quakespasm/Quake/snd_mem.c
Ozkan Sezer dd9f105e42 * snd_codec: store the samplebits value in the snd_info_t struct,
and add a new blocksize field to it which the flac decoder can
use. updated decoder sources for the snd_info_t changes, where
I made minor tidy-ups too, tightening several format checks and
fixing a few gotchas in snd_wave.c and snd_mem.c.
* snd_flac.c: adjusted for snd_info_t changes. no longed storing
metadata->data.stream_info in our private data, but just storing
a pointer to the stream->info structure. No longer checking the
metadata total_samples field (the FLAC__StreamMetadata_StreamInfo
doesn't seem to have any alignment or pack attributes and I don't
safe with its offset across different compilers), but added check
to make sure that we hit and parsed a STREAMINFO metadata instead,
and our new state seems just fine for validating the file.


git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@859 af15c1b1-3010-417e-b628-4374ebc0bcbd
2013-07-22 11:41:14 +00:00

352 lines
7 KiB
C

/*
Copyright (C) 1996-2001 Id Software, Inc.
Copyright (C) 2010-2011 O. Sezer <sezero@users.sourceforge.net>
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
See the GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
// snd_mem.c: sound caching
#include "quakedef.h"
/*
================
ResampleSfx
================
*/
static void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, byte *data)
{
int outcount;
int srcsample;
float stepscale;
int i;
int sample, samplefrac, fracstep;
sfxcache_t *sc;
sc = (sfxcache_t *) Cache_Check (&sfx->cache);
if (!sc)
return;
stepscale = (float)inrate / shm->speed; // this is usually 0.5, 1, or 2
outcount = sc->length / stepscale;
sc->length = outcount;
if (sc->loopstart != -1)
sc->loopstart = sc->loopstart / stepscale;
sc->speed = shm->speed;
if (loadas8bit.value)
sc->width = 1;
else
sc->width = inwidth;
sc->stereo = 0;
// resample / decimate to the current source rate
if (stepscale == 1 && inwidth == 1 && sc->width == 1)
{
// fast special case
for (i = 0; i < outcount; i++)
((signed char *)sc->data)[i] = (int)( (unsigned char)(data[i]) - 128);
}
else
{
// general case
samplefrac = 0;
fracstep = stepscale*256;
for (i = 0; i < outcount; i++)
{
srcsample = samplefrac >> 8;
samplefrac += fracstep;
if (inwidth == 2)
sample = LittleShort ( ((short *)data)[srcsample] );
else
sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8;
if (sc->width == 2)
((short *)sc->data)[i] = sample;
else
((signed char *)sc->data)[i] = sample >> 8;
}
}
}
//=============================================================================
/*
==============
S_LoadSound
==============
*/
sfxcache_t *S_LoadSound (sfx_t *s)
{
char namebuffer[256];
byte *data;
wavinfo_t info;
int len;
float stepscale;
sfxcache_t *sc;
byte stackbuf[1*1024]; // avoid dirtying the cache heap
// see if still in memory
sc = (sfxcache_t *) Cache_Check (&s->cache);
if (sc)
return sc;
// Con_Printf ("S_LoadSound: %x\n", (int)stackbuf);
// load it in
q_strlcpy(namebuffer, "sound/", sizeof(namebuffer));
q_strlcat(namebuffer, s->name, sizeof(namebuffer));
// Con_Printf ("loading %s\n",namebuffer);
data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf), NULL);
if (!data)
{
Con_Printf ("Couldn't load %s\n", namebuffer);
return NULL;
}
info = GetWavinfo (s->name, data, com_filesize);
if (info.channels != 1)
{
Con_Printf ("%s is a stereo sample\n",s->name);
return NULL;
}
if (info.width != 1 && info.width != 2)
{
Con_Printf("%s is not 8 or 16 bit\n", s->name);
return NULL;
}
stepscale = (float)info.rate / shm->speed;
len = info.samples / stepscale;
len = len * info.width * info.channels;
if (info.samples == 0 || len == 0)
{
Con_Printf("%s has zero samples\n", s->name);
return NULL;
}
sc = (sfxcache_t *) Cache_Alloc ( &s->cache, len + sizeof(sfxcache_t), s->name);
if (!sc)
return NULL;
sc->length = info.samples;
sc->loopstart = info.loopstart;
sc->speed = info.rate;
sc->width = info.width;
sc->stereo = info.channels;
ResampleSfx (s, sc->speed, sc->width, data + info.dataofs);
return sc;
}
/*
===============================================================================
WAV loading
===============================================================================
*/
static byte *data_p;
static byte *iff_end;
static byte *last_chunk;
static byte *iff_data;
static int iff_chunk_len;
static short GetLittleShort (void)
{
short val = 0;
val = *data_p;
val = val + (*(data_p+1)<<8);
data_p += 2;
return val;
}
static 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;
}
static void FindNextChunk (const char *name)
{
while (1)
{
// Need at least 8 bytes for a chunk
if (last_chunk + 8 >= iff_end)
{
data_p = NULL;
return;
}
data_p = last_chunk + 4;
iff_chunk_len = GetLittleLong();
if (iff_chunk_len < 0 || iff_chunk_len > iff_end - data_p)
{
data_p = NULL;
Con_DPrintf("bad \"%s\" chunk length (%d)\n", name, iff_chunk_len);
return;
}
last_chunk = data_p + ((iff_chunk_len + 1) & ~1);
data_p -= 8;
if (!Q_strncmp((char *)data_p, name, 4))
return;
}
}
static void FindChunk (const char *name)
{
last_chunk = iff_data;
FindNextChunk (name);
}
#if 0
static 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);
}
#endif
/*
============
GetWavinfo
============
*/
wavinfo_t GetWavinfo (const char *name, byte *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 && !Q_strncmp((char *)data_p + 8, "WAVE", 4)))
{
Con_Printf("%s missing RIFF/WAVE chunks\n", name);
return info;
}
// get "fmt " chunk
iff_data = data_p + 12;
#if 0
DumpChunks ();
#endif
FindChunk("fmt ");
if (!data_p)
{
Con_Printf("%s is missing fmt chunk\n", name);
return info;
}
data_p += 8;
format = GetLittleShort();
if (format != WAV_FORMAT_PCM)
{
Con_Printf("%s is not Microsoft PCM format\n", name);
return info;
}
info.channels = GetLittleShort();
info.rate = GetLittleLong();
data_p += 4 + 2;
i = GetLittleShort();
if (i != 8 && i != 16)
return info;
info.width = i / 8;
// get cue chunk
FindChunk("cue ");
if (data_p)
{
data_p += 32;
info.loopstart = GetLittleLong();
// Con_Printf("loopstart=%d\n", sfx->loopstart);
// if the next chunk is a LIST chunk, look for a cue length marker
FindNextChunk ("LIST");
if (data_p)
{
if (!strncmp((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
info.samples = info.loopstart + i;
// Con_Printf("looped length: %i\n", i);
}
}
}
else
info.loopstart = -1;
// find data chunk
FindChunk("data");
if (!data_p)
{
Con_Printf("%s is missing data chunk\n", name);
return info;
}
data_p += 4;
samples = GetLittleLong() / info.width;
if (info.samples)
{
if (samples < info.samples)
Sys_Error ("%s has a bad loop length", name);
}
else
info.samples = samples;
info.dataofs = data_p - wav;
return info;
}