83 | INSTANTIATE_TEST_CASE_P( |
| 84 | AudioRingBufferTest, AudioRingBufferTest, |
| 85 | ::testing::Combine(::testing::Values(10, 20, 42), // num_write_chunk_frames |
| 86 | ::testing::Values(1, 10, 17), // num_read_chunk_frames |
| 87 | ::testing::Values(100, 256), // buffer_frames |
| 88 | ::testing::Values(1, 4))); // num_channels |
| 89 | |
| 90 | TEST_F(AudioRingBufferTest, MoveReadPosition) { |
| 91 | const size_t kNumChannels = 1; |
| 92 | const float kInputArray[] = {1, 2, 3, 4}; |
| 93 | const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray); |
| 94 | ChannelBuffer<float> input(kInputArray, kNumFrames, kNumChannels); |
| 95 | AudioRingBuffer buf(kNumChannels, kNumFrames); |
| 96 | buf.Write(input.channels(), kNumChannels, kNumFrames); |
| 97 | |
| 98 | buf.MoveReadPosition(3); |
| 99 | ChannelBuffer<float> output(1, kNumChannels); |
| 100 | buf.Read(output.channels(), kNumChannels, 1); |
| 101 | EXPECT_EQ(4, output.data()[0]); |
| 102 | buf.MoveReadPosition(-3); |
| 103 | buf.Read(output.channels(), kNumChannels, 1); |
| 104 | EXPECT_EQ(2, output.data()[0]); |
| 105 | } |
| 106 | |
| 107 | } // namespace webrtc |