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;
+ int samplefrac, fracstep;
+ size_t i, srcsample, srclength, outcount;
// this is usually 0.5 (128), 1 (256), or 2 (512)
- fracstep = ((double) sc->speed / (double) shm->speed) * 256.0;
+ fracstep = ((double) in_format->speed / (double) shm->format.speed) * 256.0;
- srclength = sc->length << sc->stereo;
+ srclength = in_length * in_format->channels;
- outcount = (double) sc->length * (double) shm->speed / (double) sc->speed;
- Con_DPrintf("ResampleSfx: resampling sound %s from %dhz to %dhz (%d samples to %d samples)\n", name, sc->speed, shm->speed, sc->length, outcount);
- sc->length = outcount;
- if (sc->loopstart != -1)
- sc->loopstart = (double) sc->loopstart * (double) shm->speed / (double) sc->speed;
-
- sc->speed = shm->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)
{
// fast case for direct transfer
- if (sc->width == 1) // 8bit
+ if (in_format->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
+ ((signed char *)out_data)[i] = ((unsigned char *)in_data)[i] - 128;
+ else //if (sb->width == 2) // 16bit
for (i = 0;i < srclength;i++)
- ((short *)sc->data)[i] = ((short *)data)[i];
+ ((short *)out_data)[i] = ((short *)in_data)[i];
}
else
{
{
srcsample = 0;
fracstep >>= 8;
- if (sc->width == 2)
+ if (in_format->width == 2)
{
- short *out = (void *)sc->data, *in = (void *)data;
- if (sc->stereo) // LordHavoc: stereo sound support
+ 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++)
}
else
{
- signed char *out = (void *)sc->data;
- unsigned char *in = (void *)data;
- if (sc->stereo) // LordHavoc: stereo sound support
+ signed char *out = out_data;
+ const unsigned char *in = in_data;
+ if (in_format->channels == 2)
{
fracstep <<= 1;
for (i=0 ; i<outcount ; i++)
{
int sample;
int a, b;
- if (sc->width == 2)
+ if (in_format->width == 2)
{
- short *out = (void *)sc->data, *in = (void *)data;
- if (sc->stereo) // LordHavoc: stereo sound support
+ short *out = (short*)out_data;
+ const short *in = (const short*)in_data;
+ if (in_format->channels == 2)
{
for (i=0 ; i<outcount ; i++)
{
}
else
{
- signed char *out = (void *)sc->data;
- unsigned char *in = (void *)data;
- if (sc->stereo) // LordHavoc: stereo sound support
+ signed char *out = out_data;
+ const unsigned char *in = in_data;
+ if (in_format->channels == 2)
{
for (i=0 ; i<outcount ; i++)
{
}
}
- // 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;
}
//=============================================================================
+wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength);
+
+/*
+====================
+WAV_FetchSound
+====================
+*/
+static const sfxbuffer_t* WAV_FetchSound (channel_t* ch, unsigned int start, unsigned int nbsamples)
+{
+ return ch->sfx->fetcher_data;
+}
+
+
+snd_fetcher_t wav_fetcher = { WAV_FetchSound, NULL };
+
+
/*
==============
S_LoadWavFile
==============
*/
-sfxcache_t *S_LoadWavFile (const char *filename, sfx_t *s)
+qboolean S_LoadWavFile (const char *filename, sfx_t *s)
{
qbyte *data;
wavinfo_t info;
int len;
- sfxcache_t *sc;
+ sfxbuffer_t* sb;
+
+ Mem_FreePool (&s->mempool);
+ s->mempool = Mem_AllocPool(s->name);
// Load the file
- data = FS_LoadFile(filename, false);
+ data = FS_LoadFile(filename, s->mempool, false);
if (!data)
- return NULL;
+ {
+ Mem_FreePool (&s->mempool);
+ return false;
+ }
// Don't try to load it if it's not a WAV file
if (memcmp (data, "RIFF", 4) || memcmp (data + 8, "WAVE", 4))
- return NULL;
+ {
+ Mem_FreePool (&s->mempool);
+ return false;
+ }
+
+ Con_DPrintf ("Loading WAV file \"%s\"\n", filename);
info = GetWavinfo (s->name, data, fs_filesize);
// Stereo sounds are 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;
+ Mem_FreePool (&s->mempool);
+ return false;
}
// calculate resampled length
- len = (int) ((double) info.samples * (double) shm->speed / (double) info.rate);
+ len = (int) ((double) info.samples * (double) shm->format.speed / (double) info.rate);
len = len * info.width * info.channels;
- Mem_FreePool(&s->mempool);
- s->mempool = Mem_AllocPool(s->name);
- sc = s->sfxcache = Mem_Alloc(s->mempool, len + sizeof(sfxcache_t));
- if (!sc)
+ sb = Mem_Alloc (s->mempool, len + sizeof (*sb) - sizeof (sb->data));
+ if (sb == NULL)
{
Con_Printf("failed to allocate memory for sound \"%s\"\n", s->name);
Mem_FreePool(&s->mempool);
- Mem_Free(data);
- return NULL;
+ return false;
}
- sc->length = info.samples;
- sc->loopstart = info.loopstart;
- sc->speed = info.rate;
- sc->width = info.width;
- sc->stereo = info.channels == 2;
+ s->fetcher = &wav_fetcher;
+ s->fetcher_data = sb;
+ s->format.speed = info.rate;
+ s->format.width = info.width;
+ s->format.channels = info.channels;
+ if (info.loopstart < 0)
+ s->loopstart = -1;
+ else
+ s->loopstart = (double)info.loopstart * (double)shm->format.speed / (double)s->format.speed;
#if BYTE_ORDER != LITTLE_ENDIAN
// We must convert the WAV data from little endian
}
#endif
- ResampleSfx(sc, data + info.dataofs, s->name);
+ sb->length = ResampleSfx (data + info.dataofs, info.samples, &s->format, sb->data, s->name);
+ s->format.speed = shm->format.speed;
+ s->total_length = sb->length;
+ sb->offset = 0;
- Mem_Free(data);
- return sc;
+ Mem_Free (data);
+ return true;
}
S_LoadSound
==============
*/
-sfxcache_t *S_LoadSound (sfx_t *s, int complain)
+qboolean S_LoadSound (sfx_t *s, int complain)
{
char namebuffer[MAX_QPATH];
size_t len;
- sfxcache_t *sc;
qboolean modified_name = false;
// see if still in memory
- if (!shm || !shm->speed)
- return NULL;
- if (s->sfxcache && (s->sfxcache->speed == shm->speed))
- return s->sfxcache;
+ 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);
+ return true;
+ }
len = snprintf (namebuffer, sizeof (namebuffer), "sound/%s", s->name);
if (len >= sizeof (namebuffer))
- return NULL;
+ return false;
// Try to load it as a WAV file
- sc = S_LoadWavFile (namebuffer, s);
- if (sc != NULL)
- return sc;
+ if (S_LoadWavFile (namebuffer, s))
+ return true;
// Else, try to load it as an Ogg Vorbis file
if (!strcasecmp (namebuffer + len - 4, ".wav"))
strcpy (namebuffer + len - 3, "ogg");
modified_name = true;
}
- sc = OGG_LoadVorbisFile (namebuffer, s);
- if (sc != NULL)
- return sc;
+ if (OGG_LoadVorbisFile (namebuffer, s))
+ return true;
// Can't load the sound!
if (!complain)
strcpy (namebuffer + len - 3, "wav");
Con_Printf("Couldn't load %s\n", namebuffer);
}
- return NULL;
+ return false;
}
void S_UnloadSound(sfx_t *s)
{
- if (s->sfxcache)
+ if (s->fetcher != NULL)
{
- s->sfxcache = NULL;
+ unsigned int i;
+
+ s->fetcher = NULL;
+ s->fetcher_data = NULL;
Mem_FreePool(&s->mempool);
+
+ // 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;
}
}
projects / xonotic / darkplaces.git / blobdiff