common_audio/resampler/push_sinc_resampler_unittest.cc

/*
 *  Copyright (c) 2013 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.
 */

#include <cmath>

#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"

namespace webrtc {

typedef std::tr1::tuple<int, int, double, double> PushSincResamplerTestData;
class PushSincResamplerTest
    : public testing::TestWithParam<PushSincResamplerTestData> {
 public:
  PushSincResamplerTest()
      : input_rate_(std::tr1::get<0>(GetParam())),
        output_rate_(std::tr1::get<1>(GetParam())),
        rms_error_(std::tr1::get<2>(GetParam())),
        low_freq_error_(std::tr1::get<3>(GetParam())) {
  }

  virtual ~PushSincResamplerTest() {}

 protected:
  int input_rate_;
  int output_rate_;
  double rms_error_;
  double low_freq_error_;
};

// Tests resampling using a given input and output sample rate.
TEST_P(PushSincResamplerTest, Resample) {
  // Make comparisons using one second of data.
  static const double kTestDurationSecs = 1;
  // 10 ms blocks.
  const int kNumBlocks = kTestDurationSecs * 100;
  const int input_block_size = input_rate_ / 100;
  const int output_block_size = output_rate_ / 100;
  const int input_samples = kTestDurationSecs * input_rate_;
  const int output_samples = kTestDurationSecs * output_rate_;

  // Nyquist frequency for the input sampling rate.
  const double input_nyquist_freq = 0.5 * input_rate_;

  // Source for data to be resampled.
  SinusoidalLinearChirpSource resampler_source(
      input_rate_, input_samples, input_nyquist_freq, 0);

  PushSincResampler resampler(input_block_size, output_block_size);

  // TODO(dalecurtis): If we switch to AVX/SSE optimization, we'll need to
  // allocate these on 32-byte boundaries and ensure they're sized % 32 bytes.
  scoped_array<float> resampled_destination(new float[output_samples]);
  scoped_array<float> pure_destination(new float[output_samples]);
  scoped_array<float> source(new float[input_samples]);
  scoped_array<int16_t> source_int(new int16_t[input_block_size]);
  scoped_array<int16_t> destination_int(new int16_t[output_block_size]);

  // Generate resampled signal.
  // With the PushSincResampler, we produce the signal block-by-10ms-block
  // rather than in a single pass, to exercise how it will be used in WebRTC.
  resampler_source.Run(source.get(), input_samples);
  for (int i = 0; i < kNumBlocks; ++i) {
    for (int j = 0; j < input_block_size; ++j) {
      source_int[j] = static_cast<int16_t>(std::floor(32767 *
          source[i * input_block_size + j] + 0.5));
    }
    EXPECT_EQ(output_block_size,
              resampler.Resample(source_int.get(), input_block_size,
                                 destination_int.get(), output_block_size));
    for (int j = 0; j < output_block_size; ++j) {
      resampled_destination[i * output_block_size + j] =
          static_cast<float>(destination_int[j]) / 32767;
    }
  }

  // Generate pure signal.
  // The sinc resampler has an implicit delay of half the kernel size (32) at
  // the input sample rate. By moving to a push model, this delay becomes
  // explicit and is managed by zero-stuffing in PushSincResampler. This delay
  // can be a fractional sample amount, so we deal with it in the test by
  // delaying the "pure" source to match.
  static const int kInputKernelDelaySamples = 16;
  double output_delay_samples = static_cast<double>(output_rate_)
      / input_rate_ * kInputKernelDelaySamples;
  SinusoidalLinearChirpSource pure_source(
      output_rate_, output_samples, input_nyquist_freq, output_delay_samples);
  pure_source.Run(pure_destination.get(), output_samples);

  // Range of the Nyquist frequency (0.5 * min(input rate, output_rate)) which
  // we refer to as low and high.
  static const double kLowFrequencyNyquistRange = 0.7;
  static const double kHighFrequencyNyquistRange = 0.9;

  // Calculate Root-Mean-Square-Error and maximum error for the resampling.
  double sum_of_squares = 0;
  double low_freq_max_error = 0;
  double high_freq_max_error = 0;
  int minimum_rate = std::min(input_rate_, output_rate_);
  double low_frequency_range = kLowFrequencyNyquistRange * 0.5 * minimum_rate;
  double high_frequency_range = kHighFrequencyNyquistRange * 0.5 * minimum_rate;

  for (int i = 0; i < output_samples; ++i) {
    double error = fabs(resampled_destination[i] - pure_destination[i]);

    if (pure_source.Frequency(i) < low_frequency_range) {
      if (error > low_freq_max_error)
        low_freq_max_error = error;
    } else if (pure_source.Frequency(i) < high_frequency_range) {
      if (error > high_freq_max_error)
        high_freq_max_error = error;
    }
    // TODO(dalecurtis): Sanity check frequencies > kHighFrequencyNyquistRange.

    sum_of_squares += error * error;
  }

  double rms_error = sqrt(sum_of_squares / output_samples);

  // Convert each error to dbFS.
  #define DBFS(x) 20 * log10(x)
  rms_error = DBFS(rms_error);
  // In order to keep the thresholds in this test identical to SincResamplerTest
  // we must account for the quantization error introduced by truncating from
  // float to int. This happens twice (once at input and once at output) and we
  // allow for the maximum possible error (1 / 32767) for each step.
  //
  // The quantization error is insignificant in the RMS calculation so does not
  // need to be accounted for there.
  low_freq_max_error = DBFS(low_freq_max_error - 2.0 / 32767);
  high_freq_max_error = DBFS(high_freq_max_error - 2.0 / 32767);

  EXPECT_LE(rms_error, rms_error_);
  EXPECT_LE(low_freq_max_error, low_freq_error_);

  // All conversions currently have a high frequency error around -6 dbFS.
  static const double kHighFrequencyMaxError = -6.02;
  EXPECT_LE(high_freq_max_error, kHighFrequencyMaxError);
}

// Almost all conversions have an RMS error of around -14 dbFS.
static const double kResamplingRMSError = -14.42;

// Thresholds chosen arbitrarily based on what each resampling reported during
// testing.  All thresholds are in dbFS, http://en.wikipedia.org/wiki/DBFS.
INSTANTIATE_TEST_CASE_P(
    PushSincResamplerTest, PushSincResamplerTest, testing::Values(
        // First run through the rates tested in SincResamplerTest. The
        // thresholds are identical.
        //
        // We don't test rates which fail to provide an integer number of
        // samples in a 10 ms block (22050 and 11025 Hz). WebRTC doesn't support
        // these rates in any case (for the same reason).

        // To 44.1kHz
        std::tr1::make_tuple(8000, 44100, kResamplingRMSError, -62.73),
        std::tr1::make_tuple(16000, 44100, kResamplingRMSError, -62.54),
        std::tr1::make_tuple(32000, 44100, kResamplingRMSError, -63.32),
        std::tr1::make_tuple(44100, 44100, kResamplingRMSError, -73.53),
        std::tr1::make_tuple(48000, 44100, -15.01, -64.04),
        std::tr1::make_tuple(96000, 44100, -18.49, -25.51),
        std::tr1::make_tuple(192000, 44100, -20.50, -13.31),

        // To 48kHz
        std::tr1::make_tuple(8000, 48000, kResamplingRMSError, -63.43),
        std::tr1::make_tuple(16000, 48000, kResamplingRMSError, -63.96),
        std::tr1::make_tuple(32000, 48000, kResamplingRMSError, -64.04),
        std::tr1::make_tuple(44100, 48000, kResamplingRMSError, -62.63),
        std::tr1::make_tuple(48000, 48000, kResamplingRMSError, -73.52),
        std::tr1::make_tuple(96000, 48000, -18.40, -28.44),
        std::tr1::make_tuple(192000, 48000, -20.43, -14.11),

        // To 96kHz
        std::tr1::make_tuple(8000, 96000, kResamplingRMSError, -63.19),
        std::tr1::make_tuple(16000, 96000, kResamplingRMSError, -63.39),
        std::tr1::make_tuple(32000, 96000, kResamplingRMSError, -63.95),
        std::tr1::make_tuple(44100, 96000, kResamplingRMSError, -62.63),
        std::tr1::make_tuple(48000, 96000, kResamplingRMSError, -73.52),
        std::tr1::make_tuple(96000, 96000, kResamplingRMSError, -73.52),
        std::tr1::make_tuple(192000, 96000, kResamplingRMSError, -28.41),

        // To 192kHz
        std::tr1::make_tuple(8000, 192000, kResamplingRMSError, -63.10),
        std::tr1::make_tuple(16000, 192000, kResamplingRMSError, -63.14),
        std::tr1::make_tuple(32000, 192000, kResamplingRMSError, -63.38),
        std::tr1::make_tuple(44100, 192000, kResamplingRMSError, -62.63),
        std::tr1::make_tuple(48000, 192000, kResamplingRMSError, -73.44),
        std::tr1::make_tuple(96000, 192000, kResamplingRMSError, -73.52),
        std::tr1::make_tuple(192000, 192000, kResamplingRMSError, -73.52),

        // Next run through some additional cases interesting for WebRTC.
        // We skip some extreme downsampled cases (192 -> {8, 16}, 96 -> 8)
        // because they violate |kHighFrequencyMaxError|, which is not
        // unexpected. It's very unlikely that we'll see these conversions in
        // practice anyway.

        // To 8 kHz
        std::tr1::make_tuple(8000, 8000, kResamplingRMSError, -75.51),
        std::tr1::make_tuple(16000, 8000, -18.56, -28.79),
        std::tr1::make_tuple(32000, 8000, -20.36, -14.13),
        std::tr1::make_tuple(44100, 8000, -21.00, -11.39),
        std::tr1::make_tuple(48000, 8000, -20.96, -11.04),

        // To 16 kHz
        std::tr1::make_tuple(8000, 16000, kResamplingRMSError, -70.30),
        std::tr1::make_tuple(16000, 16000, kResamplingRMSError, -75.51),
        std::tr1::make_tuple(32000, 16000, -18.48, -28.59),
        std::tr1::make_tuple(44100, 16000, -19.59, -19.77),
        std::tr1::make_tuple(48000, 16000, -20.01, -18.11),
        std::tr1::make_tuple(96000, 16000, -20.95, -10.99),

        // To 32 kHz
        std::tr1::make_tuple(8000, 32000, kResamplingRMSError, -70.30),
        std::tr1::make_tuple(16000, 32000, kResamplingRMSError, -75.51),
        std::tr1::make_tuple(32000, 32000, kResamplingRMSError, -75.56),
        std::tr1::make_tuple(44100, 32000, -16.52, -51.10),
        std::tr1::make_tuple(48000, 32000, -16.90, -44.17),
        std::tr1::make_tuple(96000, 32000, -19.80, -18.05),
        std::tr1::make_tuple(192000, 32000, -21.02, -10.94)));

}  // namespace webrtc