| /* |
| * 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 |