/* * 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(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(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(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