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/*
* Copyright (c) 2012 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 "modules/audio_coding/neteq/sync_buffer.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
namespace webrtc {
TEST(SyncBuffer, CreateAndDestroy) {
// Create a SyncBuffer with two channels and 10 samples each.
static const size_t kLen = 10;
static const size_t kChannels = 2;
SyncBuffer sync_buffer(kChannels, kLen);
EXPECT_EQ(kChannels, sync_buffer.Channels());
EXPECT_EQ(kLen, sync_buffer.Size());
// When the buffer is empty, the next index to play out is at the end.
EXPECT_EQ(kLen, sync_buffer.next_index());
// Verify that all elements are zero.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kLen; ++i) {
EXPECT_EQ(0, sync_buffer[channel][i]);
}
}
}
TEST(SyncBuffer, SetNextIndex) {
// Create a SyncBuffer with two channels and 100 samples each.
static const size_t kLen = 100;
static const size_t kChannels = 2;
SyncBuffer sync_buffer(kChannels, kLen);
sync_buffer.set_next_index(0);
EXPECT_EQ(0u, sync_buffer.next_index());
sync_buffer.set_next_index(kLen / 2);
EXPECT_EQ(kLen / 2, sync_buffer.next_index());
sync_buffer.set_next_index(kLen);
EXPECT_EQ(kLen, sync_buffer.next_index());
// Try to set larger than the buffer size; should cap at buffer size.
sync_buffer.set_next_index(kLen + 1);
EXPECT_EQ(kLen, sync_buffer.next_index());
}
TEST(SyncBuffer, PushBackAndFlush) {
// Create a SyncBuffer with two channels and 100 samples each.
static const size_t kLen = 100;
static const size_t kChannels = 2;
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
AudioMultiVector new_data(kChannels, kNewLen);
// Populate `new_data`.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
new_data[channel][i] = rtc::checked_cast<int16_t>(i);
}
}
// Push back `new_data` into `sync_buffer`. This operation should pop out
// data from the front of `sync_buffer`, so that the size of the buffer
// remains the same. The `next_index_` should also move with the same length.
sync_buffer.PushBack(new_data);
ASSERT_EQ(kLen, sync_buffer.Size());
// Verify that `next_index_` moved accordingly.
EXPECT_EQ(kLen - kNewLen, sync_buffer.next_index());
// Verify the new contents.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
EXPECT_EQ(new_data[channel][i],
sync_buffer[channel][sync_buffer.next_index() + i]);
}
}
// Now flush the buffer, and verify that it is all zeros, and that next_index
// points to the end.
sync_buffer.Flush();
ASSERT_EQ(kLen, sync_buffer.Size());
EXPECT_EQ(kLen, sync_buffer.next_index());
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kLen; ++i) {
EXPECT_EQ(0, sync_buffer[channel][i]);
}
}
}
TEST(SyncBuffer, PushFrontZeros) {
// Create a SyncBuffer with two channels and 100 samples each.
static const size_t kLen = 100;
static const size_t kChannels = 2;
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
AudioMultiVector new_data(kChannels, kNewLen);
// Populate `new_data`.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
new_data[channel][i] = rtc::checked_cast<int16_t>(1000 + i);
}
}
sync_buffer.PushBack(new_data);
EXPECT_EQ(kLen, sync_buffer.Size());
// Push `kNewLen` - 1 zeros into each channel in the front of the SyncBuffer.
sync_buffer.PushFrontZeros(kNewLen - 1);
EXPECT_EQ(kLen, sync_buffer.Size()); // Size should remain the same.
// Verify that `next_index_` moved accordingly. Should be at the end - 1.
EXPECT_EQ(kLen - 1, sync_buffer.next_index());
// Verify the zeros.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen - 1; ++i) {
EXPECT_EQ(0, sync_buffer[channel][i]);
}
}
// Verify that the correct data is at the end of the SyncBuffer.
for (size_t channel = 0; channel < kChannels; ++channel) {
EXPECT_EQ(1000, sync_buffer[channel][sync_buffer.next_index()]);
}
}
TEST(SyncBuffer, GetNextAudioInterleaved) {
// Create a SyncBuffer with two channels and 100 samples each.
static const size_t kLen = 100;
static const size_t kChannels = 2;
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
AudioMultiVector new_data(kChannels, kNewLen);
// Populate `new_data`.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
new_data[channel][i] = rtc::checked_cast<int16_t>(i);
}
}
// Push back `new_data` into `sync_buffer`. This operation should pop out
// data from the front of `sync_buffer`, so that the size of the buffer
// remains the same. The `next_index_` should also move with the same length.
sync_buffer.PushBack(new_data);
// Read to interleaved output. Read in two batches, where each read operation
// should automatically update the `net_index_` in the SyncBuffer.
// Note that `samples_read` is the number of samples read from each channel.
// That is, the number of samples written to `output` is
// `samples_read` * `kChannels`.
AudioFrame output1;
sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output1);
EXPECT_EQ(kChannels, output1.num_channels_);
EXPECT_EQ(kNewLen / 2, output1.samples_per_channel_);
AudioFrame output2;
sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output2);
EXPECT_EQ(kChannels, output2.num_channels_);
EXPECT_EQ(kNewLen / 2, output2.samples_per_channel_);
// Verify the data.
const int16_t* output_ptr = output1.data();
for (size_t i = 0; i < kNewLen / 2; ++i) {
for (size_t channel = 0; channel < kChannels; ++channel) {
EXPECT_EQ(new_data[channel][i], *output_ptr);
++output_ptr;
}
}
output_ptr = output2.data();
for (size_t i = kNewLen / 2; i < kNewLen; ++i) {
for (size_t channel = 0; channel < kChannels; ++channel) {
EXPECT_EQ(new_data[channel][i], *output_ptr);
++output_ptr;
}
}
}
} // namespace webrtc