diff options
author | Chris Robinson <[email protected]> | 2010-08-03 23:19:36 -0700 |
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committer | Chris Robinson <[email protected]> | 2010-08-03 23:19:36 -0700 |
commit | 0dc3f1984e969a63f20a91dd980d0fc56fdf0b37 (patch) | |
tree | 51f6af0f69f0a0ecbd1332a77317cb6634b409da /Alc/mixer.c | |
parent | e74976e6451670c433ee09695125801c12e74e22 (diff) |
Move the core mixer functions to a separate source file
Diffstat (limited to 'Alc/mixer.c')
-rw-r--r-- | Alc/mixer.c | 791 |
1 files changed, 791 insertions, 0 deletions
diff --git a/Alc/mixer.c b/Alc/mixer.c new file mode 100644 index 00000000..e880ad3a --- /dev/null +++ b/Alc/mixer.c @@ -0,0 +1,791 @@ +/** + * OpenAL cross platform audio library + * Copyright (C) 1999-2007 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 "config.h" + +#include <math.h> +#include <stdlib.h> +#include <string.h> +#include <ctype.h> +#include <assert.h> + +#include "alMain.h" +#include "AL/al.h" +#include "AL/alc.h" +#include "alSource.h" +#include "alBuffer.h" +#include "alListener.h" +#include "alAuxEffectSlot.h" +#include "alu.h" +#include "bs2b.h" + +#define FRACTIONBITS 14 +#define FRACTIONMASK ((1L<<FRACTIONBITS)-1) +#define MAX_PITCH 65536 + +/* Minimum ramp length in milliseconds. The value below was chosen to + * adequately reduce clicks and pops from harsh gain changes. */ +#define MIN_RAMP_LENGTH 16 + + +static __inline ALfloat aluF2F(ALfloat Value) +{ + return Value; +} + +static __inline ALshort aluF2S(ALfloat Value) +{ + ALint i; + + if(Value < 0.0f) + { + i = (ALint)(Value*32768.0f); + i = max(-32768, i); + } + else + { + i = (ALint)(Value*32767.0f); + i = min( 32767, i); + } + return ((ALshort)i); +} + +static __inline ALubyte aluF2UB(ALfloat Value) +{ + ALshort i = aluF2S(Value); + return (i>>8)+128; +} + + +static __inline ALfloat point(ALfloat val1, ALfloat val2, ALint frac) +{ + return val1; + (void)val2; + (void)frac; +} +static __inline ALfloat lerp(ALfloat val1, ALfloat val2, ALint frac) +{ + return val1 + ((val2-val1)*(frac * (1.0f/(1<<FRACTIONBITS)))); +} +static __inline ALfloat cos_lerp(ALfloat val1, ALfloat val2, ALint frac) +{ + ALfloat mult = (1.0f-cos(frac * (1.0f/(1<<FRACTIONBITS)) * M_PI)) * 0.5f; + return val1 + ((val2-val1)*mult); +} + + +static void MixSource(ALsource *ALSource, ALCcontext *ALContext, + float (*DryBuffer)[OUTPUTCHANNELS], ALuint SamplesToDo) +{ + static float DummyBuffer[BUFFERSIZE]; + ALfloat *WetBuffer[MAX_SENDS]; + ALfloat DrySend[OUTPUTCHANNELS]; + ALfloat dryGainStep[OUTPUTCHANNELS]; + ALfloat wetGainStep[MAX_SENDS]; + ALuint i, j, out; + ALfloat value, outsamp; + ALbufferlistitem *BufferListItem; + ALint64 DataSize64,DataPos64; + FILTER *DryFilter, *WetFilter[MAX_SENDS]; + ALfloat WetSend[MAX_SENDS]; + ALuint rampLength; + ALboolean DuplicateStereo; + ALuint DeviceFreq; + ALint increment; + ALuint DataPosInt, DataPosFrac; + ALuint Channels, Bytes; + ALuint Frequency; + resampler_t Resampler; + ALuint BuffersPlayed; + ALboolean Looping; + ALfloat Pitch; + ALenum State; + + DuplicateStereo = ALContext->Device->DuplicateStereo; + DeviceFreq = ALContext->Device->Frequency; + + rampLength = DeviceFreq * MIN_RAMP_LENGTH / 1000; + rampLength = max(rampLength, SamplesToDo); + + /* Find buffer format */ + Frequency = 0; + Channels = 0; + Bytes = 0; + BufferListItem = ALSource->queue; + while(BufferListItem != NULL) + { + ALbuffer *ALBuffer; + if((ALBuffer=BufferListItem->buffer) != NULL) + { + Channels = aluChannelsFromFormat(ALBuffer->format); + Bytes = aluBytesFromFormat(ALBuffer->format); + Frequency = ALBuffer->frequency; + break; + } + BufferListItem = BufferListItem->next; + } + + if(ALSource->NeedsUpdate) + { + ALsource_Update(ALSource, ALContext); + ALSource->NeedsUpdate = AL_FALSE; + } + + /* Get source info */ + Resampler = ALSource->Resampler; + State = ALSource->state; + BuffersPlayed = ALSource->BuffersPlayed; + DataPosInt = ALSource->position; + DataPosFrac = ALSource->position_fraction; + Looping = ALSource->bLooping; + + /* Compute 18.14 fixed point step */ + Pitch = (ALSource->Params.Pitch*Frequency) / DeviceFreq; + if(Pitch > (float)MAX_PITCH) Pitch = (float)MAX_PITCH; + increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS)); + if(increment <= 0) increment = (1<<FRACTIONBITS); + + if(ALSource->FirstStart) + { + for(i = 0;i < OUTPUTCHANNELS;i++) + DrySend[i] = ALSource->Params.DryGains[i]; + for(i = 0;i < MAX_SENDS;i++) + WetSend[i] = ALSource->Params.WetGains[i]; + } + else + { + for(i = 0;i < OUTPUTCHANNELS;i++) + DrySend[i] = ALSource->DryGains[i]; + for(i = 0;i < MAX_SENDS;i++) + WetSend[i] = ALSource->WetGains[i]; + } + + DryFilter = &ALSource->Params.iirFilter; + for(i = 0;i < MAX_SENDS;i++) + { + WetFilter[i] = &ALSource->Params.Send[i].iirFilter; + WetBuffer[i] = (ALSource->Send[i].Slot ? + ALSource->Send[i].Slot->WetBuffer : + DummyBuffer); + } + + /* Get current buffer queue item */ + BufferListItem = ALSource->queue; + for(i = 0;i < BuffersPlayed && BufferListItem;i++) + BufferListItem = BufferListItem->next; + + j = 0; + do { + ALfloat *Data = NULL; + ALuint LoopStart = 0; + ALuint LoopEnd = 0; + ALuint DataSize = 0; + ALbuffer *ALBuffer; + ALuint BufferSize; + + /* Get buffer info */ + if((ALBuffer=BufferListItem->buffer) != NULL) + { + Data = ALBuffer->data; + DataSize = ALBuffer->size; + DataSize /= Channels * Bytes; + LoopStart = ALBuffer->LoopStart; + LoopEnd = ALBuffer->LoopEnd; + } + + if(Looping && ALSource->lSourceType == AL_STATIC) + { + /* If current offset is beyond the loop range, do not loop */ + if(DataPosInt >= LoopEnd) + Looping = AL_FALSE; + } + if(!Looping || ALSource->lSourceType != AL_STATIC) + { + /* Non-looping and non-static sources ignore loop points */ + LoopStart = 0; + LoopEnd = DataSize; + } + + if(DataPosInt >= DataSize) + goto skipmix; + + if(BufferListItem->next) + { + ALbuffer *NextBuf = BufferListItem->next->buffer; + if(NextBuf && NextBuf->size) + { + ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes; + ulExtraSamples = min(NextBuf->size, ulExtraSamples); + memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples); + } + } + else if(Looping) + { + ALbuffer *NextBuf = ALSource->queue->buffer; + if(NextBuf && NextBuf->size) + { + ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes; + ulExtraSamples = min(NextBuf->size, ulExtraSamples); + memcpy(&Data[DataSize*Channels], &NextBuf->data[LoopStart*Channels], ulExtraSamples); + } + } + else + memset(&Data[DataSize*Channels], 0, (BUFFER_PADDING*Channels*Bytes)); + + /* Compute the gain steps for each output channel */ + for(i = 0;i < OUTPUTCHANNELS;i++) + dryGainStep[i] = (ALSource->Params.DryGains[i]-DrySend[i]) / + rampLength; + for(i = 0;i < MAX_SENDS;i++) + wetGainStep[i] = (ALSource->Params.WetGains[i]-WetSend[i]) / + rampLength; + + /* Figure out how many samples we can mix. */ + DataSize64 = LoopEnd; + DataSize64 <<= FRACTIONBITS; + DataPos64 = DataPosInt; + DataPos64 <<= FRACTIONBITS; + DataPos64 += DataPosFrac; + BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment); + + BufferSize = min(BufferSize, (SamplesToDo-j)); + + /* Actual sample mixing loops */ + if(Channels == 1) /* Mono */ + { +#define DO_MIX(resampler) do { \ + while(BufferSize--) \ + { \ + for(i = 0;i < OUTPUTCHANNELS;i++) \ + DrySend[i] += dryGainStep[i]; \ + for(i = 0;i < MAX_SENDS;i++) \ + WetSend[i] += wetGainStep[i]; \ + \ + /* First order interpolator */ \ + value = (resampler)(Data[DataPosInt], Data[DataPosInt+1], \ + DataPosFrac); \ + \ + /* Direct path final mix buffer and panning */ \ + outsamp = lpFilter4P(DryFilter, 0, value); \ + DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT]; \ + DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT]; \ + DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT]; \ + DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT]; \ + DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT]; \ + DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT]; \ + DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER]; \ + DryBuffer[j][BACK_CENTER] += outsamp*DrySend[BACK_CENTER]; \ + \ + /* Room path final mix buffer and panning */ \ + for(i = 0;i < MAX_SENDS;i++) \ + { \ + outsamp = lpFilter2P(WetFilter[i], 0, value); \ + WetBuffer[i][j] += outsamp*WetSend[i]; \ + } \ + \ + DataPosFrac += increment; \ + DataPosInt += DataPosFrac>>FRACTIONBITS; \ + DataPosFrac &= FRACTIONMASK; \ + j++; \ + } \ +} while(0) + + switch(Resampler) + { + case POINT_RESAMPLER: + DO_MIX(point); break; + case LINEAR_RESAMPLER: + DO_MIX(lerp); break; + case COSINE_RESAMPLER: + DO_MIX(cos_lerp); break; + case RESAMPLER_MIN: + case RESAMPLER_MAX: + break; + } +#undef DO_MIX + } + else if(Channels == 2 && DuplicateStereo) /* Stereo */ + { + const int chans[] = { + FRONT_LEFT, FRONT_RIGHT + }; + const int chans2[] = { + BACK_LEFT, SIDE_LEFT, BACK_RIGHT, SIDE_RIGHT + }; + const ALfloat dupscaler = aluSqrt(1.0f/3.0f); + +#define DO_MIX(resampler) do { \ + const ALfloat scaler = 1.0f/Channels; \ + while(BufferSize--) \ + { \ + for(i = 0;i < OUTPUTCHANNELS;i++) \ + DrySend[i] += dryGainStep[i]; \ + for(i = 0;i < MAX_SENDS;i++) \ + WetSend[i] += wetGainStep[i]; \ + \ + for(i = 0;i < Channels;i++) \ + { \ + value = (resampler)(Data[DataPosInt*Channels + i], \ + Data[(DataPosInt+1)*Channels + i], \ + DataPosFrac); \ + \ + outsamp = lpFilter2P(DryFilter, chans[i]*2, value) * dupscaler; \ + DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]]; \ + DryBuffer[j][chans2[i*2+0]] += outsamp*DrySend[chans2[i*2+0]]; \ + DryBuffer[j][chans2[i*2+1]] += outsamp*DrySend[chans2[i*2+1]]; \ + \ + for(out = 0;out < MAX_SENDS;out++) \ + { \ + outsamp = lpFilter1P(WetFilter[out], chans[i], value); \ + WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \ + } \ + } \ + \ + DataPosFrac += increment; \ + DataPosInt += DataPosFrac>>FRACTIONBITS; \ + DataPosFrac &= FRACTIONMASK; \ + j++; \ + } \ +} while(0) + + switch(Resampler) + { + case POINT_RESAMPLER: + DO_MIX(point); break; + case LINEAR_RESAMPLER: + DO_MIX(lerp); break; + case COSINE_RESAMPLER: + DO_MIX(cos_lerp); break; + case RESAMPLER_MIN: + case RESAMPLER_MAX: + break; + } +#undef DO_MIX + } + else if(Channels == 2) /* Stereo */ + { + const int chans[] = { + FRONT_LEFT, FRONT_RIGHT + }; + +#define DO_MIX(resampler) do { \ + const ALfloat scaler = 1.0f/Channels; \ + while(BufferSize--) \ + { \ + for(i = 0;i < OUTPUTCHANNELS;i++) \ + DrySend[i] += dryGainStep[i]; \ + for(i = 0;i < MAX_SENDS;i++) \ + WetSend[i] += wetGainStep[i]; \ + \ + for(i = 0;i < Channels;i++) \ + { \ + value = (resampler)(Data[DataPosInt*Channels + i], \ + Data[(DataPosInt+1)*Channels + i], \ + DataPosFrac); \ + \ + outsamp = lpFilter2P(DryFilter, chans[i]*2, value); \ + DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]]; \ + \ + for(out = 0;out < MAX_SENDS;out++) \ + { \ + outsamp = lpFilter1P(WetFilter[out], chans[i], value); \ + WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \ + } \ + } \ + \ + DataPosFrac += increment; \ + DataPosInt += DataPosFrac>>FRACTIONBITS; \ + DataPosFrac &= FRACTIONMASK; \ + j++; \ + } \ +} while(0) + + switch(Resampler) + { + case POINT_RESAMPLER: + DO_MIX(point); break; + case LINEAR_RESAMPLER: + DO_MIX(lerp); break; + case COSINE_RESAMPLER: + DO_MIX(cos_lerp); break; + case RESAMPLER_MIN: + case RESAMPLER_MAX: + break; + } + } + else if(Channels == 4) /* Quad */ + { + const int chans[] = { + FRONT_LEFT, FRONT_RIGHT, + BACK_LEFT, BACK_RIGHT + }; + + switch(Resampler) + { + case POINT_RESAMPLER: + DO_MIX(point); break; + case LINEAR_RESAMPLER: + DO_MIX(lerp); break; + case COSINE_RESAMPLER: + DO_MIX(cos_lerp); break; + case RESAMPLER_MIN: + case RESAMPLER_MAX: + break; + } + } + else if(Channels == 6) /* 5.1 */ + { + const int chans[] = { + FRONT_LEFT, FRONT_RIGHT, + FRONT_CENTER, LFE, + BACK_LEFT, BACK_RIGHT + }; + + switch(Resampler) + { + case POINT_RESAMPLER: + DO_MIX(point); break; + case LINEAR_RESAMPLER: + DO_MIX(lerp); break; + case COSINE_RESAMPLER: + DO_MIX(cos_lerp); break; + case RESAMPLER_MIN: + case RESAMPLER_MAX: + break; + } + } + else if(Channels == 7) /* 6.1 */ + { + const int chans[] = { + FRONT_LEFT, FRONT_RIGHT, + FRONT_CENTER, LFE, + BACK_CENTER, + SIDE_LEFT, SIDE_RIGHT + }; + + switch(Resampler) + { + case POINT_RESAMPLER: + DO_MIX(point); break; + case LINEAR_RESAMPLER: + DO_MIX(lerp); break; + case COSINE_RESAMPLER: + DO_MIX(cos_lerp); break; + case RESAMPLER_MIN: + case RESAMPLER_MAX: + break; + } + } + else if(Channels == 8) /* 7.1 */ + { + const int chans[] = { + FRONT_LEFT, FRONT_RIGHT, + FRONT_CENTER, LFE, + BACK_LEFT, BACK_RIGHT, + SIDE_LEFT, SIDE_RIGHT + }; + + switch(Resampler) + { + case POINT_RESAMPLER: + DO_MIX(point); break; + case LINEAR_RESAMPLER: + DO_MIX(lerp); break; + case COSINE_RESAMPLER: + DO_MIX(cos_lerp); break; + case RESAMPLER_MIN: + case RESAMPLER_MAX: + break; + } +#undef DO_MIX + } + else /* Unknown? */ + { + for(i = 0;i < OUTPUTCHANNELS;i++) + DrySend[i] += dryGainStep[i]*BufferSize; + for(i = 0;i < MAX_SENDS;i++) + WetSend[i] += wetGainStep[i]*BufferSize; + while(BufferSize--) + { + DataPosFrac += increment; + DataPosInt += DataPosFrac>>FRACTIONBITS; + DataPosFrac &= FRACTIONMASK; + j++; + } + } + + skipmix: + /* Handle looping sources */ + if(DataPosInt >= LoopEnd) + { + if(BuffersPlayed < (ALSource->BuffersInQueue-1)) + { + BufferListItem = BufferListItem->next; + BuffersPlayed++; + DataPosInt -= DataSize; + } + else if(Looping) + { + BufferListItem = ALSource->queue; + BuffersPlayed = 0; + if(ALSource->lSourceType == AL_STATIC) + DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart; + else + DataPosInt -= DataSize; + } + else + { + State = AL_STOPPED; + BufferListItem = ALSource->queue; + BuffersPlayed = ALSource->BuffersInQueue; + DataPosInt = 0; + DataPosFrac = 0; + } + } + } while(State == AL_PLAYING && j < SamplesToDo); + + /* Update source info */ + ALSource->state = State; + ALSource->BuffersPlayed = BuffersPlayed; + ALSource->position = DataPosInt; + ALSource->position_fraction = DataPosFrac; + ALSource->Buffer = BufferListItem->buffer; + + for(i = 0;i < OUTPUTCHANNELS;i++) + ALSource->DryGains[i] = DrySend[i]; + for(i = 0;i < MAX_SENDS;i++) + ALSource->WetGains[i] = WetSend[i]; + + ALSource->FirstStart = AL_FALSE; +} + +ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size) +{ + float (*DryBuffer)[OUTPUTCHANNELS]; + ALfloat (*Matrix)[OUTPUTCHANNELS]; + const ALuint *ChanMap; + ALuint SamplesToDo; + ALeffectslot *ALEffectSlot; + ALCcontext *ALContext; + ALfloat samp; + int fpuState; + ALuint i, j, c; + ALsizei e, s; + +#if defined(HAVE_FESETROUND) + fpuState = fegetround(); + fesetround(FE_TOWARDZERO); +#elif defined(HAVE__CONTROLFP) + fpuState = _controlfp(0, 0); + _controlfp(_RC_CHOP, _MCW_RC); +#else + (void)fpuState; +#endif + + DryBuffer = device->DryBuffer; + while(size > 0) + { + /* Setup variables */ + SamplesToDo = min(size, BUFFERSIZE); + + /* Clear mixing buffer */ + memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat)); + + SuspendContext(NULL); + for(c = 0;c < device->NumContexts;c++) + { + ALContext = device->Contexts[c]; + SuspendContext(ALContext); + + s = 0; + while(s < ALContext->ActiveSourceCount) + { + ALsource *Source = ALContext->ActiveSources[s]; + if(Source->state != AL_PLAYING) + { + ALsizei end = --(ALContext->ActiveSourceCount); + ALContext->ActiveSources[s] = ALContext->ActiveSources[end]; + continue; + } + MixSource(Source, ALContext, DryBuffer, SamplesToDo); + s++; + } + + /* effect slot processing */ + for(e = 0;e < ALContext->EffectSlotMap.size;e++) + { + ALEffectSlot = ALContext->EffectSlotMap.array[e].value; + if(ALEffectSlot->EffectState) + ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot, SamplesToDo, ALEffectSlot->WetBuffer, DryBuffer); + + for(i = 0;i < SamplesToDo;i++) + ALEffectSlot->WetBuffer[i] = 0.0f; + } + ProcessContext(ALContext); + } + device->SamplesPlayed += SamplesToDo; + ProcessContext(NULL); + + //Post processing loop + ChanMap = device->DevChannels; + Matrix = device->ChannelMatrix; + switch(device->Format) + { +#define CHECK_WRITE_FORMAT(bits, type, func) \ + case AL_FORMAT_MONO##bits: \ + for(i = 0;i < SamplesToDo;i++) \ + { \ + samp = 0.0f; \ + for(c = 0;c < OUTPUTCHANNELS;c++) \ + samp += DryBuffer[i][c] * Matrix[c][FRONT_CENTER]; \ + ((type*)buffer)[ChanMap[FRONT_CENTER]] = (func)(samp); \ + buffer = ((type*)buffer) + 1; \ + } \ + break; \ + case AL_FORMAT_STEREO##bits: \ + if(device->Bs2b) \ + { \ + for(i = 0;i < SamplesToDo;i++) \ + { \ + float samples[2] = { 0.0f, 0.0f }; \ + for(c = 0;c < OUTPUTCHANNELS;c++) \ + { \ + samples[0] += DryBuffer[i][c]*Matrix[c][FRONT_LEFT]; \ + samples[1] += DryBuffer[i][c]*Matrix[c][FRONT_RIGHT]; \ + } \ + bs2b_cross_feed(device->Bs2b, samples); \ + ((type*)buffer)[ChanMap[FRONT_LEFT]] = (func)(samples[0]);\ + ((type*)buffer)[ChanMap[FRONT_RIGHT]]= (func)(samples[1]);\ + buffer = ((type*)buffer) + 2; \ + } \ + } \ + else \ + { \ + for(i = 0;i < SamplesToDo;i++) \ + { \ + static const Channel chans[] = { \ + FRONT_LEFT, FRONT_RIGHT \ + }; \ + for(j = 0;j < 2;j++) \ + { \ + samp = 0.0f; \ + for(c = 0;c < OUTPUTCHANNELS;c++) \ + samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \ + ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \ + } \ + buffer = ((type*)buffer) + 2; \ + } \ + } \ + break; \ + case AL_FORMAT_QUAD##bits: \ + for(i = 0;i < SamplesToDo;i++) \ + { \ + static const Channel chans[] = { \ + FRONT_LEFT, FRONT_RIGHT, \ + BACK_LEFT, BACK_RIGHT, \ + }; \ + for(j = 0;j < 4;j++) \ + { \ + samp = 0.0f; \ + for(c = 0;c < OUTPUTCHANNELS;c++) \ + samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \ + ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \ + } \ + buffer = ((type*)buffer) + 4; \ + } \ + break; \ + case AL_FORMAT_51CHN##bits: \ + for(i = 0;i < SamplesToDo;i++) \ + { \ + static const Channel chans[] = { \ + FRONT_LEFT, FRONT_RIGHT, \ + FRONT_CENTER, LFE, \ + BACK_LEFT, BACK_RIGHT, \ + }; \ + for(j = 0;j < 6;j++) \ + { \ + samp = 0.0f; \ + for(c = 0;c < OUTPUTCHANNELS;c++) \ + samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \ + ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \ + } \ + buffer = ((type*)buffer) + 6; \ + } \ + break; \ + case AL_FORMAT_61CHN##bits: \ + for(i = 0;i < SamplesToDo;i++) \ + { \ + static const Channel chans[] = { \ + FRONT_LEFT, FRONT_RIGHT, \ + FRONT_CENTER, LFE, BACK_CENTER, \ + SIDE_LEFT, SIDE_RIGHT, \ + }; \ + for(j = 0;j < 7;j++) \ + { \ + samp = 0.0f; \ + for(c = 0;c < OUTPUTCHANNELS;c++) \ + samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \ + ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \ + } \ + buffer = ((type*)buffer) + 7; \ + } \ + break; \ + case AL_FORMAT_71CHN##bits: \ + for(i = 0;i < SamplesToDo;i++) \ + { \ + static const Channel chans[] = { \ + FRONT_LEFT, FRONT_RIGHT, \ + FRONT_CENTER, LFE, \ + BACK_LEFT, BACK_RIGHT, \ + SIDE_LEFT, SIDE_RIGHT \ + }; \ + for(j = 0;j < 8;j++) \ + { \ + samp = 0.0f; \ + for(c = 0;c < OUTPUTCHANNELS;c++) \ + samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \ + ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \ + } \ + buffer = ((type*)buffer) + 8; \ + } \ + break; + +#define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32 +#define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32 + CHECK_WRITE_FORMAT(8, ALubyte, aluF2UB) + CHECK_WRITE_FORMAT(16, ALshort, aluF2S) + CHECK_WRITE_FORMAT(32, ALfloat, aluF2F) +#undef AL_FORMAT_STEREO32 +#undef AL_FORMAT_MONO32 +#undef CHECK_WRITE_FORMAT + + default: + break; + } + + size -= SamplesToDo; + } + +#if defined(HAVE_FESETROUND) + fesetround(fpuState); +#elif defined(HAVE__CONTROLFP) + _controlfp(fpuState, 0xfffff); +#endif +} |