/* * 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 #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 > { }; void ReadAndWriteTest(const ChannelBuffer& input, size_t num_write_chunk_frames, size_t num_read_chunk_frames, size_t buffer_frames, ChannelBuffer* 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 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 input(kFrames, static_cast(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 output(kFrames, static_cast(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 input(kNumFrames, kNumChannels); input.SetDataForTesting(kInputArray, kNumFrames); AudioRingBuffer buf(kNumChannels, kNumFrames); buf.Write(input.channels(), kNumChannels, kNumFrames); buf.MoveReadPositionForward(3); ChannelBuffer 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