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clean up mixer code, should solve some problems with popping noises, AVI sound recording needs testing

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git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@2258 fc73d0e0-1445-4013-8a0c-d673dee63da5
This commit is contained in:
TimeServ 2006-05-09 07:26:14 +00:00
parent ca601f236b
commit f38e614106
4 changed files with 47 additions and 277 deletions
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3
engine/client/m_mp3.c
View file

@ -1578,6 +1578,9 @@ void Media_RecordAudioFrame (short *sample_buffer, int samples)
if (capturetype != CT_AVI)
return;
if (samples <= 0)
return;
if (!recordavi_uncompressed_audio_stream)
return;

1
engine/client/snd_dma.c
View file

@ -85,6 +85,7 @@ cvar_t snd_linearresample = SCVAR("snd_linearresample", "1");
cvar_t snd_usemultipledevices = SCVAR("snd_multipledevices", "0");
extern vfsfile_t *rawwritefile;
// ====================================================================
// User-setable variables

318
engine/client/snd_mix.c
View file

@ -26,300 +26,50 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
#endif
#define PAINTBUFFER_SIZE 2048
portable_samplegroup_t paintbuffer[PAINTBUFFER_SIZE];
int *snd_p, snd_vol;
short *snd_out;
void Snd_WriteLinearBlastStereo16 (soundcardinfo_t *sc);
#if defined(NOSOUNDASM) || !id386
void Snd_WriteLinearBlastStereo16 (soundcardinfo_t *sc)
static int paintskip[6][6] =
{
int i, i2;
int val;
{6},
{1, 5},
{1, 1, 4},
{1, 1, 1, 3},
{1, 1, 1, 1, 2},
{1, 1, 1, 1, 1, 1}
};
for (i=0, i2=0; i<sc->snd_linear_count ; i+=2, i2+=6)
{
val = (snd_p[i2]*snd_vol)>>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[i2+1]*snd_vol)>>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;
}
}
#endif
void S_TransferStereo16 (soundcardinfo_t *sc, int endtime)
static int chnskip[6][6] =
{
int lpos;
int lpaintedtime;
short *pbuf;
snd_vol = volume.value*256;
snd_p = (int *) paintbuffer;
lpaintedtime = sc->paintedtime;
pbuf = sc->Lock(sc);
if (!pbuf)
return;
while (lpaintedtime < endtime)
{
// handle recirculating buffer issues
lpos = lpaintedtime % ((sc->sn.samples>>1));
snd_out = (short *) pbuf + (lpos<<1);
sc->snd_linear_count = (sc->sn.samples>>1) - lpos;
if (lpaintedtime + sc->snd_linear_count > endtime)
sc->snd_linear_count = endtime - lpaintedtime;
sc->snd_linear_count <<= 1;
// write a linear blast of samples
Snd_WriteLinearBlastStereo16 (sc);
if (sc == sndcardinfo) //only do this for one sound card.
Media_RecordAudioFrame(snd_out, sc->snd_linear_count);
snd_p += sc->snd_linear_count;
lpaintedtime += (sc->snd_linear_count>>1);
}
sc->Unlock(sc, pbuf);
}
void Snd_WriteLinearBlastStereo16_4Speaker (soundcardinfo_t *sc)
{
int i, i2;
int val;
for (i=0, i2=0; i<sc->snd_linear_count ; i+=4, i2+=6)
{
val = (snd_p[i2]*snd_vol)>>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[i2+1]*snd_vol)>>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;
val = (snd_p[i2+2]*snd_vol)>>8;
if (val > 0x7fff)
snd_out[i+2] = 0x7fff;
else if (val < (short)0x8000)
snd_out[i+2] = (short)0x8000;
else
snd_out[i+2] = val;
val = (snd_p[i2+3]*snd_vol)>>8;
if (val > 0x7fff)
snd_out[i+3] = 0x7fff;
else if (val < (short)0x8000)
snd_out[i+3] = (short)0x8000;
else
snd_out[i+3] = val;
// snd_out[i+0] = rand();
// snd_out[i+1] = rand();
// snd_out[i+2] = rand();
// snd_out[i+3] = rand();
}
}
void S_Transfer4Speaker16 (soundcardinfo_t *sc, int endtime)
{
int lpos;
int lpaintedtime;
short *pbuf;
snd_vol = volume.value*256;
snd_p = (int *) paintbuffer;
lpaintedtime = sc->paintedtime;
pbuf = sc->Lock(sc);
if (!pbuf)
return;
while (lpaintedtime < endtime)
{
// handle recirculating buffer issues
lpos = lpaintedtime % ((sc->sn.samples>>2));
snd_out = (short *) pbuf + (lpos<<2);
sc->snd_linear_count = (sc->sn.samples>>2) - lpos;
if (lpaintedtime + sc->snd_linear_count > endtime)
sc->snd_linear_count = endtime - lpaintedtime;
sc->snd_linear_count <<= 2;
// write a linear blast of samples
Snd_WriteLinearBlastStereo16_4Speaker (sc);
if (sc == sndcardinfo) //only do this for one sound card.
Media_RecordAudioFrame(snd_out, sc->snd_linear_count);
snd_p += sc->snd_linear_count;
lpaintedtime += (sc->snd_linear_count>>2);
}
sc->Unlock(sc, pbuf);
}
void Snd_WriteLinearBlast6Speaker16 (soundcardinfo_t *sc)
{
int i;
int val;
for (i=0 ; i<sc->snd_linear_count ; i+=6)
{
val = (snd_p[i]*snd_vol)>>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]*snd_vol)>>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;
val = (snd_p[i+2]*snd_vol)>>8;
if (val > 0x7fff)
snd_out[i+2] = 0x7fff;
else if (val < (short)0x8000)
snd_out[i+2] = (short)0x8000;
else
snd_out[i+2] = val;
val = (snd_p[i+3]*snd_vol)>>8;
if (val > 0x7fff)
snd_out[i+3] = 0x7fff;
else if (val < (short)0x8000)
snd_out[i+3] = (short)0x8000;
else
snd_out[i+3] = val;
val = (snd_p[i+4]*snd_vol)>>8;
if (val > 0x7fff)
snd_out[i+4] = 0x7fff;
else if (val < (short)0x8000)
snd_out[i+4] = (short)0x8000;
else
snd_out[i+4] = val;
val = (snd_p[i+5]*snd_vol)>>8;
if (val > 0x7fff)
snd_out[i+5] = 0x7fff;
else if (val < (short)0x8000)
snd_out[i+5] = (short)0x8000;
else
snd_out[i+5] = val;
}
}
void S_Transfer6Speaker16 (soundcardinfo_t *sc, int endtime)
{
int lpos;
int lpaintedtime;
short *pbuf;
snd_vol = volume.value*256;
snd_p = (int *) paintbuffer;
lpaintedtime = sc->paintedtime;
pbuf = sc->Lock(sc);
if (!pbuf)
return;
while (lpaintedtime < endtime)
{
// handle recirculating buffer issues
lpos = (lpaintedtime % ((sc->sn.samples/6)));
snd_out = (short *) pbuf + (lpos*6);
sc->snd_linear_count = (sc->sn.samples/6) - lpos;
if (lpaintedtime + sc->snd_linear_count > endtime)
sc->snd_linear_count = endtime - lpaintedtime;
sc->snd_linear_count *= 6;
// write a linear blast of samples
Snd_WriteLinearBlast6Speaker16 (sc);
if (sc == sndcardinfo) //only do this for one sound card.
Media_RecordAudioFrame(snd_out, sc->snd_linear_count);
snd_p += sc->snd_linear_count;
lpaintedtime += (sc->snd_linear_count/6);
}
sc->Unlock(sc, pbuf);
}
{0},
{1, -1},
{1, 1, -2},
{1, 1, 1, -3},
{1, 1, 1, 1, -4},
{1, 1, 1, 1, 1, -5}
};
void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime)
{
int out_idx;
int startidx, out_idx;
int count;
int out_mask;
int *p;
int step;
int *skip;
int *cskip;
int val;
int snd_vol;
short *pbuf;
if (sc->sn.samplebits == 16 && sc->sn.numchannels == 2)
{
S_TransferStereo16 (sc, endtime);
return;
}
if (sc->sn.samplebits == 16 && sc->sn.numchannels == 6)
{
S_Transfer6Speaker16 (sc, endtime);
return;
}
if (sc->sn.samplebits == 16 && sc->sn.numchannels == 4)
{
S_Transfer4Speaker16 (sc, endtime);
return;
}
p = (int *) paintbuffer;
skip = paintskip[sc->sn.numchannels-1];
cskip = chnskip[sc->sn.numchannels-1];
count = (endtime - sc->paintedtime) * sc->sn.numchannels;
out_mask = sc->sn.samples - 1;
out_idx = (sc->paintedtime * sc->sn.numchannels) & out_mask;
if (sc->sn.numchannels>2)
step = 1;
else
step = 6;
startidx = out_idx = (sc->paintedtime * sc->sn.numchannels) & out_mask;
snd_vol = volume.value*256;
pbuf = sc->Lock(sc);
@ -332,13 +82,27 @@ void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime)
while (count--)
{
val = (*p * snd_vol) >> 8;
p+= step;
p += *skip;
if (val > 0x7fff)
val = 0x7fff;
else if (val < (short)0x8000)
val = (short)0x8000;
out[out_idx] = val;
out_idx = (out_idx + 1) & out_mask;
skip += *cskip;
cskip += *cskip;
}
// Only do this for 1 sound card with 2 channels, because
// this function is hacky
if (sc == sndcardinfo && sc->sn.numchannels == 2)
{
if (out_idx <= startidx) // buffer looped
{
Media_RecordAudioFrame(out + startidx, (sc->sn.samples - startidx) / 2);
Media_RecordAudioFrame(out, out_idx / (2*2));
}
else
Media_RecordAudioFrame(out + startidx, (out_idx - startidx) / 2);
}
}
else if (sc->sn.samplebits == 8)
@ -347,13 +111,15 @@ void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime)
while (count--)
{
val = (*p * snd_vol) >> 8;
p+= step;
p += *skip;
if (val > 0x7fff)
val = 0x7fff;
else if (val < (short)0x8000)
val = (short)0x8000;
out[out_idx] = (val>>8) + 128;
out_idx = (out_idx + 1) & out_mask;
skip += *cskip;
cskip += *cskip;
}
}

2
engine/client/snd_mixa.s
View file

@ -140,7 +140,6 @@ LDone:
popl %esi
ret
#endif
//----------------------------------------------------------------------
// Transfer of stereo buffer to 16-bit DMA buffer code
@ -224,6 +223,7 @@ LClampDone2:
popl %esi
ret
#endif
#endif // id386