/* * 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 #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 PushSincResamplerTestData; class PushSincResamplerTest : public testing::TestWithParam { 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 resampled_destination(new float[output_samples]); scoped_array pure_destination(new float[output_samples]); scoped_array source(new float[input_samples]); scoped_array source_int(new int16_t[input_block_size]); scoped_array 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(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(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(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