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Used fixed-point arithmetic in ResampleAudio

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Brick
2023-08-20 16:08:19 +01:00
committed by Ryan C. Gordon
parent 7bb4e806ea
commit 47fea7f06b
4 changed files with 37 additions and 42 deletions
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6
build-scripts/gen_audio_resampler_filter.c
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@@ -47,7 +47,8 @@ gcc -o genfilter build-scripts/gen_audio_resampler_filter.c -lm && ./genfilter >
#define RESAMPLER_ZERO_CROSSINGS 5
#define RESAMPLER_BITS_PER_SAMPLE 16
#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1))
#define RESAMPLER_BITS_PER_ZERO_CROSSING ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1)
#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << RESAMPLER_BITS_PER_ZERO_CROSSING)
#define RESAMPLER_FILTER_SIZE (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS)
/* This is a "modified" bessel function, so you can't use POSIX j0() */
@@ -135,7 +136,8 @@ int main(void)
"\n"
"#define RESAMPLER_ZERO_CROSSINGS %d\n"
"#define RESAMPLER_BITS_PER_SAMPLE %d\n"
"#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1))\n"
"#define RESAMPLER_BITS_PER_ZERO_CROSSING ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1)\n"
"#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << RESAMPLER_BITS_PER_ZERO_CROSSING)\n"
"#define RESAMPLER_FILTER_SIZE (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS)\n"
"\n", RESAMPLER_ZERO_CROSSINGS, RESAMPLER_BITS_PER_SAMPLE
);

3
src/audio/SDL_audio_resampler_filter.h
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@@ -23,7 +23,8 @@
#define RESAMPLER_ZERO_CROSSINGS 5
#define RESAMPLER_BITS_PER_SAMPLE 16
#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1))
#define RESAMPLER_BITS_PER_ZERO_CROSSING ((RESAMPLER_BITS_PER_SAMPLE / 2) + 1)
#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING (1 << RESAMPLER_BITS_PER_ZERO_CROSSING)
#define RESAMPLER_FILTER_SIZE (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * RESAMPLER_ZERO_CROSSINGS)
static const float ResamplerFilter[RESAMPLER_FILTER_SIZE] = {

68
src/audio/SDL_audiocvt.c
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@@ -32,22 +32,21 @@
#include "SDL_audio_resampler_filter.h"
static int GetResamplerPaddingFrames(const int iinrate, const int ioutrate)
static Sint64 GetResampleRate(const int inrate, const int 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. */
const Sint32 inrate = (Sint32) iinrate;
const Sint32 outrate = (Sint32) ioutrate;
SDL_assert(inrate >= 0);
SDL_assert(outrate >= 0);
return ((Sint64)inrate << 32) / (Sint64)outrate;
}
static int GetResamplerPaddingFrames(const int inrate, const int outrate)
{
SDL_assert(inrate > 0);
SDL_assert(outrate > 0);
if (inrate == outrate) {
return 0;
} else if (inrate > outrate) {
return (int) (((RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) + (outrate - 1)) / outrate);
}
return RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
return RESAMPLER_ZERO_CROSSINGS + 1;
}
static int GetHistoryBufferSampleFrames(const Sint32 required_resampler_frames)
@@ -68,36 +67,28 @@ static int GetHistoryBufferSampleFrames(const Sint32 required_resampler_frames)
static void ResampleAudio(const int chans, const int inrate, const int outrate,
const float *lpadding, const float *rpadding,
const float *inbuf, const int inframes,
float *outbuf, const int outframes, const int offset)
float *outbuf, const int outframes, const Sint64 offset)
{
/* 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 = GetResamplerPaddingFrames(inrate, outrate);
float *dst = outbuf;
int i, j, chan;
const Sint64 srcstep = GetResampleRate(inrate, outrate);
Sint64 srcpos = offset;
for (i = 0; i < outframes; i++) {
/* Offset by outrate to avoid negative numbers (which don't round correctly) */
Sint64 srcpos = ((Sint64)i * inrate) + offset + outrate;
int srcindex = (int)(srcpos / outrate) - 1;
int srcindex = (int)(Sint32)(srcpos >> 32);
Uint32 srcfraction = (Uint32)(srcpos & 0xFFFFFFFF);
srcpos += srcstep;
SDL_assert(srcindex >= -1);
SDL_assert(srcindex < inframes);
/* 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 = (int)(srcpos % outrate);
const float interpolation1 = ((float)srcfraction) / ((float)outrate);
const int filterindex1 = (((Sint32)srcfraction) * RESAMPLER_SAMPLES_PER_ZERO_CROSSING / outrate) * RESAMPLER_ZERO_CROSSINGS;
const float interpolation1 = (float)srcfraction * 0x1p-32f;
const int filterindex1 = (int)(srcfraction >> (32 - RESAMPLER_BITS_PER_ZERO_CROSSING)) * RESAMPLER_ZERO_CROSSINGS;
const float interpolation2 = 1.0f - interpolation1;
const int filterindex2 = RESAMPLER_FILTER_SIZE - filterindex1 - RESAMPLER_ZERO_CROSSINGS;
const int filterindex2 = RESAMPLER_FILTER_SIZE - RESAMPLER_ZERO_CROSSINGS - filterindex1;
for (chan = 0; chan < chans; chan++) {
float outsample = 0.0f;
@@ -834,7 +825,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int le
int future_buffer_filled_frames = stream->future_buffer_filled_frames;
Uint8 *future_buffer = stream->future_buffer;
Uint8 *history_buffer = stream->history_buffer;
int resample_offset = stream->resample_offset;
Sint64 resample_offset = stream->resample_offset;
float *resample_outbuf;
int input_frames;
int output_frames;
@@ -861,13 +852,14 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int le
input_frames = output_frames; // total sample frames caller wants
if (dst_rate != src_rate) {
Sint64 last_offset = ((Sint64)(input_frames - 1) * src_rate) + resample_offset;
input_frames = (int)(last_offset / dst_rate) + 1;
// Make sure this matches the logic used in ResampleAudio
const Sint64 srcstep = GetResampleRate(src_rate, dst_rate);
Sint64 next_offset = last_offset + src_rate;
stream->resample_offset = (int)(next_offset - ((Sint64)input_frames * dst_rate));
Sint64 nextpos = (output_frames * srcstep) + resample_offset;
Sint64 lastpos = nextpos - srcstep;
// SDL_Log("Fraction: %i, %i", resampler_fraction, stream->resampler_fraction);
input_frames = (int)(Sint32)(lastpos >> 32) + 1;
stream->resample_offset = nextpos - ((Sint64)input_frames << 32);
if (input_frames == 0) { // uhoh, not enough input frames!
// if they are upsampling and we end up needing less than a frame of input, we reject it because it would cause artifacts on future reads to eat a full input frame.

2
src/audio/SDL_sysaudio.h
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@@ -179,7 +179,7 @@ struct SDL_AudioStream
int resampler_padding_frames;
int history_buffer_frames;
int future_buffer_filled_frames;
int resample_offset;
Sint64 resample_offset;
SDL_AudioSpec src_spec;
SDL_AudioSpec dst_spec;