teambeef/qvr
0
0
Fork 0
mirror of https://github.com/DrBeef/QVR.git synced 2024-11-26 05:42:01 +00:00
Code Issues Projects Releases Packages Wiki Activity
qvr/app/jni/snd_mix.c
Simon 90d0c76d59 Initial Commit of Source
2016-01-29 22:43:55 +00:00

532 lines
18 KiB
C
Raw Blame History

/*
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.
*/
#include "quakedef.h"
#include "snd_main.h"
extern cvar_t snd_softclip;
static portable_sampleframe_t paintbuffer[PAINTBUFFER_SIZE];
static portable_sampleframe_t paintbuffer_unswapped[PAINTBUFFER_SIZE];
extern speakerlayout_t snd_speakerlayout; // for querying the listeners
static void S_CaptureAVISound(const portable_sampleframe_t *paintbuffer, size_t length)
{
size_t i;
unsigned int j;
if (!cls.capturevideo.active)
return;
// undo whatever swapping the channel layout (swapstereo, ALSA) did
for(j = 0; j < snd_speakerlayout.channels; ++j)
{
unsigned int j0 = snd_speakerlayout.listeners[j].channel_unswapped;
for(i = 0; i < length; ++i)
paintbuffer_unswapped[i].sample[j0] = paintbuffer[i].sample[j];
}
SCR_CaptureVideo_SoundFrame(paintbuffer_unswapped, length);
}
extern cvar_t snd_softclip;
static void S_SoftClipPaintBuffer(portable_sampleframe_t *painted_ptr, int nbframes, int width, int channels)
{
int i;
if((snd_softclip.integer == 1 && width <= 2) || snd_softclip.integer > 1)
{
portable_sampleframe_t *p = painted_ptr;
#if 0
/* Soft clipping, the sound of a dream, thanks to Jon Wattes
post to Musicdsp.org */
#define SOFTCLIP(x) (x) = sin(bound(-M_PI/2, (x), M_PI/2)) * 0.25
#endif
// let's do a simple limiter instead, seems to sound better
static float maxvol = 0;
maxvol = max(1.0f, maxvol * (1.0f - nbframes / (0.4f * snd_renderbuffer->format.speed)));
#define SOFTCLIP(x) if(fabs(x)>maxvol) maxvol=fabs(x); (x) /= maxvol;
if (channels == 8) // 7.1 surround
{
for (i = 0;i < nbframes;i++, p++)
{
SOFTCLIP(p->sample[0]);
SOFTCLIP(p->sample[1]);
SOFTCLIP(p->sample[2]);
SOFTCLIP(p->sample[3]);
SOFTCLIP(p->sample[4]);
SOFTCLIP(p->sample[5]);
SOFTCLIP(p->sample[6]);
SOFTCLIP(p->sample[7]);
}
}
else if (channels == 6) // 5.1 surround
{
for (i = 0; i < nbframes; i++, p++)
{
SOFTCLIP(p->sample[0]);
SOFTCLIP(p->sample[1]);
SOFTCLIP(p->sample[2]);
SOFTCLIP(p->sample[3]);
SOFTCLIP(p->sample[4]);
SOFTCLIP(p->sample[5]);
}
}
else if (channels == 4) // 4.0 surround
{
for (i = 0; i < nbframes; i++, p++)
{
SOFTCLIP(p->sample[0]);
SOFTCLIP(p->sample[1]);
SOFTCLIP(p->sample[2]);
SOFTCLIP(p->sample[3]);
}
}
else if (channels == 2) // 2.0 stereo
{
for (i = 0; i < nbframes; i++, p++)
{
SOFTCLIP(p->sample[0]);
SOFTCLIP(p->sample[1]);
}
}
else if (channels == 1) // 1.0 mono
{
for (i = 0; i < nbframes; i++, p++)
{
SOFTCLIP(p->sample[0]);
}
}
#undef SOFTCLIP
}
}
static void S_ConvertPaintBuffer(portable_sampleframe_t *painted_ptr, void *rb_ptr, int nbframes, int width, int channels)
{
int i, val;
// FIXME: add 24bit and 32bit float formats
// FIXME: optimize with SSE intrinsics?
if (width == 2) // 16bit
{
short *snd_out = (short*)rb_ptr;
if (channels == 8) // 7.1 surround
{
for (i = 0;i < nbframes;i++, painted_ptr++)
{
val = (int)(painted_ptr->sample[0] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[1] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[2] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[3] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[4] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[5] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[6] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[7] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
}
}
else if (channels == 6) // 5.1 surround
{
for (i = 0; i < nbframes; i++, painted_ptr++)
{
val = (int)(painted_ptr->sample[0] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[1] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[2] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[3] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[4] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[5] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
}
}
else if (channels == 4) // 4.0 surround
{
for (i = 0; i < nbframes; i++, painted_ptr++)
{
val = (int)(painted_ptr->sample[0] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[1] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[2] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[3] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
}
}
else if (channels == 2) // 2.0 stereo
{
for (i = 0; i < nbframes; i++, painted_ptr++)
{
val = (int)(painted_ptr->sample[0] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
val = (int)(painted_ptr->sample[1] * 32768.0f);*snd_out++ = bound(-32768, val, 32767);
}
}
else if (channels == 1) // 1.0 mono
{
for (i = 0; i < nbframes; i++, painted_ptr++)
{
val = (int)((painted_ptr->sample[0] + painted_ptr->sample[1]) * 16384.0f);*snd_out++ = bound(-32768, val, 32767);
}
}
// noise is really really annoying
if (cls.timedemo)
memset(rb_ptr, 0, nbframes * channels * width);
}
else // 8bit
{
unsigned char *snd_out = (unsigned char*)rb_ptr;
if (channels == 8) // 7.1 surround
{
for (i = 0; i < nbframes; i++, painted_ptr++)
{
val = (int)(painted_ptr->sample[0] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[1] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[2] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[3] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[4] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[5] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[6] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[7] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
}
}
else if (channels == 6) // 5.1 surround
{
for (i = 0; i < nbframes; i++, painted_ptr++)
{
val = (int)(painted_ptr->sample[0] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[1] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[2] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[3] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[4] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[5] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
}
}
else if (channels == 4) // 4.0 surround
{
for (i = 0; i < nbframes; i++, painted_ptr++)
{
val = (int)(painted_ptr->sample[0] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[1] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[2] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[3] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
}
}
else if (channels == 2) // 2.0 stereo
{
for (i = 0; i < nbframes; i++, painted_ptr++)
{
val = (int)(painted_ptr->sample[0] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
val = (int)(painted_ptr->sample[1] * 128.0f) + 128; *snd_out++ = bound(0, val, 255);
}
}
else if (channels == 1) // 1.0 mono
{
for (i = 0;i < nbframes;i++, painted_ptr++)
{
val = (int)((painted_ptr->sample[0] + painted_ptr->sample[1]) * 64.0f) + 128; *snd_out++ = bound(0, val, 255);
}
}
// noise is really really annoying
if (cls.timedemo)
memset(rb_ptr, 128, nbframes * channels);
}
}
/*
===============================================================================
CHANNEL MIXING
===============================================================================
*/
void S_MixToBuffer(void *stream, unsigned int bufferframes)
{
int channelindex;
channel_t *ch;
int totalmixframes;
unsigned char *outbytes = (unsigned char *) stream;
sfx_t *sfx;
portable_sampleframe_t *paint;
int wantframes;
int i;
int count;
int fetched;
int fetch;
int istartframe;
int iendframe;
int ilengthframes;
int totallength;
int loopstart;
int indexfrac;
int indexfracstep;
#define S_FETCHBUFFERSIZE 4096
float fetchsampleframes[S_FETCHBUFFERSIZE*2];
const float *fetchsampleframe;
float vol[SND_LISTENERS];
float lerp[2];
float sample[3];
double posd;
double speedd;
float maxvol;
qboolean looping;
qboolean silent;
// mix as many times as needed to fill the requested buffer
while (bufferframes)
{
// limit to the size of the paint buffer
totalmixframes = min(bufferframes, PAINTBUFFER_SIZE);
// clear the paint buffer
memset(paintbuffer, 0, totalmixframes * sizeof(paintbuffer[0]));
// paint in the channels.
// channels with zero volumes still advance in time but don't paint.
ch = channels; // cppcheck complains here but it is wrong, channels is a channel_t[MAX_CHANNELS] and not an int
for (channelindex = 0;channelindex < (int)total_channels;channelindex++, ch++)
{
sfx = ch->sfx;
if (sfx == NULL)
continue;
if (!S_LoadSound (sfx, true))
continue;
if (ch->flags & CHANNELFLAG_PAUSED)
continue;
if (!sfx->total_length)
continue;
// copy the channel information to the stack for reference, otherwise the
// values might change during a mix if the spatializer is updating them
// (note: this still may get some old and some new values!)
posd = ch->position;
speedd = ch->mixspeed * sfx->format.speed / snd_renderbuffer->format.speed;
for (i = 0;i < SND_LISTENERS;i++)
vol[i] = ch->volume[i];
// check total volume level, because we can skip some code on silent sounds but other code must still run (position updates mainly)
maxvol = 0;
for (i = 0;i < SND_LISTENERS;i++)
if(vol[i] > maxvol)
maxvol = vol[i];
switch(snd_renderbuffer->format.width)
{
case 1: // 8bpp
silent = maxvol < (1.0f / (256.0f));
// so silent it has zero effect
break;
case 2: // 16bpp
silent = maxvol < (1.0f / (65536.0f));
// so silent it has zero effect
break;
default: // floating point
silent = maxvol < 1.0e-13f;
// 130 dB is difference between hearing
// threshold and a jackhammer from
// working distance.
// therefore, anyone who turns up
// volume so much they notice this
// cutoff, likely already has their
// ear-drums blown out anyway.
break;
}
// when doing prologic mixing, some channels invert one side
if (ch->prologic_invert == -1)
vol[1] *= -1.0f;
// get some sfx info in a consistent form
totallength = sfx->total_length;
loopstart = (int)sfx->loopstart < totallength ? (int)sfx->loopstart : ((ch->flags & CHANNELFLAG_FORCELOOP) ? 0 : totallength);
looping = loopstart < totallength;
// do the actual paint now (may skip work if silent)
paint = paintbuffer;
istartframe = 0;
for (wantframes = totalmixframes;wantframes > 0;posd += count * speedd, wantframes -= count)
{
// check if this is a delayed sound
if (posd < 0)
{
// for a delayed sound we have to eat into the delay first
count = (int)floor(-posd / speedd) + 1;
count = bound(1, count, wantframes);
// let the for loop iterator apply the skip
continue;
}
// compute a fetch size that won't overflow our buffer
count = wantframes;
for (;;)
{
istartframe = (int)floor(posd);
iendframe = (int)floor(posd + (count-1) * speedd);
ilengthframes = count > 1 ? (iendframe - istartframe + 2) : 2;
if (ilengthframes <= S_FETCHBUFFERSIZE)
break;
// reduce count by 25% and try again
count -= count >> 2;
}
// zero whole fetch buffer for safety
// (floating point noise from uninitialized memory = HORRIBLE)
// otherwise we would only need to clear the excess
if (!silent)
memset(fetchsampleframes, 0, ilengthframes*sfx->format.channels*sizeof(fetchsampleframes[0]));
// if looping, do multiple fetches
fetched = 0;
for (;;)
{
fetch = min(ilengthframes - fetched, totallength - istartframe);
if (fetch > 0)
{
if (!silent)
sfx->fetcher->getsamplesfloat(ch, sfx, istartframe, fetch, fetchsampleframes + fetched*sfx->format.channels);
istartframe += fetch;
fetched += fetch;
}
if (istartframe == totallength && looping && fetched < ilengthframes)
{
// loop and fetch some more
posd += loopstart - totallength;
istartframe = loopstart;
}
else
{
break;
}
}
// set up our fixedpoint resampling variables (float to int conversions are expensive so do not do one per sampleframe)
fetchsampleframe = fetchsampleframes;
indexfrac = (int)floor((posd - floor(posd)) * 65536.0);
indexfracstep = (int)floor(speedd * 65536.0);
if (!silent)
{
if (sfx->format.channels == 2)
{
// music is stereo
#if SND_LISTENERS != 8
#error the following code only supports up to 8 channels, update it
#endif
if (snd_speakerlayout.channels > 2)
{
// surround mixing
for (i = 0;i < count;i++, paint++)
{
lerp[1] = indexfrac * (1.0f / 65536.0f);
lerp[0] = 1.0f - lerp[1];
sample[0] = fetchsampleframe[0] * lerp[0] + fetchsampleframe[2] * lerp[1];
sample[1] = fetchsampleframe[1] * lerp[0] + fetchsampleframe[3] * lerp[1];
sample[2] = (sample[0] + sample[1]) * 0.5f;
paint->sample[0] += sample[0] * vol[0];
paint->sample[1] += sample[1] * vol[1];
paint->sample[2] += sample[0] * vol[2];
paint->sample[3] += sample[1] * vol[3];
paint->sample[4] += sample[2] * vol[4];
paint->sample[5] += sample[2] * vol[5];
paint->sample[6] += sample[0] * vol[6];
paint->sample[7] += sample[1] * vol[7];
indexfrac += indexfracstep;
fetchsampleframe += 2 * (indexfrac >> 16);
indexfrac &= 0xFFFF;
}
}
else
{
// stereo mixing
for (i = 0;i < count;i++, paint++)
{
lerp[1] = indexfrac * (1.0f / 65536.0f);
lerp[0] = 1.0f - lerp[1];
sample[0] = fetchsampleframe[0] * lerp[0] + fetchsampleframe[2] * lerp[1];
sample[1] = fetchsampleframe[1] * lerp[0] + fetchsampleframe[3] * lerp[1];
paint->sample[0] += sample[0] * vol[0];
paint->sample[1] += sample[1] * vol[1];
indexfrac += indexfracstep;
fetchsampleframe += 2 * (indexfrac >> 16);
indexfrac &= 0xFFFF;
}
}
}
else if (sfx->format.channels == 1)
{
// most sounds are mono
#if SND_LISTENERS != 8
#error the following code only supports up to 8 channels, update it
#endif
if (snd_speakerlayout.channels > 2)
{
// surround mixing
for (i = 0;i < count;i++, paint++)
{
lerp[1] = indexfrac * (1.0f / 65536.0f);
lerp[0] = 1.0f - lerp[1];
sample[0] = fetchsampleframe[0] * lerp[0] + fetchsampleframe[1] * lerp[1];
paint->sample[0] += sample[0] * vol[0];
paint->sample[1] += sample[0] * vol[1];
paint->sample[2] += sample[0] * vol[2];
paint->sample[3] += sample[0] * vol[3];
paint->sample[4] += sample[0] * vol[4];
paint->sample[5] += sample[0] * vol[5];
paint->sample[6] += sample[0] * vol[6];
paint->sample[7] += sample[0] * vol[7];
indexfrac += indexfracstep;
fetchsampleframe += (indexfrac >> 16);
indexfrac &= 0xFFFF;
}
}
else
{
// stereo mixing
for (i = 0;i < count;i++, paint++)
{
lerp[1] = indexfrac * (1.0f / 65536.0f);
lerp[0] = 1.0f - lerp[1];
sample[0] = fetchsampleframe[0] * lerp[0] + fetchsampleframe[1] * lerp[1];
paint->sample[0] += sample[0] * vol[0];
paint->sample[1] += sample[0] * vol[1];
indexfrac += indexfracstep;
fetchsampleframe += (indexfrac >> 16);
indexfrac &= 0xFFFF;
}
}
}
}
}
ch->position = posd;
if (!looping && istartframe == totallength)
S_StopChannel(ch - channels, false, false);
}
S_SoftClipPaintBuffer(paintbuffer, totalmixframes, snd_renderbuffer->format.width, snd_renderbuffer->format.channels);
if (!snd_usethreadedmixing)
S_CaptureAVISound(paintbuffer, totalmixframes);
S_ConvertPaintBuffer(paintbuffer, outbytes, totalmixframes, snd_renderbuffer->format.width, snd_renderbuffer->format.channels);
// advance the output pointer
outbytes += totalmixframes * snd_renderbuffer->format.width * snd_renderbuffer->format.channels;
bufferframes -= totalmixframes;
}
}
Reference in a new issue View git blame Copy permalink