|  | /* | 
|  | *  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 "webrtc/common_audio/audio_ring_buffer.h" | 
|  |  | 
|  | #include "testing/gtest/include/gtest/gtest.h" | 
|  | #include "webrtc/common_audio/channel_buffer.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_CASE_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 |