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Convert ALu.c to C++

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Required changes to bsincgen to generate C++-friendly structures.
This commit is contained in:
Chris Robinson
2018-11-16 20:32:19 -08:00
parent 317acd6ae2
commit 53373a43b8
7 changed files with 75 additions and 54 deletions
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Alc/ALu.c → Alc/alu.cpp
+18 -13
View File
@@ -82,7 +82,7 @@ static HrtfDirectMixerFunc MixDirectHrtf = MixDirectHrtf_C;
void DeinitVoice(ALvoice *voice)
{
al_free(ATOMIC_EXCHANGE_PTR_SEQ(&voice->Update, NULL));
al_free(ATOMIC_EXCHANGE_PTR_SEQ(&voice->Update, static_cast<ALvoiceProps*>(nullptr)));
}
@@ -322,7 +322,8 @@ static bool CalcContextParams(ALCcontext *Context)
ALlistener *Listener = Context->Listener;
struct ALcontextProps *props;
props = ATOMIC_EXCHANGE_PTR(&Context->Update, NULL, almemory_order_acq_rel);
props = static_cast<ALcontextProps*>(ATOMIC_EXCHANGE_PTR(&Context->Update,
static_cast<ALcontextProps*>(nullptr), almemory_order_acq_rel));
if(!props) return false;
Listener->Params.MetersPerUnit = props->MetersPerUnit;
@@ -347,7 +348,8 @@ static bool CalcListenerParams(ALCcontext *Context)
struct ALlistenerProps *props;
aluVector vel;
props = ATOMIC_EXCHANGE_PTR(&Listener->Update, NULL, almemory_order_acq_rel);
props = static_cast<ALlistenerProps*>(ATOMIC_EXCHANGE_PTR(&Listener->Update,
static_cast<ALlistenerProps*>(nullptr), almemory_order_acq_rel));
if(!props) return false;
/* AT then UP */
@@ -390,7 +392,8 @@ static bool CalcEffectSlotParams(ALeffectslot *slot, ALCcontext *context, bool f
struct ALeffectslotProps *props;
ALeffectState *state;
props = ATOMIC_EXCHANGE_PTR(&slot->Update, NULL, almemory_order_acq_rel);
props = static_cast<ALeffectslotProps*>(ATOMIC_EXCHANGE_PTR(&slot->Update,
static_cast<ALeffectslotProps*>(nullptr), almemory_order_acq_rel));
if(!props && !force) return false;
if(props)
@@ -1459,7 +1462,8 @@ static void CalcSourceParams(ALvoice *voice, ALCcontext *context, bool force)
ALbufferlistitem *BufferListItem;
struct ALvoiceProps *props;
props = ATOMIC_EXCHANGE_PTR(&voice->Update, NULL, almemory_order_acq_rel);
props = static_cast<ALvoiceProps*>(ATOMIC_EXCHANGE_PTR(&voice->Update,
static_cast<ALvoiceProps*>(nullptr), almemory_order_acq_rel));
if(!props && !force) return;
if(props)
@@ -1523,8 +1527,8 @@ static void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*RESTRICT Buffer)
int lidx, int ridx, int cidx, ALsizei SamplesToDo,
ALsizei NumChannels)
{
ALfloat (*RESTRICT lsplit)[BUFFERSIZE] = ASSUME_ALIGNED(Stablizer->LSplit, 16);
ALfloat (*RESTRICT rsplit)[BUFFERSIZE] = ASSUME_ALIGNED(Stablizer->RSplit, 16);
ALfloat (*RESTRICT lsplit)[BUFFERSIZE] = Stablizer->LSplit;
ALfloat (*RESTRICT rsplit)[BUFFERSIZE] = Stablizer->RSplit;
ALsizei i;
/* Apply an all-pass to all channels, except the front-left and front-
@@ -1572,13 +1576,12 @@ static void ApplyDistanceComp(ALfloat (*RESTRICT Samples)[BUFFERSIZE], DistanceC
{
ALsizei i, c;
Values = ASSUME_ALIGNED(Values, 16);
for(c = 0;c < numchans;c++)
{
ALfloat *RESTRICT inout = ASSUME_ALIGNED(Samples[c], 16);
ALfloat *RESTRICT inout = Samples[c];
const ALfloat gain = distcomp[c].Gain;
const ALsizei base = distcomp[c].Length;
ALfloat *RESTRICT distbuf = ASSUME_ALIGNED(distcomp[c].Buffer, 16);
ALfloat *RESTRICT distbuf = distcomp[c].Buffer;
if(base == 0)
{
@@ -1679,7 +1682,7 @@ static void Write##A(const ALfloat (*RESTRICT InBuffer)[BUFFERSIZE], \
\
for(j = 0;j < numchans;j++) \
{ \
const ALfloat *RESTRICT in = ASSUME_ALIGNED(InBuffer[j], 16); \
const ALfloat *RESTRICT in = InBuffer[j]; \
T *RESTRICT out = (T*)OutBuffer + Offset*numchans + j; \
\
for(i = 0;i < SamplesToDo;i++) \
@@ -1745,7 +1748,8 @@ void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples)
{
if(!MixSource(voice, source->id, ctx, SamplesToDo))
{
ATOMIC_STORE(&voice->Source, NULL, almemory_order_relaxed);
ATOMIC_STORE(&voice->Source, static_cast<ALsource*>(nullptr),
almemory_order_relaxed);
ATOMIC_STORE(&voice->Playing, false, almemory_order_release);
SendSourceStoppedEvent(ctx, source->id);
}
@@ -1862,7 +1866,8 @@ void aluHandleDisconnect(ALCdevice *device, const char *msg, ...)
ALvoice *voice = ctx->Voices[i];
ALsource *source;
source = ATOMIC_EXCHANGE_PTR(&voice->Source, NULL, almemory_order_relaxed);
source = static_cast<ALsource*>(ATOMIC_EXCHANGE_PTR(&voice->Source,
static_cast<ALsource*>(nullptr), almemory_order_relaxed));
if(source && ATOMIC_LOAD(&voice->Playing, almemory_order_relaxed))
{
/* If the source's voice was playing, it's now effectively
Alc/filters/defs.h
+8
View File
@@ -4,6 +4,10 @@
#include "AL/al.h"
#include "math_defs.h"
#ifdef __cplusplus
extern "C" {
#endif
/* Filters implementation is based on the "Cookbook formulae for audio
* EQ biquad filter coefficients" by Robert Bristow-Johnson
* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
@@ -109,4 +113,8 @@ inline void BiquadFilter_passthru(BiquadFilter *filter, ALsizei numsamples)
}
}
#ifdef __cplusplus
} // extern "C"
#endif
#endif /* ALC_FILTER_H */
Alc/mixer/defs.h
+8
View File
@@ -6,6 +6,10 @@
#include "alMain.h"
#include "alu.h"
#ifdef __cplusplus
extern "C" {
#endif
struct MixGains;
struct MixHrtfParams;
@@ -116,4 +120,8 @@ const ALfloat *Resample_bsinc_Neon(const InterpState *state, const ALfloat *REST
ALsizei frac, ALint increment, ALfloat *RESTRICT dst,
ALsizei dstlen);
#ifdef __cplusplus
} // extern "C"
#endif
#endif /* MIXER_DEFS_H */
Alc/mixvoice.c
+1 -1
View File
@@ -261,7 +261,7 @@ static void LoadSamples(ALfloat *RESTRICT dst, const ALvoid *RESTRICT src, ALint
static const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter,
ALfloat *RESTRICT dst, const ALfloat *RESTRICT src,
ALsizei numsamples, enum ActiveFilters type)
ALsizei numsamples, int type)
{
ALsizei i;
switch(type)
CMakeLists.txt
+1 -1
View File
@@ -799,7 +799,7 @@ SET(OPENAL_OBJS
)
SET(ALC_OBJS
Alc/alc.cpp
Alc/ALu.c
Alc/alu.cpp
Alc/alconfig.cpp
Alc/alconfig.h
Alc/bs2b.c
OpenAL32/Include/alu.h
+3 -3
View File
@@ -136,7 +136,7 @@ inline void aluMatrixfSet(aluMatrixf *matrix, ALfloat m00, ALfloat m01, ALfloat
}
enum ActiveFilters {
enum {
AF_None = 0,
AF_LowPass = 1,
AF_HighPass = 2,
@@ -285,7 +285,7 @@ typedef struct ALvoice {
InterpState ResampleState;
struct {
enum ActiveFilters FilterType;
int FilterType;
DirectParams Params[MAX_INPUT_CHANNELS];
ALfloat (*Buffer)[BUFFERSIZE];
@@ -294,7 +294,7 @@ typedef struct ALvoice {
} Direct;
struct {
enum ActiveFilters FilterType;
int FilterType;
SendParams Params[MAX_INPUT_CHANNELS];
ALfloat (*Buffer)[BUFFERSIZE];
native-tools/bsincgen.c
+36 -36
View File
@@ -249,6 +249,11 @@ static void BsiGenerateTables(FILE *output, const char *tabname, const double re
for(si = 0; si < BSINC_SCALE_COUNT; si++)
mt[si] = (mt[si]+3) & ~3;
// Calculate the table size.
i = 0;
for(si = 0; si < BSINC_SCALE_COUNT; si++)
i += 4 * BSINC_PHASE_COUNT * mt[si];
fprintf(output,
"/* This %d%s order filter has a rejection of -%.0fdB, yielding a transition width\n"
" * of ~%.3f (normalized frequency). Order increases when downsampling to a\n"
@@ -256,12 +261,36 @@ static void BsiGenerateTables(FILE *output, const char *tabname, const double re
" * width) suffers to reduce the CPU cost. The bandlimiting will cut all sound\n"
" * after downsampling by ~%.2f octaves.\n"
" */\n"
"const BSincTable %s = {\n",
"alignas(16) constexpr float %s_tab[%d] = {\n",
order, (((order%100)/10) == 1) ? "th" :
((order%10) == 1) ? "st" :
((order%10) == 2) ? "nd" :
((order%10) == 3) ? "rd" : "th",
rejection, width, log2(1.0/scaleBase), tabname);
rejection, width, log2(1.0/scaleBase), tabname, i);
for(si = 0; si < BSINC_SCALE_COUNT; si++)
{
const int m = mt[si];
const int o = num_points_min - (m / 2);
for(pi = 0; pi < BSINC_PHASE_COUNT; pi++)
{
fprintf(output, " /* %2d,%2d (%d) */", si, pi, m);
fprintf(output, "\n ");
for(i = 0; i < m; i++)
fprintf(output, " %+14.9ef,", filter[si][pi][o + i]);
fprintf(output, "\n ");
for(i = 0; i < m; i++)
fprintf(output, " %+14.9ef,", scDeltas[si][pi][o + i]);
fprintf(output, "\n ");
for(i = 0; i < m; i++)
fprintf(output, " %+14.9ef,", phDeltas[si][pi][o + i]);
fprintf(output, "\n ");
for(i = 0; i < m; i++)
fprintf(output, " %+14.9ef,", spDeltas[si][pi][o + i]);
fprintf(output, "\n");
}
}
fprintf(output, "};\nconstexpr BSincTable %s = {\n", tabname);
/* The scaleBase is calculated from the Kaiser window transition width.
It represents the absolute limit to the filter before it fully cuts
@@ -286,37 +315,8 @@ static void BsiGenerateTables(FILE *output, const char *tabname, const double re
}
fprintf(output, " },\n");
// Calculate the table size.
i = 0;
for(si = 0; si < BSINC_SCALE_COUNT; si++)
i += 4 * BSINC_PHASE_COUNT * mt[si];
fprintf(output, "\n /* Tab (%d entries) */ {\n", i);
for(si = 0; si < BSINC_SCALE_COUNT; si++)
{
const int m = mt[si];
const int o = num_points_min - (m / 2);
for(pi = 0; pi < BSINC_PHASE_COUNT; pi++)
{
fprintf(output, " /* %2d,%2d (%d) */", si, pi, m);
fprintf(output, "\n ");
for(i = 0; i < m; i++)
fprintf(output, " %+14.9ef,", filter[si][pi][o + i]);
fprintf(output, "\n ");
for(i = 0; i < m; i++)
fprintf(output, " %+14.9ef,", scDeltas[si][pi][o + i]);
fprintf(output, "\n ");
for(i = 0; i < m; i++)
fprintf(output, " %+14.9ef,", phDeltas[si][pi][o + i]);
fprintf(output, "\n ");
for(i = 0; i < m; i++)
fprintf(output, " %+14.9ef,", spDeltas[si][pi][o + i]);
fprintf(output, "\n");
}
}
fprintf(output, " }\n};\n\n");
fprintf(output, " %s_tab\n", tabname);
fprintf(output, "};\n\n");
}
@@ -348,12 +348,12 @@ int main(int argc, char *argv[])
"static_assert(BSINC_SCALE_COUNT == %d, \"Unexpected BSINC_SCALE_COUNT value!\");\n"
"static_assert(BSINC_PHASE_COUNT == %d, \"Unexpected BSINC_PHASE_COUNT value!\");\n"
"static_assert(FRACTIONONE == %d, \"Unexpected FRACTIONONE value!\");\n\n"
"typedef struct BSincTable {\n"
"struct BSincTable {\n"
" const float scaleBase, scaleRange;\n"
" const int m[BSINC_SCALE_COUNT];\n"
" const int filterOffset[BSINC_SCALE_COUNT];\n"
" alignas(16) const float Tab[];\n"
"} BSincTable;\n\n", BSINC_SCALE_COUNT, BSINC_PHASE_COUNT, FRACTIONONE);
" const float *Tab;\n"
"};\n\n", BSINC_SCALE_COUNT, BSINC_PHASE_COUNT, FRACTIONONE);
/* A 23rd order filter with a -60dB drop at nyquist. */
BsiGenerateTables(output, "bsinc24", 60.0, 23);
/* An 11th order filter with a -40dB drop at nyquist. */