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