From 01beb9919b95479d8be040bec74abc5cc67a5e43 Mon Sep 17 00:00:00 2001 From: Mikko Tiusanen Date: Sun, 4 May 2014 01:18:52 +0300 Subject: Initial import. --- src/client/snd_mix.c | 742 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 742 insertions(+) create mode 100644 src/client/snd_mix.c (limited to 'src/client/snd_mix.c') diff --git a/src/client/snd_mix.c b/src/client/snd_mix.c new file mode 100644 index 0000000..2dd4428 --- /dev/null +++ b/src/client/snd_mix.c @@ -0,0 +1,742 @@ +/* +=========================================================================== +Copyright (C) 1999-2005 Id Software, Inc. +Copyright (C) 2000-2009 Darklegion Development + +This file is part of Tremulous. + +Tremulous is free software; you can redistribute it +and/or modify it under the terms of the GNU General Public License as +published by the Free Software Foundation; either version 2 of the License, +or (at your option) any later version. + +Tremulous 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 General Public License for more details. + +You should have received a copy of the GNU General Public License +along with Tremulous; if not, write to the Free Software +Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA +=========================================================================== +*/ +// snd_mix.c -- portable code to mix sounds for snd_dma.c + +#include "client.h" +#include "snd_local.h" +#if idppc_altivec && !defined(MACOS_X) +#include +#endif + +static portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE]; +static int snd_vol; + +int* snd_p; +int snd_linear_count; +short* snd_out; + +#if !id386 // if configured not to use asm + +void S_WriteLinearBlastStereo16 (void) +{ + int i; + int val; + + for (i=0 ; i>8; + if (val > 0x7fff) + snd_out[i] = 0x7fff; + else if (val < -32768) + snd_out[i] = -32768; + else + snd_out[i] = val; + + val = snd_p[i+1]>>8; + if (val > 0x7fff) + snd_out[i+1] = 0x7fff; + else if (val < -32768) + snd_out[i+1] = -32768; + else + snd_out[i+1] = val; + } +} +#elif defined(__GNUC__) +// uses snd_mixa.s +void S_WriteLinearBlastStereo16 (void); +#else + +__declspec( naked ) void S_WriteLinearBlastStereo16 (void) +{ + __asm { + + push edi + push ebx + mov ecx,ds:dword ptr[snd_linear_count] + mov ebx,ds:dword ptr[snd_p] + mov edi,ds:dword ptr[snd_out] +LWLBLoopTop: + mov eax,ds:dword ptr[-8+ebx+ecx*4] + sar eax,8 + cmp eax,07FFFh + jg LClampHigh + cmp eax,0FFFF8000h + jnl LClampDone + mov eax,0FFFF8000h + jmp LClampDone +LClampHigh: + mov eax,07FFFh +LClampDone: + mov edx,ds:dword ptr[-4+ebx+ecx*4] + sar edx,8 + cmp edx,07FFFh + jg LClampHigh2 + cmp edx,0FFFF8000h + jnl LClampDone2 + mov edx,0FFFF8000h + jmp LClampDone2 +LClampHigh2: + mov edx,07FFFh +LClampDone2: + shl edx,16 + and eax,0FFFFh + or edx,eax + mov ds:dword ptr[-4+edi+ecx*2],edx + sub ecx,2 + jnz LWLBLoopTop + pop ebx + pop edi + ret + } +} + +#endif + +void S_TransferStereo16 (unsigned long *pbuf, int endtime) +{ + int lpos; + int ls_paintedtime; + + snd_p = (int *) paintbuffer; + ls_paintedtime = s_paintedtime; + + while (ls_paintedtime < endtime) + { + // handle recirculating buffer issues + lpos = ls_paintedtime & ((dma.samples>>1)-1); + + snd_out = (short *) pbuf + (lpos<<1); + + snd_linear_count = (dma.samples>>1) - lpos; + if (ls_paintedtime + snd_linear_count > endtime) + snd_linear_count = endtime - ls_paintedtime; + + snd_linear_count <<= 1; + + // write a linear blast of samples + S_WriteLinearBlastStereo16 (); + + snd_p += snd_linear_count; + ls_paintedtime += (snd_linear_count>>1); + + if( CL_VideoRecording( ) ) + CL_WriteAVIAudioFrame( (byte *)snd_out, snd_linear_count << 1 ); + } +} + +/* +=================== +S_TransferPaintBuffer + +=================== +*/ +void S_TransferPaintBuffer(int endtime) +{ + int out_idx; + int count; + int out_mask; + int *p; + int step; + int val; + unsigned long *pbuf; + + pbuf = (unsigned long *)dma.buffer; + + + if ( s_testsound->integer ) { + int i; + int count; + + // write a fixed sine wave + count = (endtime - s_paintedtime); + for (i=0 ; i> 8; + p+= step; + if (val > 0x7fff) + val = 0x7fff; + else if (val < -32768) + val = -32768; + out[out_idx] = val; + out_idx = (out_idx + 1) & out_mask; + } + } + else if (dma.samplebits == 8) + { + unsigned char *out = (unsigned char *) pbuf; + while (count--) + { + val = *p >> 8; + p+= step; + if (val > 0x7fff) + val = 0x7fff; + else if (val < -32768) + val = -32768; + out[out_idx] = (val>>8) + 128; + out_idx = (out_idx + 1) & out_mask; + } + } + } +} + + +/* +=============================================================================== + +CHANNEL MIXING + +=============================================================================== +*/ + +#if idppc_altivec +static void S_PaintChannelFrom16_altivec( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) { + int data, aoff, boff; + int leftvol, rightvol; + int i, j; + portable_samplepair_t *samp; + sndBuffer *chunk; + short *samples; + float ooff, fdata, fdiv, fleftvol, frightvol; + + samp = &paintbuffer[ bufferOffset ]; + + if (ch->doppler) { + sampleOffset = sampleOffset*ch->oldDopplerScale; + } + + chunk = sc->soundData; + while (sampleOffset>=SND_CHUNK_SIZE) { + chunk = chunk->next; + sampleOffset -= SND_CHUNK_SIZE; + if (!chunk) { + chunk = sc->soundData; + } + } + + if (!ch->doppler || ch->dopplerScale==1.0f) { + vector signed short volume_vec; + vector unsigned int volume_shift; + int vectorCount, samplesLeft, chunkSamplesLeft; + leftvol = ch->leftvol*snd_vol; + rightvol = ch->rightvol*snd_vol; + samples = chunk->sndChunk; + ((short *)&volume_vec)[0] = leftvol; + ((short *)&volume_vec)[1] = leftvol; + ((short *)&volume_vec)[4] = leftvol; + ((short *)&volume_vec)[5] = leftvol; + ((short *)&volume_vec)[2] = rightvol; + ((short *)&volume_vec)[3] = rightvol; + ((short *)&volume_vec)[6] = rightvol; + ((short *)&volume_vec)[7] = rightvol; + volume_shift = vec_splat_u32(8); + i = 0; + + while(i < count) { + /* Try to align destination to 16-byte boundary */ + while(i < count && (((unsigned long)&samp[i] & 0x1f) || ((count-i) < 8) || ((SND_CHUNK_SIZE - sampleOffset) < 8))) { + data = samples[sampleOffset++]; + samp[i].left += (data * leftvol)>>8; + samp[i].right += (data * rightvol)>>8; + + if (sampleOffset == SND_CHUNK_SIZE) { + chunk = chunk->next; + samples = chunk->sndChunk; + sampleOffset = 0; + } + i++; + } + /* Destination is now aligned. Process as many 8-sample + chunks as we can before we run out of room from the current + sound chunk. We do 8 per loop to avoid extra source data reads. */ + samplesLeft = count - i; + chunkSamplesLeft = SND_CHUNK_SIZE - sampleOffset; + if(samplesLeft > chunkSamplesLeft) + samplesLeft = chunkSamplesLeft; + + vectorCount = samplesLeft / 8; + + if(vectorCount) + { + vector unsigned char tmp; + vector short s0, s1, sampleData0, sampleData1; + vector signed int merge0, merge1; + vector signed int d0, d1, d2, d3; + vector unsigned char samplePermute0 = + VECCONST_UINT8(0, 1, 4, 5, 0, 1, 4, 5, 2, 3, 6, 7, 2, 3, 6, 7); + vector unsigned char samplePermute1 = + VECCONST_UINT8(8, 9, 12, 13, 8, 9, 12, 13, 10, 11, 14, 15, 10, 11, 14, 15); + vector unsigned char loadPermute0, loadPermute1; + + // Rather than permute the vectors after we load them to do the sample + // replication and rearrangement, we permute the alignment vector so + // we do everything in one step below and avoid data shuffling. + tmp = vec_lvsl(0,&samples[sampleOffset]); + loadPermute0 = vec_perm(tmp,tmp,samplePermute0); + loadPermute1 = vec_perm(tmp,tmp,samplePermute1); + + s0 = *(vector short *)&samples[sampleOffset]; + while(vectorCount) + { + /* Load up source (16-bit) sample data */ + s1 = *(vector short *)&samples[sampleOffset+7]; + + /* Load up destination sample data */ + d0 = *(vector signed int *)&samp[i]; + d1 = *(vector signed int *)&samp[i+2]; + d2 = *(vector signed int *)&samp[i+4]; + d3 = *(vector signed int *)&samp[i+6]; + + sampleData0 = vec_perm(s0,s1,loadPermute0); + sampleData1 = vec_perm(s0,s1,loadPermute1); + + merge0 = vec_mule(sampleData0,volume_vec); + merge0 = vec_sra(merge0,volume_shift); /* Shift down to proper range */ + + merge1 = vec_mulo(sampleData0,volume_vec); + merge1 = vec_sra(merge1,volume_shift); + + d0 = vec_add(merge0,d0); + d1 = vec_add(merge1,d1); + + merge0 = vec_mule(sampleData1,volume_vec); + merge0 = vec_sra(merge0,volume_shift); /* Shift down to proper range */ + + merge1 = vec_mulo(sampleData1,volume_vec); + merge1 = vec_sra(merge1,volume_shift); + + d2 = vec_add(merge0,d2); + d3 = vec_add(merge1,d3); + + /* Store destination sample data */ + *(vector signed int *)&samp[i] = d0; + *(vector signed int *)&samp[i+2] = d1; + *(vector signed int *)&samp[i+4] = d2; + *(vector signed int *)&samp[i+6] = d3; + + i += 8; + vectorCount--; + s0 = s1; + sampleOffset += 8; + } + if (sampleOffset == SND_CHUNK_SIZE) { + chunk = chunk->next; + samples = chunk->sndChunk; + sampleOffset = 0; + } + } + } + } else { + fleftvol = ch->leftvol*snd_vol; + frightvol = ch->rightvol*snd_vol; + + ooff = sampleOffset; + samples = chunk->sndChunk; + + for ( i=0 ; idopplerScale; + boff = ooff; + fdata = 0; + for (j=aoff; jnext; + if (!chunk) { + chunk = sc->soundData; + } + samples = chunk->sndChunk; + ooff -= SND_CHUNK_SIZE; + } + fdata += samples[j&(SND_CHUNK_SIZE-1)]; + } + fdiv = 256 * (boff-aoff); + samp[i].left += (fdata * fleftvol)/fdiv; + samp[i].right += (fdata * frightvol)/fdiv; + } + } +} +#endif + +static void S_PaintChannelFrom16_scalar( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) { + int data, aoff, boff; + int leftvol, rightvol; + int i, j; + portable_samplepair_t *samp; + sndBuffer *chunk; + short *samples; + float ooff, fdata, fdiv, fleftvol, frightvol; + + samp = &paintbuffer[ bufferOffset ]; + + if (ch->doppler) { + sampleOffset = sampleOffset*ch->oldDopplerScale; + } + + chunk = sc->soundData; + while (sampleOffset>=SND_CHUNK_SIZE) { + chunk = chunk->next; + sampleOffset -= SND_CHUNK_SIZE; + if (!chunk) { + chunk = sc->soundData; + } + } + + if (!ch->doppler || ch->dopplerScale==1.0f) { + leftvol = ch->leftvol*snd_vol; + rightvol = ch->rightvol*snd_vol; + samples = chunk->sndChunk; + for ( i=0 ; i>8; + samp[i].right += (data * rightvol)>>8; + + if (sampleOffset == SND_CHUNK_SIZE) { + chunk = chunk->next; + samples = chunk->sndChunk; + sampleOffset = 0; + } + } + } else { + fleftvol = ch->leftvol*snd_vol; + frightvol = ch->rightvol*snd_vol; + + ooff = sampleOffset; + samples = chunk->sndChunk; + + + + + for ( i=0 ; idopplerScale; + boff = ooff; + fdata = 0; + for (j=aoff; jnext; + if (!chunk) { + chunk = sc->soundData; + } + samples = chunk->sndChunk; + ooff -= SND_CHUNK_SIZE; + } + fdata += samples[j&(SND_CHUNK_SIZE-1)]; + } + fdiv = 256 * (boff-aoff); + samp[i].left += (fdata * fleftvol)/fdiv; + samp[i].right += (fdata * frightvol)/fdiv; + } + } +} + +static void S_PaintChannelFrom16( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) { +#if idppc_altivec + if (com_altivec->integer) { + // must be in a seperate function or G3 systems will crash. + S_PaintChannelFrom16_altivec( ch, sc, count, sampleOffset, bufferOffset ); + return; + } +#endif + S_PaintChannelFrom16_scalar( ch, sc, count, sampleOffset, bufferOffset ); +} + +void S_PaintChannelFromWavelet( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) { + int data; + int leftvol, rightvol; + int i; + portable_samplepair_t *samp; + sndBuffer *chunk; + short *samples; + + leftvol = ch->leftvol*snd_vol; + rightvol = ch->rightvol*snd_vol; + + i = 0; + samp = &paintbuffer[ bufferOffset ]; + chunk = sc->soundData; + while (sampleOffset>=(SND_CHUNK_SIZE_FLOAT*4)) { + chunk = chunk->next; + sampleOffset -= (SND_CHUNK_SIZE_FLOAT*4); + i++; + } + + if (i!=sfxScratchIndex || sfxScratchPointer != sc) { + S_AdpcmGetSamples( chunk, sfxScratchBuffer ); + sfxScratchIndex = i; + sfxScratchPointer = sc; + } + + samples = sfxScratchBuffer; + + for ( i=0 ; i>8; + samp[i].right += (data * rightvol)>>8; + + if (sampleOffset == SND_CHUNK_SIZE*2) { + chunk = chunk->next; + decodeWavelet(chunk, sfxScratchBuffer); + sfxScratchIndex++; + sampleOffset = 0; + } + } +} + +void S_PaintChannelFromADPCM( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) { + int data; + int leftvol, rightvol; + int i; + portable_samplepair_t *samp; + sndBuffer *chunk; + short *samples; + + leftvol = ch->leftvol*snd_vol; + rightvol = ch->rightvol*snd_vol; + + i = 0; + samp = &paintbuffer[ bufferOffset ]; + chunk = sc->soundData; + + if (ch->doppler) { + sampleOffset = sampleOffset*ch->oldDopplerScale; + } + + while (sampleOffset>=(SND_CHUNK_SIZE*4)) { + chunk = chunk->next; + sampleOffset -= (SND_CHUNK_SIZE*4); + i++; + } + + if (i!=sfxScratchIndex || sfxScratchPointer != sc) { + S_AdpcmGetSamples( chunk, sfxScratchBuffer ); + sfxScratchIndex = i; + sfxScratchPointer = sc; + } + + samples = sfxScratchBuffer; + + for ( i=0 ; i>8; + samp[i].right += (data * rightvol)>>8; + + if (sampleOffset == SND_CHUNK_SIZE*4) { + chunk = chunk->next; + S_AdpcmGetSamples( chunk, sfxScratchBuffer); + sampleOffset = 0; + sfxScratchIndex++; + } + } +} + +void S_PaintChannelFromMuLaw( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) { + int data; + int leftvol, rightvol; + int i; + portable_samplepair_t *samp; + sndBuffer *chunk; + byte *samples; + float ooff; + + leftvol = ch->leftvol*snd_vol; + rightvol = ch->rightvol*snd_vol; + + samp = &paintbuffer[ bufferOffset ]; + chunk = sc->soundData; + while (sampleOffset>=(SND_CHUNK_SIZE*2)) { + chunk = chunk->next; + sampleOffset -= (SND_CHUNK_SIZE*2); + if (!chunk) { + chunk = sc->soundData; + } + } + + if (!ch->doppler) { + samples = (byte *)chunk->sndChunk + sampleOffset; + for ( i=0 ; i>8; + samp[i].right += (data * rightvol)>>8; + samples++; + if (samples == (byte *)chunk->sndChunk+(SND_CHUNK_SIZE*2)) { + chunk = chunk->next; + samples = (byte *)chunk->sndChunk; + } + } + } else { + ooff = sampleOffset; + samples = (byte *)chunk->sndChunk; + for ( i=0 ; idopplerScale; + samp[i].left += (data * leftvol)>>8; + samp[i].right += (data * rightvol)>>8; + if (ooff >= SND_CHUNK_SIZE*2) { + chunk = chunk->next; + if (!chunk) { + chunk = sc->soundData; + } + samples = (byte *)chunk->sndChunk; + ooff = 0.0; + } + } + } +} + +/* +=================== +S_PaintChannels +=================== +*/ +void S_PaintChannels( int endtime ) { + int i; + int end; + int stream; + channel_t *ch; + sfx_t *sc; + int ltime, count; + int sampleOffset; + + if(s_muted->integer) + snd_vol = 0; + else + snd_vol = s_volume->value*255; + +//Com_Printf ("%i to %i\n", s_paintedtime, endtime); + while ( s_paintedtime < endtime ) { + // if paintbuffer is smaller than DMA buffer + // we may need to fill it multiple times + end = endtime; + if ( endtime - s_paintedtime > PAINTBUFFER_SIZE ) { + end = s_paintedtime + PAINTBUFFER_SIZE; + } + + // clear the paint buffer and mix any raw samples... + Com_Memset(paintbuffer, 0, sizeof (paintbuffer)); + for (stream = 0; stream < MAX_RAW_STREAMS; stream++) { + if ( s_rawend[stream] >= s_paintedtime ) { + // copy from the streaming sound source + const portable_samplepair_t *rawsamples = s_rawsamples[stream]; + const int stop = (end < s_rawend[stream]) ? end : s_rawend[stream]; + for ( i = s_paintedtime ; i < stop ; i++ ) { + const int s = i&(MAX_RAW_SAMPLES-1); + paintbuffer[i-s_paintedtime].left += rawsamples[s].left; + paintbuffer[i-s_paintedtime].right += rawsamples[s].right; + } + } + } + + // paint in the channels. + ch = s_channels; + for ( i = 0; i < MAX_CHANNELS ; i++, ch++ ) { + if ( !ch->thesfx || (ch->leftvol<0.25 && ch->rightvol<0.25 )) { + continue; + } + + ltime = s_paintedtime; + sc = ch->thesfx; + + sampleOffset = ltime - ch->startSample; + count = end - ltime; + if ( sampleOffset + count > sc->soundLength ) { + count = sc->soundLength - sampleOffset; + } + + if ( count > 0 ) { + if( sc->soundCompressionMethod == 1) { + S_PaintChannelFromADPCM (ch, sc, count, sampleOffset, ltime - s_paintedtime); + } else if( sc->soundCompressionMethod == 2) { + S_PaintChannelFromWavelet (ch, sc, count, sampleOffset, ltime - s_paintedtime); + } else if( sc->soundCompressionMethod == 3) { + S_PaintChannelFromMuLaw (ch, sc, count, sampleOffset, ltime - s_paintedtime); + } else { + S_PaintChannelFrom16 (ch, sc, count, sampleOffset, ltime - s_paintedtime); + } + } + } + + // paint in the looped channels. + ch = loop_channels; + for ( i = 0; i < numLoopChannels ; i++, ch++ ) { + if ( !ch->thesfx || (!ch->leftvol && !ch->rightvol )) { + continue; + } + + ltime = s_paintedtime; + sc = ch->thesfx; + + if (sc->soundData==NULL || sc->soundLength==0) { + continue; + } + // we might have to make two passes if it + // is a looping sound effect and the end of + // the sample is hit + do { + sampleOffset = (ltime % sc->soundLength); + + count = end - ltime; + if ( sampleOffset + count > sc->soundLength ) { + count = sc->soundLength - sampleOffset; + } + + if ( count > 0 ) { + if( sc->soundCompressionMethod == 1) { + S_PaintChannelFromADPCM (ch, sc, count, sampleOffset, ltime - s_paintedtime); + } else if( sc->soundCompressionMethod == 2) { + S_PaintChannelFromWavelet (ch, sc, count, sampleOffset, ltime - s_paintedtime); + } else if( sc->soundCompressionMethod == 3) { + S_PaintChannelFromMuLaw (ch, sc, count, sampleOffset, ltime - s_paintedtime); + } else { + S_PaintChannelFrom16 (ch, sc, count, sampleOffset, ltime - s_paintedtime); + } + ltime += count; + } + } while ( ltime < end); + } + + // transfer out according to DMA format + S_TransferPaintBuffer( end ); + s_paintedtime = end; + } +} -- cgit