Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
-// snd_mix.c -- portable code to mix sounds for snd_dma.c
+// snd_mix.c -- portable code to mix sounds
#include "quakedef.h"
+#include "snd_main.h"
+
+typedef struct
+{
+ int left;
+ int right;
+} portable_samplepair_t;
// LordHavoc: was 512, expanded to 2048
#define PAINTBUFFER_SIZE 2048
portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
-int snd_scaletable[32][256];
-/*
-// LordHavoc: disabled this because it desyncs with the video too easily
-extern cvar_t cl_avidemo;
-static FILE *cl_avidemo_soundfile = NULL;
-void S_CaptureAVISound(portable_samplepair_t *buf, int length)
+// FIXME: this desyncs with the video too easily
+extern qboolean cl_capturevideo_active;
+extern void SCR_CaptureVideo_SoundFrame(qbyte *bufstereo16le, size_t length, int rate);
+void S_CaptureAVISound(portable_samplepair_t *buf, size_t length)
{
- int i, n;
+ int n;
+ size_t i;
qbyte out[PAINTBUFFER_SIZE * 4];
- char filename[MAX_OSPATH];
-
- if (cl_avidemo.value >= 0.1f)
- {
- if (cl_avidemo_soundfile == NULL)
- {
- cl_avidemo_soundfile = FS_Open ("dpavi.wav", "wb", false);
- memset(out, 0, 44);
- fwrite(out, 1, 44, cl_avidemo_soundfile);
- // header will be filled out when file is closed
- }
- fseek(cl_avidemo_soundfile, 0, SEEK_END);
- // write the sound buffer as little endian 16bit interleaved stereo
- for(i = 0;i < length;i++)
- {
- n = buf[i].left >> 2; // quiet enough to prevent clipping most of the time
- n = bound(-32768, n, 32767);
- out[i*4+0] = n & 0xFF;
- out[i*4+1] = (n >> 8) & 0xFF;
- n = buf[i].right >> 2; // quiet enough to prevent clipping most of the time
- n = bound(-32768, n, 32767);
- out[i*4+2] = n & 0xFF;
- out[i*4+3] = (n >> 8) & 0xFF;
- }
- if (fwrite(out, 4, length, cl_avidemo_soundfile) < length)
- {
- Cvar_SetValueQuick(&cl_avidemo, 0);
- Con_Print("avi saving sound failed, out of disk space? stopping avi demo capture.\n");
- }
- }
- else if (cl_avidemo_soundfile)
+ if (!cl_capturevideo_active)
+ return;
+ // write the sound buffer as little endian 16bit interleaved stereo
+ for(i = 0;i < length;i++)
{
- // file has not been closed yet, close it
- fseek(cl_avidemo_soundfile, 0, SEEK_END);
- i = ftell(cl_avidemo_soundfile);
-
- //"RIFF", (int) unknown (chunk size), "WAVE",
- //"fmt ", (int) 16 (chunk size), (short) format 1 (uncompressed PCM), (short) 2 channels, (int) unknown rate, (int) unknown bytes per second, (short) 4 bytes per sample (channels * bytes per channel), (short) 16 bits per channel
- //"data", (int) unknown (chunk size)
- memcpy(out, "RIFF****WAVEfmt \x10\x00\x00\x00\x01\x00\x02\x00********\x04\x00\x10\x00data****", 44);
- // the length of the whole RIFF chunk
- n = i - 8;
- out[4] = (n) & 0xFF;
- out[5] = (n >> 8) & 0xFF;
- out[6] = (n >> 16) & 0xFF;
- out[7] = (n >> 24) & 0xFF;
- // rate
- n = shm->speed;
- out[24] = (n) & 0xFF;
- out[25] = (n >> 8) & 0xFF;
- out[26] = (n >> 16) & 0xFF;
- out[27] = (n >> 24) & 0xFF;
- // bytes per second (rate * channels * bytes per channel)
- n = shm->speed * 4;
- out[28] = (n) & 0xFF;
- out[29] = (n >> 8) & 0xFF;
- out[30] = (n >> 16) & 0xFF;
- out[31] = (n >> 24) & 0xFF;
- // the length of the data chunk
- n = i - 44;
- out[40] = (n) & 0xFF;
- out[41] = (n >> 8) & 0xFF;
- out[42] = (n >> 16) & 0xFF;
- out[43] = (n >> 24) & 0xFF;
-
- fseek(cl_avidemo_soundfile, 0, SEEK_SET);
- fwrite(out, 1, 44, cl_avidemo_soundfile);
- fclose(cl_avidemo_soundfile);
- cl_avidemo_soundfile = NULL;
+ n = buf[i].left >> 2; // quiet enough to prevent clipping most of the time
+ n = bound(-32768, n, 32767);
+ out[i*4+0] = n & 0xFF;
+ out[i*4+1] = (n >> 8) & 0xFF;
+ n = buf[i].right >> 2; // quiet enough to prevent clipping most of the time
+ 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;
int *snd_p;
- int snd_vol;
int lpaintedtime;
int snd_linear_count;
int val;
snd_p = (int *) paintbuffer;
- snd_vol = volume.value*256;
lpaintedtime = paintedtime;
if (shm->format.width == 2)
{
{
for (i = 0;i < snd_linear_count;i += 2)
{
- val = (snd_p[i + 1] * snd_vol) >> 8;
- snd_out[i ] = bound(-32768, val, 32767);
- val = (snd_p[i ] * snd_vol) >> 8;
- snd_out[i + 1] = bound(-32768, val, 32767);
+ snd_out[i ] = bound(-32768, snd_p[i + 1], 32767);
+ snd_out[i + 1] = bound(-32768, snd_p[i ], 32767);
}
}
else
{
for (i = 0;i < snd_linear_count;i += 2)
{
- val = (snd_p[i ] * snd_vol) >> 8;
- snd_out[i ] = bound(-32768, val, 32767);
- val = (snd_p[i + 1] * snd_vol) >> 8;
- snd_out[i + 1] = bound(-32768, val, 32767);
+ snd_out[i ] = bound(-32768, snd_p[i ], 32767);
+ snd_out[i + 1] = bound(-32768, snd_p[i + 1], 32767);
}
}
snd_p += snd_linear_count;
snd_linear_count = endtime - lpaintedtime;
for (i = 0;i < snd_linear_count;i++)
{
- val = ((snd_p[i * 2 + 0] + snd_p[i * 2 + 1]) * snd_vol) >> 9;
+ val = (snd_p[i * 2 + 0] + snd_p[i * 2 + 1]) >> 1;
snd_out[i] = bound(-32768, val, 32767);
}
snd_p += snd_linear_count << 1;
{
for (i = 0;i < snd_linear_count;i += 2)
{
- val = ((snd_p[i + 1] * snd_vol) >> 16) + 128;
+ val = (snd_p[i + 1] >> 8) + 128;
snd_out[i ] = bound(0, val, 255);
- val = ((snd_p[i ] * snd_vol) >> 16) + 128;
+ val = (snd_p[i ] >> 8) + 128;
snd_out[i + 1] = bound(0, val, 255);
}
}
{
for (i = 0;i < snd_linear_count;i += 2)
{
- val = ((snd_p[i ] * snd_vol) >> 16) + 128;
+ val = (snd_p[i ] >> 8) + 128;
snd_out[i ] = bound(0, val, 255);
- val = ((snd_p[i + 1] * snd_vol) >> 16) + 128;
+ val = (snd_p[i + 1] >> 8) + 128;
snd_out[i + 1] = bound(0, val, 255);
}
}
snd_linear_count = endtime - lpaintedtime;
for (i = 0;i < snd_linear_count;i++)
{
- val = (((snd_p[i * 2] + snd_p[i * 2 + 1]) * snd_vol) >> 17) + 128;
+ val = ((snd_p[i * 2] + snd_p[i * 2 + 1]) >> 9) + 128;
snd_out[i ] = bound(0, val, 255);
}
snd_p += snd_linear_count << 1;
if (endtime - paintedtime > PAINTBUFFER_SIZE)
end = paintedtime + PAINTBUFFER_SIZE;
- // clear the paint buffer, filling it with data from rawsamples (music/video/whatever)
- S_RawSamples_Dequeue(&paintbuffer->left, end - paintedtime);
+ // clear the paint buffer
+ memset (&paintbuffer, 0, (end - paintedtime) * sizeof (paintbuffer[0]));
// paint in the channels.
ch = channels;
if (!S_LoadSound (sfx, true))
continue;
- ltime = paintedtime;
+ // if the channel is paused
+ if (ch->flags & CHANNELFLAG_PAUSED)
+ {
+ size_t pausedtime;
+
+ 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;
if (count > 0)
{
+ if (ch->leftvol > 255)
+ ch->leftvol = 255;
+ if (ch->rightvol > 255)
+ ch->rightvol = 255;
+
if (sfx->format.width == 1)
- stop_paint = !SND_PaintChannelFrom8(ch, count);
+ stop_paint = !SND_PaintChannelFrom8 (ch, count);
else
- stop_paint = !SND_PaintChannelFrom16(ch, count);
+ stop_paint = !SND_PaintChannelFrom16 (ch, count);
- ltime += 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->forceloop) && !stop_paint)
+ 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;
if (stop_paint)
{
- if (ch->sfx->fetcher->end != NULL)
- ch->sfx->fetcher->end (ch);
- ch->sfx = NULL;
+ S_StopChannel (ch - channels);
break;
}
}
}
// transfer out according to DMA format
- //S_CaptureAVISound(paintbuffer, end - paintedtime);
+ S_CaptureAVISound (paintbuffer, end - paintedtime);
S_TransferPaintBuffer(end);
paintedtime = end;
}
}
-void SND_InitScaletable (void)
-{
- int i, j;
-
- for (i = 0;i < 32;i++)
- for (j = 0;j < 256;j++)
- snd_scaletable[i][j] = ((signed char)j) * i * 8;
-}
-
+// TODO: Try to merge SND_PaintChannelFrom8 and SND_PaintChannelFrom16
qboolean SND_PaintChannelFrom8 (channel_t *ch, int count)
{
- int *lscale, *rscale;
- unsigned char *sfx;
+ int snd_vol, leftvol, rightvol;
+ const signed char *sfx;
const sfxbuffer_t *sb;
- int i, n;
+ int i;
- if (ch->leftvol > 255)
- ch->leftvol = 255;
- if (ch->rightvol > 255)
- ch->rightvol = 255;
+ // If this channel manages its own volume
+ if (ch->flags & CHANNELFLAG_FULLVOLUME)
+ snd_vol = 256;
+ else
+ snd_vol = volume.value * 256;
- lscale = snd_scaletable[ch->leftvol >> 3];
- rscale = snd_scaletable[ch->rightvol >> 3];
+ leftvol = ch->leftvol * snd_vol;
+ rightvol = ch->rightvol * snd_vol;
sb = ch->sfx->fetcher->getsb (ch, ch->pos, count);
if (sb == NULL)
return false;
+ // Stereo sound support
if (ch->sfx->format.channels == 2)
{
- // LordHavoc: stereo sound support, and optimizations
- sfx = (unsigned char *)sb->data + (ch->pos - sb->offset) * 2;
+ sfx = sb->data + (ch->pos - sb->offset) * 2;
for (i = 0;i < count;i++)
{
- paintbuffer[i].left += lscale[*sfx++];
- paintbuffer[i].right += rscale[*sfx++];
+ paintbuffer[i].left += (*sfx++ * leftvol) >> 8;
+ paintbuffer[i].right += (*sfx++ * rightvol) >> 8;
}
}
else
{
- sfx = (unsigned char *)sb->data + ch->pos - sb->offset;
+ sfx = sb->data + ch->pos - sb->offset;
for (i = 0;i < count;i++)
{
- n = *sfx++;
- paintbuffer[i].left += lscale[n];
- paintbuffer[i].right += rscale[n];
+ paintbuffer[i].left += (*sfx * leftvol) >> 8;
+ paintbuffer[i].right += (*sfx++ * rightvol) >> 8;
}
}
qboolean SND_PaintChannelFrom16 (channel_t *ch, int count)
{
- int leftvol, rightvol;
+ int snd_vol, leftvol, rightvol;
signed short *sfx;
const sfxbuffer_t *sb;
int i;
- leftvol = ch->leftvol;
- rightvol = ch->rightvol;
+ // If this channel manages its own volume
+ if (ch->flags & CHANNELFLAG_FULLVOLUME)
+ snd_vol = 256;
+ else
+ snd_vol = volume.value * 256;
+
+ leftvol = ch->leftvol * snd_vol;
+ rightvol = ch->rightvol * snd_vol;
sb = ch->sfx->fetcher->getsb (ch, ch->pos, count);
if (sb == NULL)
return false;
+ // Stereo sound support
if (ch->sfx->format.channels == 2)
{
- // LordHavoc: stereo sound support, and optimizations
sfx = (signed short *)sb->data + (ch->pos - sb->offset) * 2;
for (i=0 ; i<count ; i++)
{
- paintbuffer[i].left += (*sfx++ * leftvol) >> 8;
- paintbuffer[i].right += (*sfx++ * rightvol) >> 8;
+ paintbuffer[i].left += (*sfx++ * leftvol) >> 16;
+ paintbuffer[i].right += (*sfx++ * rightvol) >> 16;
}
}
else
for (i=0 ; i<count ; i++)
{
- paintbuffer[i].left += (*sfx * leftvol) >> 8;
- paintbuffer[i].right += (*sfx++ * rightvol) >> 8;
+ paintbuffer[i].left += (*sfx * leftvol) >> 16;
+ paintbuffer[i].right += (*sfx++ * rightvol) >> 16;
}
}