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((const char *)data_p, name, 4))
}
}
-static void FindChunk(char *name)
+static void FindChunk(const char *name)
{
last_chunk = iff_data;
FindNextChunk (name);
/*
====================
-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 (sfxbuffer_t *)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)
+static void WAV_FreeSfx(sfx_t *sfx)
{
- sfxbuffer_t* sb = (sfxbuffer_t *)sfx->fetcher_data;
-
- // Free the sound buffer
- sfx->memsize -= (sb->length * sfx->format.channels * sfx->format.width) + sizeof (*sb) - sizeof (sb->data);
- Mem_Free(sb);
-
- sfx->fetcher_data = NULL;
- sfx->fetcher = NULL;
+ // free the loaded sound data
+ Mem_Free(sfx->fetcher_data);
}
-const snd_fetcher_t wav_fetcher = { WAV_FetchSound, NULL, WAV_FreeSfx };
+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)
{
fs_offset_t filesize;
unsigned char *data;
wavinfo_t info;
- int len;
- size_t memsize;
- sfxbuffer_t* sb;
+ int i, len;
+ const unsigned char *inb;
+ unsigned char *outb;
// Already loaded?
- if (s->fetcher != NULL)
+ if (sfx->fetcher != NULL)
return true;
// Load the file
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)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);
+ 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;
-
- memsize = len + sizeof (*sb) - sizeof (sb->data);
- sb = (sfxbuffer_t *)Mem_Alloc (snd_mempool, memsize);
- 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_Free(data);
- 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->memsize += memsize;
-
- 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 = (int)((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;
}
projects / xonotic / darkplaces.git / blobdiff