mirror of
https://github.com/UberGames/lilium-voyager.git
synced 2024-12-14 22:20:58 +00:00
0e412276c5
Based on Thilo's Elite Force patch for ioq3. Updated Makefile changes. Left out in-tree win32 lib support. Striped trailing whitespace from snd_codec_mp3.c. Changed two #if USE_CODEC_MP3 to use #ifdef.
716 lines
17 KiB
C
716 lines
17 KiB
C
/*
|
|
===========================================================================
|
|
Copyright (C) 1999-2005 Id Software, Inc.
|
|
Copyright (C) 2005 Stuart Dalton (badcdev@gmail.com)
|
|
Copyright (C) 2005-2006 Joerg Dietrich <dietrich_joerg@gmx.de>
|
|
Copyright (C) 2006 Thilo Schulz <arny@ats.s.bawue.de>
|
|
|
|
This file is part of Quake III Arena source code.
|
|
|
|
Quake III Arena source code 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.
|
|
|
|
Quake III Arena source code 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 Quake III Arena source code; if not, write to the Free Software
|
|
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
|
|
===========================================================================
|
|
*/
|
|
|
|
// MP3 support is enabled by this define
|
|
#ifdef USE_CODEC_MP3
|
|
|
|
// includes for the Q3 sound system
|
|
#include "client.h"
|
|
#include "snd_codec.h"
|
|
|
|
// includes for the MP3 codec
|
|
#include <mad.h>
|
|
|
|
#define MP3_SAMPLE_WIDTH 2
|
|
#define MP3_PCMSAMPLES_PERSLICE 32
|
|
|
|
// buffer size used when reading through the mp3
|
|
#define MP3_DATA_BUFSIZ 128*1024
|
|
|
|
// undefine this if you don't want any dithering.
|
|
#define MP3_DITHERING
|
|
|
|
// Q3 MP3 codec
|
|
snd_codec_t mp3_codec =
|
|
{
|
|
"mp3",
|
|
S_MP3_CodecLoad,
|
|
S_MP3_CodecOpenStream,
|
|
S_MP3_CodecReadStream,
|
|
S_MP3_CodecCloseStream,
|
|
NULL
|
|
};
|
|
|
|
// structure used for info purposes
|
|
struct snd_codec_mp3_info
|
|
{
|
|
byte encbuf[MP3_DATA_BUFSIZ]; // left over bytes not consumed
|
|
// by the decoder.
|
|
struct mad_stream madstream; // uses encbuf as buffer.
|
|
struct mad_frame madframe; // control structures for libmad.
|
|
struct mad_synth madsynth;
|
|
|
|
byte *pcmbuf; // buffer for not-used samples.
|
|
int buflen; // length of buffer data.
|
|
int pcmbufsize; // amount of allocated memory for
|
|
// pcmbuf. This should have at least
|
|
// the size of a decoded mp3 frame.
|
|
|
|
byte *dest; // copy decoded data here.
|
|
int destlen; // amount of already copied data.
|
|
int destsize; // amount of bytes we must decode.
|
|
};
|
|
|
|
/*************** MP3 utility functions ***************/
|
|
|
|
/*
|
|
=================
|
|
S_MP3_ReadData
|
|
=================
|
|
*/
|
|
|
|
// feed libmad with data
|
|
int S_MP3_ReadData(snd_stream_t *stream, struct mad_stream *madstream, byte *encbuf, int encbufsize)
|
|
{
|
|
int retval;
|
|
int leftover;
|
|
|
|
if(!stream)
|
|
return -1;
|
|
|
|
leftover = madstream->bufend - madstream->next_frame;
|
|
if(leftover > 0)
|
|
memmove(encbuf, madstream->this_frame, leftover);
|
|
|
|
|
|
// Fill the buffer right to the end
|
|
|
|
retval = FS_Read(&encbuf[leftover], encbufsize - leftover, stream->file);
|
|
|
|
if(retval <= 0)
|
|
{
|
|
// EOF reached, that's ok.
|
|
return 0;
|
|
}
|
|
|
|
mad_stream_buffer(madstream, encbuf, retval + leftover);
|
|
|
|
return retval;
|
|
}
|
|
|
|
|
|
/*
|
|
=================
|
|
S_MP3_Scanfile
|
|
|
|
to determine the samplecount, we apparently must get *all* headers :(
|
|
I basically used the xmms-mad plugin source to see how this stuff works.
|
|
|
|
returns a value < 0 on error.
|
|
=================
|
|
*/
|
|
|
|
int S_MP3_Scanfile(snd_stream_t *stream)
|
|
{
|
|
struct mad_stream madstream;
|
|
struct mad_header madheader;
|
|
int retval;
|
|
int samplecount;
|
|
byte encbuf[MP3_DATA_BUFSIZ];
|
|
|
|
// error out on invalid input.
|
|
if(!stream)
|
|
return -1;
|
|
|
|
mad_stream_init(&madstream);
|
|
mad_header_init(&madheader);
|
|
|
|
while(1)
|
|
{
|
|
retval = S_MP3_ReadData(stream, &madstream, encbuf, sizeof(encbuf));
|
|
if(retval < 0)
|
|
return -1;
|
|
else if(retval == 0)
|
|
break;
|
|
|
|
// Start decoding the headers.
|
|
while(1)
|
|
{
|
|
if((retval = mad_header_decode(&madheader, &madstream)) < 0)
|
|
{
|
|
if(madstream.error == MAD_ERROR_BUFLEN)
|
|
{
|
|
// We need to read more data
|
|
break;
|
|
}
|
|
|
|
if(!MAD_RECOVERABLE (madstream.error))
|
|
{
|
|
// unrecoverable error... we must bail out.
|
|
return retval;
|
|
}
|
|
|
|
mad_stream_skip(&madstream, madstream.skiplen);
|
|
continue;
|
|
}
|
|
|
|
// we got a valid header.
|
|
|
|
if(madheader.layer != MAD_LAYER_III)
|
|
{
|
|
// we don't support non-mp3s
|
|
return -1;
|
|
}
|
|
|
|
if(!stream->info.samples)
|
|
{
|
|
// This here is the very first frame. Set initial values now,
|
|
// that we expect to stay constant throughout the whole mp3.
|
|
|
|
stream->info.rate = madheader.samplerate;
|
|
stream->info.width = MP3_SAMPLE_WIDTH;
|
|
stream->info.channels = MAD_NCHANNELS(&madheader);
|
|
stream->info.samples = 0;
|
|
stream->info.size = 0; // same here.
|
|
stream->info.dataofs = 0;
|
|
}
|
|
else
|
|
{
|
|
// Check whether something changed that shouldn't.
|
|
|
|
if(stream->info.rate != madheader.samplerate ||
|
|
stream->info.channels != MAD_NCHANNELS(&madheader))
|
|
return -1;
|
|
}
|
|
|
|
// Update the counters
|
|
samplecount = MAD_NSBSAMPLES(&madheader) * MP3_PCMSAMPLES_PERSLICE;
|
|
stream->info.samples += samplecount;
|
|
stream->info.size += samplecount * stream->info.channels * stream->info.width;
|
|
}
|
|
}
|
|
|
|
// Reset the file pointer so we can do the real decoding.
|
|
FS_Seek(stream->file, 0, FS_SEEK_SET);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/************************ dithering functions ***************************/
|
|
|
|
#ifdef MP3_DITHERING
|
|
|
|
// All dithering done here is taken from the GPL'ed xmms-mad plugin.
|
|
|
|
/* Copyright (C) 1997 Makoto Matsumoto and Takuji Nishimura. */
|
|
/* Any feedback is very welcome. For any question, comments, */
|
|
/* see http://www.math.keio.ac.jp/matumoto/emt.html or email */
|
|
/* matumoto@math.keio.ac.jp */
|
|
|
|
/* Period parameters */
|
|
#define MP3_DITH_N 624
|
|
#define MP3_DITH_M 397
|
|
#define MATRIX_A 0x9908b0df /* constant vector a */
|
|
#define UPPER_MASK 0x80000000 /* most significant w-r bits */
|
|
#define LOWER_MASK 0x7fffffff /* least significant r bits */
|
|
|
|
/* Tempering parameters */
|
|
#define TEMPERING_MASK_B 0x9d2c5680
|
|
#define TEMPERING_MASK_C 0xefc60000
|
|
#define TEMPERING_SHIFT_U(y) (y >> 11)
|
|
#define TEMPERING_SHIFT_S(y) (y << 7)
|
|
#define TEMPERING_SHIFT_T(y) (y << 15)
|
|
#define TEMPERING_SHIFT_L(y) (y >> 18)
|
|
|
|
static unsigned long mt[MP3_DITH_N]; /* the array for the state vector */
|
|
static int mti=MP3_DITH_N+1; /* mti==MP3_DITH_N+1 means mt[MP3_DITH_N] is not initialized */
|
|
|
|
/* initializing the array with a NONZERO seed */
|
|
void sgenrand(unsigned long seed)
|
|
{
|
|
/* setting initial seeds to mt[MP3_DITH_N] using */
|
|
/* the generator Line 25 of Table 1 in */
|
|
/* [KNUTH 1981, The Art of Computer Programming */
|
|
/* Vol. 2 (2nd Ed.), pp102] */
|
|
mt[0]= seed & 0xffffffff;
|
|
for (mti=1; mti<MP3_DITH_N; mti++)
|
|
mt[mti] = (69069 * mt[mti-1]) & 0xffffffff;
|
|
}
|
|
|
|
unsigned long genrand(void)
|
|
{
|
|
unsigned long y;
|
|
static unsigned long mag01[2]={0x0, MATRIX_A};
|
|
/* mag01[x] = x * MATRIX_A for x=0,1 */
|
|
|
|
if (mti >= MP3_DITH_N) { /* generate MP3_DITH_N words at one time */
|
|
int kk;
|
|
|
|
if (mti == MP3_DITH_N+1) /* if sgenrand() has not been called, */
|
|
sgenrand(4357); /* a default initial seed is used */
|
|
|
|
for (kk=0;kk<MP3_DITH_N-MP3_DITH_M;kk++) {
|
|
y = (mt[kk]&UPPER_MASK)|(mt[kk+1]&LOWER_MASK);
|
|
mt[kk] = mt[kk+MP3_DITH_M] ^ (y >> 1) ^ mag01[y & 0x1];
|
|
}
|
|
for (;kk<MP3_DITH_N-1;kk++) {
|
|
y = (mt[kk]&UPPER_MASK)|(mt[kk+1]&LOWER_MASK);
|
|
mt[kk] = mt[kk+(MP3_DITH_M-MP3_DITH_N)] ^ (y >> 1) ^ mag01[y & 0x1];
|
|
}
|
|
y = (mt[MP3_DITH_N-1]&UPPER_MASK)|(mt[0]&LOWER_MASK);
|
|
mt[MP3_DITH_N-1] = mt[MP3_DITH_M-1] ^ (y >> 1) ^ mag01[y & 0x1];
|
|
|
|
mti = 0;
|
|
}
|
|
|
|
y = mt[mti++];
|
|
y ^= TEMPERING_SHIFT_U(y);
|
|
y ^= TEMPERING_SHIFT_S(y) & TEMPERING_MASK_B;
|
|
y ^= TEMPERING_SHIFT_T(y) & TEMPERING_MASK_C;
|
|
y ^= TEMPERING_SHIFT_L(y);
|
|
|
|
return y;
|
|
}
|
|
|
|
long triangular_dither_noise(int nbits) {
|
|
// parameter nbits : the peak-to-peak amplitude desired (in bits)
|
|
// use with nbits set to 2 + nber of bits to be trimmed.
|
|
// (because triangular is made from two uniformly distributed processes,
|
|
// it starts at 2 bits peak-to-peak amplitude)
|
|
// see The Theory of Dithered Quantization by Robert Alexander Wannamaker
|
|
// for complete proof of why that's optimal
|
|
|
|
long v = (genrand()/2 - genrand()/2); // in ]-2^31, 2^31[
|
|
//int signe = (v>0) ? 1 : -1;
|
|
long P = 1 << (32 - nbits); // the power of 2
|
|
v /= P;
|
|
// now v in ]-2^(nbits-1), 2^(nbits-1) [
|
|
|
|
return v;
|
|
}
|
|
|
|
#endif // MP3_DITHERING
|
|
|
|
/************************ decoder functions ***************************/
|
|
|
|
/*
|
|
=================
|
|
S_MP3_Scale
|
|
|
|
Converts the signal to 16 bit LE-PCM data and does dithering.
|
|
|
|
- borrowed from xmms-mad plugin source.
|
|
=================
|
|
*/
|
|
|
|
/*
|
|
* xmms-mad - mp3 plugin for xmms
|
|
* Copyright (C) 2001-2002 Sam Clegg
|
|
*/
|
|
|
|
signed int S_MP3_Scale(mad_fixed_t sample)
|
|
{
|
|
int n_bits_to_loose = MAD_F_FRACBITS + 1 - 16;
|
|
#ifdef MP3_DITHERING
|
|
int dither;
|
|
#endif
|
|
|
|
// round
|
|
sample += (1L << (n_bits_to_loose - 1));
|
|
|
|
#ifdef MP3_DITHERING
|
|
dither = triangular_dither_noise(n_bits_to_loose + 1);
|
|
sample += dither;
|
|
#endif
|
|
|
|
/* clip */
|
|
if (sample >= MAD_F_ONE)
|
|
sample = MAD_F_ONE - 1;
|
|
else if (sample < -MAD_F_ONE)
|
|
sample = -MAD_F_ONE;
|
|
|
|
/* quantize */
|
|
return sample >> n_bits_to_loose;
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
=================
|
|
S_MP3_PCMCopy
|
|
|
|
Copy and convert pcm data until bytecount bytes have been written.
|
|
return the position in pcm->samples.
|
|
indicate the amount of actually written bytes in wrotecnt.
|
|
=================
|
|
*/
|
|
|
|
int S_MP3_PCMCopy(byte *buf, struct mad_pcm *pcm, int bufofs,
|
|
int sampleofs, int bytecount, int *wrotecnt)
|
|
{
|
|
int written = 0;
|
|
signed int sample;
|
|
int framesize = pcm->channels * MP3_SAMPLE_WIDTH;
|
|
|
|
// add new pcm data.
|
|
while(written < bytecount && sampleofs < pcm->length)
|
|
{
|
|
sample = S_MP3_Scale(pcm->samples[0][sampleofs]);
|
|
|
|
#ifdef Q3_BIG_ENDIAN
|
|
// output to 16 bit big endian PCM
|
|
buf[bufofs++] = (sample >> 8) & 0xff;
|
|
buf[bufofs++] = sample & 0xff;
|
|
#else
|
|
// output to 16 bit little endian PCM
|
|
buf[bufofs++] = sample & 0xff;
|
|
buf[bufofs++] = (sample >> 8) & 0xff;
|
|
#endif
|
|
|
|
if(pcm->channels == 2)
|
|
{
|
|
sample = S_MP3_Scale(pcm->samples[1][sampleofs]);
|
|
|
|
#ifdef Q3_BIG_ENDIAN
|
|
buf[bufofs++] = (sample >> 8) & 0xff;
|
|
buf[bufofs++] = sample & 0xff;
|
|
#else
|
|
buf[bufofs++] = sample & 0xff;
|
|
buf[bufofs++] = (sample >> 8) & 0xff;
|
|
#endif
|
|
}
|
|
|
|
sampleofs++;
|
|
written += framesize;
|
|
}
|
|
|
|
if(wrotecnt)
|
|
*wrotecnt = written;
|
|
|
|
return sampleofs;
|
|
}
|
|
|
|
|
|
/*
|
|
=================
|
|
S_MP3_Decode
|
|
=================
|
|
*/
|
|
|
|
// gets executed for every decoded frame.
|
|
int S_MP3_Decode(snd_stream_t *stream)
|
|
{
|
|
struct snd_codec_mp3_info *mp3info;
|
|
struct mad_stream *madstream;
|
|
struct mad_frame *madframe;
|
|
struct mad_synth *madsynth;
|
|
struct mad_pcm *pcm;
|
|
int cursize;
|
|
int samplecount;
|
|
int needcount;
|
|
int wrote;
|
|
int retval;
|
|
|
|
if(!stream)
|
|
return -1;
|
|
|
|
mp3info = stream->ptr;
|
|
madstream = &mp3info->madstream;
|
|
madframe = &mp3info->madframe;
|
|
|
|
if(mad_frame_decode(madframe, madstream))
|
|
{
|
|
if(madstream->error == MAD_ERROR_BUFLEN)
|
|
{
|
|
// we need more data. Read another chunk.
|
|
retval = S_MP3_ReadData(stream, madstream, mp3info->encbuf, sizeof(mp3info->encbuf));
|
|
|
|
// call myself again now that buffer is full.
|
|
if(retval > 0)
|
|
retval = S_MP3_Decode(stream);
|
|
}
|
|
else if(MAD_RECOVERABLE(madstream->error))
|
|
{
|
|
mad_stream_skip(madstream, madstream->skiplen);
|
|
return S_MP3_Decode(stream);
|
|
}
|
|
else
|
|
retval = -1;
|
|
|
|
return retval;
|
|
}
|
|
|
|
// check whether this really is an mp3
|
|
if(madframe->header.layer != MAD_LAYER_III)
|
|
return -1;
|
|
|
|
// generate pcm data
|
|
madsynth = &mp3info->madsynth;
|
|
mad_synth_frame(madsynth, madframe);
|
|
|
|
pcm = &madsynth->pcm;
|
|
|
|
// perform a few checks to see whether something changed that shouldn't.
|
|
|
|
if(stream->info.rate != pcm->samplerate ||
|
|
stream->info.channels != pcm->channels)
|
|
{
|
|
return -1;
|
|
}
|
|
// see whether we have got enough data now.
|
|
cursize = pcm->length * pcm->channels * stream->info.width;
|
|
needcount = mp3info->destsize - mp3info->destlen;
|
|
|
|
// Copy exactly as many samples as required.
|
|
samplecount = S_MP3_PCMCopy(mp3info->dest, pcm,
|
|
mp3info->destlen, 0, needcount, &wrote);
|
|
mp3info->destlen += wrote;
|
|
|
|
if(samplecount < pcm->length)
|
|
{
|
|
// Not all samples got copied. Copy the rest into the pcm buffer.
|
|
samplecount = S_MP3_PCMCopy(mp3info->pcmbuf, pcm,
|
|
mp3info->buflen,
|
|
samplecount,
|
|
mp3info->pcmbufsize - mp3info->buflen,
|
|
&wrote);
|
|
mp3info->buflen += wrote;
|
|
|
|
|
|
if(samplecount < pcm->length)
|
|
{
|
|
// The pcm buffer was not large enough. Make it bigger.
|
|
byte *newbuf = Z_Malloc(cursize);
|
|
|
|
if(mp3info->pcmbuf)
|
|
{
|
|
memcpy(newbuf, mp3info->pcmbuf, mp3info->buflen);
|
|
Z_Free(mp3info->pcmbuf);
|
|
}
|
|
|
|
mp3info->pcmbuf = newbuf;
|
|
mp3info->pcmbufsize = cursize;
|
|
|
|
samplecount = S_MP3_PCMCopy(mp3info->pcmbuf, pcm,
|
|
mp3info->buflen,
|
|
samplecount,
|
|
mp3info->pcmbufsize - mp3info->buflen,
|
|
&wrote);
|
|
mp3info->buflen += wrote;
|
|
}
|
|
|
|
// we're definitely done.
|
|
retval = 0;
|
|
}
|
|
else if(mp3info->destlen >= mp3info->destsize)
|
|
retval = 0;
|
|
else
|
|
retval = 1;
|
|
|
|
return retval;
|
|
}
|
|
|
|
/*************** Callback functions for quake3 ***************/
|
|
|
|
/*
|
|
=================
|
|
S_MP3_CodecOpenStream
|
|
=================
|
|
*/
|
|
|
|
snd_stream_t *S_MP3_CodecOpenStream(const char *filename)
|
|
{
|
|
snd_stream_t *stream;
|
|
struct snd_codec_mp3_info *mp3info;
|
|
|
|
// Open the stream
|
|
stream = S_CodecUtilOpen(filename, &mp3_codec);
|
|
if(!stream || stream->length <= 0)
|
|
return NULL;
|
|
|
|
// We have to scan through the MP3 to determine the important mp3 info.
|
|
if(S_MP3_Scanfile(stream) < 0)
|
|
{
|
|
// scanning didn't work out...
|
|
S_CodecUtilClose(&stream);
|
|
return NULL;
|
|
}
|
|
|
|
// Initialize the mp3 info structure we need for streaming
|
|
mp3info = Z_Malloc(sizeof(*mp3info));
|
|
if(!mp3info)
|
|
{
|
|
S_CodecUtilClose(&stream);
|
|
return NULL;
|
|
}
|
|
|
|
stream->ptr = mp3info;
|
|
|
|
// initialize the libmad control structures.
|
|
mad_stream_init(&mp3info->madstream);
|
|
mad_frame_init(&mp3info->madframe);
|
|
mad_synth_init(&mp3info->madsynth);
|
|
|
|
if(S_MP3_ReadData(stream, &mp3info->madstream, mp3info->encbuf, sizeof(mp3info->encbuf)) <= 0)
|
|
{
|
|
// we didnt read anything, that's bad.
|
|
S_MP3_CodecCloseStream(stream);
|
|
return NULL;
|
|
}
|
|
|
|
return stream;
|
|
}
|
|
|
|
/*
|
|
=================
|
|
S_MP3_CodecCloseStream
|
|
=================
|
|
*/
|
|
|
|
// free all memory we allocated.
|
|
void S_MP3_CodecCloseStream(snd_stream_t *stream)
|
|
{
|
|
struct snd_codec_mp3_info *mp3info;
|
|
|
|
if(!stream)
|
|
return;
|
|
|
|
// free all data in our mp3info tree
|
|
|
|
if(stream->ptr)
|
|
{
|
|
mp3info = stream->ptr;
|
|
|
|
if(mp3info->pcmbuf)
|
|
Z_Free(mp3info->pcmbuf);
|
|
|
|
mad_synth_finish(&mp3info->madsynth);
|
|
mad_frame_finish(&mp3info->madframe);
|
|
mad_stream_finish(&mp3info->madstream);
|
|
|
|
Z_Free(stream->ptr);
|
|
}
|
|
|
|
S_CodecUtilClose(&stream);
|
|
}
|
|
|
|
/*
|
|
=================
|
|
S_MP3_CodecReadStream
|
|
=================
|
|
*/
|
|
int S_MP3_CodecReadStream(snd_stream_t *stream, int bytes, void *buffer)
|
|
{
|
|
struct snd_codec_mp3_info *mp3info;
|
|
int retval;
|
|
|
|
if(!stream)
|
|
return -1;
|
|
|
|
mp3info = stream->ptr;
|
|
|
|
// Make sure we get complete frames all the way through.
|
|
bytes -= bytes % (stream->info.channels * stream->info.width);
|
|
|
|
if(mp3info->buflen)
|
|
{
|
|
if(bytes < mp3info->buflen)
|
|
{
|
|
// we still have enough bytes in our decoded pcm buffer
|
|
memcpy(buffer, mp3info->pcmbuf, bytes);
|
|
|
|
// remove the portion from our buffer.
|
|
mp3info->buflen -= bytes;
|
|
memmove(mp3info->pcmbuf, &mp3info->pcmbuf[bytes], mp3info->buflen);
|
|
return bytes;
|
|
}
|
|
else
|
|
{
|
|
// copy over the samples we already have.
|
|
memcpy(buffer, mp3info->pcmbuf, mp3info->buflen);
|
|
mp3info->destlen = mp3info->buflen;
|
|
mp3info->buflen = 0;
|
|
}
|
|
}
|
|
else
|
|
mp3info->destlen = 0;
|
|
|
|
mp3info->dest = buffer;
|
|
mp3info->destsize = bytes;
|
|
|
|
do
|
|
{
|
|
retval = S_MP3_Decode(stream);
|
|
} while(retval > 0);
|
|
|
|
// if there was an error return nothing.
|
|
if(retval < 0)
|
|
return 0;
|
|
|
|
return mp3info->destlen;
|
|
}
|
|
|
|
/*
|
|
=====================================================================
|
|
S_MP3_CodecLoad
|
|
|
|
We handle S_MP3_CodecLoad as a special case of the streaming functions
|
|
where we read the whole stream at once.
|
|
======================================================================
|
|
*/
|
|
void *S_MP3_CodecLoad(const char *filename, snd_info_t *info)
|
|
{
|
|
snd_stream_t *stream;
|
|
byte *pcmbuffer;
|
|
|
|
// check if input is valid
|
|
if(!filename)
|
|
return NULL;
|
|
|
|
stream = S_MP3_CodecOpenStream(filename);
|
|
|
|
if(!stream)
|
|
return NULL;
|
|
|
|
// copy over the info
|
|
info->rate = stream->info.rate;
|
|
info->width = stream->info.width;
|
|
info->channels = stream->info.channels;
|
|
info->samples = stream->info.samples;
|
|
info->dataofs = stream->info.dataofs;
|
|
|
|
// allocate enough buffer for all pcm data
|
|
pcmbuffer = Hunk_AllocateTempMemory(stream->info.size);
|
|
if(!pcmbuffer)
|
|
{
|
|
S_MP3_CodecCloseStream(stream);
|
|
return NULL;
|
|
}
|
|
|
|
info->size = S_MP3_CodecReadStream(stream, stream->info.size, pcmbuffer);
|
|
|
|
if(info->size <= 0)
|
|
{
|
|
// we didn't read anything at all. darn.
|
|
Hunk_FreeTempMemory(pcmbuffer);
|
|
pcmbuffer = NULL;
|
|
}
|
|
|
|
S_MP3_CodecCloseStream(stream);
|
|
|
|
return pcmbuffer;
|
|
}
|
|
|
|
#endif // USE_CODEC_MP3
|