webrtc/modules/audio_coding/neteq/normal_unittest.cc

/*
 *  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 Normal class.

#include "webrtc/modules/audio_coding/neteq/normal.h"

#include <memory>
#include <vector>

#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_expand.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/test/gtest.h"

using ::testing::_;
using ::testing::Invoke;

namespace webrtc {

namespace {

int ExpandProcess120ms(AudioMultiVector* output) {
  AudioMultiVector dummy_audio(1, 11520u);
  dummy_audio.CopyTo(output);
  return 0;
}

} // namespace

TEST(Normal, CreateAndDestroy) {
  MockDecoderDatabase db;
  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);
  Normal normal(fs, &db, bgn, &expand);
  EXPECT_CALL(db, Die());  // Called when |db| goes out of scope.
}

TEST(Normal, AvoidDivideByZero) {
  WebRtcSpl_Init();
  MockDecoderDatabase db;
  int fs = 8000;
  size_t channels = 1;
  BackgroundNoise bgn(channels);
  SyncBuffer sync_buffer(1, 1000);
  RandomVector random_vector;
  StatisticsCalculator statistics;
  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
                    channels);
  Normal normal(fs, &db, bgn, &expand);

  int16_t input[1000] = {0};
  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
  for (size_t i = 0; i < channels; ++i) {
    mute_factor_array[i] = 16384;
  }
  AudioMultiVector output(channels);

  // Zero input length.
  EXPECT_EQ(
      0,
      normal.Process(input, 0, kModeExpand, mute_factor_array.get(), &output));
  EXPECT_EQ(0u, output.Size());

  // Try to make energy_length >> scaling = 0;
  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
  EXPECT_CALL(expand, Process(_));
  EXPECT_CALL(expand, Reset());
  // If input_size_samples < 64, then energy_length in Normal::Process() will
  // be equal to input_size_samples. Since the input is all zeros, decoded_max
  // will be zero, and scaling will be >= 6. Thus, energy_length >> scaling = 0,
  // and using this as a denominator would lead to problems.
  int input_size_samples = 63;
  EXPECT_EQ(input_size_samples,
            normal.Process(input,
                           input_size_samples,
                           kModeExpand,
                           mute_factor_array.get(),
                           &output));

  EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
  EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
}

TEST(Normal, InputLengthAndChannelsDoNotMatch) {
  WebRtcSpl_Init();
  MockDecoderDatabase db;
  int fs = 8000;
  size_t channels = 2;
  BackgroundNoise bgn(channels);
  SyncBuffer sync_buffer(channels, 1000);
  RandomVector random_vector;
  StatisticsCalculator statistics;
  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
                    channels);
  Normal normal(fs, &db, bgn, &expand);

  int16_t input[1000] = {0};
  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
  for (size_t i = 0; i < channels; ++i) {
    mute_factor_array[i] = 16384;
  }
  AudioMultiVector output(channels);

  // Let the number of samples be one sample less than 80 samples per channel.
  size_t input_len = 80 * channels - 1;
  EXPECT_EQ(
      0,
      normal.Process(
          input, input_len, kModeExpand, mute_factor_array.get(), &output));
  EXPECT_EQ(0u, output.Size());

  EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
  EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
}

TEST(Normal, LastModeExpand120msPacket) {
  WebRtcSpl_Init();
  MockDecoderDatabase db;
  const int kFs = 48000;
  const size_t kPacketsizeBytes = 11520u;
  const size_t kChannels = 1;
  BackgroundNoise bgn(kChannels);
  SyncBuffer sync_buffer(kChannels, 1000);
  RandomVector random_vector;
  StatisticsCalculator statistics;
  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, kFs,
                    kChannels);
  Normal normal(kFs, &db, bgn, &expand);

  int16_t input[kPacketsizeBytes] = {0};

  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[kChannels]);
  for (size_t i = 0; i < kChannels; ++i) {
    mute_factor_array[i] = 16384;
  }

  AudioMultiVector output(kChannels);

  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
  EXPECT_CALL(expand, Process(_)).WillOnce(Invoke(ExpandProcess120ms));
  EXPECT_CALL(expand, Reset());
  EXPECT_EQ(static_cast<int>(kPacketsizeBytes),
            normal.Process(input,
                           kPacketsizeBytes,
                           kModeExpand,
                           mute_factor_array.get(),
                           &output));

  EXPECT_EQ(kPacketsizeBytes, output.Size());

  EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
  EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
}

// TODO(hlundin): Write more tests.

}  // namespace webrtc