blob: 3671ce4dfbdb3284bc114d40ae3cfb328ac24d9b [file] [log] [blame]
/*
* 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 "webrtc/common_audio/audio_ring_buffer.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_processing/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.samples_per_channel();
AudioRingBuffer buf(num_channels, buffer_frames);
scoped_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(), static_cast<int>(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(), static_cast<int>(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(), static_cast<int>(input_pos)),
num_channels, total_frames - input_pos);
if (buf.ReadFramesAvailable())
buf.Read(output->Slice(slice.get(), static_cast<int>(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(kInputArray, kNumFrames, kNumChannels);
AudioRingBuffer buf(kNumChannels, kNumFrames);
buf.Write(input.channels(), kNumChannels, kNumFrames);
buf.MoveReadPosition(3);
ChannelBuffer<float> output(1, kNumChannels);
buf.Read(output.channels(), kNumChannels, 1);
EXPECT_EQ(4, output.data()[0]);
buf.MoveReadPosition(-3);
buf.Read(output.channels(), kNumChannels, 1);
EXPECT_EQ(2, output.data()[0]);
}
} // namespace webrtc