/* * 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 "common_audio/resampler/push_sinc_resampler.h" #include #include #include #include #include "common_audio/include/audio_util.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 T DBFS(T x) { return 20 * std::log10(x); } } // namespace class PushSincResamplerTest : public ::testing::TestWithParam< ::testing::tuple> { 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(input_rate_ / 100); const size_t output_samples = static_cast(output_rate_ / 100); const int kResampleIterations = 500000; // Source for data to be resampled. ZeroSource resampler_source; std::unique_ptr resampled_destination(new float[output_samples]); std::unique_ptr source(new float[input_samples]); std::unique_ptr source_int(new int16_t[input_samples]); std::unique_ptr 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(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(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(kTestDurationSecs * 100); const size_t input_block_size = static_cast(input_rate_ / 100); const size_t output_block_size = static_cast(output_rate_ / 100); const size_t input_samples = static_cast(kTestDurationSecs * input_rate_); const size_t output_samples = static_cast(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 resampled_destination(new float[output_samples]); std::unique_ptr pure_destination(new float[output_samples]); std::unique_ptr source(new float[input_samples]); std::unique_ptr source_int(new int16_t[input_block_size]); std::unique_ptr 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 = -6.01; 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 directly test rates which fail to provide an integer number // of samples in a 10 ms block (22050 and 11025 Hz), they are replaced // by nearby rates in order to simplify testing. // // The PushSincResampler is in practice sample rate agnostic and derives // resampling ratios from the block size, which for WebRTC purposes are // blocks of floor(sample_rate/100) samples. So the 22050 Hz case is // treated identically to the 22000 Hz case. Direct tests of 22050 Hz // have to account for the simulated clock drift induced by the // resampler inferring an incorrect sample rate ratio, without testing // anything new within the resampler itself. // To 22kHz std::make_tuple(8000, 22000, kResamplingRMSError, -62.73), std::make_tuple(11000, 22000, kResamplingRMSError, -74.17), std::make_tuple(16000, 22000, kResamplingRMSError, -62.54), std::make_tuple(22000, 22000, kResamplingRMSError, -73.53), std::make_tuple(32000, 22000, kResamplingRMSError, -46.45), std::make_tuple(44100, 22000, kResamplingRMSError, -28.34), std::make_tuple(48000, 22000, -15.01, -25.56), std::make_tuple(96000, 22000, -18.49, -13.30), std::make_tuple(192000, 22000, -20.50, -9.20), // To 44.1kHz ::testing::make_tuple(8000, 44100, kResamplingRMSError, -62.73), ::testing::make_tuple(11000, 44100, kResamplingRMSError, -63.57), ::testing::make_tuple(16000, 44100, kResamplingRMSError, -62.54), ::testing::make_tuple(22000, 44100, kResamplingRMSError, -62.73), ::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(11000, 48000, kResamplingRMSError, -63.96), ::testing::make_tuple(16000, 48000, kResamplingRMSError, -63.96), ::testing::make_tuple(22000, 48000, kResamplingRMSError, -63.80), ::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(11000, 96000, kResamplingRMSError, -63.89), ::testing::make_tuple(16000, 96000, kResamplingRMSError, -63.39), ::testing::make_tuple(22000, 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(11000, 192000, kResamplingRMSError, -63.17), ::testing::make_tuple(16000, 192000, kResamplingRMSError, -63.14), ::testing::make_tuple(22000, 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(11000, 16000, kResamplingRMSError, -72.31), ::testing::make_tuple(16000, 16000, kResamplingRMSError, -75.51), ::testing::make_tuple(22000, 16000, kResamplingRMSError, -52.08), ::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, -10.9596), // To 32 kHz ::testing::make_tuple(8000, 32000, kResamplingRMSError, -70.30), ::testing::make_tuple(11000, 32000, kResamplingRMSError, -71.34), ::testing::make_tuple(16000, 32000, kResamplingRMSError, -75.51), ::testing::make_tuple(22000, 32000, kResamplingRMSError, -72.05), ::testing::make_tuple(32000, 32000, kResamplingRMSError, -75.51), ::testing::make_tuple(44100, 32000, -16.44, -51.0349), ::testing::make_tuple(48000, 32000, -16.90, -43.9967), ::testing::make_tuple(96000, 32000, -19.61, -18.04), ::testing::make_tuple(192000, 32000, -21.02, -10.94))); } // namespace webrtc