?? avi2mp2.c
字號:
/* avi2mp2.c -
音頻編碼文件*/
#include <dos.h>
#include "common.h"
#include "encoder.h"
#include "global.h"
#include "video.h"
/* 定義全局變量 */
Bit_stream_struc bs;
char programName[] = "avi2mpg1";
int channels;
FILE *hOutput;
unsigned long length, last_sample = 0;
/************************************************************************
/*
/* set_parms
/*
/* 功能: 根據命令行的輸入為編碼設置參數。如果命令行中沒有指定，則使用缺省參數
/*
/************************************************************************/
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;
int file;
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(4);
}
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(5);
}
open_bit_stream_w(&bs, tempfile, BUFFER_SIZE);
*num_samples=0;
for(file=0;file<numAviFiles;file++)
{
if(pWavFormat.nChannels == 2)
*num_samples += pAudioStreamInfo[file].dwLength * 2;
else
*num_samples += pAudioStreamInfo[file].dwLength;
}
if(pWavFormat.nSamplesPerSec == 22050)
*num_samples = *num_samples * 2;
if(pWavFormat.nSamplesPerSec == 11025)
*num_samples = *num_samples * 4;
}
/************************************************************************
/*
/* avi2mp2
/*
/* 功能 : MPEG I編碼器，支持層1和2以及心理聲學模型2 (AT&T)
/*
/************************************************************************/
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;
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");
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;
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; }
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;
frac_SpF = avg_slots_per_frame - (double)whole_SpF;
slot_lag = -frac_SpF;
/
if (frac_SpF != 0)
{}
else
info.padding = 0;
while (get_audio(buffer, num_samples, stereo, info.lay) > 0) {
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;
}
else {
extra_slot = 1;
info.padding = 1;
slot_lag += (1-frac_SpF);
}
}
adb = (whole_SpF+extra_slot) * bitsPerSlot;
switch (info.lay)
{
/***************************** Layer I **********************************/
case 1 :
for (j=0;j<SCALE_BLOCK;j++)
for (k=0;k<stereo;k++)
{
window_subband(&win_buf[k], &(*win_que)[k][0], k);
filter_subband(&(*win_que)[k][0], &(*sb_sample)[k][0][j][0]);
}
I_scale_factor_calc(*sb_sample, scalar, stereo);
if(fr_ps.actual_mode == MPG_MD_JOINT_STEREO)
{
I_combine_LR(*sb_sample, *j_sample);
I_scale_factor_calc(j_sample, &j_scale, 1);
}
put_scale(scalar, &fr_ps, max_sc);
for (k=0;k<stereo;k++)
{
psycho_anal(&buffer[k][0],&sam[k][0], k, info.lay, snr32,
(FLOAT)s_freq[info.sampling_frequency]*1000);
for (i=0;i<SBLIMIT;i++)
ltmin[k][i] = (double) snr32[i];
}
I_main_bit_allocation(ltmin, bit_alloc, &adb, &fr_ps);
if (error_protection)
I_CRC_calc(&fr_ps, bit_alloc, &crc);
encode_info(&fr_ps, &bs);
if (error_protection)
encode_CRC(crc, &bs);
I_encode_bit_alloc(bit_alloc, &fr_ps, &bs);
I_encode_scale(scalar, bit_alloc, &fr_ps, &bs);
I_subband_quantization(scalar, *sb_sample, j_scale, *j_sample,
bit_alloc, *subband, &fr_ps);
I_sample_encoding(*subband, bit_alloc, &fr_ps, &bs);
for (i=0;i<adb;i++)
put1bit(&bs, 0);
break;
/***************************** Layer 2 **********************************/
case 2 :
for (i=0;i<3;i++)
for (j=0;j<SCALE_BLOCK;j++)
for (k=0;k<stereo;k++)
{
window_subband(&win_buf[k], &(*win_que)[k][0], k);
filter_subband(&(*win_que)[k][0], &(*sb_sample)[k][i][j][0]);
}
II_scale_factor_calc(*sb_sample, scalar, stereo, fr_ps.sblimit);
pick_scale(scalar, &fr_ps, max_sc);
if(fr_ps.actual_mode == MPG_MD_JOINT_STEREO)
{
II_combine_LR(*sb_sample, *j_sample, fr_ps.sblimit);
II_scale_factor_calc(j_sample, &j_scale, 1, fr_ps.sblimit);
} /* this way we calculate more mono than we need */
/* but it is cheap */
for (k=0;k<stereo;k++)
{
psycho_anal(&buffer[k][0],&sam[k][0], k,
info.lay, snr32,
(FLOAT)s_freq[info.sampling_frequency]*1000);
for (i=0;i<SBLIMIT;i++)
ltmin[k][i] = (double) snr32[i];
}
II_transmission_pattern(scalar, scfsi, &fr_ps);
II_main_bit_allocation(ltmin, scfsi, bit_alloc, &adb, &fr_ps);
if (error_protection)
II_CRC_calc(&fr_ps, bit_alloc, scfsi, &crc);
encode_info(&fr_ps, &bs);
if (error_protection)
encode_CRC(crc, &bs);
II_encode_bit_alloc(bit_alloc, &fr_ps, &bs);
II_encode_scale(bit_alloc, scfsi, scalar, &fr_ps, &bs);
II_subband_quantization(scalar, *sb_sample, j_scale,
*j_sample, bit_alloc, *subband, &fr_ps);
II_sample_encoding(*subband, bit_alloc, &fr_ps, &bs);
for (i=0;i<adb;i++)
put1bit(&bs, 0);
break;
}
frameBits = sstell(&bs) - sentBits;
if(frameBits%bitsPerSlot)
fprintf(stderr,"Sent %ld bits = %ld slots plus %ld\n",
frameBits, frameBits/bitsPerSlot,
frameBits%bitsPerSlot);
sentBits += frameBits;
}
close_bit_stream_w(&bs);
return(0);
}
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