blob: d5a55eb056b47ff653be00b0124a5119354d12f4 [file] [log] [blame]
* 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.
// Unit tests for Merge class.
#include "modules/audio_coding/neteq/merge.h"
#include <algorithm>
#include <vector>
#include "modules/audio_coding/neteq/background_noise.h"
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/random_vector.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
TEST(Merge, CreateAndDestroy) {
int fs = 8000;
size_t channels = 1;
BackgroundNoise bgn(channels);
SyncBuffer sync_buffer(1, 1000);
RandomVector random_vector;
StatisticsCalculator statistics;
Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
Merge merge(fs, channels, &expand, &sync_buffer);
namespace {
// This is the same size that is given to the SyncBuffer object in NetEq.
const size_t kNetEqSyncBufferLengthMs = 720;
} // namespace
class MergeTest : public testing::TestWithParam<size_t> {
: input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) {
void SetUp() override {
// Fast-forward the input file until there is speech (about 1.1 second into
// the file).
const int speech_start_samples =
static_cast<int>(test_sample_rate_hz_ * 1.1f);
// Pre-load the sync buffer with speech data.
std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
// Move index such that the sync buffer appears to have 5 ms left to play.
sync_buffer_.set_next_index(sync_buffer_.next_index() -
test_sample_rate_hz_ * 5 / 1000);
ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
ASSERT_GT(sync_buffer_.FutureLength(), 0u);
test::ResampleInputAudioFile input_file_;
int test_sample_rate_hz_;
size_t num_channels_;
BackgroundNoise background_noise_;
SyncBuffer sync_buffer_;
RandomVector random_vector_;
StatisticsCalculator statistics_;
Expand expand_;
Merge merge_;
TEST_P(MergeTest, Process) {
AudioMultiVector output(num_channels_);
// Start by calling Expand once, to prime the state.
EXPECT_EQ(0, expand_.Process(&output));
EXPECT_GT(output.Size(), 0u);
// Now call Merge, but with a very short decoded input. Try different length
// if the input.
const size_t input_len = GetParam();
std::vector<int16_t> input(input_len, 17);
merge_.Process(, input_len, &output);
EXPECT_GT(output.Size(), 0u);
// Instantiate with values for the input length that are interesting in
// Merge::Downsample. Why are these values interesting?
// - In 8000 Hz sample rate, signal_offset in Merge::Downsample will be 2, so
// the values 1, 2, 3 are just around that value.
// - Also in 8000 Hz, the variable length_limit in the same method will be 80,
// so values 80 and 81 will be on either side of the branch point
// "input_length <= length_limit".
// - Finally, 160 is simply 20 ms in 8000 Hz, which is a common packet size.
testing::Values(1, 2, 3, 80, 81, 160));
// TODO(hlundin): Write more tests.
} // namespace webrtc