/* Copyright (C) 1996-1997 Id Software, Inc. This program 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. This program 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 this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ // snd_mix.c -- portable code to mix sounds #include "quakedef.h" #include "snd_main.h" typedef struct portable_samplepair_s { int sample[SND_LISTENERS]; } portable_sampleframe_t; // LordHavoc: was 512, expanded to 2048 #define PAINTBUFFER_SIZE 2048 portable_sampleframe_t paintbuffer[PAINTBUFFER_SIZE]; // FIXME: this desyncs with the video too easily extern void SCR_CaptureVideo_SoundFrame(unsigned char *bufstereo16le, size_t length, int rate); void S_CaptureAVISound(portable_sampleframe_t *buf, size_t length) { int n; size_t i; unsigned char out[PAINTBUFFER_SIZE * 4]; if (!cls.capturevideo_active) return; // write the sound buffer as little endian 16bit interleaved stereo for(i = 0;i < length;i++) { n = buf[i].sample[0]; n = bound(-32768, n, 32767); out[i*4+0] = n & 0xFF; out[i*4+1] = (n >> 8) & 0xFF; n = buf[i].sample[1]; n = bound(-32768, n, 32767); out[i*4+2] = n & 0xFF; out[i*4+3] = (n >> 8) & 0xFF; } SCR_CaptureVideo_SoundFrame(out, length, shm->format.speed); } // TODO: rewrite this function void S_TransferPaintBuffer(int endtime) { void *pbuf; int i; portable_sampleframe_t *snd_p; int lpaintedtime; int snd_frames; int val; if ((pbuf = S_LockBuffer())) { snd_p = paintbuffer; lpaintedtime = paintedtime; for (lpaintedtime = paintedtime;lpaintedtime < endtime;lpaintedtime += snd_frames) { // handle recirculating buffer issues i = lpaintedtime & (shm->sampleframes - 1); snd_frames = shm->sampleframes - i; if (snd_frames > endtime - lpaintedtime) snd_frames = endtime - lpaintedtime; if (shm->format.width == 2) { // 16bit short *snd_out = (short *) pbuf + i * shm->format.channels; if (shm->format.channels == 8) { // 7.1 surround if (snd_swapstereo.value) { for (i = 0;i < snd_frames;i++, snd_p++) { *snd_out++ = bound(-32768, snd_p->sample[1], 32767); *snd_out++ = bound(-32768, snd_p->sample[0], 32767); *snd_out++ = bound(-32768, snd_p->sample[3], 32767); *snd_out++ = bound(-32768, snd_p->sample[2], 32767); *snd_out++ = bound(-32768, snd_p->sample[4], 32767); *snd_out++ = bound(-32768, snd_p->sample[5], 32767); *snd_out++ = bound(-32768, snd_p->sample[6], 32767); *snd_out++ = bound(-32768, snd_p->sample[7], 32767); } } else { for (i = 0;i < snd_frames;i++, snd_p++) { *snd_out++ = bound(-32768, snd_p->sample[0], 32767); *snd_out++ = bound(-32768, snd_p->sample[1], 32767); *snd_out++ = bound(-32768, snd_p->sample[2], 32767); *snd_out++ = bound(-32768, snd_p->sample[3], 32767); *snd_out++ = bound(-32768, snd_p->sample[4], 32767); *snd_out++ = bound(-32768, snd_p->sample[5], 32767); *snd_out++ = bound(-32768, snd_p->sample[6], 32767); *snd_out++ = bound(-32768, snd_p->sample[7], 32767); } } } else if (shm->format.channels == 6) { // 5.1 surround if (snd_swapstereo.value) { for (i = 0;i < snd_frames;i++, snd_p++) { *snd_out++ = bound(-32768, snd_p->sample[1], 32767); *snd_out++ = bound(-32768, snd_p->sample[0], 32767); *snd_out++ = bound(-32768, snd_p->sample[3], 32767); *snd_out++ = bound(-32768, snd_p->sample[2], 32767); *snd_out++ = bound(-32768, snd_p->sample[4], 32767); *snd_out++ = bound(-32768, snd_p->sample[5], 32767); } } else { for (i = 0;i < snd_frames;i++, snd_p++) { *snd_out++ = bound(-32768, snd_p->sample[0], 32767); *snd_out++ = bound(-32768, snd_p->sample[1], 32767); *snd_out++ = bound(-32768, snd_p->sample[2], 32767); *snd_out++ = bound(-32768, snd_p->sample[3], 32767); *snd_out++ = bound(-32768, snd_p->sample[4], 32767); *snd_out++ = bound(-32768, snd_p->sample[5], 32767); } } } else if (shm->format.channels == 4) { // 4.0 surround if (snd_swapstereo.value) { for (i = 0;i < snd_frames;i++, snd_p++) { *snd_out++ = bound(-32768, snd_p->sample[1], 32767); *snd_out++ = bound(-32768, snd_p->sample[0], 32767); *snd_out++ = bound(-32768, snd_p->sample[3], 32767); *snd_out++ = bound(-32768, snd_p->sample[2], 32767); } } else { for (i = 0;i < snd_frames;i++, snd_p++) { *snd_out++ = bound(-32768, snd_p->sample[0], 32767); *snd_out++ = bound(-32768, snd_p->sample[1], 32767); *snd_out++ = bound(-32768, snd_p->sample[2], 32767); *snd_out++ = bound(-32768, snd_p->sample[3], 32767); } } } else if (shm->format.channels == 2) { // 2.0 stereo if (snd_swapstereo.value) { for (i = 0;i < snd_frames;i++, snd_p++) { *snd_out++ = bound(-32768, snd_p->sample[1], 32767); *snd_out++ = bound(-32768, snd_p->sample[0], 32767); } } else { for (i = 0;i < snd_frames;i++, snd_p++) { *snd_out++ = bound(-32768, snd_p->sample[0], 32767); *snd_out++ = bound(-32768, snd_p->sample[1], 32767); } } } else if (shm->format.channels == 1) { // 1.0 mono for (i = 0;i < snd_frames;i++, snd_p++) *snd_out++ = bound(-32768, (snd_p->sample[0] + snd_p->sample[1]) >> 1, 32767); } } else { // 8bit unsigned char *snd_out = (unsigned char *) pbuf + i * shm->format.channels; if (shm->format.channels == 8) { // 7.1 surround if (snd_swapstereo.value) { for (i = 0;i < snd_frames;i++, snd_p++) { val = (snd_p->sample[1] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[0] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[3] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[2] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[4] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[5] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[6] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[7] >> 8) + 128;*snd_out++ = bound(0, val, 255); } } else { for (i = 0;i < snd_frames;i++, snd_p++) { val = (snd_p->sample[0] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[1] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[2] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[3] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[4] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[5] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[6] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[7] >> 8) + 128;*snd_out++ = bound(0, val, 255); } } } else if (shm->format.channels == 6) { // 5.1 surround if (snd_swapstereo.value) { for (i = 0;i < snd_frames;i++, snd_p++) { val = (snd_p->sample[1] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[0] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[3] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[2] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[4] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[5] >> 8) + 128;*snd_out++ = bound(0, val, 255); } } else { for (i = 0;i < snd_frames;i++, snd_p++) { val = (snd_p->sample[0] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[1] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[2] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[3] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[4] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[5] >> 8) + 128;*snd_out++ = bound(0, val, 255); } } } else if (shm->format.channels == 4) { // 4.0 surround if (snd_swapstereo.value) { for (i = 0;i < snd_frames;i++, snd_p++) { val = (snd_p->sample[1] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[0] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[3] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[2] >> 8) + 128;*snd_out++ = bound(0, val, 255); } } else { for (i = 0;i < snd_frames;i++, snd_p++) { val = (snd_p->sample[0] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[1] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[2] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[3] >> 8) + 128;*snd_out++ = bound(0, val, 255); } } } else if (shm->format.channels == 2) { // 2.0 stereo if (snd_swapstereo.value) { for (i = 0;i < snd_frames;i++, snd_p++) { val = (snd_p->sample[1] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[0] >> 8) + 128;*snd_out++ = bound(0, val, 255); } } else { for (i = 0;i < snd_frames;i++, snd_p++) { val = (snd_p->sample[0] >> 8) + 128;*snd_out++ = bound(0, val, 255); val = (snd_p->sample[1] >> 8) + 128;*snd_out++ = bound(0, val, 255); } } } else if (shm->format.channels == 1) { // 1.0 mono for (i = 0;i < snd_frames;i++, snd_p++) { val = ((snd_p->sample[0]+snd_p->sample[1]) >> 9) + 128;*snd_out++ = bound(0, val, 255); } } } } S_UnlockBuffer(); } } /* =============================================================================== CHANNEL MIXING =============================================================================== */ qboolean SND_PaintChannel (channel_t *ch, int endtime); void S_PaintChannels(int endtime) { unsigned int i, j; int end; channel_t *ch; sfx_t *sfx; int ltime, count; while (paintedtime < endtime) { // if paintbuffer is smaller than DMA buffer end = endtime; if (endtime - paintedtime > PAINTBUFFER_SIZE) end = paintedtime + PAINTBUFFER_SIZE; // clear the paint buffer memset (&paintbuffer, 0, (end - paintedtime) * sizeof (paintbuffer[0])); // paint in the channels. ch = channels; for (i=0; isfx; if (!sfx) continue; for (j = 0;j < SND_LISTENERS;j++) if (ch->listener_volume[j]) break; if (j == SND_LISTENERS) continue; if (!S_LoadSound (sfx, true)) continue; // if the channel is paused if (ch->flags & CHANNELFLAG_PAUSED) { int pausedtime = end - paintedtime; ch->lastptime += pausedtime; ch->end += pausedtime; continue; } // if the sound hasn't been painted last time, update his position if (ch->lastptime < paintedtime) { ch->pos += paintedtime - ch->lastptime; // If the sound should have ended by then if ((unsigned int)ch->pos > sfx->total_length) { int loopstart; if (ch->flags & CHANNELFLAG_FORCELOOP) loopstart = 0; else loopstart = -1; if (sfx->loopstart >= 0) loopstart = sfx->loopstart; // If the sound is looped if (loopstart >= 0) ch->pos = (ch->pos - sfx->total_length) % (sfx->total_length - loopstart) + loopstart; else ch->pos = sfx->total_length; ch->end = paintedtime + sfx->total_length - ch->pos; } } ltime = paintedtime; while (ltime < end) { qboolean stop_paint; // paint up to end if (ch->end < end) count = (int)ch->end - ltime; else count = end - ltime; if (count > 0) { for (j = 0;j < SND_LISTENERS;j++) ch->listener_volume[j] = bound(0, ch->listener_volume[j], 255); stop_paint = !SND_PaintChannel (ch, count); if (!stop_paint) { ltime += count; ch->lastptime = ltime; } } else stop_paint = false; if (ltime >= ch->end) { // if at end of loop, restart if ((sfx->loopstart >= 0 || (ch->flags & CHANNELFLAG_FORCELOOP)) && !stop_paint) { ch->pos = bound(0, sfx->loopstart, (int)sfx->total_length - 1); ch->end = ltime + sfx->total_length - ch->pos; } // channel just stopped else stop_paint = true; } if (stop_paint) { S_StopChannel (ch - channels); break; } } } // transfer out according to DMA format S_CaptureAVISound (paintbuffer, end - paintedtime); S_TransferPaintBuffer(end); paintedtime = end; } } qboolean SND_PaintChannel (channel_t *ch, int count) { int snd_vol, vol[SND_LISTENERS]; const sfxbuffer_t *sb; int i; // If this channel manages its own volume if (ch->flags & CHANNELFLAG_FULLVOLUME) snd_vol = 256; else snd_vol = (int)(volume.value * 256); for (i = 0;i < SND_LISTENERS;i++) vol[i] = ch->listener_volume[i] * snd_vol; sb = ch->sfx->fetcher->getsb (ch, ch->pos, count); if (sb == NULL) return false; #if SND_LISTENERS != 8 #error this code only supports up to 8 channels, update it #endif if (ch->sfx->format.width == 1) { const signed char *sfx = (signed char *)sb->data + (ch->pos - sb->offset) * ch->sfx->format.channels; // Stereo sound support if (ch->sfx->format.channels == 2) { if (vol[6] + vol[7] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 8; paintbuffer[i].sample[1] += (sfx[1] * vol[1]) >> 8; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 8; paintbuffer[i].sample[3] += (sfx[1] * vol[3]) >> 8; paintbuffer[i].sample[4] += ((sfx[0]+sfx[1]) * vol[4]) >> 9; paintbuffer[i].sample[5] += ((sfx[0]+sfx[1]) * vol[5]) >> 9; paintbuffer[i].sample[6] += (sfx[0] * vol[6]) >> 8; paintbuffer[i].sample[7] += (sfx[1] * vol[7]) >> 8; sfx += 2; } } else if (vol[4] + vol[5] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 8; paintbuffer[i].sample[1] += (sfx[1] * vol[1]) >> 8; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 8; paintbuffer[i].sample[3] += (sfx[1] * vol[3]) >> 8; paintbuffer[i].sample[4] += ((sfx[0]+sfx[1]) * vol[4]) >> 9; paintbuffer[i].sample[5] += ((sfx[0]+sfx[1]) * vol[5]) >> 9; sfx += 2; } } else if (vol[2] + vol[3] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 8; paintbuffer[i].sample[1] += (sfx[1] * vol[1]) >> 8; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 8; paintbuffer[i].sample[3] += (sfx[1] * vol[3]) >> 8; sfx += 2; } } else if (vol[0] + vol[1] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 8; paintbuffer[i].sample[1] += (sfx[1] * vol[1]) >> 8; sfx += 2; } } } else if (ch->sfx->format.channels == 1) { if (vol[6] + vol[7] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 8; paintbuffer[i].sample[1] += (sfx[0] * vol[1]) >> 8; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 8; paintbuffer[i].sample[3] += (sfx[0] * vol[3]) >> 8; paintbuffer[i].sample[4] += (sfx[0] * vol[4]) >> 8; paintbuffer[i].sample[5] += (sfx[0] * vol[5]) >> 8; paintbuffer[i].sample[6] += (sfx[0] * vol[6]) >> 8; paintbuffer[i].sample[7] += (sfx[0] * vol[7]) >> 8; sfx += 1; } } else if (vol[4] + vol[5] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 8; paintbuffer[i].sample[1] += (sfx[0] * vol[1]) >> 8; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 8; paintbuffer[i].sample[3] += (sfx[0] * vol[3]) >> 8; paintbuffer[i].sample[4] += (sfx[0] * vol[4]) >> 8; paintbuffer[i].sample[5] += (sfx[0] * vol[5]) >> 8; sfx += 1; } } else if (vol[2] + vol[3] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 8; paintbuffer[i].sample[1] += (sfx[0] * vol[1]) >> 8; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 8; paintbuffer[i].sample[3] += (sfx[0] * vol[3]) >> 8; sfx += 1; } } else if (vol[0] + vol[1] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 8; paintbuffer[i].sample[1] += (sfx[0] * vol[1]) >> 8; sfx += 1; } } } else return true; // unsupported number of channels in sound } else if (ch->sfx->format.width == 2) { const signed short *sfx = (signed short *)sb->data + (ch->pos - sb->offset) * ch->sfx->format.channels; // Stereo sound support if (ch->sfx->format.channels == 2) { if (vol[6] + vol[7] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 16; paintbuffer[i].sample[1] += (sfx[1] * vol[1]) >> 16; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 16; paintbuffer[i].sample[3] += (sfx[1] * vol[3]) >> 16; paintbuffer[i].sample[4] += ((sfx[0]+sfx[1]) * vol[4]) >> 17; paintbuffer[i].sample[5] += ((sfx[0]+sfx[1]) * vol[5]) >> 17; paintbuffer[i].sample[6] += (sfx[0] * vol[6]) >> 16; paintbuffer[i].sample[7] += (sfx[1] * vol[7]) >> 16; sfx += 2; } } else if (vol[4] + vol[5] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 16; paintbuffer[i].sample[1] += (sfx[1] * vol[1]) >> 16; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 16; paintbuffer[i].sample[3] += (sfx[1] * vol[3]) >> 16; paintbuffer[i].sample[4] += ((sfx[0]+sfx[1]) * vol[4]) >> 17; paintbuffer[i].sample[5] += ((sfx[0]+sfx[1]) * vol[5]) >> 17; sfx += 2; } } else if (vol[2] + vol[3] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 16; paintbuffer[i].sample[1] += (sfx[1] * vol[1]) >> 16; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 16; paintbuffer[i].sample[3] += (sfx[1] * vol[3]) >> 16; sfx += 2; } } else if (vol[0] + vol[1] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 16; paintbuffer[i].sample[1] += (sfx[1] * vol[1]) >> 16; sfx += 2; } } } else if (ch->sfx->format.channels == 1) { if (vol[6] + vol[7] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 16; paintbuffer[i].sample[1] += (sfx[0] * vol[1]) >> 16; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 16; paintbuffer[i].sample[3] += (sfx[0] * vol[3]) >> 16; paintbuffer[i].sample[4] += (sfx[0] * vol[4]) >> 16; paintbuffer[i].sample[5] += (sfx[0] * vol[5]) >> 16; paintbuffer[i].sample[6] += (sfx[0] * vol[6]) >> 16; paintbuffer[i].sample[7] += (sfx[0] * vol[7]) >> 16; sfx += 1; } } else if (vol[4] + vol[5] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 16; paintbuffer[i].sample[1] += (sfx[0] * vol[1]) >> 16; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 16; paintbuffer[i].sample[3] += (sfx[0] * vol[3]) >> 16; paintbuffer[i].sample[4] += (sfx[0] * vol[4]) >> 16; paintbuffer[i].sample[5] += (sfx[0] * vol[5]) >> 16; sfx += 1; } } else if (vol[2] + vol[3] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 16; paintbuffer[i].sample[1] += (sfx[0] * vol[1]) >> 16; paintbuffer[i].sample[2] += (sfx[0] * vol[2]) >> 16; paintbuffer[i].sample[3] += (sfx[0] * vol[3]) >> 16; sfx += 1; } } else if (vol[0] + vol[1] > 0) { for (i = 0;i < count;i++) { paintbuffer[i].sample[0] += (sfx[0] * vol[0]) >> 16; paintbuffer[i].sample[1] += (sfx[0] * vol[1]) >> 16; sfx += 1; } } } else return true; // unsupported number of channels in sound } ch->pos += count; return true; }