748 lines
22 KiB
C
748 lines
22 KiB
C
/*____________________________________________________________________________
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FreeAmp - The Free MP3 Player
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MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
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Corp. http://www.xingtech.com
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Portions Copyright (C) 1998-1999 EMusic.com
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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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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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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., 675 Mass Ave, Cambridge, MA 02139, USA.
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$Id: towave.c,v 1.3 1999/10/19 07:13:09 elrod Exp $
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____________________________________________________________________________*/
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/* ------------------------------------------------------------------------
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NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE
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This file exists for reference only. It is not actually used
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in the FreeAmp project. There is no need to mess with this
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file. There is no need to flatten the beavers, either.
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NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE NOTE
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/*---- towave.c --------------------------------------------
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32 bit version only
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decode mpeg Layer I/II/III file using portable ANSI C decoder,
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output to pcm wave file.
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mod 8/19/98 decode 22 sf bands
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mod 5/14/98 allow mpeg25 (dec8 not supported for mpeg25 samp rate)
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mod 3/4/98 bs_trigger bs_bufbytes made signed, unsigned may
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not terminate properly. Also extra test in bs_fill.
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mod 8/6/96 add 8 bit output to standard decoder
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ver 1.4 mods 7/18/96 32 bit and add asm option
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mods 6/29/95 allow MS wave file for u-law. bugfix u-law table dec8.c
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mods 2/95 add sample rate reduction, freq_limit and conversions.
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add _decode8 for 8Ks output, 16bit 8bit, u-law output.
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add additional control parameters to init.
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add _info function
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mod 5/12/95 add quick window cwinq.c
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mod 5/19/95 change from stream io to handle io
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mod 11/16/95 add Layer I
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mod 1/5/95 integer overflow mod iup.c
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ver 1.3
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mod 2/5/96 portability mods
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drop Tom and Gloria pcm file types
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ver 2.0
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mod 1/7/97 Layer 3 (float mpeg-1 only)
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2/6/97 Layer 3 MPEG-2
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ver 3.01 Layer III bugfix crc problem 8/18/97
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ver 3.02 Layer III fix wannabe.mp3 problem 10/9/97
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ver 3.03 allow mpeg 2.5 5/14/98
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Decoder functions for _decode8 are defined in dec8.c. Useage
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is same as regular decoder.
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Towave illustrates use of decoder. Towave converts
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mpeg audio files to 16 bit (short) pcm. Default pcm file
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format is wave. Other formats can be accommodated by
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adding alternative write_pcm_header and write_pcm_tailer
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functions. The functions kbhit and getch used in towave.c
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may not port to other systems.
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The decoder handles all mpeg1 and mpeg2 Layer I/II bitstreams.
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For compatability with the asm decoder and future C versions,
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source code users are discouraged from making modifications
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to the decoder proper. MS Windows applications can use wrapper
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functions in a separate module if decoder functions need to
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be exported.
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NOTE additional control parameters.
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mod 8/6/96 standard decoder adds 8 bit output
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decode8 (8Ks output) convert_code:
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convert_code = 4*bit_code + chan_code
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bit_code: 1 = 16 bit linear pcm
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2 = 8 bit (unsigned) linear pcm
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3 = u-law (8 bits unsigned)
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chan_code: 0 = convert two chan to mono
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1 = convert two chan to mono
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2 = convert two chan to left chan
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3 = convert two chan to right chan
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decode (standard decoder) convert_code:
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0 = two chan output
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1 = convert two chan to mono
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2 = convert two chan to left chan
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3 = convert two chan to right chan
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or with 8 = 8 bit output
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(other bits ignored)
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decode (standard decoder) reduction_code:
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0 = full sample rate output
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1 = half rate
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2 = quarter rate
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-----------------------------------------------------------*/
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#include <stdlib.h>
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#include <stdio.h>
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#include <float.h>
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#include <math.h>
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#include <string.h>
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#ifdef WIN32
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#include <io.h>
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#endif
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#include <fcntl.h> /* file open flags */
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#include <sys/types.h> /* someone wants for port */
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#include <sys/stat.h> /* forward slash for portability */
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#include "mhead.h" /* mpeg header structure, decode protos */
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#include "port.h"
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// JDW
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#ifdef __linux__
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#include <sys/ioctl.h>
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#include <sys/soundcard.h>
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#include <fcntl.h>
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#include <errno.h>
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#endif
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// JDW
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#include "mp3struct.h"
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#include <assert.h>
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typedef struct id3v1_1 {
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char id[3];
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char title[30]; // <file basename>
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char artist[30]; // "Raven Software"
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char album[30]; // "#UNCOMP %d" // needed
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char year[4]; // "2000"
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char comment[28]; // "#MAXVOL %g" // needed
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char zero;
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char track;
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char genre;
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} id3v1_1; // 128 bytes in size
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id3v1_1 *gpTAG;
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#define BYTESREMAINING_ACCOUNT_FOR_REAR_TAG(_pvData, _iBytesRemaining) \
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\
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/* account for trailing ID3 tag in _iBytesRemaining */ \
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gpTAG = (id3v1_1*) (((byte *)_pvData + _iBytesRemaining)-sizeof(id3v1_1)); /* sizeof = 128 */ \
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if (!strncmp(gpTAG->id, "TAG", 3)) \
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{ \
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_iBytesRemaining -= sizeof(id3v1_1); \
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}
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/******** pcm buffer ********/
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#define PCM_BUFBYTES 60000U // more than enough to cover the largest that one packet will ever expand to
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char PCM_Buffer[PCM_BUFBYTES]; // better off being declared, so we don't do mallocs in this module (MAC reasons)
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typedef struct
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{
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int (*decode_init) (MPEG_HEAD * h, int framebytes_arg,
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int reduction_code, int transform_code,
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int convert_code, int freq_limit);
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void (*decode_info) (DEC_INFO * info);
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IN_OUT(*decode) (unsigned char *bs, short *pcm, unsigned char *pNextByteAfterData);
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}
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AUDIO;
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#if 0
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// stuff this...
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static AUDIO audio_table[2][2] =
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{
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{
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{audio_decode_init, audio_decode_info, audio_decode},
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{audio_decode8_init, audio_decode8_info, audio_decode8},
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},
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{
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{i_audio_decode_init, i_audio_decode_info, i_audio_decode},
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{audio_decode8_init, audio_decode8_info, audio_decode8},
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}
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};
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#else
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static AUDIO audio_table[2][2] =
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{
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{
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{audio_decode_init, audio_decode_info, audio_decode},
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{audio_decode_init, audio_decode_info, audio_decode},
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},
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{
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{audio_decode_init, audio_decode_info, audio_decode},
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{audio_decode_init, audio_decode_info, audio_decode},
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}
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};
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#endif
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static const AUDIO audio = {audio_decode_init, audio_decode_info, audio_decode}; //audio_table[0][0];
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// Do NOT change these, ever!!!!!!!!!!!!!!!!!!
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//
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const int reduction_code = 0; // unpack at full sample rate output
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const int convert_code_mono = 1;
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const int convert_code_stereo = 0;
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const int freq_limit = 24000; // no idea what this is about, but it's always this value so...
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// the entire decode mechanism uses this now...
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//
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MP3STREAM _MP3Stream;
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LP_MP3STREAM pMP3Stream = &_MP3Stream;
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int bFastEstimateOnly = 0; // MUST DEFAULT TO THIS VALUE!!!!!!!!!
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// char *return is NZ for any errors (no trailing CR!)
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//
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char *C_MP3_IsValid(void *pvData, int iDataLen, int bStereoDesired)
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{
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// char sTemp[1024]; /////////////////////////////////////////////////
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unsigned int iRealDataStart;
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MPEG_HEAD head;
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DEC_INFO decinfo;
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int iBitRate;
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int iFrameBytes;
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//#ifdef _DEBUG
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// int iIgnoreThisForNowIJustNeedItToBreakpointOnToReadAValue = sizeof(MP3STREAM);
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//#endif
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memset(pMP3Stream,0,sizeof(*pMP3Stream));
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iFrameBytes = head_info3( pvData, iDataLen/2, &head, &iBitRate, &iRealDataStart);
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if (iFrameBytes == 0)
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{
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return "MP3ERR: Bad or unsupported file!";
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}
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// check for files with bad frame unpack sizes (that would crash the game), or stereo files.
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//
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// although the decoder can convert stereo to mono (apparently), we want to know about stereo files
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// because they're a waste of source space... (all FX are mono, and moved via panning)
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//
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if (audio.decode_init(&head, iFrameBytes, reduction_code, iRealDataStart, bStereoDesired?convert_code_stereo:convert_code_mono, freq_limit))
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{
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if (bStereoDesired)
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{
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if (pMP3Stream->outbytes > 4608)
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{
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return "MP3ERR: Source file has output packet size > 2304 (*2 for stereo) bytes!";
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}
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}
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else
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{
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if (pMP3Stream->outbytes > 2304)
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{
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return "MP3ERR: Source file has output packet size > 2304 bytes!";
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}
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}
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audio.decode_info(&decinfo);
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if (decinfo.bits != 16)
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{
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return "MP3ERR: Source file is not 16bit!"; // will this ever happen? oh well...
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}
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if (decinfo.samprate != 44100)
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{
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return "MP3ERR: Source file is not sampled @ 44100!";
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}
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if (bStereoDesired && decinfo.channels != 2)
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{
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return "MP3ERR: Source file is not stereo!"; // sod it, I'm going to count this as an error now
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}
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}
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else
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{
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return "MP3ERR: Decoder failed to initialise";
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}
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// file seems to be valid...
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//
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return NULL;
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}
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// char *return is NZ for any errors (no trailing CR!)
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//
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char* C_MP3_GetHeaderData(void *pvData, int iDataLen, int *piRate, int *piWidth, int *piChannels, int bStereoDesired)
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{
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unsigned int iRealDataStart;
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MPEG_HEAD head;
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DEC_INFO decinfo;
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int iBitRate;
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int iFrameBytes;
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memset(pMP3Stream,0,sizeof(*pMP3Stream));
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iFrameBytes = head_info3( pvData, iDataLen/2, &head, &iBitRate, &iRealDataStart);
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if (iFrameBytes == 0)
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{
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return "MP3ERR: Bad or unsupported file!";
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}
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if (audio.decode_init(&head, iFrameBytes, reduction_code, iRealDataStart, bStereoDesired?convert_code_stereo:convert_code_mono, freq_limit))
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{
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audio.decode_info(&decinfo);
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*piRate = decinfo.samprate; // rate (eg 22050, 44100 etc)
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*piWidth = decinfo.bits/8; // 1 for 8bit, 2 for 16 bit
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*piChannels = decinfo.channels; // 1 for mono, 2 for stereo
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}
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else
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{
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return "MP3ERR: Decoder failed to initialise";
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}
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// everything ok...
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//
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return NULL;
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}
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// this duplicates work done in C_MP3_IsValid(), but it avoids global structs, and means that you can call this anytime
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// if you just want info for some reason
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//
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// ( size is now workd out just by decompressing each packet header, not the whole stream. MUCH faster :-)
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//
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// char *return is NZ for any errors (no trailing CR!)
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//
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char *C_MP3_GetUnpackedSize(void *pvData, int iSourceBytesRemaining, int *piUnpackedSize, int bStereoDesired )
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{
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int iReadLimit;
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unsigned int iRealDataStart;
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MPEG_HEAD head;
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int iBitRate;
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char *pPCM_Buffer = PCM_Buffer;
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char *psReturn = NULL;
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// int iSourceReadIndex = 0;
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int iDestWriteIndex = 0;
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int iFrameBytes;
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int iFrameCounter;
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DEC_INFO decinfo;
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IN_OUT x;
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memset(pMP3Stream,0,sizeof(*pMP3Stream));
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#define iSourceReadIndex iRealDataStart
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// iFrameBytes = head_info2( pvData, 0, &head, &iBitRate);
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iFrameBytes = head_info3( pvData, iSourceBytesRemaining/2, &head, &iBitRate, &iRealDataStart);
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BYTESREMAINING_ACCOUNT_FOR_REAR_TAG(pvData, iSourceBytesRemaining)
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iSourceBytesRemaining -= iRealDataStart;
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iReadLimit = iSourceReadIndex + iSourceBytesRemaining;
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if (iFrameBytes)
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{
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//pPCM_Buffer = malloc(PCM_BUFBYTES);
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//if (pPCM_Buffer)
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{
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// init decoder...
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if (audio.decode_init(&head, iFrameBytes, reduction_code, iRealDataStart, bStereoDesired?convert_code_stereo:convert_code_mono, freq_limit))
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{
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audio.decode_info(&decinfo);
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// decode...
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//
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for (iFrameCounter = 0;;iFrameCounter++)
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{
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if ( iSourceBytesRemaining == 0 || iSourceBytesRemaining < iFrameBytes)
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break; // end of file
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bFastEstimateOnly = 1; ///////////////////////////////
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x = audio.decode((unsigned char *)pvData + iSourceReadIndex, (short *) ((char *)pPCM_Buffer
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//+ iDestWriteIndex // keep decoding over the same spot since we're only counting bytes in this function
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),
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(unsigned char *)pvData + iReadLimit
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);
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bFastEstimateOnly = 0; ///////////////////////////////
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iSourceReadIndex += x.in_bytes;
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iSourceBytesRemaining -= x.in_bytes;
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iDestWriteIndex += x.out_bytes;
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if (x.in_bytes <= 0)
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{
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//psReturn = "MP3ERR: Bad sync in file";
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break;
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}
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}
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*piUnpackedSize = iDestWriteIndex; // yeeehaaa!
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}
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else
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{
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psReturn = "MP3ERR: Decoder failed to initialise";
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}
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}
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// else
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// {
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// psReturn = "MP3ERR: Unable to alloc temp decomp buffer";
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// }
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}
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else
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{
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psReturn = "MP3ERR: Bad or Unsupported MP3 file!";
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}
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// if (pPCM_Buffer)
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// {
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// free(pPCM_Buffer);
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// pPCM_Buffer = NULL; // I know, I know...
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// }
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return psReturn;
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#undef iSourceReadIndex
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}
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char *C_MP3_UnpackRawPCM( void *pvData, int iSourceBytesRemaining, int *piUnpackedSize, void *pbUnpackBuffer, int bStereoDesired)
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{
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int iReadLimit;
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unsigned int iRealDataStart;
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MPEG_HEAD head;
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int iBitRate;
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char *psReturn = NULL;
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// int iSourceReadIndex = 0;
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int iDestWriteIndex = 0;
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int iFrameBytes;
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int iFrameCounter;
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DEC_INFO decinfo;
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IN_OUT x;
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memset(pMP3Stream,0,sizeof(*pMP3Stream));
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#define iSourceReadIndex iRealDataStart
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// iFrameBytes = head_info2( pvData, 0, &head, &iBitRate);
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iFrameBytes = head_info3( pvData, iSourceBytesRemaining/2, &head, &iBitRate, &iRealDataStart);
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BYTESREMAINING_ACCOUNT_FOR_REAR_TAG(pvData, iSourceBytesRemaining)
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iSourceBytesRemaining -= iRealDataStart;
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iReadLimit = iSourceReadIndex + iSourceBytesRemaining;
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if (iFrameBytes)
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{
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// if (1)////////////////////////pPCM_Buffer)
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{
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// init decoder...
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if (audio.decode_init(&head, iFrameBytes, reduction_code, iRealDataStart, bStereoDesired?convert_code_stereo:convert_code_mono, freq_limit))
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{
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audio.decode_info(&decinfo);
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// printf("\n output samprate = %6ld",decinfo.samprate);
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// printf("\n output channels = %6d", decinfo.channels);
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// printf("\n output bits = %6d", decinfo.bits);
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// printf("\n output type = %6d", decinfo.type);
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//===============
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// decode...
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//
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for (iFrameCounter = 0;;iFrameCounter++)
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{
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if ( iSourceBytesRemaining == 0 || iSourceBytesRemaining < iFrameBytes)
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break; // end of file
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x = audio.decode((unsigned char *)pvData + iSourceReadIndex, (short *) ((char *)pbUnpackBuffer + iDestWriteIndex),
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(unsigned char *)pvData + iReadLimit
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);
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iSourceReadIndex += x.in_bytes;
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iSourceBytesRemaining -= x.in_bytes;
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iDestWriteIndex += x.out_bytes;
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if (x.in_bytes <= 0)
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{
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//psReturn = "MP3ERR: Bad sync in file";
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break;
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}
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}
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*piUnpackedSize = iDestWriteIndex; // yeeehaaa!
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}
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else
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{
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psReturn = "MP3ERR: Decoder failed to initialise";
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}
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}
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}
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else
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{
|
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psReturn = "MP3ERR: Bad or Unsupported MP3 file!";
|
|
}
|
|
|
|
return psReturn;
|
|
|
|
#undef iSourceReadIndex
|
|
}
|
|
|
|
|
|
// called once, after we've decided to keep something as MP3. This just sets up the decoder for subsequent stream-calls.
|
|
//
|
|
// (the struct pSFX_MP3Stream is cleared internally, so pass as args anything you want stored in it)
|
|
//
|
|
// char * return is NULL for ok, else error string
|
|
//
|
|
// NEW CODE NOTE: Now that this function is being called for raw data that's going to be runtime-stored in a link-list
|
|
// chunk format for SOF2 instead of just one alloc then you must re-init pMP3Stream->pbSourceData after you've called
|
|
// this, or it'll be pointing at the deallocated original raw data, not the first chunk of the link list
|
|
//
|
|
char *C_MP3Stream_DecodeInit( LP_MP3STREAM pSFX_MP3Stream, void *pvSourceData, int iSourceBytesRemaining,
|
|
int iGameAudioSampleRate, int iGameAudioSampleBits, int bStereoDesired )
|
|
{
|
|
char *psReturn = NULL;
|
|
MPEG_HEAD head; // only relevant within this function during init
|
|
DEC_INFO decinfo; // " "
|
|
int iBitRate; // not used after being filled in by head_info3()
|
|
|
|
pMP3Stream = pSFX_MP3Stream;
|
|
|
|
memset(pMP3Stream,0,sizeof(*pMP3Stream));
|
|
|
|
pMP3Stream->pbSourceData = (byte *) pvSourceData; // this MUST be re-initialised to link-mem outside here for SOF2, since raw data is now link-listed
|
|
pMP3Stream->iSourceBytesRemaining = iSourceBytesRemaining;
|
|
pMP3Stream->iSourceFrameBytes = head_info3( (byte *) pvSourceData, iSourceBytesRemaining/2, &head, &iBitRate, (unsigned int*)&pMP3Stream->iSourceReadIndex );
|
|
|
|
// hack, do NOT do this for stereo, since music files are now streamed and therefore the data isn't actually fully
|
|
// loaded at this point, only about 4k or so for the header is actually in memory!!!...
|
|
//
|
|
if (!bStereoDesired)
|
|
{
|
|
BYTESREMAINING_ACCOUNT_FOR_REAR_TAG(pvSourceData, pMP3Stream->iSourceBytesRemaining);
|
|
pMP3Stream->iSourceBytesRemaining -= pMP3Stream->iSourceReadIndex;
|
|
}
|
|
|
|
// backup a couple of fields so we can play this again later...
|
|
//
|
|
pMP3Stream->iRewind_SourceReadIndex = pMP3Stream->iSourceReadIndex;
|
|
pMP3Stream->iRewind_SourceBytesRemaining= pMP3Stream->iSourceBytesRemaining;
|
|
|
|
assert(pMP3Stream->iSourceFrameBytes);
|
|
if (pMP3Stream->iSourceFrameBytes)
|
|
{
|
|
if (audio.decode_init(&head, pMP3Stream->iSourceFrameBytes, reduction_code, pMP3Stream->iSourceReadIndex, bStereoDesired?convert_code_stereo:convert_code_mono, freq_limit))
|
|
{
|
|
pMP3Stream->iRewind_FinalReductionCode = reduction_code; // default = 0 (no reduction), 1=half, 2 = quarter
|
|
|
|
pMP3Stream->iRewind_FinalConvertCode = bStereoDesired?convert_code_stereo:convert_code_mono;
|
|
// default = 1 (mono), OR with 8 for 8-bit output
|
|
|
|
// only now can we ask what kind of properties this file has, and then adjust to fit what the game wants...
|
|
//
|
|
audio.decode_info(&decinfo);
|
|
|
|
// printf("\n output samprate = %6ld",decinfo.samprate);
|
|
// printf("\n output channels = %6d", decinfo.channels);
|
|
// printf("\n output bits = %6d", decinfo.bits);
|
|
// printf("\n output type = %6d", decinfo.type);
|
|
|
|
// decoder offers half or quarter rate adjustement only...
|
|
//
|
|
if (iGameAudioSampleRate == decinfo.samprate>>1)
|
|
pMP3Stream->iRewind_FinalReductionCode = 1;
|
|
else
|
|
if (iGameAudioSampleRate == decinfo.samprate>>2)
|
|
pMP3Stream->iRewind_FinalReductionCode = 2;
|
|
|
|
if (iGameAudioSampleBits == decinfo.bits>>1) // if game wants 8 bit sounds, then setup for that
|
|
pMP3Stream->iRewind_FinalConvertCode |= 8;
|
|
|
|
if (audio.decode_init(&head, pMP3Stream->iSourceFrameBytes, pMP3Stream->iRewind_FinalReductionCode, pMP3Stream->iSourceReadIndex, pMP3Stream->iRewind_FinalConvertCode, freq_limit))
|
|
{
|
|
audio.decode_info(&decinfo);
|
|
#ifdef _DEBUG
|
|
assert( iGameAudioSampleRate == decinfo.samprate );
|
|
assert( iGameAudioSampleBits == decinfo.bits );
|
|
#endif
|
|
|
|
// sod it, no harm in one last check... (should never happen)
|
|
//
|
|
if ( iGameAudioSampleRate != decinfo.samprate || iGameAudioSampleBits != decinfo.bits )
|
|
{
|
|
psReturn = "MP3ERR: Decoder unable to convert to current game audio settings";
|
|
}
|
|
}
|
|
else
|
|
{
|
|
psReturn = "MP3ERR: Decoder failed to initialise for pass 2 sample adjust";
|
|
}
|
|
}
|
|
else
|
|
{
|
|
psReturn = "MP3ERR: Decoder failed to initialise";
|
|
}
|
|
}
|
|
else
|
|
{
|
|
psReturn = "MP3ERR: Errr.... something's broken with this MP3 file"; // should never happen by this point
|
|
}
|
|
|
|
// restore global stream ptr before returning to normal functions (so the rest of the MP3 code still works)...
|
|
//
|
|
pMP3Stream = &_MP3Stream;
|
|
|
|
return psReturn;
|
|
}
|
|
|
|
// return value is decoded bytes for this packet, which is effectively a BOOL, NZ for not finished decoding yet...
|
|
//
|
|
unsigned int C_MP3Stream_Decode( LP_MP3STREAM pSFX_MP3Stream )
|
|
{
|
|
unsigned int uiDecoded = 0; // default to "finished"
|
|
IN_OUT x;
|
|
|
|
pMP3Stream = pSFX_MP3Stream;
|
|
|
|
do
|
|
{
|
|
if ( pSFX_MP3Stream->iSourceBytesRemaining == 0)// || pSFX_MP3Stream->iSourceBytesRemaining < pSFX_MP3Stream->iSourceFrameBytes)
|
|
{
|
|
uiDecoded = 0; // finished
|
|
break;
|
|
}
|
|
|
|
x = audio.decode(pSFX_MP3Stream->pbSourceData + pSFX_MP3Stream->iSourceReadIndex, (short *) (pSFX_MP3Stream->bDecodeBuffer),
|
|
pSFX_MP3Stream->pbSourceData + pSFX_MP3Stream->iRewind_SourceReadIndex + pSFX_MP3Stream->iRewind_SourceBytesRemaining
|
|
);
|
|
|
|
#ifdef _DEBUG
|
|
pSFX_MP3Stream->iSourceFrameCounter++;
|
|
#endif
|
|
|
|
pSFX_MP3Stream->iSourceReadIndex += x.in_bytes;
|
|
pSFX_MP3Stream->iSourceBytesRemaining -= x.in_bytes;
|
|
pSFX_MP3Stream->iBytesDecodedTotal += x.out_bytes;
|
|
pSFX_MP3Stream->iBytesDecodedThisPacket = x.out_bytes;
|
|
|
|
uiDecoded = x.out_bytes;
|
|
|
|
if (x.in_bytes <= 0)
|
|
{
|
|
//psReturn = "MP3ERR: Bad sync in file";
|
|
uiDecoded= 0; // finished
|
|
break;
|
|
}
|
|
}
|
|
#pragma warning (disable : 4127 ) // conditional expression is constant
|
|
while (0); // <g>
|
|
#pragma warning (default : 4127 ) // conditional expression is constant
|
|
|
|
// restore global stream ptr before returning to normal functions (so the rest of the MP3 code still works)...
|
|
//
|
|
pMP3Stream = &_MP3Stream;
|
|
|
|
return uiDecoded;
|
|
}
|
|
|
|
|
|
// ret is char* errstring, else NULL for ok
|
|
//
|
|
char *C_MP3Stream_Rewind( LP_MP3STREAM pSFX_MP3Stream )
|
|
{
|
|
char* psReturn = NULL;
|
|
MPEG_HEAD head; // only relevant within this function during init
|
|
int iBitRate; // ditto
|
|
int iNULL;
|
|
|
|
pMP3Stream = pSFX_MP3Stream;
|
|
|
|
pMP3Stream->iSourceReadIndex = pMP3Stream->iRewind_SourceReadIndex;
|
|
pMP3Stream->iSourceBytesRemaining = pMP3Stream->iRewind_SourceBytesRemaining; // already adjusted for tags etc
|
|
|
|
// I'm not sure that this is needed, but where else does decode_init get passed useful data ptrs?...
|
|
//
|
|
if (pMP3Stream->iSourceFrameBytes == head_info3( pMP3Stream->pbSourceData, pMP3Stream->iSourceBytesRemaining/2, &head, &iBitRate, (unsigned int*)&iNULL ) )
|
|
{
|
|
if (audio.decode_init(&head, pMP3Stream->iSourceFrameBytes, pMP3Stream->iRewind_FinalReductionCode, pMP3Stream->iSourceReadIndex, pMP3Stream->iRewind_FinalConvertCode, freq_limit))
|
|
{
|
|
// we should always get here...
|
|
//
|
|
}
|
|
else
|
|
{
|
|
psReturn = "MP3ERR: Failed to re-init decoder for rewind!";
|
|
}
|
|
}
|
|
else
|
|
{
|
|
psReturn = "MP3ERR: Frame bytes mismatch during rewind header-read!";
|
|
}
|
|
|
|
// restore global stream ptr before returning to normal functions (so the rest of the MP3 code still works)...
|
|
//
|
|
pMP3Stream = &_MP3Stream;
|
|
|
|
return psReturn;
|
|
}
|
|
|