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alSoftware.c
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alSoftware.c
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/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2000 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "alInternal.h"
#if HAVE_PRAGMA_EXPORT
#pragma export off
#endif
ALboolean __alRingBufferInit(__ALRingBuffer *ring, ALsizei size)
{
UInt8 *ptr = (UInt8 *) realloc(ring->buffer, size);
if (ptr == NULL)
return(AL_FALSE);
ring->buffer = ptr;
ring->size = size;
ring->write = 0;
ring->read = 0;
ring->used = 0;
return(AL_TRUE);
} // __alRingBufferInit
ALvoid __alRingBufferShutdown(__ALRingBuffer *ring)
{
free(ring->buffer);
ring->buffer = NULL;
} // __alRingBufferShutdown
ALsizei __alRingBufferSize(__ALRingBuffer *ring)
{
return(ring->used);
} // __alRingBufferSize
ALvoid __alRingBufferPut(__ALRingBuffer *ring, UInt8 *data, ALsizei _size)
{
register ALsizei size = _size;
register ALsizei cpy;
register ALsizei avail;
if (!size) // just in case...
return;
// Putting more data than ring buffer holds in total? Replace it all.
if (size > ring->size)
{
ring->write = 0;
ring->read = 0;
ring->used = ring->size;
memcpy(ring->buffer, data + (size - ring->size), ring->size);
return;
} // if
// Buffer overflow? Push read pointer to oldest sample not overwritten...
avail = ring->size - ring->used;
if (size > avail)
{
ring->read += size - avail;
if (ring->read > ring->size)
ring->read -= ring->size;
} // if
// Clip to end of buffer and copy first block...
cpy = ring->size - ring->write;
if (size < cpy)
cpy = size;
if (cpy) memcpy(ring->buffer + ring->write, data, cpy);
// Wrap around to front of ring buffer and copy remaining data...
avail = size - cpy;
if (avail) memcpy(ring->buffer, data + cpy, avail);
// Update write pointer...
ring->write += size;
if (ring->write > ring->size)
ring->write -= ring->size;
ring->used += size;
if (ring->used > ring->size)
ring->used = ring->size;
} // __alRingBufferPut
ALsizei __alRingBufferGet(__ALRingBuffer *ring, UInt8 *data, ALsizei _size)
{
register ALsizei cpy;
register ALsizei size = _size;
register ALsizei avail = ring->used;
// Clamp amount to read to available data...
if (size > avail)
size = avail;
// Clip to end of buffer and copy first block...
cpy = ring->size - ring->read;
if (cpy > size) cpy = size;
if (cpy) memcpy(data, ring->buffer + ring->read, cpy);
// Wrap around to front of ring buffer and copy remaining data...
avail = size - cpy;
if (avail) memcpy(data + cpy, ring->buffer, avail);
// Update read pointer...
ring->read += size;
if (ring->read > ring->size)
ring->read -= ring->size;
ring->used -= size;
return(size); // may have been clamped if there wasn't enough data...
} // __alRingBufferGet
// Converters ...
ALboolean __alConvertFromMonoFloat32(Float32 *s, UInt8 *d,
ALenum fmt, ALsizei samples)
{
// !!! FIXME: Altivec this!
register Float32 *src = s;
register ALsizei i;
switch (fmt)
{
#if SUPPORTS_AL_EXT_FLOAT32
case AL_FORMAT_MONO_FLOAT32:
memcpy(d, src, samples * sizeof (Float32));
return(AL_TRUE);
case AL_FORMAT_STEREO_FLOAT32:
{
register Float32 samp;
register Float32 *dst = (Float32 *) d;
for (i = 0; i < samples; i++)
{
samp = *src;
dst[0] = samp;
dst[1] = samp;
dst += 2;
src++;
} // for
return(AL_TRUE);
} // case
#endif
case AL_FORMAT_MONO8:
{
register UInt8 *dst = (UInt8 *) d;
for (i = 0; i < samples; i++)
{
*dst = (UInt8) ((*src * 127.0f) + 127.0f);
dst++;
src++;
} // for
return(AL_TRUE);
} // case
case AL_FORMAT_STEREO8:
{
register UInt8 *dst = (UInt8 *) d;
register UInt8 val;
for (i = 0; i < samples; i++)
{
val = (UInt8) ((*src * 127.0f) + 127.0f);
dst[0] = val;
dst[1] = val;
dst += 2;
src++;
} // for
return(AL_TRUE);
} // case
case AL_FORMAT_MONO16:
{
register SInt16 *dst = (SInt16 *) d;
for (i = 0; i < samples; i++)
{
*dst = (SInt16) (*src * 32767.0f);
dst++;
src++;
} // for
return(AL_TRUE);
} // case
case AL_FORMAT_STEREO16:
{
register SInt16 *dst = (SInt16 *) d;
register SInt16 val;
for (i = 0; i < samples; i++)
{
val = (SInt16) (*src * 32767.0f);
dst[0] = val;
dst[1] = val;
dst += 2;
src++;
} // for
return(AL_TRUE);
} // case
} // switch
return(AL_FALSE);
} // __alConvertFromMonoFloat32
ALboolean __alConvertFromStereoFloat32(Float32 *s, UInt8 *d,
ALenum fmt, ALsizei samples)
{
// !!! FIXME: Altivec this!
register Float32 samp;
register Float32 *src = s;
register ALsizei i;
switch (fmt)
{
#if SUPPORTS_AL_EXT_FLOAT32
case AL_FORMAT_MONO_FLOAT32:
{
register Float32 *dst = (Float32 *) d;
for (i = 0; i < samples; i++)
{
*dst = (src[0] + src[1]) * 0.5f;
dst++;
src += 2;
} // for
return(AL_TRUE);
} // case
case AL_FORMAT_STEREO_FLOAT32:
memcpy(d, src, samples * (sizeof (Float32) * 2));
return(AL_TRUE);
#endif
case AL_FORMAT_MONO8:
{
register UInt8 *dst = (UInt8 *) d;
for (i = 0; i < samples; i++)
{
samp = ((src[0] + src[1]) * 0.5f);
*dst = (UInt8) ((samp * 127.0f) + 127.0f);
dst++;
src += 2;
} // for
return(AL_TRUE);
} // case
case AL_FORMAT_STEREO8:
{
register UInt8 *dst = (UInt8 *) d;
samples <<= 1;
for (i = 0; i < samples; i++)
{
*dst = (UInt8) ((*src * 127.0f) + 127.0f);
dst++;
src++;
} // for
return(AL_TRUE);
} // case
case AL_FORMAT_MONO16:
{
register SInt16 *dst = (SInt16 *) d;
for (i = 0; i < samples; i++)
{
samp = ((src[0] + src[1]) * 0.5f);
*dst = (SInt16) (samp * 32767.0f);
dst++;
src += 2;
} // for
return(AL_TRUE);
} // case
case AL_FORMAT_STEREO16:
{
register SInt16 *dst = (SInt16 *) d;
samples <<= 1;
for (i = 0; i < samples; i++)
{
*dst = (SInt16) (*src * 32767.0f);
dst++;
src++;
} // for
return(AL_TRUE);
} // case
} // switch
return(AL_FALSE);
} // __alConvertFromStereoFloat32
ALvoid __alRecalcMonoSource(ALcontext *ctx, ALsource *src)
{
#if DO_DOPPLER
ALvector SourceListenerDir;
#endif
ALvector Position;
ALvector Velocity;
ALvector U;
ALvector V;
ALvector N;
ALfloat Matrix[3][3];
ALfloat Distance;
ALfloat minGain, maxGain;
ALfloat CalcVolume;
ALenum distanceModel;
ALlistener *listener;
// Only process if some source attribute has changed.
// if (!src->needsRecalculation)
// return;
src->needsRecalculation = AL_FALSE;
distanceModel = ctx->distanceModel;
listener = &ctx->listener;
// Convert source position to listener's coordinate system
minGain = src->minGain;
maxGain = src->maxGain;
__alVectorAssign3(Position, src->Position);
__alVectorAssign3(Velocity, src->Velocity);
// Translate Listener to origin if not in AL_SOURCE_RELATIVE mode
if (src->srcRelative == AL_FALSE)
__alVectorSubtract3(Position, listener->Position);
#if DO_DOPPLER
// Set SourceListenerDir to be used later for doppler
__alVectorAssign3(SourceListenerDir, Position);
#endif
// Align coordinate system axis
__alCrossproduct(listener->Up,listener->Forward,U);
__alNormalize(U);
__alCrossproduct(listener->Forward,U,V);
__alNormalize(V);
__alVectorAssign3(N, listener->Forward);
__alNormalize(N);
Matrix[0][0]=U[0]; Matrix[0][1]=V[0]; Matrix[0][2]=-N[0];
Matrix[1][0]=U[1]; Matrix[1][1]=V[1]; Matrix[1][2]=-N[1];
Matrix[2][0]=U[2]; Matrix[2][1]=V[2]; Matrix[2][2]=-N[2];
__alMatrixVector(Matrix,Position);
// Convert into falloff and panning
Distance = __alSqrt(Position[0]*Position[0]+Position[1]*Position[1]+Position[2]*Position[2]);
if ((distanceModel != AL_NONE) && (Distance > 0.0f))
{
if (distanceModel == AL_INVERSE_DISTANCE_CLAMPED)
{
if (Distance < src->referenceDistance)
Distance = src->referenceDistance;
if (Distance > src->maxDistance)
Distance = src->maxDistance;
} // if
CalcVolume=(src->gain*listener->Gain*src->referenceDistance)/(src->referenceDistance+src->rolloffFactor*(Distance-src->referenceDistance));
} // if
else
{
CalcVolume=(src->gain*listener->Gain);
} // else
// cap Volume by min/max gains
if (CalcVolume < minGain)
CalcVolume = minGain;
else if (CalcVolume > maxGain)
CalcVolume = maxGain;
if (CalcVolume > 1.0f)
CalcVolume = 1.0f;
// !!! FIXME: Where the hell did this come from?
//CalcVolume *= FULL_VOLUME;
if (ctx->device->speakerConfig == SPKCFG_STDSTEREO)
{
if (!EPSILON_EQUAL(Distance, 0.0f))
{
register ALfloat Panning;
// this is broken, pushes everything to the right.
//Panning = 2.0f * ( (0.5f + (0.5f * 0.707f)) *
// cosf(atan2f(Position[2],Position[0])) );
// Jonas Echterhoff caught the problem...too many parentheses!
Panning = 2.0f * ( 0.5f + (0.5f * 0.707f) *
(ALfloat) (cos(atan2f(Position[2],Position[0]))) );
src->channelGain0 = CalcVolume * Panning * ctx->device->speakergains[0];
src->channelGain1 = CalcVolume * (2.0f - Panning) * ctx->device->speakergains[1];
} // if
else
{
src->channelGain0 = CalcVolume * ctx->device->speakergains[0];
src->channelGain1 = CalcVolume * ctx->device->speakergains[1];
} // else
CLAMP(src->channelGain0, 0.0f, 1.0f);
CLAMP(src->channelGain1, 0.0f, 1.0f);
} // if
else if (ctx->device->speakerConfig == SPKCFG_POSATTENUATION)
{
assert(0); // !!! FIXME
} // else if
else
{
#if USE_VBAP
__alDoVBAP(ctx, src, Position, CalcVolume);
#else
assert(0); // houston, we have a problem. Not stereo, but no VBAP?!
#endif
} // else
#if DO_DOPPLER
// Calculate doppler
if (!((SourceListenerDir[0]==0)&&(SourceListenerDir[1]==0)&&(SourceListenerDir[2]==0)))
{
ALfloat CalcPitch = src->pitch;
ALfloat dopplerVelocity = ctx->dopplerVelocity;
ALfloat ListenerSpeed, SourceSpeed;
alNormalize(SourceListenerDir);
ListenerSpeed = -alDotproduct(SourceListenerDir, listener->Velocity);
SourceSpeed = alDotproduct(SourceListenerDir, src->Velocity);
CalcPitch=CalcPitch+CalcPitch*ctx->dopplerFactor*((dopplerVelocity-ListenerSpeed)/(dopplerVelocity+SourceSpeed)-1);
if (CalcPitch < 0.5f)
CalcPitch = 0.5f;
else if (CalcPitch > 2.0f)
CalcPitch = 2.0f;
src->pitch = CalcPitch;
} // if
#endif
} // __alRecalcMonoSource
// Mixing...
// !!! FIXME: Make these mixers not suck.
// !!! FIXME: ... and do altivec versions of them, too.
UInt32 __alMixMono8(ALcontext *ctx, ALbuffer *buf, ALsource *src, Float32 *_dst, UInt32 frames)
{
register UInt32 buflen = buf->allocatedSpace - src->bufferReadIndex;
register UInt32 channels;
register ALsizei overflow = 0;
register SInt8 *in;
register Float32 *dst;
register Float32 *max;
register Float32 sample;
register Float32 gain0;
register Float32 gain1;
register Float32 gain2;
register ALsizei channel0;
register ALsizei channel1;
register ALsizei channel2;
__alRecalcMonoSource(ctx, src);
gain0 = src->channelGain0 * DIVBY127;
gain1 = src->channelGain1 * DIVBY127;
gain2 = src->channelGain2 * DIVBY127;
channel0 = src->channel0;
channel1 = src->channel1;
channel2 = src->channel2;
overflow = frames - buflen;
if (overflow < 0)
overflow = 0;
else
frames = buflen;
// Too quiet to be worth playing? Just skip ahead in the buffer appropriately.
if ((gain0 > MINVOL) || (gain1 > MINVOL) || (gain2 > MINVOL))
{
// Mix to the output buffer...
channels = ctx->device->streamFormat.mChannelsPerFrame;
dst = _dst;
max = dst + (frames * channels);
in = ((SInt8 *) buf->mixData) + src->bufferReadIndex;
while (dst < max)
{
sample = ((Float32) *in);
in++;
dst[channel0] += sample * gain0;
dst[channel1] += sample * gain1;
dst[channel2] += sample * gain2;
dst += channels; // skip rest of channels.
} // while
} // if
src->bufferReadIndex += frames;
return(overflow);
} // __alMixMono8
UInt32 __alMixMono16(ALcontext *ctx, ALbuffer *buf, ALsource *src, Float32 *_dst, UInt32 frames)
{
register UInt32 buflen = (buf->allocatedSpace >> 1) - src->bufferReadIndex;
register UInt32 channels;
register ALsizei overflow = 0;
register SInt16 *in;
register Float32 *dst;
register Float32 *max;
register Float32 sample;
register Float32 gain0;
register Float32 gain1;
register Float32 gain2;
register ALsizei channel0;
register ALsizei channel1;
register ALsizei channel2;
__alRecalcMonoSource(ctx, src);
gain0 = src->channelGain0;
gain1 = src->channelGain1;
gain2 = src->channelGain2;
channel0 = src->channel0;
channel1 = src->channel1;
channel2 = src->channel2;
overflow = frames - buflen;
if (overflow < 0)
overflow = 0;
else
frames = buflen;
// Too quiet to be worth playing? Just skip ahead in the buffer appropriately.
if ((gain0 > MINVOL) || (gain1 > MINVOL) || (gain2 > MINVOL))
{
// Mix to the output buffer...
gain0 *= DIVBY32767;
gain1 *= DIVBY32767;
gain2 *= DIVBY32767;
channels = ctx->device->streamFormat.mChannelsPerFrame;
dst = _dst;
max = dst + (frames * channels);
in = ((SInt16 *) buf->mixData) + src->bufferReadIndex;
while (dst < max)
{
sample = ((Float32) *in);
in++;
dst[channel0] += sample * gain0;
dst[channel1] += sample * gain1;
dst[channel2] += sample * gain2;
dst += channels; // skip rest of channels.
} // while
} // if
src->bufferReadIndex += frames;
return(overflow);
} // __alMixMono16
// !!! FIXME: Stereo samples don't spatialize; what speakers should be used
// !!! FIXME: in a > stereo configurations?
UInt32 __alMixStereo8(ALcontext *ctx, ALbuffer *buf, ALsource *src, Float32 *_dst, UInt32 frames)
{
register UInt32 buflen = (buf->allocatedSpace >> 1) - src->bufferReadIndex;
register UInt32 channels;
register ALsizei overflow = 0;
register SInt8 *in;
register Float32 *dst;
register Float32 *max;
register Float32 gainLeft;
register Float32 gainRight;
gainLeft = gainRight = src->gain * ctx->listener.Gain;
gainLeft *= ctx->device->speakergains[0];
gainRight *= ctx->device->speakergains[1];
overflow = frames - buflen;
if (overflow < 0)
overflow = 0;
else
frames = buflen;
// Too quiet to be worth playing? Just skip ahead in the buffer appropriately.
if ((gainLeft > MINVOL) || (gainRight > MINVOL))
{
// Mix to the output buffer...
channels = ctx->device->streamFormat.mChannelsPerFrame;
dst = _dst;
max = dst + (frames * channels);
in = ((SInt8 *) buf->mixData) + (src->bufferReadIndex << 1);
gainLeft *= DIVBY127;
gainRight *= DIVBY127;
while (dst < max)
{
dst[0] += ((Float32) in[0]) * gainLeft;
dst[1] += ((Float32) in[1]) * gainRight;
in += 2;
dst += channels; // skip rest of channels.
} // while
} // if
src->bufferReadIndex += frames;
return(overflow);
} // __alMixStereo8
UInt32 __alMixStereo16(ALcontext *ctx, ALbuffer *buf, ALsource *src, Float32 *_dst, UInt32 frames)
{
register UInt32 buflen = (buf->allocatedSpace >> 2) - src->bufferReadIndex;
register UInt32 channels;
register ALsizei overflow = 0;
register SInt16 *in;
register Float32 *dst;
register Float32 *max;
register Float32 gainLeft;
register Float32 gainRight;
gainLeft = gainRight = src->gain * ctx->listener.Gain;
gainLeft *= ctx->device->speakergains[0];
gainRight *= ctx->device->speakergains[1];
overflow = frames - buflen;
if (overflow < 0)
overflow = 0;
else
frames = buflen;
// Too quiet to be worth playing? Just skip ahead in the buffer appropriately.
if ((gainLeft > MINVOL) || (gainRight > MINVOL))
{
// Mix to the output buffer...
channels = ctx->device->streamFormat.mChannelsPerFrame;
dst = _dst;
max = dst + (frames * channels);
in = ((SInt16 *) buf->mixData) + (src->bufferReadIndex << 1);
gainLeft *= DIVBY32767;
gainRight *= DIVBY32767;
while (dst < max)
{
dst[0] += ((Float32) in[0]) * gainLeft;
dst[1] += ((Float32) in[1]) * gainRight;
in += 2;
dst += channels; // skip rest of channels.
} // while
} // if
src->bufferReadIndex += frames;
return(overflow);
} // __alMixStereo16
// !!! FIXME: Altivec mixers only work in stereo configurations.
UInt32 __alMixMono8_altivec(ALcontext *ctx, ALbuffer *buf, ALsource *src, Float32 *_dst, UInt32 frames)
{
register ALsizei overflow = 0;
#if __POWERPC__
register UInt32 buflen = buf->allocatedSpace - src->bufferReadIndex;
register SInt8 *in = ((SInt8 *) buf->mixData) + src->bufferReadIndex;
register Float32 *dst = _dst;
register Float32 *max;
register Float32 gainLeft;
register Float32 gainRight;
register Float32 sample;
register size_t dsti = ((size_t) dst) & 0x0000000F;
assert(ctx->device->streamFormat.mChannelsPerFrame == 2);
__alRecalcMonoSource(ctx, src);
gainLeft = src->channelGain0;
gainRight = src->channelGain1;
if (dsti) // Destination isn't 16-byte aligned?
{
if ((dsti % 2) && (frames > dsti)) // bump dst to 16-byte alignment.
{
register Float32 convGainLeft = gainLeft * DIVBY127;
register Float32 convGainRight = gainRight * DIVBY127;
while (UNALIGNED_PTR(dst))
{
sample = ((Float32) *in);
in++;
dst[0] += sample * convGainLeft;
dst[1] += sample * convGainRight;
dst += 2;
frames--;
} // while
} // if
else // dst will never align in this run; use scalar version. :(
{
return(__alMixMono8(ctx, buf, src, dst, frames));
} // else
} // if
overflow = frames - buflen;
if (overflow < 0)
overflow = 0;
else
frames = buflen;
// Too quiet to be worth playing? Just skip ahead in the buffer appropriately.
if ((gainLeft > MINVOL) || (gainRight > MINVOL))
{
// Permute vectors to convert mono channel to stereo.
register vector unsigned char permute1 = (vector unsigned char)
( 0x00, 0x00, 0x01, 0x01,
0x02, 0x02, 0x03, 0x03,
0x04, 0x04, 0x05, 0x05,
0x06, 0x06, 0x07, 0x07 );
register vector unsigned char permute2;
register vector unsigned char aligner = vec_lvsl(0, in);
register vector float gainvec = (vector float) ( DIVBY127, DIVBY127,
DIVBY127, DIVBY127 );
register vector float zero = (vector float) vec_splat_u32(0);
register vector float mixvec1;
register vector float mixvec2;
register vector float mixvec3;
register vector float mixvec4;
register vector float mixvec5;
register vector float mixvec6;
register vector float mixvec7;
register vector float mixvec8;
register vector float v1;
register vector float v2;
register vector float v3;
register vector float v4;
register vector float v5;
register vector float v6;
register vector float v7;
register vector float v8;
register vector signed short conv_short;
register vector signed int conv_int;
register vector signed char ld;
register vector signed char ld2;
register vector signed char ld_overflow;
register int extra = (frames % 32);
union
{
vector float v;
float f[4];
} gaincvt;
permute2 = vec_add(permute1, vec_splat_u8(8));
gaincvt.f[0] = gaincvt.f[2] = gainLeft;
gaincvt.f[1] = gaincvt.f[3] = gainRight;
gainvec = vec_madd(gainvec, gaincvt.v, zero);
// Mix to the output buffer...
frames -= extra;
max = dst + (frames << 1);
ld = vec_ld(0, in);
while (dst < max)
{
// Get the sample points into a vector register...
ld_overflow = vec_ld(16, in); // read 16 8-bit samples.
in += 16;
mixvec1 = vec_ld(0x00, dst);
mixvec2 = vec_ld(0x10, dst);
mixvec3 = vec_ld(0x20, dst);
mixvec4 = vec_ld(0x30, dst);
mixvec5 = vec_ld(0x40, dst);
mixvec6 = vec_ld(0x50, dst);
mixvec7 = vec_ld(0x60, dst);
mixvec8 = vec_ld(0x70, dst);
ld = vec_perm(ld, ld_overflow, aligner);
// convert to stereo, so there's 8 stereo samples per vector.
// ld == 0l 0r 1l 1r 2l 2r 3l 3r 4l 4r 5l 5r 6l 6r 7l 7r
// ld2 == 8l 8r 9l 9r Al Ar Bl Br Cl Cr Dl Dr El Er Fl Fr
ld2 = vec_perm(ld, ld, permute2);
ld = vec_perm(ld, ld, permute1);
// convert to 16-bit ints, then 32-bit ints, then floats...
conv_short = vec_unpackh(ld);
conv_int = vec_unpackh(conv_short);
v1 = vec_ctf(conv_int, 0);
conv_int = vec_unpackl(conv_short);
v2 = vec_ctf(conv_int, 0);
conv_short = vec_unpackl(ld);
conv_int = vec_unpackh(conv_short);
v3 = vec_ctf(conv_int, 0);
conv_int = vec_unpackl(conv_short);
v4 = vec_ctf(conv_int, 0);
conv_short = vec_unpackh(ld2);
conv_int = vec_unpackh(conv_short);
v5 = vec_ctf(conv_int, 0);
conv_int = vec_unpackl(conv_short);
v6 = vec_ctf(conv_int, 0);
conv_short = vec_unpackl(ld2);
conv_int = vec_unpackh(conv_short);
v7 = vec_ctf(conv_int, 0);
conv_int = vec_unpackl(conv_short);
v8 = vec_ctf(conv_int, 0);
// change to float scale, apply gain, and mix with dst...
v1 = vec_madd(v1, gainvec, mixvec1);
v2 = vec_madd(v2, gainvec, mixvec2);
v3 = vec_madd(v3, gainvec, mixvec3);
v4 = vec_madd(v4, gainvec, mixvec4);
v5 = vec_madd(v5, gainvec, mixvec5);
v6 = vec_madd(v6, gainvec, mixvec6);
v7 = vec_madd(v7, gainvec, mixvec7);
v8 = vec_madd(v8, gainvec, mixvec8);
// store converted version back to RAM...
vec_st(v1, 0x00, dst);
vec_st(v2, 0x10, dst);
vec_st(v3, 0x20, dst);
vec_st(v4, 0x30, dst);
vec_st(v5, 0x40, dst);
vec_st(v6, 0x50, dst);
vec_st(v7, 0x60, dst);
vec_st(v8, 0x70, dst);
dst += 32;
ld = ld_overflow;
} // while
// Handle overflow as scalar data.
if (extra)
{
gainLeft *= DIVBY127;
gainRight *= DIVBY127;
max += extra;
frames += extra;
while (dst < max)
{
sample = ((Float32) *in);
in++;
dst[0] += sample * gainLeft;
dst[1] += sample * gainRight;
dst += 2;
} // while
} // if
} // if
src->bufferReadIndex += frames;
#endif // __POWERPC__
return(overflow);
} // __alMixMono8_altivec
UInt32 __alMixMono16_altivec(ALcontext *ctx, ALbuffer *buf, ALsource *src, Float32 *_dst, UInt32 frames)
{
register ALsizei overflow = 0;
#if __POWERPC__
//register UInt32 ctrl_in = DST_BLOCK_CTRL(1, 2, 16);
//register UInt32 ctrl_out = DST_BLOCK_CTRL(1, 8, 16);
register UInt32 buflen = (buf->allocatedSpace >> 1) - src->bufferReadIndex;
register SInt16 *in = ((SInt16 *) buf->mixData) + src->bufferReadIndex;
register Float32 *dst = _dst;
register Float32 *max;
register Float32 gainLeft;
register Float32 gainRight;
register Float32 sample;
register size_t dsti = ((size_t) dst) & 0x0000000F;
assert(ctx->device->streamFormat.mChannelsPerFrame == 2);
__alRecalcMonoSource(ctx, src);
gainLeft = src->channelGain0;
gainRight = src->channelGain1;
if (dsti) // Destination isn't 16-byte aligned?
{
if ((dsti % 2) && (frames > dsti)) // bump dst to 16-byte alignment.
{
register Float32 convGainLeft = gainLeft * DIVBY32767;
register Float32 convGainRight = gainRight * DIVBY32767;
while (UNALIGNED_PTR(dst))
{
sample = ((Float32) *in);
in++;
dst[0] += sample * convGainLeft;
dst[1] += sample * convGainRight;
dst += 2;
frames--;
} // while
} // if
else // dst will never align in this run; use scalar version. :(
{
return(__alMixMono16(ctx, buf, src, dst, frames));
} // else
} // if
overflow = frames - buflen;
if (overflow < 0)
overflow = 0;
else
frames = buflen;
// Too quiet to be worth playing? Just skip ahead in the buffer appropriately.
if ((gainLeft > MINVOL) || (gainRight > MINVOL))
{
// Permute vectors to convert mono channel to stereo.
register vector unsigned char permute1 = (vector unsigned char)
( 0x00, 0x01, 0x00, 0x01,
0x02, 0x03, 0x02, 0x03,
0x04, 0x05, 0x04, 0x05,
0x06, 0x07, 0x06, 0x07 );
register vector unsigned char permute2;
register vector unsigned char aligner = vec_lvsl(0, in);
register vector float divby = (vector float) ( DIVBY32767, DIVBY32767,
DIVBY32767, DIVBY32767 );
register vector float zero = (vector float) vec_splat_u32(0);
register vector float gainvec;
register vector signed int conv1;
register vector signed int conv2;
register vector signed int conv3;
register vector signed int conv4;
register vector float mixvec1;
register vector float mixvec2;
register vector float mixvec3;
register vector float mixvec4;
register vector float v1;
register vector float v2;
register vector float v3;
register vector float v4;
register vector signed short ld;
register vector signed short ld2;
register vector signed short ld_overflow;
register int extra = (frames % 16);
union
{
vector float v;
float f[4];
} gaincvt;
permute2 = vec_add(permute1, vec_splat_u8(8));
gaincvt.f[0] = gaincvt.f[2] = gainLeft;
gaincvt.f[1] = gaincvt.f[3] = gainRight;
gainvec = vec_madd(divby, gaincvt.v, zero);
// Mix to the output buffer...
frames -= extra;
max = dst + (frames << 1);
ld = vec_ld(0, in); // read eight 16-bit samples.
while (dst < max)
{
//vec_dstt(in, ctrl_in, 0);
//vec_dstst(dst, ctrl_out, 1);
// Get the sample points into a vector register...
ld_overflow = vec_ld(16, in); // read eight 16-bit samples.
mixvec1 = vec_ld(0x00, dst);
mixvec2 = vec_ld(0x10, dst);
mixvec3 = vec_ld(0x20, dst);
mixvec4 = vec_ld(0x30, dst);
in += 8;
ld = vec_perm(ld, ld_overflow, aligner);
// convert to stereo, so there's 4 stereo samples per vector.
// ld == 0l 0r 1l 1r 2l 2r 3l 3r
// ld2 == 4l 4r 5l 5r 6l 6r 7l 6r
ld2 = vec_perm(ld, ld, permute2);
ld = vec_perm(ld, ld, permute1);
// convert to 32-bit integers for vec_ctf...
conv1 = vec_unpackh(ld);
conv2 = vec_unpackl(ld);
conv3 = vec_unpackh(ld2);
conv4 = vec_unpackl(ld2);
// convert from int to float...
v1 = vec_ctf(conv1, 0);
v2 = vec_ctf(conv2, 0);