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Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc | 164 |
1 files changed, 164 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc new file mode 100644 index 00000000000..1a3d0fe781c --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc @@ -0,0 +1,164 @@ +/* + * 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 "webrtc/modules/audio_coding/neteq/sync_buffer.h" + +#include "gtest/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] = 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] = 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] = 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. + int16_t output[kChannels * kNewLen]; + // 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|. + size_t samples_read = sync_buffer.GetNextAudioInterleaved(kNewLen / 2, + output); + samples_read += + sync_buffer.GetNextAudioInterleaved(kNewLen / 2, + &output[samples_read * kChannels]); + EXPECT_EQ(kNewLen, samples_read); + + // Verify the data. + int16_t* output_ptr = output; + for (size_t i = 0; i < kNewLen; ++i) { + for (size_t channel = 0; channel < kChannels; ++channel) { + EXPECT_EQ(new_data[channel][i], *output_ptr); + ++output_ptr; + } + } +} + +} // namespace webrtc |