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8184225473
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@15 fc73d0e0-1445-4013-8a0c-d673dee63da5
497 lines
12 KiB
C
497 lines
12 KiB
C
//source file by david walton, derived from minimad.c
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//modified slightly to use new library functions instead for single threaded asyncronous mp3 sound.
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//in Quake!
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//Be aware that MP3s are patented and thus not GPLable. This file is an interface to madlib which contains the decoder.
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//This file is excluded from the main source tree because of this, and madlib is not distributed - in binary or source form.
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//The GPL disallows distribution of patented software. You may reactive, but do not distribute binaries.
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//Really madlib should come with the exception of the mp3 patent.
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#include "quakedef.h"
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#ifdef AVAIL_MP3
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#include "winquake.h"
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#undef channels
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#ifdef _WIN32
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#define inline _inline
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#endif
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//the problem with this is that we store the entire decoded file in memory.
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//this takes a lot of mem.
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//uses madlib
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/*
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* mad - MPEG audio decoder
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* Copyright (C) 2000-2001 Robert Leslie
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*
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*/
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# include "../../mp3/libmad/mad.h"
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/*
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* This is perhaps the simplest example use of the MAD high-level API.
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* Standard input is mapped into memory via mmap(), then the high-level API
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* is invoked with three callbacks: input, output, and error. The output
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* callback converts MAD's high-resolution PCM samples to 16 bits, then
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* writes them to standard output in little-endian, stereo-interleaved
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* format.
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*/
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/*
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* This is a private message structure. A generic pointer to this structure
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* is passed to each of the callback functions. Put here any data you need
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* to access from within the callbacks.
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*/
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typedef struct {
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unsigned char *start;
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unsigned long length;
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qboolean failed;
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sfxcache_t mp3sc;
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char *mp3aswavdata;
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int mp3aswavpos;
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int mp3aswavbuflen;
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struct mad_decoder decoder;
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sfx_t *s;
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} decoderbuffer_t;
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int mymad_run(struct mad_decoder *decoder);
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int mymad_reset(struct mad_decoder *decoder);
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void mymad_finish(struct mad_decoder *decoder);
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int startdecode(unsigned char *start, unsigned long length, decoderbuffer_t *buffer);
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#define BUFFERSIZEINC (2*1024*1024)
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int DecodeSomeMP3(sfx_t *s, int minlength);
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byte *COM_LoadFile (char *path, int usehunk);
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int MP3_decode(unsigned char *start, unsigned long length, decoderbuffer_t *buffer);
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void CancelDecoder(sfx_t *s);
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sfxcache_t *S_LoadMP3Sound (sfx_t *s)
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{
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char namebuffer[MAX_OSPATH];
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char *name;
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decoderbuffer_t *buffer;
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char *data;
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qboolean telluser;
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int len;
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name = s->name;
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if (name[0] == '#')
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strcpy(namebuffer, &name[1]);
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else
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sprintf (namebuffer, "sound/%s", name);
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len = strlen(namebuffer);
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telluser = strcmp(namebuffer+len-4, ".wav");
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if (!telluser)
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strcpy(namebuffer+len-4, ".mp3");
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//try opening from a quake path
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data = COM_LoadFile(namebuffer, 5);
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if (!data)
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{//if that didn't work, try opening direct from exe - this is media after all.
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FILE *f;
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if (!telluser)
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return NULL; //never go out of the quake path for a wav replacement.
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#ifndef _WIN32
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char unixname[128];
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if (name[1] == ':' && name[2] == '\\') //convert from windows to a suitable alternative.
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{
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sprintf(unixname, "/mnt/%c/%s", name[0]-'A'+'a', name+3);
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name = unixname;
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while (*name)
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{
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if (*name == '\\')
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*name = '/';
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name++;
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}
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name = unixname;
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}
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#endif
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if ((f = fopen(name, "rb")))
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{
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com_filesize = COM_filelength(f);
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data = BZ_Malloc(com_filesize);
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fread(data, 1, com_filesize, f);
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fclose(f);
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}
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else
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{
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Con_SafePrintf ("Couldn't load %s\n", namebuffer);
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return NULL;
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}
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}
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if (!s->decoder)
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s->decoder = Z_Malloc(sizeof(decoderbuffer_t) + sizeof(sfxdecode_t));
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buffer = (decoderbuffer_t*)(s->decoder+1);
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buffer->mp3aswavpos=0;
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buffer->mp3sc.length=0;
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buffer->s = s;
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s->decoder->buf = buffer;
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s->decoder->decodemore = DecodeSomeMP3;
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s->decoder->abort = CancelDecoder;
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if (!startdecode(data, com_filesize, buffer))
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{
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if (buffer->mp3aswavdata)
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{
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BZ_Free(buffer->mp3aswavdata);
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buffer->mp3aswavdata=NULL;
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}
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Con_SafePrintf ("Couldn't load %s - corrupt.\n", namebuffer);
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return NULL;
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}
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s->decoder->decodemore(s, 100);
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s->cache.fake=true;
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return buffer->s->cache.data;
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}
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/*
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* This is the input callback. The purpose of this callback is to (re)fill
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* the stream buffer which is to be decoded. In this example, an entire file
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* has been mapped into memory, so we just call mad_stream_buffer() with the
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* address and length of the mapping. When this callback is called a second
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* time, we are finished decoding.
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*/
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static
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enum mad_flow input(void *data,
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struct mad_stream *stream)
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{
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decoderbuffer_t *buffer = data;
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if (!buffer->length)
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return MAD_FLOW_STOP;
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mad_stream_buffer(stream, buffer->start, buffer->length);
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buffer->length = 0;
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return MAD_FLOW_CONTINUE;
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}
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/*
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* The following utility routine performs simple rounding, clipping, and
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* scaling of MAD's high-resolution samples down to 16 bits. It does not
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* perform any dithering or noise shaping, which would be recommended to
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* obtain any exceptional audio quality. It is therefore not recommended to
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* use this routine if high-quality output is desired.
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*/
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static inline
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signed int scale(mad_fixed_t sample)
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{
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/* round */
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sample += (1L << (MAD_F_FRACBITS - 16));
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/* clip */
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if (sample >= MAD_F_ONE)
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sample = MAD_F_ONE - 1;
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else if (sample < -MAD_F_ONE)
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sample = -MAD_F_ONE;
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/* quantize */
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return sample >> (MAD_F_FRACBITS + 1 - 16);
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}
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/*
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* This is the output callback function. It is called after each frame of
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* MPEG audio data has been completely decoded. The purpose of this callback
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* is to output (or play) the decoded PCM audio.
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*/
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static
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enum mad_flow output(void *data,
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struct mad_header const *header,
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struct mad_pcm *pcm)
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{
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decoderbuffer_t *buffer = data;
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sfxcache_t *cache;
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unsigned int nchannels, nsamples;
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mad_fixed_t const *left_ch, *right_ch;
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char *outpos;
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signed int sample;
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float speedfactor;
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speedfactor = (float)snd_speed/buffer->mp3sc.speed;
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/* pcm->samplerate contains the sampling frequency */
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nchannels = pcm->channels;
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nsamples = pcm->length*speedfactor;
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left_ch = pcm->samples[0];
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right_ch = pcm->samples[1];
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if (buffer->mp3aswavpos + nsamples*nchannels*sizeof(short) > buffer->mp3aswavbuflen || !buffer->mp3aswavdata)
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{
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int newsize = buffer->mp3aswavpos + nsamples*nchannels*sizeof(short) + BUFFERSIZEINC+sizeof(sfxcache_t);
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char *newbuf = BZ_Malloc(newsize);
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if (!newbuf)
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return MAD_FLOW_STOP; //damn and blast!
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if (buffer->mp3aswavdata) //if we had an old buffer, fill the new one and free the old
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{
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// memset(newbuf, 0, newsize);
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memcpy(newbuf, buffer->mp3aswavdata, buffer->mp3aswavpos+sizeof(sfxcache_t));
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// newbuf[newsize] = '\0';
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BZ_Free(buffer->mp3aswavdata);
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}
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buffer->mp3aswavdata = newbuf;
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buffer->mp3aswavbuflen = newsize;
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buffer->s->cache.data = buffer->mp3aswavdata;
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}
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cache = (sfxcache_t *)buffer->mp3aswavdata;
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outpos = buffer->mp3aswavdata+sizeof(sfxcache_t) + buffer->mp3aswavpos;
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if (nchannels == 1)
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{
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if (speedfactor==1) //fast
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{
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while (nsamples--)
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{
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sample = scale(*left_ch++);
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*outpos++ = ((sample >> 0) & 0xff);
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*outpos++ = ((sample >> 8) & 0xff);
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}
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}
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else
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{
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int src=0;
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int pos=0;
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while (pos < nsamples)
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{
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src = pos/speedfactor;
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sample = scale(left_ch[src]);
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*outpos++ = (((sample >> 0) & 0xff));
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*outpos++ = (((sample >> 8) & 0xff));
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pos++;
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}
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}
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cache->stereo = 0;
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}
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else
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{
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if (speedfactor==1) //fast
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{
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while (nsamples--)
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{ //merge channels?
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sample = scale(*left_ch++);
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*outpos++ = (((sample >> 0) & 0xff));
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*outpos++ = (((sample >> 8) & 0xff));
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sample = scale(*right_ch++)/2;
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*outpos++ = (((sample >> 0) & 0xff));
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*outpos++ = (((sample >> 8) & 0xff));
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}
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}
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else
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{
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int src=0;
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int pos=0;
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while (pos < nsamples)
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{
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src = pos/speedfactor;
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sample = scale(left_ch[src]);
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*outpos++ = (((sample >> 0) & 0xff));
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*outpos++ = (((sample >> 8) & 0xff));
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sample = scale(right_ch[src]);
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*outpos++ = (((sample >> 0) & 0xff));
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*outpos++ = (((sample >> 8) & 0xff));
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pos++;
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}
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}
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cache->stereo = 1;
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}
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buffer->mp3aswavpos += pcm->length*speedfactor*sizeof(short)*nchannels;
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buffer->mp3sc.length+= pcm->length*speedfactor;
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cache->speed = snd_speed;
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cache->length = buffer->s->decoder->decodedlen = buffer->mp3sc.length;
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cache->loopstart = -1;//buffer->mp3sc.loopstart;
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cache->width = buffer->mp3sc.width;
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return MAD_FLOW_CONTINUE;
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}
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/*
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* This is the error callback function. It is called whenever a decoding
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* error occurs. The error is indicated by stream->error; the list of
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* possible MAD_ERROR_* errors can be found in the mad.h (or
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* libmad/stream.h) header file.
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*/
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static
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enum mad_flow error(void *data,
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struct mad_stream *stream,
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struct mad_frame *frame)
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{
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decoderbuffer_t *buffer = data;
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if (stream->error == 257)
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return MAD_FLOW_IGNORE;
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if (developer.value)
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Con_SafePrintf("MP3Error: decoding error 0x%04x (%s) at byte offset %u\n",
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stream->error, mad_stream_errorstr(stream),
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stream->this_frame - buffer->start);
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// buffer->failed = true;
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return MAD_FLOW_IGNORE;
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}
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static enum mad_flow header(void *data,
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struct mad_header const *header)
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{
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decoderbuffer_t *buffer = data;
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buffer->mp3sc.speed = header->samplerate;
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buffer->mp3sc.width = 2;
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return MAD_FLOW_CONTINUE;
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}
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void CancelDecoder(sfx_t *s)
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{
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decoderbuffer_t *dec = s->decoder->buf;
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mymad_finish(&dec->decoder);
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mad_decoder_finish(&dec->decoder);
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s->cache.fake = false; //give it a true cache now, and hope that we don't need to free it while it's still playing.
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s->cache.data=NULL;
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BZ_Free(dec->mp3aswavdata);
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if (dec->start)
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BZ_Free(dec->start);
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Z_Free(s->decoder);
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s->decoder = NULL;
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}
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/*
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* This is the function called by main() above to perform all the
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* decoding. It instantiates a decoder object and configures it with the
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* input, output, and error callback functions above. A single call to
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* mad_decoder_run() continues until a callback function returns
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* MAD_FLOW_STOP (to stop decoding) or MAD_FLOW_BREAK (to stop decoding and
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* signal an error).
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*/
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int DecodeSomeMP3(sfx_t *s, int minlength)
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{
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decoderbuffer_t *dec = s->decoder->buf;
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// int oldlen;
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if (!dec->start)
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return 1;
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while(dec->mp3sc.length < minlength)
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{
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if (!mymad_run(&dec->decoder) || dec->failed)
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{
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if (dec->mp3aswavbuflen>441000)//10 secs at highest quality
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{ //big fat heffers of mp3s are kept like this and are flushed when they finish playing fully.
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BZ_Free(dec->start);
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dec->start = NULL;
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}
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else
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{ //small mp3 files are now treated like wavs
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void *newmem;
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mymad_finish(&dec->decoder);
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mad_decoder_finish(&dec->decoder);
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s->cache.fake = false; //give it a true cache now, and hope that we don't need to free it while it's still playing.
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s->cache.data=NULL;
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newmem = Cache_Alloc(&s->cache, dec->mp3aswavbuflen+sizeof(sfxcache_t), s->name);
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if (dec->mp3aswavdata)
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{
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memcpy(newmem, dec->mp3aswavdata, dec->mp3aswavbuflen+sizeof(sfxcache_t));
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BZ_Free(dec->mp3aswavdata);
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}
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BZ_Free(dec->start);
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Z_Free(s->decoder);
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s->decoder = NULL;
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}
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return 1;
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}
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}
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return 0;
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}
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int mymad_decoder_run(struct mad_decoder *decoder, enum mad_decoder_mode mode);
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static
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int startdecode(unsigned char *start, unsigned long length, decoderbuffer_t *buffer)
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{
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/* initialize our private message structure */
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buffer->start = start;
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buffer->length = length;
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buffer->failed = false;
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/* configure input, output, and error functions */
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mad_decoder_init(&buffer->decoder, buffer,
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input, header, 0 /* filter */, output,
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error, 0 /* message */);
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/* start decoding */
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mymad_reset(&buffer->decoder);
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if (buffer->failed)
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return false;
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return true;
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}
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#endif
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