#include "quakedef.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;
+ 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;
}
//=============================================================================
-/*
-==============
-S_LoadWavFile
-==============
-*/
-sfxcache_t *S_LoadWavFile (const char *filename, sfx_t *s)
-{
- qbyte *data;
- wavinfo_t info;
- int len;
- sfxcache_t *sc;
-
- // Load the file
- data = FS_LoadFile(filename, tempmempool, false);
- if (!data)
- return NULL;
-
- // 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;
-
- 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;
- }
-
- // calculate resampled length
- len = (int) ((double) info.samples * (double) shm->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)
- {
- Con_Printf("failed to allocate memory for sound \"%s\"\n", s->name);
- 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;
-
-#if BYTE_ORDER != LITTLE_ENDIAN
- // We must convert the WAV data from little endian
- // to the machine endianess before resampling it
- if (info.width == 2)
- {
- int i;
- short* ptr;
-
- len = info.samples * info.channels;
- ptr = (short*)(data + info.dataofs);
- for (i = 0; i < len; i++)
- ptr[i] = LittleShort (ptr[i]);
- }
-#endif
-
- ResampleSfx(sc, data + info.dataofs, s->name);
-
- Mem_Free(data);
- return sc;
-}
-
-
/*
==============
S_LoadSound
==============
*/
-sfxcache_t *S_LoadSound (sfx_t *s, int complain)
+qboolean S_LoadSound (sfx_t *s, qboolean 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);
+ len = strlcpy (namebuffer, s->name, sizeof (namebuffer));
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;
- Mem_FreePool(&s->mempool);
- }
-}
-
-
-/*
-===============================================================================
-
-WAV loading
-
-===============================================================================
-*/
-
-
-static qbyte *data_p;
-static qbyte *iff_end;
-static qbyte *last_chunk;
-static qbyte *iff_data;
-static int iff_chunk_len;
+ unsigned int i;
+ s->fetcher = NULL;
+ s->fetcher_data = NULL;
+ Mem_FreePool(&s->mempool);
-short GetLittleShort(void)
-{
- short val;
-
- val = BuffLittleShort (data_p);
- data_p += 2;
-
- return val;
-}
-
-int GetLittleLong(void)
-{
- int val = 0;
-
- val = BuffLittleLong (data_p);
- data_p += 4;
-
- return val;
-}
-
-void FindNextChunk(char *name)
-{
- while (1)
- {
- 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;
- }
- data_p -= 8;
- last_chunk = data_p + 8 + ( (iff_chunk_len + 1) & ~1 );
- if (!strncmp(data_p, name, 4))
- return;
- }
-}
-
-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)))
- {
- Con_Print("Missing RIFF/WAVE chunks\n");
- return info;
- }
-
- // get "fmt " chunk
- iff_data = data_p + 12;
- //DumpChunks ();
-
- FindChunk("fmt ");
- if (!data_p)
- {
- Con_Print("Missing fmt chunk\n");
- return info;
- }
- data_p += 8;
- format = GetLittleShort();
- if (format != 1)
- {
- Con_Print("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)
- {
- 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;
- }
- }
- }
- else
- info.loopstart = -1;
-
- // find data chunk
- FindChunk("data");
- if (!data_p)
- {
- Con_Print("Missing data chunk\n");
- return info;
- }
-
- 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);
+ // 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.samples = samples;
-
- info.dataofs = data_p - wav;
-
- return info;
}
-
projects / xonotic / darkplaces.git / blobdiff