webrtc_m130/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc
yujo 36b1a5fcec Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
const int16_t* data() const;
int16_t* mutable_data();

- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.

These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.

This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.

BUG=webrtc:7343
TBR=henrika

Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
2017-06-12 19:45:32 +00:00

174 lines
6.3 KiB
C++

/*
* 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 "webrtc/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] = 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.
// 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