www.pudn.com > avi 到 mpeg 的转换程序及源代码.zip > AVI2MP2.C
/* avi2mp2.c - originally musicin.c */
/**********************************************************************
Copyright (c) 1991 MPEG/audio software simulation group, All Rights Reserved
musicin.c
**********************************************************************/
/**********************************************************************
* MPEG/audio coding/decoding software, work in progress *
* NOT for public distribution until verified and approved by the *
* MPEG/audio committee. For further information, please contact *
* Davis Pan, 508-493-2241, e-mail: pan@3d.enet.dec.com *
* *
* VERSION 4.0 *
* changes made since last update: *
* date programmers comment *
* 3/01/91 Douglas Wong, start of version 1.1 records *
* Davis Pan *
* 3/06/91 Douglas Wong, rename: setup.h to endef.h *
* removed extraneous variables *
* 3/21/91 J.Georges Fritsch introduction of the bit-stream *
* package. This package allows you *
* to generate the bit-stream in a *
* binary or ascii format *
* 3/31/91 Bill Aspromonte replaced the read of the SB matrix *
* by an "code generated" one *
* 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. *
* Incorporated Jean-Georges Fritsch's *
* "bitstream.c" package. *
* Modified to strictly adhere to *
* encoded bitstream specs, including *
* "Berlin changes". *
* Modified user interface dialog & code *
* to accept any input & output *
* filenames desired. Also added *
* de-emphasis prompt and final bail-out *
* opportunity before encoding. *
* Added AIFF PCM sound file reading *
* capability. *
* Modified PCM sound file handling to *
* process all incoming samples and fill *
* out last encoded frame with zeros *
* (silence) if needed. *
* Located and fixed numerous software *
* bugs and table data errors. *
* 27jun91 dpwe (Aware Inc) Used new frame_params struct. *
* Clear all automatic arrays. *
* Changed some variable names, *
* simplified some code. *
* Track number of bits actually sent. *
* Fixed padding slot, stereo bitrate *
* Added joint-stereo : scales L+R. *
* 6/12/91 Earle Jennings added fix for MS_DOS in obtain_param *
* 6/13/91 Earle Jennings added stack length adjustment before *
* main for MS_DOS *
* 7/10/91 Earle Jennings conversion of all float to FLOAT *
* port to MsDos from MacIntosh completed*
* 8/ 8/91 Jens Spille Change for MS-C6.00 *
* 8/22/91 Jens Spille new obtain_parameters() *
*10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. *
* Don H. Lee, *
* Peter W. Farrett *
*10/ 3/91 Don H. Lee implemented CRC-16 error protection *
* newly introduced functions are *
* I_CRC_calc, II_CRC_calc and encode_CRC*
* Additions and revisions are marked *
* with "dhl" for clarity *
*11/11/91 Katherine Wang Documentation of code. *
* (variables in documentation are *
* surround by the # symbol, and an '*'*
* denotes layer I or II versions) *
* 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most *
* important fixes involved changing *
* 16-bit ints to long or unsigned in *
* bit alloc routines for quant of 65535 *
* and passing proper function args. *
* Removed "Other Joint Stereo" option *
* and made bitrate be total channel *
* bitrate, irrespective of the mode. *
* Fixed many small bugs & reorganized. *
* 2/25/92 Masahiro Iwadare made code cleaner and more consistent *
* 8/07/92 Mike Coleman make exit() codes return error status *
* made slight changes for portability *
*19 aug 92 Soren H. Nielsen Changed MS-DOS file name extensions. *
* 8/25/92 Shaun Astarabadi Replaced rint() function with explicit*
* rounding for portability with MSDOS. *
* 9/22/92 jddevine@aware.com Fixed _scale_factor_calc() calls. *
*10/19/92 Masahiro Iwadare added info->mode and info->mode_ext *
* updates for AIFF format files *
* 3/10/93 Kevin Peterson In parse_args, only set non default *
* bit rate if specified in arg list. *
* Use return value from aiff_read_hdrs *
* to fseek to start of sound data *
* 7/26/93 Davis Pan fixed bug in printing info->mode_ext *
* value for joint stereo condition *
* 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, *
* Daniel Lauzon, and *
* Bill Truerniet *
**********************************************************************/
/*
* 4/4/97 - John Schlichther
*
* extensively altered to create avi2mpg1 - avi to mpeg-1 encoder
*
* Since avi file, and the avi subsystem are platform dependant, cross
* platform compatibility removed, many optional features disabled or
* removed, code generally trimmed to a minimum.
*
* 7/15/97 - John Schlichther - added support for 8 and 16 bit audio at
* 11.025, 22.05, and 44.1 KHz sample rates
*/
#include
#include "common.h"
#include "encoder.h"
#include "global.h"
/* Global variable definitions for "avi2mp2.c" */
Bit_stream_struc bs;
char programName[] = "avi2mpg1";
int channels;
FILE *hOutput;
unsigned long length, last_sample = 0;
/* Implementations */
/************************************************************************
/*
/* set_parms
/*
/* PURPOSE: Sets encoding parameters to the specifications of the
/* command line. Default settings are used for parameters
/* not specified in the command line.
/*
/************************************************************************/
void set_parms(fr_ps, num_samples, tempfile)
frame_params *fr_ps;
unsigned long *num_samples;
char * tempfile;
{
layer *info = fr_ps->header;
int err = 0, i = 0;
/* preset defaults */
info->lay = audio_layer_parm;
if(pWavFormat.nChannels == 1)
{
info->mode = MPG_MD_MONO;
info->mode_ext = 0;
}
else
{
if(joint_stereo_parm == 1)
{
info->mode = MPG_MD_JOINT_STEREO;
}
else
{
info->mode = MPG_MD_STEREO;
info->mode_ext = 0;
}
}
if((info->sampling_frequency = SmpFrqIndex((long)(1000*DFLT_SFQ))) < 0)
{
fprintf(stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ);
abort();
}
if((info->bitrate_index = BitrateIndex(info->lay, audio_bitrate_parm)) < 0)
{
fprintf(stderr, "%s: bad default bitrate %u\n", programName, DFLT_BRT);
abort();
}
switch(DFLT_EMP)
{
case 'n': info->emphasis = 0; break;
case '5': info->emphasis = 1; break;
case 'c': info->emphasis = 3; break;
default:
fprintf(stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP);
abort();
}
info->copyright = 0; info->original = 0; info->error_protection = FALSE;
length = sizeof(pWavFormat);
channels = pWavFormat.nChannels; // store for later
if ((pWavFormat.nSamplesPerSec != 11025)&&(pWavFormat.nSamplesPerSec != 22050)&&(pWavFormat.nSamplesPerSec != 44100))
{
fprintf(stderr, "Samples per Sec = %i\n", pWavFormat.nSamplesPerSec);
fprintf(stderr, "input avi audio sample rate must = 11.025, 22.05, or 44.1 kHz!");
exit(1);
}
if ((pWavFormat.wBitsPerSample != 8)&&(pWavFormat.wBitsPerSample != 16))
{
fprintf(stderr, "Bits per Sample = %i\n", pWavFormat.wBitsPerSample);
fprintf(stderr, "input avi audio must be 8 or 16 bit!");
exit(1);
}
open_bit_stream_w(&bs, tempfile, BUFFER_SIZE);
if(pWavFormat.nChannels == 2)
*num_samples = pAudioStreamInfo.dwLength * 2;
else
*num_samples = pAudioStreamInfo.dwLength;
if(pWavFormat.nSamplesPerSec == 22050)
*num_samples = *num_samples * 2;
if(pWavFormat.nSamplesPerSec == 11025)
*num_samples = *num_samples * 4;
}
/************************************************************************
/*
/* avi2mp2
/*
/* PURPOSE: MPEG I Encoder supporting layers 1 and 2, and
/* psychoacoustic model 2 (AT&T)
/*
/* SEMANTICS: One overlapping frame of audio of up to 2 channels are
/* processed at a time in the following order:
/* (associated routines are in parentheses)
/*
/* 1. Filter sliding window of data to get 32 subband
/* samples per channel.
/* (window_subband,filter_subband)
/*
/* 2. If joint stereo mode, combine left and right channels
/* for subbands above #jsbound#.
/* (*_combine_LR)
/*
/* 3. Calculate scalefactors for the frame, and if layer 2,
/* also calculate scalefactor select information.
/* (*_scale_factor_calc)
/*
/* 4. Calculate psychoacoustic masking levels using selected
/* psychoacoustic model.
/* (*_Psycho_One, psycho_anal)
/*
/* 5. Perform iterative bit allocation for subbands with low
/* mask_to_noise ratios using masking levels from step 4.
/* (*_main_bit_allocation)
/*
/* 6. If error protection flag is active, add redundancy for
/* error protection.
/* (*_CRC_calc)
/*
/* 7. Pack bit allocation, scalefactors, and scalefactor select
/* information (layer 2) onto bitstream.
/* (*_encode_bit_alloc,*_encode_scale,II_transmission_pattern)
/*
/* 8. Quantize subbands and pack them into bitstream
/* (*_subband_quantization, *_sample_encoding)
/*
/************************************************************************/
avi2mp2(tempfile)
char * tempfile;
{
typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT];
SBS *sb_sample;
typedef double JSBS[3][SCALE_BLOCK][SBLIMIT];
JSBS *j_sample;
typedef double INN[2][HAN_SIZE];
INN *win_que;
typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT];
SUB *subband;
frame_params fr_ps;
layer info;
short **win_buf;
static short buffer[2][1152];
static unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT];
static unsigned int scalar[2][3][SBLIMIT], j_scale[3][SBLIMIT];
static double ltmin[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT];
FLOAT snr32[32];
short sam[2][1056];
int whole_SpF, extra_slot = 0;
double avg_slots_per_frame, frac_SpF, slot_lag;
int stereo, error_protection;
static unsigned int crc;
int i, j, k, adb;
unsigned long bitsPerSlot, samplesPerFrame, frameNum = 0;
unsigned long frameBits, sentBits = 0;
unsigned long num_samples;
/* Most large variables are declared dynamically to ensure
compatibility with smaller machines */
sb_sample = (SBS *) mem_alloc(sizeof(SBS), "sb_sample");
j_sample = (JSBS *) mem_alloc(sizeof(JSBS), "j_sample");
win_que = (INN *) mem_alloc(sizeof(INN), "Win_que");
subband = (SUB *) mem_alloc(sizeof(SUB),"subband");
win_buf = (short **) mem_alloc(sizeof(short *)*2, "win_buf");
/* clear buffers */
memset((char *) buffer, 0, sizeof(buffer));
memset((char *) bit_alloc, 0, sizeof(bit_alloc));
memset((char *) scalar, 0, sizeof(scalar));
memset((char *) j_scale, 0, sizeof(j_scale));
memset((char *) scfsi, 0, sizeof(scfsi));
memset((char *) ltmin, 0, sizeof(ltmin));
memset((char *) lgmin, 0, sizeof(lgmin));
memset((char *) max_sc, 0, sizeof(max_sc));
memset((char *) snr32, 0, sizeof(snr32));
memset((char *) sam, 0, sizeof(sam));
fr_ps.header = &info;
fr_ps.tab_num = -1; /* no table loaded */
fr_ps.alloc = NULL;
info.version = MPEG_AUDIO_ID;
set_parms(&fr_ps, &num_samples, tempfile);
hdr_to_frps(&fr_ps);
stereo = fr_ps.stereo;
error_protection = info.error_protection;
if (info.lay == 1) { bitsPerSlot = 32; samplesPerFrame = 384; }
else { bitsPerSlot = 8; samplesPerFrame = 1152; }
/* Figure average number of 'slots' per frame. */
/* Bitrate means TOTAL for both channels, not per side. */
avg_slots_per_frame = ((double)samplesPerFrame /
s_freq[info.sampling_frequency]) *
((double)bitrate[info.lay-1][info.bitrate_index] /
(double)bitsPerSlot);
whole_SpF = (int) avg_slots_per_frame;
//printf("slots/frame = %d\n",whole_SpF);
frac_SpF = avg_slots_per_frame - (double)whole_SpF;
slot_lag = -frac_SpF;
//fprintf(stderr, "frac SpF=%.3f, tot bitrate=%d kbps, s freq=%.1f kHz\n",
// frac_SpF, bitrate[info.lay-1][info.bitrate_index],
// s_freq[info.sampling_frequency]);
if (frac_SpF != 0)
//printf("Fractional number of slots, padding required\n");
{}
else
info.padding = 0;
while (get_audio(buffer, num_samples, stereo, info.lay) > 0) {
//fprintf(stderr, "\raudio frame %u", frameNum++); fflush(stderr);
fprintf(stderr, "\r%5.1f%% complete", ((float)(num_samples-bytes_processed)/(float)num_samples)*100.0);
win_buf[0] = &buffer[0][0];
win_buf[1] = &buffer[1][0];
if (frac_SpF != 0) {
if (slot_lag > (frac_SpF-1.0) ) {
slot_lag -= frac_SpF;
extra_slot = 0;
info.padding = 0;
/* printf("No padding for this frame\n"); */
}
else {
extra_slot = 1;
info.padding = 1;
slot_lag += (1-frac_SpF);
/* printf("Padding for this frame\n"); */
}
}
adb = (whole_SpF+extra_slot) * bitsPerSlot;
switch (info.lay)
{
/***************************** Layer I **********************************/
case 1 :
for (j=0;j