From 20e17559e545c5d3cfe86c1c4772365e70090779 Mon Sep 17 00:00:00 2001
From: Qrox <[EMAIL REDACTED]>
Date: Thu, 9 Mar 2023 17:34:51 +0800
Subject: [PATCH] Uses integer arithmetics in SDL_ResampleAudio
- Avoids precision loss caused by large floating point numbers.
- Adds unit test to test the signal-to-noise ratio and maximum error of resampler.
- Code cleanup
---
src/audio/SDL_audiocvt.c | 56 +++++++++--------
test/testautomation_audio.c | 118 +++++++++++++++++++++++++++++++++++-
2 files changed, 148 insertions(+), 26 deletions(-)
diff --git a/src/audio/SDL_audiocvt.c b/src/audio/SDL_audiocvt.c
index de97beacc385..a30ae5c9e47c 100644
--- a/src/audio/SDL_audiocvt.c
+++ b/src/audio/SDL_audiocvt.c
@@ -177,14 +177,18 @@ SDL_ConvertMonoToStereo_SSE(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#include "SDL_audio_resampler_filter.h"
-static int
-ResamplerPadding(const int inrate, const int outrate)
+static Sint32
+ResamplerPadding(const Sint32 inrate, const Sint32 outrate)
{
+ /* This function uses integer arithmetics to avoid precision loss caused
+ * by large floating point numbers. Sint32 is needed for the large number
+ * multiplication. The integers are assumed to be non-negative so that
+ * division rounds by truncation. */
if (inrate == outrate) {
return 0;
}
if (inrate > outrate) {
- return (int) SDL_ceilf(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate)));
+ return (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate + outrate - 1) / outrate;
}
return RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
}
@@ -196,57 +200,59 @@ SDL_ResampleAudio(const int chans, const int inrate, const int outrate,
const float *inbuf, const int inbuflen,
float *outbuf, const int outbuflen)
{
- /* Note that this used to be double, but it looks like we can get by with float in most cases at
- almost twice the speed on Intel processors, and orders of magnitude more
- on CPUs that need a software fallback for double calculations. */
- typedef float ResampleFloatType;
-
- const ResampleFloatType finrate = (ResampleFloatType) inrate;
- const ResampleFloatType ratio = ((float) outrate) / ((float) inrate);
+ /* This function uses integer arithmetics to avoid precision loss caused
+ * by large floating point numbers. For some operations, Sint32 or Sint64
+ * are needed for the large number multiplications. The input integers are
+ * assumed to be non-negative so that division rounds by truncation and
+ * modulo is always non-negative. Note that the operator order is important
+ * for these integer divisions. */
const int paddinglen = ResamplerPadding(inrate, outrate);
const int framelen = chans * (int)sizeof (float);
const int inframes = inbuflen / framelen;
- const int wantedoutframes = (int) ((inbuflen / framelen) * ratio); /* outbuflen isn't total to write, it's total available. */
+ /* outbuflen isn't total to write, it's total available. */
+ const int wantedoutframes = ((Sint64) inframes) * outrate / inrate;
const int maxoutframes = outbuflen / framelen;
const int outframes = SDL_min(wantedoutframes, maxoutframes);
- ResampleFloatType outtime = 0.0f;
float *dst = outbuf;
int i, j, chan;
for (i = 0; i < outframes; i++) {
- const int srcindex = (int) (outtime * inrate);
- const ResampleFloatType intime = ((ResampleFloatType) srcindex) / finrate;
- const ResampleFloatType innexttime = ((ResampleFloatType) (srcindex + 1)) / finrate;
- const ResampleFloatType indeltatime = innexttime - intime;
- const ResampleFloatType interpolation1 = (indeltatime == 0.0f) ? 1.0f : (1.0f - ((innexttime - outtime) / indeltatime));
- const int filterindex1 = (int) (interpolation1 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING);
- const ResampleFloatType interpolation2 = 1.0f - interpolation1;
- const int filterindex2 = (int) (interpolation2 * RESAMPLER_SAMPLES_PER_ZERO_CROSSING);
+ const int srcindex = ((Sint64) i) * inrate / outrate;
+ /* Calculating the following way avoids subtraction or modulo of large
+ * floats which have low result precision.
+ * interpolation1
+ * = (i / outrate * inrate) - floor(i / outrate * inrate)
+ * = mod(i / outrate * inrate, 1)
+ * = mod(i * inrate, outrate) / outrate */
+ const int srcfraction = ((Sint64) i) * inrate % outrate;
+ const float interpolation1 = ((float) srcfraction) / ((float) outrate);
+ const int filterindex1 = ((Sint32) srcfraction) * RESAMPLER_SAMPLES_PER_ZERO_CROSSING / outrate;
+ const float interpolation2 = 1.0f - interpolation1;
+ const int filterindex2 = ((Sint32) (outrate - srcfraction)) * RESAMPLER_SAMPLES_PER_ZERO_CROSSING / outrate;
for (chan = 0; chan < chans; chan++) {
float outsample = 0.0f;
/* do this twice to calculate the sample, once for the "left wing" and then same for the right. */
for (j = 0; (filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
+ const int filt_ind = filterindex1 + j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
const int srcframe = srcindex - j;
/* !!! FIXME: we can bubble this conditional out of here by doing a pre loop. */
const float insample = (srcframe < 0) ? lpadding[((paddinglen + srcframe) * chans) + chan] : inbuf[(srcframe * chans) + chan];
- outsample += (float)(insample * (ResamplerFilter[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)] + (interpolation1 * ResamplerFilterDifference[filterindex1 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)])));
+ outsample += (float)(insample * (ResamplerFilter[filt_ind] + (interpolation1 * ResamplerFilterDifference[filt_ind])));
}
/* Do the right wing! */
for (j = 0; (filterindex2 + (j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING)) < RESAMPLER_FILTER_SIZE; j++) {
- const int jsamples = j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
+ const int filt_ind = filterindex2 + j * RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
const int srcframe = srcindex + 1 + j;
/* !!! FIXME: we can bubble this conditional out of here by doing a post loop. */
const float insample = (srcframe >= inframes) ? rpadding[((srcframe - inframes) * chans) + chan] : inbuf[(srcframe * chans) + chan];
- outsample += (float)(insample * (ResamplerFilter[filterindex2 + jsamples] + (interpolation2 * ResamplerFilterDifference[filterindex2 + jsamples])));
+ outsample += (float)(insample * (ResamplerFilter[filt_ind] + (interpolation2 * ResamplerFilterDifference[filt_ind])));
}
*(dst++) = outsample;
}
-
- outtime = ((ResampleFloatType) i) / ((ResampleFloatType) outrate);
}
return outframes * chans * sizeof (float);
diff --git a/test/testautomation_audio.c b/test/testautomation_audio.c
index e6127b573001..22ee2e6b3a8a 100644
--- a/test/testautomation_audio.c
+++ b/test/testautomation_audio.c
@@ -8,6 +8,7 @@
# define _CRT_SECURE_NO_WARNINGS
#endif
+#include <math.h>
#include <stdio.h>
#include <string.h>
@@ -969,6 +970,118 @@ int audio_openCloseAudioDeviceConnected()
return TEST_COMPLETED;
}
+static double sine_wave_sample(const Sint64 idx, const Sint64 rate, const Sint64 freq, const double phase)
+{
+ /* Using integer modulo to avoid precision loss caused by large floating
+ * point numbers. Sint64 is needed for the large integer multiplication.
+ * The integers are assumed to be non-negative so that modulo is always
+ * non-negative.
+ * sin(i / rate * freq * 2 * M_PI + phase)
+ * = sin(mod(i / rate * freq, 1) * 2 * M_PI + phase)
+ * = sin(mod(i * freq, rate) / rate * 2 * M_PI + phase) */
+ return SDL_sin(((double) (idx * freq % rate)) / ((double) rate) * (M_PI * 2) + phase);
+}
+
+/**
+ * \brief Check signal-to-noise ratio and maximum error of audio resampling.
+ *
+ * \sa https://wiki.libsdl.org/SDL_BuildAudioCVT
+ * \sa https://wiki.libsdl.org/SDL_ConvertAudio
+ */
+int audio_resampleLoss()
+{
+ /* Note: always test long input time (>= 5s from experience) in some test
+ * cases because an improper implementation may suffer from low resampling
+ * precision with long input due to e.g. doing subtraction with large floats. */
+ struct test_spec_t {
+ int time;
+ int freq;
+ double phase;
+ int rate_in;
+ int rate_out;
+ double signal_to_noise;
+ double max_error;
+ } test_specs[] = {
+ { 50, 440, 0, 44100, 48000, 60, 0.0025 },
+ { 50, 5000, M_PI / 2, 20000, 10000, 65, 0.0010 },
+ { 0 }
+ };
+
+ int spec_idx = 0;
+
+ for (spec_idx = 0; test_specs[spec_idx].time > 0; ++spec_idx) {
+ const struct test_spec_t *spec = &test_specs[spec_idx];
+ const int frames_in = spec->time * spec->rate_in;
+ const int frames_target = spec->time * spec->rate_out;
+ const int len_in = frames_in * (int) sizeof (float);
+ const int len_target = frames_target * (int) sizeof (float);
+
+ Uint64 tick_beg = 0;
+ Uint64 tick_end = 0;
+ SDL_AudioCVT cvt;
+ int i = 0;
+ int ret = 0;
+ double max_error = 0;
+ double sum_squared_error = 0;
+ double sum_squared_value = 0;
+ double signal_to_noise = 0;
+
+ SDLTest_AssertPass("Test resampling of %i s %i Hz %f phase sine wave from sampling rate of %i Hz to %i Hz",
+ spec->time, spec->freq, spec->phase, spec->rate_in, spec->rate_out);
+
+ ret = SDL_BuildAudioCVT(&cvt, AUDIO_F32, 1, spec->rate_in, AUDIO_F32, 1, spec->rate_out);
+ SDLTest_AssertPass("Call to SDL_BuildAudioCVT(&cvt, AUDIO_F32, 1, %i, AUDIO_F32, 1, %i)", spec->rate_in, spec->rate_out);
+ SDLTest_AssertCheck(ret == 1, "Expected SDL_BuildAudioCVT to succeed and conversion to be needed.");
+ if (ret != 1) {
+ return TEST_ABORTED;
+ }
+
+ cvt.buf = (Uint8 *) SDL_malloc(len_in * cvt.len_mult);
+ SDLTest_AssertCheck(cvt.buf != NULL, "Expected input buffer to be created.");
+ if (cvt.buf == NULL) {
+ return TEST_ABORTED;
+ }
+
+ cvt.len = len_in;
+ for (i = 0; i < frames_in; ++i) {
+ *(((float *) cvt.buf) + i) = (float) sine_wave_sample(i, spec->rate_in, spec->freq, spec->phase);
+ }
+
+ tick_beg = SDL_GetPerformanceCounter();
+ ret = SDL_ConvertAudio(&cvt);
+ tick_end = SDL_GetPerformanceCounter();
+ SDLTest_AssertPass("Call to SDL_ConvertAudio(&cvt)");
+ SDLTest_AssertCheck(ret == 0, "Expected SDL_ConvertAudio to succeed.");
+ SDLTest_AssertCheck(cvt.len_cvt == len_target, "Expected output length %i, got %i.", len_target, cvt.len_cvt);
+ if (ret != 0 || cvt.len_cvt != len_target) {
+ SDL_free(cvt.buf);
+ return TEST_ABORTED;
+ }
+ SDLTest_Log("Resampling used %f seconds.", ((double) (tick_end - tick_beg)) / SDL_GetPerformanceFrequency());
+
+ for (i = 0; i < frames_target; ++i) {
+ const float output = *(((float *) cvt.buf) + i);
+ const double target = sine_wave_sample(i, spec->rate_out, spec->freq, spec->phase);
+ const double error = SDL_fabs(target - output);
+ max_error = SDL_max(max_error, error);
+ sum_squared_error += error * error;
+ sum_squared_value += target * target;
+ }
+ SDL_free(cvt.buf);
+ signal_to_noise = 10 * SDL_log10(sum_squared_value / sum_squared_error); /* decibel */
+ SDLTest_AssertCheck(isfinite(sum_squared_value), "Sum of squared target should be finite.");
+ SDLTest_AssertCheck(isfinite(sum_squared_error), "Sum of squared error should be finite.");
+ /* Infinity is theoretically possible when there is very little to no noise */
+ SDLTest_AssertCheck(!isnan(signal_to_noise), "Signal-to-noise ratio should not be NaN.");
+ SDLTest_AssertCheck(isfinite(max_error), "Maximum conversion error should be finite.");
+ SDLTest_AssertCheck(signal_to_noise >= spec->signal_to_noise, "Conversion signal-to-noise ratio %f dB should be no less than %f dB.",
+ signal_to_noise, spec->signal_to_noise);
+ SDLTest_AssertCheck(max_error <= spec->max_error, "Maximum conversion error %f should be no more than %f.",
+ max_error, spec->max_error);
+ }
+
+ return TEST_COMPLETED;
+}
/* ================= Test Case References ================== */
@@ -1024,11 +1137,14 @@ static const SDLTest_TestCaseReference audioTest14 =
static const SDLTest_TestCaseReference audioTest15 =
{ (SDLTest_TestCaseFp)audio_pauseUnpauseAudio, "audio_pauseUnpauseAudio", "Pause and Unpause audio for various audio specs while testing callback.", TEST_ENABLED };
+static const SDLTest_TestCaseReference audioTest16 =
+ { (SDLTest_TestCaseFp)audio_resampleLoss, "audio_resampleLoss", "Check signal-to-noise ratio and maximum error of audio resampling.", TEST_ENABLED };
+
/* Sequence of Audio test cases */
static const SDLTest_TestCaseReference *audioTests[] = {
&audioTest1, &audioTest2, &audioTest3, &audioTest4, &audioTest5, &audioTest6,
&audioTest7, &audioTest8, &audioTest9, &audioTest10, &audioTest11,
- &audioTest12, &audioTest13, &audioTest14, &audioTest15, NULL
+ &audioTest12, &audioTest13, &audioTest14, &audioTest15, &audioTest16, NULL
};
/* Audio test suite (global) */