/* =========================================================================== Copyright (C) 1999 - 2005, Id Software, Inc. Copyright (C) 2000 - 2013, Raven Software, Inc. Copyright (C) 2001 - 2013, Activision, Inc. Copyright (C) 2013 - 2015, OpenJK contributors This file is part of the OpenJK source code. OpenJK is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License version 2 as published by the Free Software Foundation. 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, see . =========================================================================== */ // snd_mix.c -- portable code to mix sounds for snd_dma.c #include "../server/exe_headers.h" #include "snd_local.h" portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE]; int *snd_p, snd_linear_count, snd_vol; short *snd_out; void S_WriteLinearBlastStereo16 (void) { int i; int val; for (i=0 ; i>8; if (val > 0x7fff) snd_out[i] = 0x7fff; else if (val < (short)0x8000) snd_out[i] = (short)0x8000; else snd_out[i] = val; val = snd_p[i+1]>>8; if (val > 0x7fff) snd_out[i+1] = 0x7fff; else if (val < (short)0x8000) snd_out[i+1] = (short)0x8000; else snd_out[i+1] = val; } } 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); } } /* =================== 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 < (short)0x8000) val = (short)0x8000; out[out_idx] = (short)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 < (short)0x8000) val = (short)0x8000; out[out_idx] = (short)((val>>8) + 128); out_idx = (out_idx + 1) & out_mask; } } } } /* =============================================================================== CHANNEL MIXING =============================================================================== */ static void S_PaintChannelFrom16( channel_t *ch, const sfx_t *sfx, int count, int sampleOffset, int bufferOffset ) { portable_samplepair_t *pSamplesDest; int iData; int iLeftVol = ch->leftvol * snd_vol; int iRightVol = ch->rightvol * snd_vol; pSamplesDest = &paintbuffer[ bufferOffset ]; for ( int i=0 ; ipSoundData[ sampleOffset++ ]; pSamplesDest[i].left += (iData * iLeftVol )>>8; pSamplesDest[i].right += (iData * iRightVol)>>8; } } void S_PaintChannelFromMP3( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) { int data; int leftvol, rightvol; signed short *sfx; int i; portable_samplepair_t *samp; static short tempMP3Buffer[PAINTBUFFER_SIZE]; MP3Stream_GetSamples( ch, sampleOffset, count, tempMP3Buffer, qfalse ); // qfalse = not stereo leftvol = ch->leftvol*snd_vol; rightvol = ch->rightvol*snd_vol; sfx = tempMP3Buffer; samp = &paintbuffer[ bufferOffset ]; while ( count & 3 ) { data = *sfx; samp->left += (data * leftvol)>>8; samp->right += (data * rightvol)>>8; sfx++; samp++; count--; } for ( i=0 ; i>8; samp[i].right += (data * rightvol)>>8; data = sfx[i+1]; samp[i+1].left += (data * leftvol)>>8; samp[i+1].right += (data * rightvol)>>8; data = sfx[i+2]; samp[i+2].left += (data * leftvol)>>8; samp[i+2].right += (data * rightvol)>>8; data = sfx[i+3]; samp[i+3].left += (data * leftvol)>>8; samp[i+3].right += (data * rightvol)>>8; } } // subroutinised to save code dup (called twice) -ste // void ChannelPaint(channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset) { switch (sc->eSoundCompressionMethod) { case ct_16: S_PaintChannelFrom16 (ch, sc, count, sampleOffset, bufferOffset); break; case ct_MP3: S_PaintChannelFromMP3 (ch, sc, count, sampleOffset, bufferOffset); break; default: assert(0); // debug aid, ignored in release. FIXME: Should we ERR_DROP here for badness-catch? break; } } void S_PaintChannels( int endtime ) { int i; int end; channel_t *ch; sfx_t *sc; int ltime, count; int sampleOffset; int normal_vol,voice_vol; snd_vol = normal_vol = s_volume->value*256.0f; voice_vol = (s_volumeVoice->value*256.0f); //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 to either music or zeros if ( s_rawend < s_paintedtime ) { if ( s_rawend ) { //Com_DPrintf ("background sound underrun\n"); } memset(paintbuffer, 0, (end - s_paintedtime) * sizeof(portable_samplepair_t)); } else { // copy from the streaming sound source int s; int stop; stop = (end < s_rawend) ? end : s_rawend; for ( i = s_paintedtime ; i < stop ; i++ ) { s = i&(MAX_RAW_SAMPLES-1); paintbuffer[i-s_paintedtime] = s_rawsamples[s]; } // if (i != end) // Com_Printf ("partial stream\n"); // else // Com_Printf ("full stream\n"); for ( ; i < end ; i++ ) { paintbuffer[i-s_paintedtime].left = paintbuffer[i-s_paintedtime].right = 0; } } // 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; } if ( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL ) snd_vol = voice_vol; else snd_vol = normal_vol; ltime = s_paintedtime; sc = ch->thesfx; // we might have to make 2 passes if it is // a looping sound effect and the end of // the sameple is hit... // do { if (ch->loopSound) { sampleOffset = ltime % sc->iSoundLengthInSamples; } else { sampleOffset = ltime - ch->startSample; } count = end - ltime; if ( sampleOffset + count > sc->iSoundLengthInSamples ) { count = sc->iSoundLengthInSamples - sampleOffset; } if ( count > 0 ) { ChannelPaint(ch, sc, count, sampleOffset, ltime - s_paintedtime); ltime += count; } } while ( ltime < end && ch->loopSound ); } /* temprem // 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 ) { ChannelPaint(ch, sc, count, sampleOffset, ltime - s_paintedtime); ltime += count; } } while ( ltime < end); } } */ // transfer out according to DMA format S_TransferPaintBuffer( end ); s_paintedtime = end; } }