Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
-// snd_mem.c: sound caching
+
#include "quakedef.h"
*/
size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t* in_format, qbyte *out_data, const char* sfxname)
{
- int samplefrac, fracstep;
- size_t i, srcsample, srclength, outcount;
-
- // this is usually 0.5 (128), 1 (256), or 2 (512)
- fracstep = ((double) in_format->speed / (double) shm->format.speed) * 256.0;
+ size_t srclength, outcount, i;
srclength = in_length * in_format->channels;
+ outcount = (double)in_length * shm->format.speed / in_format->speed;
- outcount = (double) in_length * (double) shm->format.speed / (double) in_format->speed;
Con_DPrintf("ResampleSfx: resampling sound \"%s\" from %dHz to %dHz (%d samples to %d samples)\n",
sfxname, in_format->speed, shm->format.speed, in_length, outcount);
-// resample / decimate to the current source rate
-
- if (fracstep == 256)
+ // Trivial case (direct transfer)
+ if (in_format->speed == shm->format.speed)
{
- // fast case for direct transfer
- if (in_format->width == 1) // 8bit
- for (i = 0;i < srclength;i++)
- ((signed char *)out_data)[i] = ((unsigned char *)in_data)[i] - 128;
- else //if (sb->width == 2) // 16bit
- for (i = 0;i < srclength;i++)
- ((short *)out_data)[i] = ((short *)in_data)[i];
+ if (in_format->width == 1)
+ {
+ for (i = 0; i < srclength; i++)
+ ((signed char*)out_data)[i] = in_data[i] - 128;
+ }
+ else // if (in_format->width == 2)
+ memcpy (out_data, in_data, srclength * in_format->width);
}
+
+ // General case (linear interpolation with a fixed-point fractional
+ // step, 18-bit integer part and 14-bit fractional part)
+ // Can handle up to 2^18 (262144) samples per second (> 96KHz stereo)
+ #define FRACTIONAL_BITS 14
+ #define FRACTIONAL_MASK ((1 << FRACTIONAL_BITS) - 1)
+ #define INTEGER_BITS (sizeof(samplefrac)*8 - FRACTIONAL_BITS)
else
{
- // general case
- samplefrac = 0;
- if ((fracstep & 255) == 0) // skipping points on perfect multiple
+ const unsigned int fracstep = (double)in_format->speed / shm->format.speed * (1 << FRACTIONAL_BITS);
+ size_t remain_in = srclength, total_out = 0;
+ unsigned int samplefrac;
+ const qbyte *in_ptr = in_data;
+ qbyte *out_ptr = out_data;
+
+ // Check that we can handle one second of that sound
+ if (in_format->speed * in_format->channels > (1 << INTEGER_BITS))
+ Sys_Error ("ResampleSfx: sound quality too high for resampling (%uHz, %u channel(s))",
+ in_format->speed, in_format->channels);
+
+ // We work 1 sec at a time to make sure we don't accumulate any
+ // significant error when adding "fracstep" over several seconds, and
+ // also to be able to handle very long sounds.
+ while (total_out < outcount)
{
- srcsample = 0;
- fracstep >>= 8;
- if (in_format->width == 2)
- {
- short *out = (short*)out_data;
- const short *in = (const short*)in_data;
- if (in_format->channels == 2) // LordHavoc: stereo sound support
- {
- fracstep <<= 1;
- for (i=0 ; i<outcount ; i++)
- {
- *out++ = in[srcsample ];
- *out++ = in[srcsample+1];
- srcsample += fracstep;
- }
- }
- else
- {
- for (i=0 ; i<outcount ; i++)
- {
- *out++ = in[srcsample];
- srcsample += fracstep;
- }
- }
- }
- else
- {
- signed char *out = out_data;
- const unsigned char *in = in_data;
- if (in_format->channels == 2)
- {
- fracstep <<= 1;
- for (i=0 ; i<outcount ; i++)
- {
- *out++ = in[srcsample ] - 128;
- *out++ = in[srcsample+1] - 128;
- srcsample += fracstep;
- }
- }
- else
- {
- for (i=0 ; i<outcount ; i++)
- {
- *out++ = in[srcsample ] - 128;
- srcsample += fracstep;
- }
- }
- }
- }
- else
- {
- int sample;
- int a, b;
- if (in_format->width == 2)
- {
- short *out = (short*)out_data;
- const short *in = (const short*)in_data;
- if (in_format->channels == 2)
- {
- for (i=0 ; i<outcount ; i++)
- {
- srcsample = (samplefrac >> 8) << 1;
- a = in[srcsample ];
- if (srcsample+2 >= srclength)
- b = 0;
- else
- b = in[srcsample+2];
- sample = (((b - a) * (samplefrac & 255)) >> 8) + a;
- *out++ = (short) sample;
- a = in[srcsample+1];
- if (srcsample+2 >= srclength)
- b = 0;
- else
- b = in[srcsample+3];
- sample = (((b - a) * (samplefrac & 255)) >> 8) + a;
- *out++ = (short) sample;
- samplefrac += fracstep;
- }
- }
- else
- {
- for (i=0 ; i<outcount ; i++)
- {
- srcsample = samplefrac >> 8;
- a = in[srcsample ];
- if (srcsample+1 >= srclength)
- b = 0;
- else
- b = in[srcsample+1];
- sample = (((b - a) * (samplefrac & 255)) >> 8) + a;
- *out++ = (short) sample;
- samplefrac += fracstep;
- }
- }
- }
+ size_t tmpcount;
+
+ samplefrac = 0;
+
+ // If more than 1 sec of sound remains to be converted
+ if (outcount - total_out > shm->format.speed)
+ tmpcount = shm->format.speed;
else
+ tmpcount = outcount - total_out;
+
+ // Convert up to 1 sec of sound
+ for (i = 0; i < tmpcount; i++)
{
- signed char *out = out_data;
- const unsigned char *in = in_data;
- if (in_format->channels == 2)
+ unsigned int j = 0;
+ unsigned int srcsample = (samplefrac >> FRACTIONAL_BITS) * in_format->channels;
+ int a, b;
+
+ // 16 bit samples
+ if (in_format->width == 2)
{
- for (i=0 ; i<outcount ; i++)
+ for (j = 0; j < in_format->channels; j++, srcsample++)
{
- srcsample = (samplefrac >> 8) << 1;
- a = (int) in[srcsample ] - 128;
- if (srcsample+2 >= srclength)
- b = 0;
- else
- b = (int) in[srcsample+2] - 128;
- sample = (((b - a) * (samplefrac & 255)) >> 8) + a;
- *out++ = (signed char) sample;
- a = (int) in[srcsample+1] - 128;
- if (srcsample+2 >= srclength)
- b = 0;
+ // No value to interpolate with?
+ if (srcsample + in_format->channels < remain_in)
+ {
+ a = ((const short*)in_ptr)[srcsample];
+ b = ((const short*)in_ptr)[srcsample + in_format->channels];
+ *((short*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a;
+ }
else
- b = (int) in[srcsample+3] - 128;
- sample = (((b - a) * (samplefrac & 255)) >> 8) + a;
- *out++ = (signed char) sample;
- samplefrac += fracstep;
+ *((short*)out_ptr) = ((const short*)in_ptr)[srcsample];
+
+ out_ptr += sizeof (short);
}
}
- else
+ // 8 bit samples
+ else // if (in_format->width == 1)
{
- for (i=0 ; i<outcount ; i++)
+ for (j = 0; j < in_format->channels; j++, srcsample++)
{
- srcsample = samplefrac >> 8;
- a = (int) in[srcsample ] - 128;
- if (srcsample+1 >= srclength)
- b = 0;
+ // No more value to interpolate with?
+ if (srcsample + in_format->channels < remain_in)
+ {
+ a = ((const qbyte*)in_ptr)[srcsample] - 128;
+ b = ((const qbyte*)in_ptr)[srcsample + in_format->channels] - 128;
+ *((signed char*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a;
+ }
else
- b = (int) in[srcsample+1] - 128;
- sample = (((b - a) * (samplefrac & 255)) >> 8) + a;
- *out++ = (signed char) sample;
- samplefrac += fracstep;
+ *((signed char*)out_ptr) = ((const qbyte*)in_ptr)[srcsample] - 128;
+
+ out_ptr += sizeof (signed char);
}
}
+
+ samplefrac += fracstep;
}
+
+ // Update the counters and the buffer position
+ remain_in -= in_format->speed * in_format->channels;
+ in_ptr += in_format->speed * in_format->channels * in_format->width;
+ total_out += tmpcount;
}
}
S_LoadSound
==============
*/
-qboolean S_LoadSound (sfx_t *s, int complain)
+qboolean S_LoadSound (sfx_t *s, qboolean complain)
{
char namebuffer[MAX_QPATH];
size_t len;
return true;
}
- len = snprintf (namebuffer, sizeof (namebuffer), "sound/%s", s->name);
+ len = strlcpy (namebuffer, s->name, sizeof (namebuffer));
if (len >= sizeof (namebuffer))
return false;
projects / xonotic / darkplaces.git / blobdiff