aboutsummaryrefslogtreecommitdiff
path: root/games/ioquake3/files/extra-patch-mp3
diff options
context:
space:
mode:
Diffstat (limited to 'games/ioquake3/files/extra-patch-mp3')
-rw-r--r--games/ioquake3/files/extra-patch-mp3753
1 files changed, 0 insertions, 753 deletions
diff --git a/games/ioquake3/files/extra-patch-mp3 b/games/ioquake3/files/extra-patch-mp3
deleted file mode 100644
index b33f2a660571..000000000000
--- a/games/ioquake3/files/extra-patch-mp3
+++ /dev/null
@@ -1,753 +0,0 @@
-Index: code/client/snd_codec.c
-===================================================================
---- code/client/snd_codec.c (revision 917)
-+++ code/client/snd_codec.c (working copy)
-@@ -105,6 +105,9 @@
- #if USE_CODEC_VORBIS
- S_CodecRegister(&ogg_codec);
- #endif
-+#if USE_CODEC_MP3
-+ S_CodecRegister(&mp3_codec);
-+#endif
- }
-
- /*
-Index: code/client/snd_codec.h
-===================================================================
---- code/client/snd_codec.h (revision 917)
-+++ code/client/snd_codec.h (working copy)
-@@ -95,4 +95,13 @@
- int S_OGG_CodecReadStream(snd_stream_t *stream, int bytes, void *buffer);
- #endif // USE_CODEC_VORBIS
-
-+// MP3 codec
-+#ifdef USE_CODEC_MP3
-+extern snd_codec_t mp3_codec;
-+void *S_MP3_CodecLoad(const char *filename, snd_info_t *info);
-+snd_stream_t *S_MP3_CodecOpenStream(const char *filename);
-+void S_MP3_CodecCloseStream(snd_stream_t *stream);
-+int S_MP3_CodecReadStream(snd_stream_t *stream, int bytes, void *buffer);
-+#endif // USE_CODEC_MP3
-+
- #endif // !_SND_CODEC_H_
-Index: code/client/snd_codec_mp3.c
-===================================================================
---- code/client/snd_codec_mp3.c (revision 0)
-+++ code/client/snd_codec_mp3.c (revision 0)
-@@ -0,0 +1,716 @@
-+/*
-+===========================================================================
-+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
-+#if 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 = Z_Malloc(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.
-+ Z_Free(pcmbuffer);
-+ pcmbuffer = NULL;
-+ }
-+
-+ S_MP3_CodecCloseStream(stream);
-+
-+ return pcmbuffer;
-+}
-+
-+#endif // USE_CODEC_MP3
generated by cgit v1.2.3 (git 2.25.1) at 2025-03-01 17:36:50 +0000