ResampleSfx
================
*/
-size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t* in_format, qbyte *out_data, const char* sfxname)
+size_t ResampleSfx (const unsigned char *in_data, size_t in_length, const snd_format_t* in_format, unsigned char *out_data, const char* sfxname)
{
- size_t srclength, outcount, i;
+ size_t srclength, outcount;
srclength = in_length * in_format->channels;
- outcount = (double)in_length * shm->format.speed / in_format->speed;
+ outcount = (int)((double)in_length * shm->format.speed / in_format->speed);
- Con_DPrintf("ResampleSfx(%s): %d samples @ %dHz -> %d samples @ %dHz\n",
- sfxname, in_length, in_format->speed, outcount, shm->format.speed);
+ //Con_DPrintf("ResampleSfx(%s): %d samples @ %dHz -> %d samples @ %dHz\n",
+ // sfxname, in_length, in_format->speed, outcount, shm->format.speed);
// Trivial case (direct transfer)
if (in_format->speed == shm->format.speed)
{
if (in_format->width == 1)
{
+ size_t i;
+
for (i = 0; i < srclength; i++)
((signed char*)out_data)[i] = in_data[i] - 128;
}
// 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)
+# define FRACTIONAL_BITS 14
+# define FRACTIONAL_MASK ((1 << FRACTIONAL_BITS) - 1)
+# define INTEGER_BITS (sizeof(samplefrac)*8 - FRACTIONAL_BITS)
else
{
- const unsigned int fracstep = (double)in_format->speed / shm->format.speed * (1 << FRACTIONAL_BITS);
+ const unsigned int fracstep = (unsigned int)((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;
+ const unsigned char *in_ptr = in_data;
+ unsigned char *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))",
+ {
+ Con_Printf ("ResampleSfx: sound quality too high for resampling (%uHz, %u channel(s))\n",
in_format->speed, in_format->channels);
+ return 0;
+ }
// 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)
{
- size_t tmpcount;
+ size_t tmpcount, interpolation_limit, i, j;
+ unsigned int srcsample;
samplefrac = 0;
// If more than 1 sec of sound remains to be converted
if (outcount - total_out > shm->format.speed)
+ {
tmpcount = shm->format.speed;
+ interpolation_limit = tmpcount; // all samples can be interpolated
+ }
else
+ {
tmpcount = outcount - total_out;
+ interpolation_limit = (int)ceil((double)(((remain_in / in_format->channels) - 1) << FRACTIONAL_BITS) / fracstep);
+ if (interpolation_limit > tmpcount)
+ interpolation_limit = tmpcount;
+ }
- // Convert up to 1 sec of sound
- for (i = 0; i < tmpcount; i++)
+ // 16 bit samples
+ if (in_format->width == 2)
{
- unsigned int j = 0;
- unsigned int srcsample = (samplefrac >> FRACTIONAL_BITS) * in_format->channels;
- int a, b;
+ const short* in_ptr_short;
- // 16 bit samples
- if (in_format->width == 2)
+ // Interpolated part
+ for (i = 0; i < interpolation_limit; i++)
{
- for (j = 0; j < in_format->channels; j++, srcsample++)
+ srcsample = (samplefrac >> FRACTIONAL_BITS) * in_format->channels;
+ in_ptr_short = &((const short*)in_ptr)[srcsample];
+
+ for (j = 0; j < in_format->channels; j++)
{
- // 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
- *((short*)out_ptr) = ((const short*)in_ptr)[srcsample];
+ int a, b;
+
+ a = *in_ptr_short;
+ b = *(in_ptr_short + in_format->channels);
+ *((short*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a;
+ in_ptr_short++;
out_ptr += sizeof (short);
}
+
+ samplefrac += fracstep;
}
- // 8 bit samples
- else // if (in_format->width == 1)
+
+ // Non-interpolated part
+ for (/* nothing */; i < tmpcount; i++)
{
- for (j = 0; j < in_format->channels; j++, srcsample++)
+ srcsample = (samplefrac >> FRACTIONAL_BITS) * in_format->channels;
+ in_ptr_short = &((const short*)in_ptr)[srcsample];
+
+ for (j = 0; j < in_format->channels; j++)
{
- // 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
- *((signed char*)out_ptr) = ((const qbyte*)in_ptr)[srcsample] - 128;
+ *((short*)out_ptr) = *in_ptr_short;
+ in_ptr_short++;
+ out_ptr += sizeof (short);
+ }
+
+ samplefrac += fracstep;
+ }
+ }
+ // 8 bit samples
+ else // if (in_format->width == 1)
+ {
+ const unsigned char* in_ptr_byte;
+
+ // Convert up to 1 sec of sound
+ for (i = 0; i < interpolation_limit; i++)
+ {
+ srcsample = (samplefrac >> FRACTIONAL_BITS) * in_format->channels;
+ in_ptr_byte = &((const unsigned char*)in_ptr)[srcsample];
+
+ for (j = 0; j < in_format->channels; j++)
+ {
+ int a, b;
+
+ a = *in_ptr_byte - 128;
+ b = *(in_ptr_byte + in_format->channels) - 128;
+ *((signed char*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a;
+
+ in_ptr_byte++;
out_ptr += sizeof (signed char);
}
+
+ samplefrac += fracstep;
}
- samplefrac += fracstep;
+ // Non-interpolated part
+ for (/* nothing */; i < tmpcount; i++)
+ {
+ srcsample = (samplefrac >> FRACTIONAL_BITS) * in_format->channels;
+ in_ptr_byte = &((const unsigned char*)in_ptr)[srcsample];
+
+ for (j = 0; j < in_format->channels; j++)
+ {
+ *((signed char*)out_ptr) = *in_ptr_byte - 128;
+
+ in_ptr_byte++;
+ out_ptr += sizeof (signed char);
+ }
+
+ samplefrac += fracstep;
+ }
}
// Update the counters and the buffer position
*/
qboolean S_LoadSound (sfx_t *s, qboolean complain)
{
- char namebuffer[MAX_QPATH];
+ char namebuffer[MAX_QPATH + 16];
size_t len;
- qboolean modified_name = false;
if (!shm || !shm->format.speed)
return false;
if (s->fetcher != NULL)
{
if (s->format.speed != shm->format.speed)
- Sys_Error ("S_LoadSound: sound %s hasn't been resampled (%uHz instead of %uHz)", s->name);
+ Con_Printf ("S_LoadSound: sound %s hasn't been resampled (%uHz instead of %uHz)\n", s->name);
return true;
}
- len = strlcpy (namebuffer, s->name, sizeof (namebuffer));
- if (len >= sizeof (namebuffer))
- return false;
+ // LordHavoc: if the sound filename does not begin with sound/, try adding it
+ if (strncasecmp(s->name, "sound/", 6))
+ {
+ len = dpsnprintf (namebuffer, sizeof(namebuffer), "sound/%s", s->name);
+ if (len < 0)
+ {
+ // name too long
+ Con_Printf("S_LoadSound: name \"%s\" is too long\n", s->name);
+ return false;
+ }
+ if (S_LoadWavFile (namebuffer, s))
+ return true;
+ if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav"))
+ strcpy (namebuffer + len - 3, "ogg");
+ if (OGG_LoadVorbisFile (namebuffer, s))
+ return true;
+ }
- // Try to load it as a WAV file
+ // LordHavoc: then try without the added sound/ as wav and ogg
+ len = dpsnprintf (namebuffer, sizeof(namebuffer), "%s", s->name);
+ if (len < 0)
+ {
+ // name too long
+ Con_Printf("S_LoadSound: name \"%s\" is too long\n", s->name);
+ return false;
+ }
if (S_LoadWavFile (namebuffer, s))
return true;
-
- // Else, try to load it as an Ogg Vorbis file
- if (!strcasecmp (namebuffer + len - 4, ".wav"))
- {
+ if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav"))
strcpy (namebuffer + len - 3, "ogg");
- modified_name = true;
- }
if (OGG_LoadVorbisFile (namebuffer, s))
return true;
// Can't load the sound!
s->flags |= SFXFLAG_FILEMISSING;
if (complain)
- {
- if (modified_name)
- strcpy (namebuffer + len - 3, "wav");
- Con_Printf("Couldn't load %s\n", namebuffer);
- }
+ Con_Printf("S_LoadSound: Couldn't load \"%s\"\n", s->name);
return false;
}
-
-void S_UnloadSound (sfx_t *s)
-{
- if (s->fetcher != NULL)
- {
- unsigned int i;
-
- // Stop all channels that use this sound
- for (i = 0; i < total_channels ; i++)
- if (channels[i].sfx == s)
- S_StopChannel (i);
-
- s->fetcher = NULL;
- s->fetcher_data = NULL;
- Mem_FreePool(&s->mempool);
- }
-}
projects / xonotic / darkplaces.git / blobdiff