/** * 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 #include #include #include #include #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<>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<Device->Frequency; /* 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<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]; } /* Compute the gain steps for each output channel */ rampLength = DeviceFreq * MIN_RAMP_LENGTH / 1000; rampLength = max(rampLength, SamplesToDo); 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; 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;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)); /* 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) /* Stereo */ { const int chans[] = { FRONT_LEFT, FRONT_RIGHT }; const int chans2[] = { BACK_LEFT, SIDE_LEFT, BACK_RIGHT, SIDE_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]]; \ 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 == 4) /* Quad */ { const int chans[] = { FRONT_LEFT, FRONT_RIGHT, BACK_LEFT, BACK_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 == 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; 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 }