clean up mixer code, should solve some problems with popping noises, AVI sound recording needs testing
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@2258 fc73d0e0-1445-4013-8a0c-d673dee63da5
This commit is contained in:
TimeServ
parent
ca601f236b
commit
f38e614106
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@ -1578,6 +1578,9 @@ void Media_RecordAudioFrame (short *sample_buffer, int samples)
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if (capturetype != CT_AVI)
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return;
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if (samples <= 0)
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return;
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if (!recordavi_uncompressed_audio_stream)
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return;
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@ -85,6 +85,7 @@ cvar_t snd_linearresample = SCVAR("snd_linearresample", "1");
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cvar_t snd_usemultipledevices = SCVAR("snd_multipledevices", "0");
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extern vfsfile_t *rawwritefile;
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// ====================================================================
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// User-setable variables
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@ -26,300 +26,50 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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#endif
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#define PAINTBUFFER_SIZE 2048
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portable_samplegroup_t paintbuffer[PAINTBUFFER_SIZE];
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int *snd_p, snd_vol;
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short *snd_out;
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void Snd_WriteLinearBlastStereo16 (soundcardinfo_t *sc);
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#if defined(NOSOUNDASM) || !id386
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void Snd_WriteLinearBlastStereo16 (soundcardinfo_t *sc)
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static int paintskip[6][6] =
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{
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int i, i2;
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int val;
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{6},
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{1, 5},
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{1, 1, 4},
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{1, 1, 1, 3},
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{1, 1, 1, 1, 2},
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{1, 1, 1, 1, 1, 1}
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};
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for (i=0, i2=0; i<sc->snd_linear_count ; i+=2, i2+=6)
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{
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val = (snd_p[i2]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i] = (short)0x8000;
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else
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snd_out[i] = val;
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val = (snd_p[i2+1]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+1] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+1] = (short)0x8000;
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else
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snd_out[i+1] = val;
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}
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}
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#endif
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void S_TransferStereo16 (soundcardinfo_t *sc, int endtime)
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static int chnskip[6][6] =
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{
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int lpos;
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int lpaintedtime;
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short *pbuf;
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snd_vol = volume.value*256;
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snd_p = (int *) paintbuffer;
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lpaintedtime = sc->paintedtime;
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pbuf = sc->Lock(sc);
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if (!pbuf)
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return;
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while (lpaintedtime < endtime)
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{
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// handle recirculating buffer issues
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lpos = lpaintedtime % ((sc->sn.samples>>1));
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snd_out = (short *) pbuf + (lpos<<1);
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sc->snd_linear_count = (sc->sn.samples>>1) - lpos;
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if (lpaintedtime + sc->snd_linear_count > endtime)
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sc->snd_linear_count = endtime - lpaintedtime;
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sc->snd_linear_count <<= 1;
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// write a linear blast of samples
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Snd_WriteLinearBlastStereo16 (sc);
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if (sc == sndcardinfo) //only do this for one sound card.
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Media_RecordAudioFrame(snd_out, sc->snd_linear_count);
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snd_p += sc->snd_linear_count;
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lpaintedtime += (sc->snd_linear_count>>1);
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}
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sc->Unlock(sc, pbuf);
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}
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void Snd_WriteLinearBlastStereo16_4Speaker (soundcardinfo_t *sc)
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{
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int i, i2;
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int val;
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for (i=0, i2=0; i<sc->snd_linear_count ; i+=4, i2+=6)
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{
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val = (snd_p[i2]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i] = (short)0x8000;
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else
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snd_out[i] = val;
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val = (snd_p[i2+1]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+1] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+1] = (short)0x8000;
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else
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snd_out[i+1] = val;
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val = (snd_p[i2+2]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+2] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+2] = (short)0x8000;
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else
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snd_out[i+2] = val;
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val = (snd_p[i2+3]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+3] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+3] = (short)0x8000;
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else
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snd_out[i+3] = val;
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// snd_out[i+0] = rand();
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// snd_out[i+1] = rand();
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// snd_out[i+2] = rand();
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// snd_out[i+3] = rand();
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}
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}
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void S_Transfer4Speaker16 (soundcardinfo_t *sc, int endtime)
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{
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int lpos;
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int lpaintedtime;
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short *pbuf;
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snd_vol = volume.value*256;
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snd_p = (int *) paintbuffer;
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lpaintedtime = sc->paintedtime;
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pbuf = sc->Lock(sc);
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if (!pbuf)
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return;
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while (lpaintedtime < endtime)
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{
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// handle recirculating buffer issues
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lpos = lpaintedtime % ((sc->sn.samples>>2));
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snd_out = (short *) pbuf + (lpos<<2);
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sc->snd_linear_count = (sc->sn.samples>>2) - lpos;
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if (lpaintedtime + sc->snd_linear_count > endtime)
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sc->snd_linear_count = endtime - lpaintedtime;
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sc->snd_linear_count <<= 2;
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// write a linear blast of samples
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Snd_WriteLinearBlastStereo16_4Speaker (sc);
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if (sc == sndcardinfo) //only do this for one sound card.
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Media_RecordAudioFrame(snd_out, sc->snd_linear_count);
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snd_p += sc->snd_linear_count;
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lpaintedtime += (sc->snd_linear_count>>2);
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}
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sc->Unlock(sc, pbuf);
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}
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void Snd_WriteLinearBlast6Speaker16 (soundcardinfo_t *sc)
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{
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int i;
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int val;
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for (i=0 ; i<sc->snd_linear_count ; i+=6)
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{
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val = (snd_p[i]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i] = (short)0x8000;
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else
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snd_out[i] = val;
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val = (snd_p[i+1]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+1] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+1] = (short)0x8000;
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else
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snd_out[i+1] = val;
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val = (snd_p[i+2]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+2] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+2] = (short)0x8000;
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else
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snd_out[i+2] = val;
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val = (snd_p[i+3]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+3] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+3] = (short)0x8000;
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else
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snd_out[i+3] = val;
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val = (snd_p[i+4]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+4] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+4] = (short)0x8000;
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else
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snd_out[i+4] = val;
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val = (snd_p[i+5]*snd_vol)>>8;
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if (val > 0x7fff)
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snd_out[i+5] = 0x7fff;
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else if (val < (short)0x8000)
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snd_out[i+5] = (short)0x8000;
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else
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snd_out[i+5] = val;
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}
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}
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void S_Transfer6Speaker16 (soundcardinfo_t *sc, int endtime)
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{
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int lpos;
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int lpaintedtime;
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short *pbuf;
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snd_vol = volume.value*256;
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snd_p = (int *) paintbuffer;
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lpaintedtime = sc->paintedtime;
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pbuf = sc->Lock(sc);
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if (!pbuf)
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return;
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while (lpaintedtime < endtime)
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{
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// handle recirculating buffer issues
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lpos = (lpaintedtime % ((sc->sn.samples/6)));
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snd_out = (short *) pbuf + (lpos*6);
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sc->snd_linear_count = (sc->sn.samples/6) - lpos;
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if (lpaintedtime + sc->snd_linear_count > endtime)
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sc->snd_linear_count = endtime - lpaintedtime;
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sc->snd_linear_count *= 6;
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// write a linear blast of samples
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Snd_WriteLinearBlast6Speaker16 (sc);
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if (sc == sndcardinfo) //only do this for one sound card.
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Media_RecordAudioFrame(snd_out, sc->snd_linear_count);
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snd_p += sc->snd_linear_count;
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lpaintedtime += (sc->snd_linear_count/6);
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}
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sc->Unlock(sc, pbuf);
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}
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{0},
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{1, -1},
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{1, 1, -2},
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{1, 1, 1, -3},
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{1, 1, 1, 1, -4},
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{1, 1, 1, 1, 1, -5}
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};
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void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime)
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{
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int out_idx;
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int startidx, out_idx;
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int count;
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int out_mask;
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int *p;
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int step;
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int *skip;
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int *cskip;
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int val;
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int snd_vol;
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short *pbuf;
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if (sc->sn.samplebits == 16 && sc->sn.numchannels == 2)
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{
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S_TransferStereo16 (sc, endtime);
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return;
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}
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if (sc->sn.samplebits == 16 && sc->sn.numchannels == 6)
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{
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S_Transfer6Speaker16 (sc, endtime);
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return;
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}
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if (sc->sn.samplebits == 16 && sc->sn.numchannels == 4)
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{
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S_Transfer4Speaker16 (sc, endtime);
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return;
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}
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p = (int *) paintbuffer;
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skip = paintskip[sc->sn.numchannels-1];
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cskip = chnskip[sc->sn.numchannels-1];
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count = (endtime - sc->paintedtime) * sc->sn.numchannels;
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out_mask = sc->sn.samples - 1;
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out_idx = (sc->paintedtime * sc->sn.numchannels) & out_mask;
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if (sc->sn.numchannels>2)
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step = 1;
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else
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step = 6;
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startidx = out_idx = (sc->paintedtime * sc->sn.numchannels) & out_mask;
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snd_vol = volume.value*256;
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pbuf = sc->Lock(sc);
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@ -332,13 +82,27 @@ void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime)
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while (count--)
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{
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val = (*p * snd_vol) >> 8;
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p+= step;
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p += *skip;
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if (val > 0x7fff)
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val = 0x7fff;
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else if (val < (short)0x8000)
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val = (short)0x8000;
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out[out_idx] = val;
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out_idx = (out_idx + 1) & out_mask;
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skip += *cskip;
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cskip += *cskip;
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}
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// Only do this for 1 sound card with 2 channels, because
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// this function is hacky
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if (sc == sndcardinfo && sc->sn.numchannels == 2)
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{
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if (out_idx <= startidx) // buffer looped
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{
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Media_RecordAudioFrame(out + startidx, (sc->sn.samples - startidx) / 2);
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Media_RecordAudioFrame(out, out_idx / (2*2));
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}
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else
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Media_RecordAudioFrame(out + startidx, (out_idx - startidx) / 2);
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}
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}
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else if (sc->sn.samplebits == 8)
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@ -347,13 +111,15 @@ void S_TransferPaintBuffer(soundcardinfo_t *sc, int endtime)
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while (count--)
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{
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val = (*p * snd_vol) >> 8;
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p+= step;
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p += *skip;
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if (val > 0x7fff)
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val = 0x7fff;
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else if (val < (short)0x8000)
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val = (short)0x8000;
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out[out_idx] = (val>>8) + 128;
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out_idx = (out_idx + 1) & out_mask;
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skip += *cskip;
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cskip += *cskip;
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}
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}
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@ -140,7 +140,6 @@ LDone:
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popl %esi
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ret
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#endif
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//----------------------------------------------------------------------
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// Transfer of stereo buffer to 16-bit DMA buffer code
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@ -224,6 +223,7 @@ LClampDone2:
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popl %esi
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ret
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#endif
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#endif // id386
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