#include "snd_wav.h"
-typedef struct
+typedef struct wavinfo_s
{
int rate;
int width;
} wavinfo_t;
-static qbyte *data_p;
-static qbyte *iff_end;
-static qbyte *last_chunk;
-static qbyte *iff_data;
+static unsigned char *data_p;
+static unsigned char *iff_end;
+static unsigned char *last_chunk;
+static unsigned char *iff_data;
static int iff_chunk_len;
return val;
}
-static void FindNextChunk(char *name)
+static void FindNextChunk(const char *name)
{
while (1)
{
data_p = NULL;
return;
}
+ if (data_p + iff_chunk_len > iff_end)
+ {
+ // truncated chunk!
+ data_p = NULL;
+ return;
+ }
data_p -= 8;
last_chunk = data_p + 8 + ( (iff_chunk_len + 1) & ~1 );
- if (!strncmp(data_p, name, 4))
+ if (!strncmp((const char *)data_p, name, 4))
return;
}
}
-static void FindChunk(char *name)
+static void FindChunk(const char *name)
{
last_chunk = iff_data;
FindNextChunk (name);
GetWavinfo
============
*/
-static wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength)
+static wavinfo_t GetWavinfo (char *name, unsigned char *wav, int wavlength)
{
wavinfo_t info;
int i;
// find "RIFF" chunk
FindChunk("RIFF");
- if (!(data_p && !strncmp(data_p+8, "WAVE", 4)))
+ if (!(data_p && !strncmp((const char *)data_p+8, "WAVE", 4)))
{
Con_Print("Missing RIFF/WAVE chunks\n");
return info;
FindNextChunk ("LIST");
if (data_p)
{
- if (!strncmp (data_p + 28, "mark", 4))
+ if (!strncmp ((const char *)data_p + 28, "mark", 4))
{ // this is not a proper parse, but it works with cooledit...
data_p += 24;
i = GetLittleLong (); // samples in loop
{
if (samples < info.samples)
{
- Con_Printf ("Sound %s has a bad loop length", name);
+ Con_Printf ("Sound %s has a bad loop length\n", name);
info.samples = samples;
}
}
/*
====================
-WAV_FetchSound
+WAV_GetSamplesFloat
====================
*/
-static const sfxbuffer_t* WAV_FetchSound (channel_t* ch, unsigned int start, unsigned int nbsamples)
+static void WAV_GetSamplesFloat(channel_t *ch, sfx_t *sfx, int firstsampleframe, int numsampleframes, float *outsamplesfloat)
{
- return ch->sfx->fetcher_data;
+ int i, len = numsampleframes * sfx->format.channels;
+ if (sfx->format.width == 2)
+ {
+ const short *bufs = (const short *)sfx->fetcher_data + firstsampleframe * sfx->format.channels;
+ for (i = 0;i < len;i++)
+ outsamplesfloat[i] = bufs[i] * (1.0f / 32768.0f);
+ }
+ else
+ {
+ const signed char *bufb = (const signed char *)sfx->fetcher_data + firstsampleframe * sfx->format.channels;
+ for (i = 0;i < len;i++)
+ outsamplesfloat[i] = bufb[i] * (1.0f / 128.0f);
+ }
}
+/*
+====================
+WAV_FreeSfx
+====================
+*/
+static void WAV_FreeSfx(sfx_t *sfx)
+{
+ // free the loaded sound data
+ Mem_Free(sfx->fetcher_data);
+}
-snd_fetcher_t wav_fetcher = { WAV_FetchSound, NULL };
+const snd_fetcher_t wav_fetcher = { WAV_GetSamplesFloat, NULL, WAV_FreeSfx };
/*
S_LoadWavFile
==============
*/
-qboolean S_LoadWavFile (const char *filename, sfx_t *s)
+qboolean S_LoadWavFile (const char *filename, sfx_t *sfx)
{
- qbyte *data;
+ fs_offset_t filesize;
+ unsigned char *data;
wavinfo_t info;
- int len;
- sfxbuffer_t* sb;
+ int i, len;
+ const unsigned char *inb;
+ unsigned char *outb;
- Mem_FreePool (&s->mempool);
- s->mempool = Mem_AllocPool(s->name, 0, NULL);
+ // Already loaded?
+ if (sfx->fetcher != NULL)
+ return true;
// Load the file
- data = FS_LoadFile(filename, s->mempool, false);
+ data = FS_LoadFile(filename, snd_mempool, false, &filesize);
if (!data)
- {
- 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))
{
- Mem_FreePool (&s->mempool);
+ Mem_Free(data);
return false;
}
- Con_DPrintf ("Loading WAV file \"%s\"\n", filename);
+ if (developer_loading.integer >= 2)
+ Con_Printf ("Loading WAV file \"%s\"\n", filename);
- info = GetWavinfo (s->name, data, (int)fs_filesize);
- // Stereo sounds are allowed (intended for music)
- if (info.channels < 1 || info.channels > 2)
+ info = GetWavinfo (sfx->name, data, (int)filesize);
+ if (info.channels < 1 || info.channels > 2) // Stereo sounds are allowed (intended for music)
{
- Con_Printf("%s has an unsupported number of channels (%i)\n",s->name, info.channels);
- Mem_FreePool (&s->mempool);
+ Con_Printf("%s has an unsupported number of channels (%i)\n",sfx->name, info.channels);
+ Mem_Free(data);
return false;
}
//if (info.channels == 2)
- // Log_Printf("stereosounds.log", "%s\n", s->name);
-
- // calculate resampled length
- len = (int) ((double) info.samples * (double) shm->format.speed / (double) info.rate);
- len = len * info.width * info.channels;
-
- sb = Mem_Alloc (s->mempool, len + sizeof (*sb) - sizeof (sb->data));
- if (sb == NULL)
+ // Log_Printf("stereosounds.log", "%s\n", sfx->name);
+
+ sfx->format.speed = info.rate;
+ sfx->format.width = info.width;
+ sfx->format.channels = info.channels;
+ sfx->fetcher = &wav_fetcher;
+ sfx->fetcher_data = Mem_Alloc(snd_mempool, info.samples * sfx->format.width * sfx->format.channels);
+ sfx->total_length = info.samples;
+ sfx->memsize += filesize;
+ len = info.samples * sfx->format.channels * sfx->format.width;
+ inb = data + info.dataofs;
+ outb = (unsigned char *)sfx->fetcher_data;
+ if (info.width == 2)
{
- Con_Printf("failed to allocate memory for sound \"%s\"\n", s->name);
- Mem_FreePool(&s->mempool);
- return false;
+ if (mem_bigendian)
+ {
+ // we have to byteswap the data at load (better than doing it while mixing)
+ for (i = 0;i < len;i += 2)
+ {
+ outb[i] = inb[i+1];
+ outb[i+1] = inb[i];
+ }
+ }
+ else
+ {
+ // we can just copy it straight
+ memcpy(outb, inb, len);
+ }
}
-
- 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;
- s->flags &= ~SFXFLAG_STREAMED;
-
-#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]);
+ // convert unsigned byte sound data to signed bytes for quicker mixing
+ for (i = 0;i < len;i++)
+ outb[i] = inb[i] - 0x80;
}
-#endif
- sb->length = (int)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;
+ if (info.loopstart < 0)
+ sfx->loopstart = sfx->total_length;
+ else
+ sfx->loopstart = info.loopstart;
+ sfx->loopstart = min(sfx->loopstart, sfx->total_length);
+ sfx->flags &= ~SFXFLAG_STREAMED;
- Mem_Free (data);
return true;
}