122 lines
4.4 KiB
C++
122 lines
4.4 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Unit tests for Merge class.
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#include "modules/audio_coding/neteq/merge.h"
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#include <algorithm>
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#include <vector>
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#include "modules/audio_coding/neteq/background_noise.h"
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#include "modules/audio_coding/neteq/expand.h"
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#include "modules/audio_coding/neteq/random_vector.h"
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#include "modules/audio_coding/neteq/statistics_calculator.h"
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#include "modules/audio_coding/neteq/sync_buffer.h"
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#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
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#include "test/gtest.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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TEST(Merge, CreateAndDestroy) {
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int fs = 8000;
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size_t channels = 1;
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BackgroundNoise bgn(channels);
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SyncBuffer sync_buffer(1, 1000);
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RandomVector random_vector;
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StatisticsCalculator statistics;
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Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
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Merge merge(fs, channels, &expand, &sync_buffer);
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}
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namespace {
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// This is the same size that is given to the SyncBuffer object in NetEq.
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const size_t kNetEqSyncBufferLengthMs = 720;
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} // namespace
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class MergeTest : public testing::TestWithParam<size_t> {
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protected:
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MergeTest()
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: input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
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32000),
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test_sample_rate_hz_(8000),
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num_channels_(1),
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background_noise_(num_channels_),
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sync_buffer_(num_channels_,
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kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
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expand_(&background_noise_,
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&sync_buffer_,
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&random_vector_,
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&statistics_,
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test_sample_rate_hz_,
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num_channels_),
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merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) {
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input_file_.set_output_rate_hz(test_sample_rate_hz_);
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}
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void SetUp() override {
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// Fast-forward the input file until there is speech (about 1.1 second into
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// the file).
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const int speech_start_samples =
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static_cast<int>(test_sample_rate_hz_ * 1.1f);
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ASSERT_TRUE(input_file_.Seek(speech_start_samples));
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// Pre-load the sync buffer with speech data.
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std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
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ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
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sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
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// Move index such that the sync buffer appears to have 5 ms left to play.
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sync_buffer_.set_next_index(sync_buffer_.next_index() -
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test_sample_rate_hz_ * 5 / 1000);
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ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
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ASSERT_GT(sync_buffer_.FutureLength(), 0u);
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}
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test::ResampleInputAudioFile input_file_;
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int test_sample_rate_hz_;
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size_t num_channels_;
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BackgroundNoise background_noise_;
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SyncBuffer sync_buffer_;
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RandomVector random_vector_;
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StatisticsCalculator statistics_;
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Expand expand_;
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Merge merge_;
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};
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TEST_P(MergeTest, Process) {
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AudioMultiVector output(num_channels_);
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// Start by calling Expand once, to prime the state.
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EXPECT_EQ(0, expand_.Process(&output));
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EXPECT_GT(output.Size(), 0u);
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output.Clear();
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// Now call Merge, but with a very short decoded input. Try different length
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// if the input.
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const size_t input_len = GetParam();
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std::vector<int16_t> input(input_len, 17);
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merge_.Process(input.data(), input_len, &output);
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EXPECT_GT(output.Size(), 0u);
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}
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// Instantiate with values for the input length that are interesting in
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// Merge::Downsample. Why are these values interesting?
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// - In 8000 Hz sample rate, signal_offset in Merge::Downsample will be 2, so
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// the values 1, 2, 3 are just around that value.
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// - Also in 8000 Hz, the variable length_limit in the same method will be 80,
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// so values 80 and 81 will be on either side of the branch point
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// "input_length <= length_limit".
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// - Finally, 160 is simply 20 ms in 8000 Hz, which is a common packet size.
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INSTANTIATE_TEST_SUITE_P(DifferentInputLengths,
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MergeTest,
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testing::Values(1, 2, 3, 80, 81, 160));
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// TODO(hlundin): Write more tests.
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} // namespace webrtc
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