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/*
* 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 Expand class.
#include "modules/audio_coding/neteq/expand.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_coding/neteq/background_noise.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 "rtc_base/safe_conversions.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
namespace webrtc {
TEST(Expand, 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);
}
TEST(Expand, CreateUsingFactory) {
int fs = 8000;
size_t channels = 1;
BackgroundNoise bgn(channels);
SyncBuffer sync_buffer(1, 1000);
RandomVector random_vector;
StatisticsCalculator statistics;
ExpandFactory expand_factory;
Expand* expand = expand_factory.Create(&bgn, &sync_buffer, &random_vector,
&statistics, fs, channels);
EXPECT_TRUE(expand != NULL);
delete expand;
}
namespace {
class FakeStatisticsCalculator : public StatisticsCalculator {
public:
void LogDelayedPacketOutageEvent(int outage_duration_ms) override {
last_outage_duration_ms_ = outage_duration_ms;
}
int last_outage_duration_ms() const { return last_outage_duration_ms_; }
private:
int last_outage_duration_ms_ = 0;
};
// This is the same size that is given to the SyncBuffer object in NetEq.
const size_t kNetEqSyncBufferLengthMs = 720;
} // namespace
class ExpandTest : public ::testing::Test {
protected:
ExpandTest()
: input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
32000),
test_sample_rate_hz_(32000),
num_channels_(1),
background_noise_(num_channels_),
sync_buffer_(num_channels_,
kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
expand_(&background_noise_,
&sync_buffer_,
&random_vector_,
&statistics_,
test_sample_rate_hz_,
num_channels_) {
WebRtcSpl_Init();
input_file_.set_output_rate_hz(test_sample_rate_hz_);
}
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);
ASSERT_TRUE(input_file_.Seek(speech_start_samples));
// 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);
ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
}
test::ResampleInputAudioFile input_file_;
int test_sample_rate_hz_;
size_t num_channels_;
BackgroundNoise background_noise_;
SyncBuffer sync_buffer_;
RandomVector random_vector_;
FakeStatisticsCalculator statistics_;
Expand expand_;
};
// This test calls the expand object to produce concealment data a few times,
// and then ends by calling SetParametersForNormalAfterExpand. This simulates
// the situation where the packet next up for decoding was just delayed, not
// lost.
TEST_F(ExpandTest, DelayedPacketOutage) {
AudioMultiVector output(num_channels_);
size_t sum_output_len_samples = 0;
for (int i = 0; i < 10; ++i) {
EXPECT_EQ(0, expand_.Process(&output));
EXPECT_GT(output.Size(), 0u);
sum_output_len_samples += output.Size();
EXPECT_EQ(0, statistics_.last_outage_duration_ms());
}
expand_.SetParametersForNormalAfterExpand();
// Convert |sum_output_len_samples| to milliseconds.
EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
(test_sample_rate_hz_ / 1000)),
statistics_.last_outage_duration_ms());
}
// This test is similar to DelayedPacketOutage, but ends by calling
// SetParametersForMergeAfterExpand. This simulates the situation where the
// packet next up for decoding was actually lost (or at least a later packet
// arrived before it).
TEST_F(ExpandTest, LostPacketOutage) {
AudioMultiVector output(num_channels_);
size_t sum_output_len_samples = 0;
for (int i = 0; i < 10; ++i) {
EXPECT_EQ(0, expand_.Process(&output));
EXPECT_GT(output.Size(), 0u);
sum_output_len_samples += output.Size();
EXPECT_EQ(0, statistics_.last_outage_duration_ms());
}
expand_.SetParametersForMergeAfterExpand();
EXPECT_EQ(0, statistics_.last_outage_duration_ms());
}
// This test is similar to the DelayedPacketOutage test above, but with the
// difference that Expand::Reset() is called after 5 calls to Expand::Process().
// This should reset the statistics, and will in the end lead to an outage of
// 5 periods instead of 10.
TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
AudioMultiVector output(num_channels_);
size_t sum_output_len_samples = 0;
for (int i = 0; i < 10; ++i) {
EXPECT_EQ(0, expand_.Process(&output));
EXPECT_GT(output.Size(), 0u);
sum_output_len_samples += output.Size();
if (i == 5) {
expand_.Reset();
sum_output_len_samples = 0;
}
EXPECT_EQ(0, statistics_.last_outage_duration_ms());
}
expand_.SetParametersForNormalAfterExpand();
// Convert |sum_output_len_samples| to milliseconds.
EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
(test_sample_rate_hz_ / 1000)),
statistics_.last_outage_duration_ms());
}
namespace {
// Runs expand until Muted() returns true. Times out after 1000 calls.
void ExpandUntilMuted(size_t num_channels, Expand* expand) {
EXPECT_FALSE(expand->Muted()) << "Instance is muted from the start";
AudioMultiVector output(num_channels);
int num_calls = 0;
while (!expand->Muted()) {
ASSERT_LT(num_calls++, 1000) << "Test timed out";
EXPECT_EQ(0, expand->Process(&output));
}
}
} // namespace
// Verifies that Muted() returns true after a long expand period. Also verifies
// that Muted() is reset to false after calling Reset(),
// SetParametersForMergeAfterExpand() and SetParametersForNormalAfterExpand().
TEST_F(ExpandTest, Muted) {
ExpandUntilMuted(num_channels_, &expand_);
expand_.Reset();
EXPECT_FALSE(expand_.Muted()); // Should be back to unmuted.
ExpandUntilMuted(num_channels_, &expand_);
expand_.SetParametersForMergeAfterExpand();
EXPECT_FALSE(expand_.Muted()); // Should be back to unmuted.
expand_.Reset(); // Must reset in order to start a new expand period.
ExpandUntilMuted(num_channels_, &expand_);
expand_.SetParametersForNormalAfterExpand();
EXPECT_FALSE(expand_.Muted()); // Should be back to unmuted.
}
// TODO(hlundin): Write more tests.
} // namespace webrtc