This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
-MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
See the GNU General Public License for more details.
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
-// snd_mem.c: sound caching
+
#include "quakedef.h"
-int cache_full_cycle;
+#include "snd_ogg.h"
+#include "snd_wav.h"
-byte *S_Alloc (int size);
/*
================
ResampleSfx
================
*/
-void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, byte *data)
+size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t* in_format, qbyte *out_data, const char* sfxname)
{
- int outcount;
- int srcsample;
- float stepscale;
- int i;
- int sample, samplefrac, fracstep;
- sfxcache_t *sc;
-
- sc = Cache_Check (&sfx->cache);
- if (!sc)
- return;
-
- stepscale = (float)inrate / shm->speed; // this is usually 0.5, 1, or 2
-
- outcount = sc->length / stepscale;
- sc->length = outcount;
- if (sc->loopstart != -1)
- sc->loopstart = sc->loopstart / stepscale;
-
- sc->speed = shm->speed;
- if (loadas8bit.value)
- sc->width = 1;
- else
- sc->width = inwidth;
-// sc->stereo = 0;
+ size_t srclength, outcount, i;
+
+ srclength = in_length * in_format->channels;
+ outcount = (double)in_length * shm->format.speed / in_format->speed;
-// resample / decimate to the current source rate
+ 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);
- if (stepscale == 1 && inwidth == 1 && sc->width == 1)
+ // Trivial case (direct transfer)
+ if (in_format->speed == shm->format.speed)
{
-// fast special case
- // LordHavoc: I do not serve the readability gods...
- int *indata, *outdata;
- int count4, count1;
- count1 = outcount << sc->stereo;
- count4 = count1 >> 2;
- indata = (void *)data;
- outdata = (void *)sc->data;
- while (count4--)
- *outdata++ = *indata++ ^ 0x80808080;
- if (count1 & 2)
- ((short*)outdata)[0] = ((short*)indata)[0] ^ 0x8080;
- if (count1 & 1)
- ((char*)outdata)[2] = ((char*)indata)[2] ^ 0x80;
- /*
- if (sc->stereo) // LordHavoc: stereo sound support
+ if (in_format->width == 1)
{
- for (i=0 ; i<(outcount<<1) ; i++)
- ((signed char *)sc->data)[i] = (int)( (unsigned char)(data[i]) - 128);
+ for (i = 0; i < srclength; i++)
+ ((signed char*)out_data)[i] = in_data[i] - 128;
}
- else
- {
- for (i=0 ; i<outcount ; i++)
- ((signed char *)sc->data)[i] = (int)( (unsigned char)(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
- Con_DPrintf("ResampleSfx: resampling sound %s\n", sfx->name);
- samplefrac = 0;
- fracstep = stepscale*256;
- if (sc->stereo) // LordHavoc: stereo sound support
- {
- for (i=0 ; i<outcount ; i+=2)
- {
- srcsample = samplefrac >> 8;
- samplefrac += fracstep;
- srcsample <<= 1;
- // left
- if (inwidth == 2)
- sample = LittleShort ( ((short *)data)[srcsample] );
- else
- sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8;
- if (sc->width == 2)
- ((short *)sc->data)[i] = sample;
- else
- ((signed char *)sc->data)[i] = sample >> 8;
- // right
- srcsample++;
- if (inwidth == 2)
- sample = LittleShort ( ((short *)data)[srcsample] );
- else
- sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8;
- if (sc->width == 2)
- ((short *)sc->data)[i+1] = sample;
- else
- ((signed char *)sc->data)[i+1] = sample >> 8;
- }
- }
- else
+ 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)
{
- for (i=0 ; i<outcount ; i++)
+ 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++)
{
- srcsample = samplefrac >> 8;
+ 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 (j = 0; j < in_format->channels; j++, srcsample++)
+ {
+ // 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];
+
+ out_ptr += sizeof (short);
+ }
+ }
+ // 8 bit samples
+ else // if (in_format->width == 1)
+ {
+ for (j = 0; j < in_format->channels; j++, srcsample++)
+ {
+ // 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;
+
+ out_ptr += sizeof (signed char);
+ }
+ }
+
samplefrac += fracstep;
- if (inwidth == 2)
- sample = LittleShort ( ((short *)data)[srcsample] );
- else
- sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8;
- if (sc->width == 2)
- ((short *)sc->data)[i] = sample;
- else
- ((signed char *)sc->data)[i] = sample >> 8;
}
+
+ // 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;
}
}
+
+ return outcount;
}
//=============================================================================
S_LoadSound
==============
*/
-sfxcache_t *S_LoadSound (sfx_t *s)
+qboolean S_LoadSound (sfx_t *s, qboolean complain)
{
- char namebuffer[256];
- byte *data;
- wavinfo_t info;
- int len;
- float stepscale;
- sfxcache_t *sc;
- byte stackbuf[1*1024]; // avoid dirtying the cache heap
-
-// see if still in memory
- sc = Cache_Check (&s->cache);
- if (sc)
- return sc;
-
-//Con_Printf ("S_LoadSound: %x\n", (int)stackbuf);
-// load it in
- strcpy(namebuffer, "sound/");
- strcat(namebuffer, s->name);
-
-// Con_Printf ("loading %s\n",namebuffer);
-
- data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf), false);
-
- if (!data)
+ char namebuffer[MAX_QPATH];
+ size_t len;
+ qboolean modified_name = false;
+
+ // see if still in memory
+ if (!shm || !shm->format.speed)
+ return false;
+ if (s->fetcher != NULL)
{
- Con_Printf ("Couldn't load %s\n", namebuffer);
- return NULL;
+ if (s->format.speed != shm->format.speed)
+ Sys_Error ("S_LoadSound: sound %s hasn't been resampled (%uHz instead of %uHz)", s->name);
+ return true;
}
- info = GetWavinfo (s->name, data, com_filesize);
- // LordHavoc: stereo sounds are now allowed (intended for music)
- if (info.channels < 1 || info.channels > 2)
- {
- Con_Printf ("%s has an unsupported number of channels (%i)\n",s->name, info.channels);
- return NULL;
- }
- /*
- if (info.channels != 1)
- {
- Con_Printf ("%s is a stereo sample\n",s->name);
- return NULL;
- }
- */
-
- stepscale = (float)info.rate / shm->speed;
- len = info.samples / stepscale;
-
- len = len * info.width * info.channels;
-
- sc = Cache_Alloc ( &s->cache, len + sizeof(sfxcache_t), s->name);
- if (!sc)
- return NULL;
-
- sc->length = info.samples;
- sc->loopstart = info.loopstart;
- sc->speed = info.rate;
- sc->width = info.width;
- sc->stereo = info.channels == 2;
-
- ResampleSfx (s, sc->speed, sc->width, data + info.dataofs);
-
- return sc;
-}
-
+ len = strlcpy (namebuffer, s->name, sizeof (namebuffer));
+ if (len >= sizeof (namebuffer))
+ return false;
+ // Try to load it as a WAV file
+ if (S_LoadWavFile (namebuffer, s))
+ return true;
-/*
-===============================================================================
-
-WAV loading
-
-===============================================================================
-*/
-
-
-byte *data_p;
-byte *iff_end;
-byte *last_chunk;
-byte *iff_data;
-int iff_chunk_len;
-
-
-short GetLittleShort(void)
-{
- short val = 0;
- val = *data_p;
- val = val + (*(data_p+1)<<8);
- data_p += 2;
- return val;
-}
-
-int GetLittleLong(void)
-{
- int val = 0;
- val = *data_p;
- val = val + (*(data_p+1)<<8);
- val = val + (*(data_p+2)<<16);
- val = val + (*(data_p+3)<<24);
- data_p += 4;
- return val;
-}
-
-void FindNextChunk(char *name)
-{
- while (1)
+ // Else, try to load it as an Ogg Vorbis file
+ if (!strcasecmp (namebuffer + len - 4, ".wav"))
{
- data_p=last_chunk;
-
- if (data_p >= iff_end)
- { // didn't find the chunk
- data_p = NULL;
- return;
- }
-
- data_p += 4;
- iff_chunk_len = GetLittleLong();
- if (iff_chunk_len < 0)
- {
- data_p = NULL;
- return;
- }
-// if (iff_chunk_len > 1024*1024)
-// Sys_Error ("FindNextChunk: %i length is past the 1 meg sanity limit", iff_chunk_len);
- data_p -= 8;
- last_chunk = data_p + 8 + ( (iff_chunk_len + 1) & ~1 );
- if (!strncmp(data_p, name, 4))
- return;
+ strcpy (namebuffer + len - 3, "ogg");
+ modified_name = true;
}
-}
+ if (OGG_LoadVorbisFile (namebuffer, s))
+ return true;
-void FindChunk(char *name)
-{
- last_chunk = iff_data;
- FindNextChunk (name);
-}
-
-
-void DumpChunks(void)
-{
- char str[5];
-
- str[4] = 0;
- data_p=iff_data;
- do
+ // Can't load the sound!
+ if (!complain)
+ s->flags |= SFXFLAG_SILENTLYMISSING;
+ else
+ s->flags &= ~SFXFLAG_SILENTLYMISSING;
+ if (complain)
{
- memcpy (str, data_p, 4);
- data_p += 4;
- iff_chunk_len = GetLittleLong();
- Con_Printf ("0x%x : %s (%d)\n", (int)(data_p - 4), str, iff_chunk_len);
- data_p += (iff_chunk_len + 1) & ~1;
- } while (data_p < iff_end);
+ if (modified_name)
+ strcpy (namebuffer + len - 3, "wav");
+ Con_Printf("Couldn't load %s\n", namebuffer);
+ }
+ return false;
}
-/*
-============
-GetWavinfo
-============
-*/
-wavinfo_t GetWavinfo (char *name, byte *wav, int wavlength)
+void S_UnloadSound(sfx_t *s)
{
- wavinfo_t info;
- int i;
- int format;
- int samples;
-
- memset (&info, 0, sizeof(info));
-
- if (!wav)
- return info;
-
- iff_data = wav;
- iff_end = wav + wavlength;
-
-// find "RIFF" chunk
- FindChunk("RIFF");
- if (!(data_p && !strncmp(data_p+8, "WAVE", 4)))
- {
- Con_Printf("Missing RIFF/WAVE chunks\n");
- return info;
- }
-
-// get "fmt " chunk
- iff_data = data_p + 12;
-// DumpChunks ();
-
- FindChunk("fmt ");
- if (!data_p)
- {
- Con_Printf("Missing fmt chunk\n");
- return info;
- }
- data_p += 8;
- format = GetLittleShort();
- if (format != 1)
+ if (s->fetcher != NULL)
{
- Con_Printf("Microsoft PCM format only\n");
- return info;
- }
+ unsigned int i;
- info.channels = GetLittleShort();
- info.rate = GetLittleLong();
- data_p += 4+2;
- info.width = GetLittleShort() / 8;
+ s->fetcher = NULL;
+ s->fetcher_data = NULL;
+ Mem_FreePool(&s->mempool);
-// get cue chunk
- FindChunk("cue ");
- if (data_p)
- {
- data_p += 32;
- info.loopstart = GetLittleLong();
-// Con_Printf("loopstart=%d\n", sfx->loopstart);
-
- // if the next chunk is a LIST chunk, look for a cue length marker
- FindNextChunk ("LIST");
- if (data_p)
- {
- if (!strncmp (data_p + 28, "mark", 4))
- { // this is not a proper parse, but it works with cooledit...
- data_p += 24;
- i = GetLittleLong (); // samples in loop
- info.samples = info.loopstart + i;
-// Con_Printf("looped length: %i\n", i);
- }
- }
+ // At this point, some per-channel data pointers may point to freed zones.
+ // Practically, it shouldn't be a problem; but it's wrong, so we fix that
+ for (i = 0; i < total_channels ; i++)
+ if (channels[i].sfx == s)
+ channels[i].fetcher_data = NULL;
}
- else
- info.loopstart = -1;
-
-// find data chunk
- FindChunk("data");
- if (!data_p)
- {
- Con_Printf("Missing data chunk\n");
- return info;
- }
-
- data_p += 4;
- samples = GetLittleLong () / info.width;
-
- if (info.samples)
- {
- if (samples < info.samples)
- Host_Error ("Sound %s has a bad loop length", name);
- }
- else
- info.samples = samples;
-
- info.dataofs = data_p - wav;
-
- return info;
}
-
projects / xonotic / darkplaces.git / blobdiff