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/*
* 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 <algorithm>
#include <cmath>
#include <cstring>
#include <memory>
#include "common_audio/include/audio_util.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "common_audio/resampler/sinusoidal_linear_chirp_source.h"
#include "rtc_base/time_utils.h"
#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
// Almost all conversions have an RMS error of around -14 dbFS.
const double kResamplingRMSError = -14.42;
// Used to convert errors to dbFS.
template <typename T>
T DBFS(T x) {
return 20 * std::log10(x);
}
} // namespace
class PushSincResamplerTest : public ::testing::TestWithParam<
::testing::tuple<int, int, double, double>> {
public:
PushSincResamplerTest()
: input_rate_(::testing::get<0>(GetParam())),
output_rate_(::testing::get<1>(GetParam())),
rms_error_(::testing::get<2>(GetParam())),
low_freq_error_(::testing::get<3>(GetParam())) {}
~PushSincResamplerTest() override {}
protected:
void ResampleBenchmarkTest(bool int_format);
void ResampleTest(bool int_format);
int input_rate_;
int output_rate_;
double rms_error_;
double low_freq_error_;
};
class ZeroSource : public SincResamplerCallback {
public:
void Run(size_t frames, float* destination) override {
std::memset(destination, 0, sizeof(float) * frames);
}
};
void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
const size_t input_samples = static_cast<size_t>(input_rate_ / 100);
const size_t output_samples = static_cast<size_t>(output_rate_ / 100);
const int kResampleIterations = 500000;
// Source for data to be resampled.
ZeroSource resampler_source;
std::unique_ptr<float[]> resampled_destination(new float[output_samples]);
std::unique_ptr<float[]> source(new float[input_samples]);
std::unique_ptr<int16_t[]> source_int(new int16_t[input_samples]);
std::unique_ptr<int16_t[]> destination_int(new int16_t[output_samples]);
resampler_source.Run(input_samples, source.get());
for (size_t i = 0; i < input_samples; ++i) {
source_int[i] = static_cast<int16_t>(floor(32767 * source[i] + 0.5));
}
printf("Benchmarking %d iterations of %d Hz -> %d Hz:\n", kResampleIterations,
input_rate_, output_rate_);
const double io_ratio = input_rate_ / static_cast<double>(output_rate_);
SincResampler sinc_resampler(io_ratio, SincResampler::kDefaultRequestSize,
&resampler_source);
int64_t start = rtc::TimeNanos();
for (int i = 0; i < kResampleIterations; ++i) {
sinc_resampler.Resample(output_samples, resampled_destination.get());
}
double total_time_sinc_us =
(rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
printf("SincResampler took %.2f us per frame.\n",
total_time_sinc_us / kResampleIterations);
PushSincResampler resampler(input_samples, output_samples);
start = rtc::TimeNanos();
if (int_format) {
for (int i = 0; i < kResampleIterations; ++i) {
EXPECT_EQ(output_samples,
resampler.Resample(source_int.get(), input_samples,
destination_int.get(), output_samples));
}
} else {
for (int i = 0; i < kResampleIterations; ++i) {
EXPECT_EQ(output_samples, resampler.Resample(source.get(), input_samples,
resampled_destination.get(),
output_samples));
}
}
double total_time_us =
(rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
printf(
"PushSincResampler took %.2f us per frame; which is a %.1f%% overhead "
"on SincResampler.\n\n",
total_time_us / kResampleIterations,
(total_time_us - total_time_sinc_us) / total_time_sinc_us * 100);
}
// Disabled because it takes too long to run routinely. Use for performance
// benchmarking when needed.
TEST_P(PushSincResamplerTest, DISABLED_BenchmarkInt) {
ResampleBenchmarkTest(true);
}
TEST_P(PushSincResamplerTest, DISABLED_BenchmarkFloat) {
ResampleBenchmarkTest(false);
}
// Tests resampling using a given input and output sample rate.
void PushSincResamplerTest::ResampleTest(bool int_format) {
// Make comparisons using one second of data.
static const double kTestDurationSecs = 1;
// 10 ms blocks.
const size_t kNumBlocks = static_cast<size_t>(kTestDurationSecs * 100);
const size_t input_block_size = static_cast<size_t>(input_rate_ / 100);
const size_t output_block_size = static_cast<size_t>(output_rate_ / 100);
const size_t input_samples =
static_cast<size_t>(kTestDurationSecs * input_rate_);
const size_t output_samples =
static_cast<size_t>(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.
std::unique_ptr<float[]> resampled_destination(new float[output_samples]);
std::unique_ptr<float[]> pure_destination(new float[output_samples]);
std::unique_ptr<float[]> source(new float[input_samples]);
std::unique_ptr<int16_t[]> source_int(new int16_t[input_block_size]);
std::unique_ptr<int16_t[]> destination_int(new int16_t[output_block_size]);
// The sinc resampler has an implicit delay of approximately half the kernel
// size at the input sample rate. By moving to a push model, this delay
// becomes explicit and is managed by zero-stuffing in PushSincResampler. We
// deal with it in the test by delaying the "pure" source to match. It must be
// checked before the first call to Resample(), because ChunkSize() will
// change afterwards.
const size_t output_delay_samples =
output_block_size - resampler.get_resampler_for_testing()->ChunkSize();
// 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(input_samples, source.get());
if (int_format) {
for (size_t i = 0; i < kNumBlocks; ++i) {
FloatToS16(&source[i * input_block_size], input_block_size,
source_int.get());
EXPECT_EQ(output_block_size,
resampler.Resample(source_int.get(), input_block_size,
destination_int.get(), output_block_size));
S16ToFloat(destination_int.get(), output_block_size,
&resampled_destination[i * output_block_size]);
}
} else {
for (size_t i = 0; i < kNumBlocks; ++i) {
EXPECT_EQ(
output_block_size,
resampler.Resample(&source[i * input_block_size], input_block_size,
&resampled_destination[i * output_block_size],
output_block_size));
}
}
// Generate pure signal.
SinusoidalLinearChirpSource pure_source(
output_rate_, output_samples, input_nyquist_freq, output_delay_samples);
pure_source.Run(output_samples, pure_destination.get());
// 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 (size_t 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);
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);
}
TEST_P(PushSincResamplerTest, ResampleInt) {
ResampleTest(true);
}
TEST_P(PushSincResamplerTest, ResampleFloat) {
ResampleTest(false);
}
// Thresholds chosen arbitrarily based on what each resampling reported during
// testing. All thresholds are in dbFS, http://en.wikipedia.org/wiki/DBFS.
INSTANTIATE_TEST_SUITE_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
::testing::make_tuple(8000, 44100, kResamplingRMSError, -62.73),
::testing::make_tuple(16000, 44100, kResamplingRMSError, -62.54),
::testing::make_tuple(32000, 44100, kResamplingRMSError, -63.32),
::testing::make_tuple(44100, 44100, kResamplingRMSError, -73.53),
::testing::make_tuple(48000, 44100, -15.01, -64.04),
::testing::make_tuple(96000, 44100, -18.49, -25.51),
::testing::make_tuple(192000, 44100, -20.50, -13.31),
// To 48kHz
::testing::make_tuple(8000, 48000, kResamplingRMSError, -63.43),
::testing::make_tuple(16000, 48000, kResamplingRMSError, -63.96),
::testing::make_tuple(32000, 48000, kResamplingRMSError, -64.04),
::testing::make_tuple(44100, 48000, kResamplingRMSError, -62.63),
::testing::make_tuple(48000, 48000, kResamplingRMSError, -73.52),
::testing::make_tuple(96000, 48000, -18.40, -28.44),
::testing::make_tuple(192000, 48000, -20.43, -14.11),
// To 96kHz
::testing::make_tuple(8000, 96000, kResamplingRMSError, -63.19),
::testing::make_tuple(16000, 96000, kResamplingRMSError, -63.39),
::testing::make_tuple(32000, 96000, kResamplingRMSError, -63.95),
::testing::make_tuple(44100, 96000, kResamplingRMSError, -62.63),
::testing::make_tuple(48000, 96000, kResamplingRMSError, -73.52),
::testing::make_tuple(96000, 96000, kResamplingRMSError, -73.52),
::testing::make_tuple(192000, 96000, kResamplingRMSError, -28.41),
// To 192kHz
::testing::make_tuple(8000, 192000, kResamplingRMSError, -63.10),
::testing::make_tuple(16000, 192000, kResamplingRMSError, -63.14),
::testing::make_tuple(32000, 192000, kResamplingRMSError, -63.38),
::testing::make_tuple(44100, 192000, kResamplingRMSError, -62.63),
::testing::make_tuple(48000, 192000, kResamplingRMSError, -73.44),
::testing::make_tuple(96000, 192000, kResamplingRMSError, -73.52),
::testing::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
::testing::make_tuple(8000, 8000, kResamplingRMSError, -75.50),
::testing::make_tuple(16000, 8000, -18.56, -28.79),
::testing::make_tuple(32000, 8000, -20.36, -14.13),
::testing::make_tuple(44100, 8000, -21.00, -11.39),
::testing::make_tuple(48000, 8000, -20.96, -11.04),
// To 16 kHz
::testing::make_tuple(8000, 16000, kResamplingRMSError, -70.30),
::testing::make_tuple(16000, 16000, kResamplingRMSError, -75.51),
::testing::make_tuple(32000, 16000, -18.48, -28.59),
::testing::make_tuple(44100, 16000, -19.30, -19.67),
::testing::make_tuple(48000, 16000, -19.81, -18.11),
::testing::make_tuple(96000, 16000, -20.95, -10.96),
// To 32 kHz
::testing::make_tuple(8000, 32000, kResamplingRMSError, -70.30),
::testing::make_tuple(16000, 32000, kResamplingRMSError, -75.51),
::testing::make_tuple(32000, 32000, kResamplingRMSError, -75.51),
::testing::make_tuple(44100, 32000, -16.44, -51.10),
::testing::make_tuple(48000, 32000, -16.90, -44.03),
::testing::make_tuple(96000, 32000, -19.61, -18.04),
::testing::make_tuple(192000, 32000, -21.02, -10.94)));
} // namespace webrtc