| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <memory> |
| |
| #include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h" |
| |
| #include "common_audio/channel_buffer.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| |
| class AudioRingBufferTest |
| : public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > { |
| }; |
| |
| void ReadAndWriteTest(const ChannelBuffer<float>& input, |
| size_t num_write_chunk_frames, |
| size_t num_read_chunk_frames, |
| size_t buffer_frames, |
| ChannelBuffer<float>* output) { |
| const size_t num_channels = input.num_channels(); |
| const size_t total_frames = input.num_frames(); |
| AudioRingBuffer buf(num_channels, buffer_frames); |
| std::unique_ptr<float* []> slice(new float*[num_channels]); |
| |
| size_t input_pos = 0; |
| size_t output_pos = 0; |
| while (input_pos + buf.WriteFramesAvailable() < total_frames) { |
| // Write until the buffer is as full as possible. |
| while (buf.WriteFramesAvailable() >= num_write_chunk_frames) { |
| buf.Write(input.Slice(slice.get(), input_pos), num_channels, |
| num_write_chunk_frames); |
| input_pos += num_write_chunk_frames; |
| } |
| // Read until the buffer is as empty as possible. |
| while (buf.ReadFramesAvailable() >= num_read_chunk_frames) { |
| EXPECT_LT(output_pos, total_frames); |
| buf.Read(output->Slice(slice.get(), output_pos), num_channels, |
| num_read_chunk_frames); |
| output_pos += num_read_chunk_frames; |
| } |
| } |
| |
| // Write and read the last bit. |
| if (input_pos < total_frames) { |
| buf.Write(input.Slice(slice.get(), input_pos), num_channels, |
| total_frames - input_pos); |
| } |
| if (buf.ReadFramesAvailable()) { |
| buf.Read(output->Slice(slice.get(), output_pos), num_channels, |
| buf.ReadFramesAvailable()); |
| } |
| EXPECT_EQ(0u, buf.ReadFramesAvailable()); |
| } |
| |
| TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) { |
| const size_t kFrames = 5000; |
| const size_t num_channels = ::testing::get<3>(GetParam()); |
| |
| // Initialize the input data to an increasing sequence. |
| ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels)); |
| for (size_t i = 0; i < num_channels; ++i) |
| for (size_t j = 0; j < kFrames; ++j) |
| input.channels()[i][j] = (i + 1) * (j + 1); |
| |
| ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels)); |
| ReadAndWriteTest(input, ::testing::get<0>(GetParam()), |
| ::testing::get<1>(GetParam()), ::testing::get<2>(GetParam()), |
| &output); |
| |
| // Verify the read data matches the input. |
| for (size_t i = 0; i < num_channels; ++i) |
| for (size_t j = 0; j < kFrames; ++j) |
| EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P( |
| AudioRingBufferTest, |
| AudioRingBufferTest, |
| ::testing::Combine(::testing::Values(10, 20, 42), // num_write_chunk_frames |
| ::testing::Values(1, 10, 17), // num_read_chunk_frames |
| ::testing::Values(100, 256), // buffer_frames |
| ::testing::Values(1, 4))); // num_channels |
| |
| TEST_F(AudioRingBufferTest, MoveReadPosition) { |
| const size_t kNumChannels = 1; |
| const float kInputArray[] = {1, 2, 3, 4}; |
| const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray); |
| ChannelBuffer<float> input(kNumFrames, kNumChannels); |
| input.SetDataForTesting(kInputArray, kNumFrames); |
| AudioRingBuffer buf(kNumChannels, kNumFrames); |
| buf.Write(input.channels(), kNumChannels, kNumFrames); |
| |
| buf.MoveReadPositionForward(3); |
| ChannelBuffer<float> output(1, kNumChannels); |
| buf.Read(output.channels(), kNumChannels, 1); |
| EXPECT_EQ(4, output.channels()[0][0]); |
| buf.MoveReadPositionBackward(3); |
| buf.Read(output.channels(), kNumChannels, 1); |
| EXPECT_EQ(2, output.channels()[0][0]); |
| } |
| |
| } // namespace webrtc |