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The Tor Browser
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merge_unittest.cc (4700B)


      1 /*
      2 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
      3 *
      4 *  Use of this source code is governed by a BSD-style license
      5 *  that can be found in the LICENSE file in the root of the source
      6 *  tree. An additional intellectual property rights grant can be found
      7 *  in the file PATENTS.  All contributing project authors may
      8 *  be found in the AUTHORS file in the root of the source tree.
      9 */
     10 
     11 // Unit tests for Merge class.
     12 
     13 #include "modules/audio_coding/neteq/merge.h"
     14 
     15 #include <cstddef>
     16 #include <cstdint>
     17 #include <memory>
     18 #include <vector>
     19 
     20 #include "api/neteq/tick_timer.h"
     21 #include "modules/audio_coding/neteq/background_noise.h"
     22 #include "modules/audio_coding/neteq/expand.h"
     23 #include "modules/audio_coding/neteq/random_vector.h"
     24 #include "modules/audio_coding/neteq/statistics_calculator.h"
     25 #include "modules/audio_coding/neteq/sync_buffer.h"
     26 #include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
     27 #include "test/gtest.h"
     28 #include "test/testsupport/file_utils.h"
     29 
     30 namespace webrtc {
     31 
     32 TEST(Merge, CreateAndDestroy) {
     33  int fs = 8000;
     34  size_t channels = 1;
     35  BackgroundNoise bgn(channels);
     36  SyncBuffer sync_buffer(1, 1000);
     37  RandomVector random_vector;
     38  TickTimer timer;
     39  StatisticsCalculator statistics(&timer);
     40  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
     41  Merge merge(fs, channels, &expand, &sync_buffer);
     42 }
     43 
     44 namespace {
     45 // This is the same size that is given to the SyncBuffer object in NetEq.
     46 constexpr size_t kNetEqSyncBufferLengthMs = 720;
     47 }  // namespace
     48 
     49 class MergeTest : public testing::TestWithParam<size_t> {
     50 protected:
     51  MergeTest()
     52      : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
     53                    32000),
     54        test_sample_rate_hz_(8000),
     55        num_channels_(1),
     56        background_noise_(num_channels_),
     57        sync_buffer_(num_channels_,
     58                     kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
     59        statistics_(&timer_),
     60        expand_(&background_noise_,
     61                &sync_buffer_,
     62                &random_vector_,
     63                &statistics_,
     64                test_sample_rate_hz_,
     65                num_channels_),
     66        merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) {
     67    input_file_.set_output_rate_hz(test_sample_rate_hz_);
     68  }
     69 
     70  void SetUp() override {
     71    // Fast-forward the input file until there is speech (about 1.1 second into
     72    // the file).
     73    const int speech_start_samples =
     74        static_cast<int>(test_sample_rate_hz_ * 1.1f);
     75    ASSERT_TRUE(input_file_.Seek(speech_start_samples));
     76 
     77    // Pre-load the sync buffer with speech data.
     78    std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
     79    ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
     80    sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
     81    // Move index such that the sync buffer appears to have 5 ms left to play.
     82    sync_buffer_.set_next_index(sync_buffer_.next_index() -
     83                                test_sample_rate_hz_ * 5 / 1000);
     84    ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
     85    ASSERT_GT(sync_buffer_.FutureLength(), 0u);
     86  }
     87 
     88  test::ResampleInputAudioFile input_file_;
     89  int test_sample_rate_hz_;
     90  size_t num_channels_;
     91  BackgroundNoise background_noise_;
     92  SyncBuffer sync_buffer_;
     93  RandomVector random_vector_;
     94  TickTimer timer_;
     95  StatisticsCalculator statistics_;
     96  Expand expand_;
     97  Merge merge_;
     98 };
     99 
    100 TEST_P(MergeTest, Process) {
    101  AudioMultiVector output(num_channels_);
    102  // Start by calling Expand once, to prime the state.
    103  EXPECT_EQ(0, expand_.Process(&output));
    104  EXPECT_GT(output.Size(), 0u);
    105  output.Clear();
    106  // Now call Merge, but with a very short decoded input. Try different length
    107  // if the input.
    108  const size_t input_len = GetParam();
    109  std::vector<int16_t> input(input_len, 17);
    110  merge_.Process(input.data(), input_len, &output);
    111  EXPECT_GT(output.Size(), 0u);
    112 }
    113 
    114 // Instantiate with values for the input length that are interesting in
    115 // Merge::Downsample. Why are these values interesting?
    116 // - In 8000 Hz sample rate, signal_offset in Merge::Downsample will be 2, so
    117 //   the values 1, 2, 3 are just around that value.
    118 // - Also in 8000 Hz, the variable length_limit in the same method will be 80,
    119 //   so values 80 and 81 will be on either side of the branch point
    120 //   "input_length <= length_limit".
    121 // - Finally, 160 is simply 20 ms in 8000 Hz, which is a common packet size.
    122 INSTANTIATE_TEST_SUITE_P(DifferentInputLengths,
    123                         MergeTest,
    124                         testing::Values(1, 2, 3, 80, 81, 160));
    125 // TODO(hlundin): Write more tests.
    126 
    127 }  // namespace webrtc