| /* |
| * 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 "webrtc/modules/audio_coding/neteq/expand.h" |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| #include "webrtc/base/safe_conversions.h" |
| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" |
| #include "webrtc/modules/audio_coding/neteq/background_noise.h" |
| #include "webrtc/modules/audio_coding/neteq/random_vector.h" |
| #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" |
| #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" |
| #include "webrtc/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 size_t speech_start_samples = |
| static_cast<size_t>(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 |