Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
-// snd_mem.c: sound caching
+
#include "quakedef.h"
-qbyte *S_Alloc (int size);
+#include "snd_main.h"
+#include "snd_ogg.h"
+#include "snd_wav.h"
+
/*
================
ResampleSfx
================
*/
-void ResampleSfx (sfxcache_t *sc, qbyte *data, char *name)
+size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t* in_format, qbyte *out_data, const char* sfxname)
{
- int i, outcount, srcsample, srclength, samplefrac, fracstep;
-
- // this is usually 0.5 (128), 1 (256), or 2 (512)
- fracstep = ((double) sc->speed / (double) shm->speed) * 256.0;
-
- srclength = sc->length << sc->stereo;
+ size_t srclength, outcount, i;
- outcount = (double) sc->length * (double) shm->speed / (double) sc->speed;
- sc->length = outcount;
- if (sc->loopstart != -1)
- sc->loopstart = (double) sc->loopstart * (double) shm->speed / (double) sc->speed;
+ srclength = in_length * in_format->channels;
+ outcount = (double)in_length * shm->format.speed / in_format->speed;
- sc->speed = shm->speed;
+ //Con_DPrintf("ResampleSfx(%s): %d samples @ %dHz -> %d samples @ %dHz\n",
+ // sfxname, in_length, in_format->speed, outcount, shm->format.speed);
-// resample / decimate to the current source rate
-
- if (fracstep == 256)
+ // Trivial case (direct transfer)
+ if (in_format->speed == shm->format.speed)
{
- if (sc->width == 1) // 8bit
- for (i = 0;i < srclength;i++)
- ((signed char *)sc->data)[i] = ((unsigned char *)data)[i] - 128;
- else //if (sc->width == 2) // 16bit
- for (i = 0;i < srclength;i++)
- ((short *)sc->data)[i] = LittleShort (((short *)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
- Con_DPrintf("ResampleSfx: resampling sound %s\n", name);
- 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))
{
- srcsample = 0;
- fracstep >>= 8;
- if (sc->width == 2)
- {
- short *out = (void *)sc->data, *in = (void *)data;
- if (sc->stereo) // LordHavoc: stereo sound support
- {
- fracstep <<= 1;
- for (i=0 ; i<outcount ; i++)
- {
- *out++ = LittleShort (in[srcsample ]);
- *out++ = LittleShort (in[srcsample+1]);
- srcsample += fracstep;
- }
- }
- else
- {
- for (i=0 ; i<outcount ; i++)
- {
- *out++ = LittleShort (in[srcsample ]);
- srcsample += fracstep;
- }
- }
- }
- else
- {
- signed char *out = (void *)sc->data;
- unsigned char *in = (void *)data;
- if (sc->stereo) // LordHavoc: stereo sound support
- {
- 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;
- }
- }
- }
+ Con_Printf ("ResampleSfx: sound quality too high for resampling (%uHz, %u channel(s))",
+ in_format->speed, in_format->channels);
+ return 0;
}
- else
+
+ // 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)
{
- int sample;
- int a, b;
- if (sc->width == 2)
- {
- short *out = (void *)sc->data, *in = (void *)data;
- if (sc->stereo) // LordHavoc: stereo sound support
- {
- 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 = (void *)sc->data;
- unsigned char *in = (void *)data;
- if (sc->stereo) // LordHavoc: stereo sound support
+ 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;
}
}
- // LordHavoc: use this for testing if it ever becomes useful again
-// COM_WriteFile (va("sound/%s.pcm", name), sc->data, (sc->length << sc->stereo) * sc->width);
+ return outcount;
}
//=============================================================================
S_LoadSound
==============
*/
-sfxcache_t *S_LoadSound (sfx_t *s, int complain)
-{
- char namebuffer[256];
- qbyte *data;
- wavinfo_t info;
- int len;
- sfxcache_t *sc;
-
-// see if still in memory
- if (s->sfxcache)
- return s->sfxcache;
-
-// load it in
- strcpy(namebuffer, "sound/");
- strcat(namebuffer, s->name);
-
- data = COM_LoadFile(namebuffer, false);
-
- if (!data)
- {
- if (complain)
- Con_Printf ("Couldn't load %s\n", namebuffer);
- return NULL;
- }
-
- 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);
- Mem_Free(data);
- return NULL;
- }
-
- // calculate resampled length
- len = (int) ((double) info.samples * (double) shm->speed / (double) info.rate);
- len = len * info.width * info.channels;
-
- // FIXME: add S_UnloadSounds or something?
- Mem_FreePool(&s->mempool);
- s->mempool = Mem_AllocPool(s->name);
- sc = s->sfxcache = Mem_Alloc(s->mempool, len + sizeof(sfxcache_t));
- if (!sc)
- {
- Mem_FreePool(&s->mempool);
- Mem_Free(data);
- return NULL;
- }
-
- sc->length = info.samples;
- sc->loopstart = info.loopstart;
- sc->speed = info.rate;
- sc->width = info.width;
- sc->stereo = info.channels == 2;
-
- ResampleSfx (sc, data + info.dataofs, s->name);
-
- Mem_Free(data);
- return sc;
-}
-
-
-
-/*
-===============================================================================
-
-WAV loading
-
-===============================================================================
-*/
-
-
-qbyte *data_p;
-qbyte *iff_end;
-qbyte *last_chunk;
-qbyte *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)
+qboolean S_LoadSound (sfx_t *s, qboolean complain)
{
- while (1)
- {
- data_p=last_chunk;
+ char namebuffer[MAX_QPATH + 16];
+ size_t len;
- if (data_p >= iff_end)
- { // didn't find the chunk
- data_p = NULL;
- return;
- }
+ if (!shm || !shm->format.speed)
+ return false;
- data_p += 4;
- iff_chunk_len = GetLittleLong();
- if (iff_chunk_len < 0)
- {
- data_p = NULL;
- return;
- }
- data_p -= 8;
- last_chunk = data_p + 8 + ( (iff_chunk_len + 1) & ~1 );
- if (!strncmp(data_p, name, 4))
- return;
- }
-}
+ // If we weren't able to load it previously, no need to retry
+ if (s->flags & SFXFLAG_FILEMISSING)
+ return false;
-void FindChunk(char *name)
-{
- last_chunk = iff_data;
- FindNextChunk (name);
-}
-
-
-void DumpChunks(void)
-{
- char str[5];
-
- str[4] = 0;
- data_p=iff_data;
- do
- {
- 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);
-}
-
-/*
-============
-GetWavinfo
-============
-*/
-wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength)
-{
- 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)))
+ // See if in memory
+ if (s->fetcher != NULL)
{
- Con_Printf("Missing RIFF/WAVE chunks\n");
- return info;
+ if (s->format.speed != shm->format.speed)
+ Con_Printf ("S_LoadSound: sound %s hasn't been resampled (%uHz instead of %uHz)", s->name);
+ return true;
}
-// 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)
+ // LordHavoc: if the sound filename does not begin with sound/, try adding it
+ if (strncasecmp(s->name, "sound/", 6))
{
- Con_Printf("Microsoft PCM format only\n");
- return info;
- }
-
- info.channels = GetLittleShort();
- info.rate = GetLittleLong();
- data_p += 4+2;
- info.width = GetLittleShort() / 8;
-
-// get cue chunk
- FindChunk("cue ");
- if (data_p)
- {
- data_p += 32;
- info.loopstart = GetLittleLong();
-
- // if the next chunk is a LIST chunk, look for a cue length marker
- FindNextChunk ("LIST");
- if (data_p)
+ len = dpsnprintf (namebuffer, sizeof(namebuffer), "sound/%s", s->name);
+ if (len < 0)
{
- 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;
- }
+ // 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;
}
- else
- info.loopstart = -1;
-// find data chunk
- FindChunk("data");
- if (!data_p)
+ // LordHavoc: then try without the added sound/ as wav and ogg
+ len = dpsnprintf (namebuffer, sizeof(namebuffer), "%s", s->name);
+ if (len < 0)
{
- Con_Printf("Missing data chunk\n");
- return info;
+ // name too long
+ Con_Printf("S_LoadSound: name \"%s\" is too long\n", s->name);
+ return false;
}
-
- data_p += 4;
- samples = GetLittleLong () / info.width / info.channels;
-
- 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;
+ 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;
+
+ // Can't load the sound!
+ s->flags |= SFXFLAG_FILEMISSING;
+ if (complain)
+ Con_Printf("S_LoadSound: Couldn't load \"%s\"\n", s->name);
+ return false;
}
-
projects / xonotic / darkplaces.git / blobdiff