实验6：MPEG音频编码实验

最新推荐文章于 2022-11-03 16:36:38 发布

妖妖yaoyao

最新推荐文章于 2022-11-03 16:36:38 发布

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本文链接：https://blog.csdn.net/weixin_48846782/article/details/118559111

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实验要求：

输出音频的采样率和目标码率、选择三个不同特性的音频文件、输出某个数据帧的性质

实验原理：

在这里插入图片描述

实验过程：

while (get_audio (musicin, buffer, num_samples, nch, &header) > 0) {//每次循环操作一个帧
if (glopts.verbosity > 1)
if (++frameNum % 10 == 0)
fprintf (stderr, “[%4u]\r”, frameNum);
fflush (stderr);
win_buf[0] = &buffer[0][0];
win_buf[1] = &buffer[1][0];

adb = available_bits (&header, &glopts);
lg_frame = adb / 8;//一帧的字节数
if (header.dab_extension) {
  /* in 24 kHz we always have 4 bytes */
  if (header.sampling_frequency == 1)
header.dab_extension = 4;

/* You must have one frame in memory if you are in DAB mode /
/ in conformity of the norme ETS 300 401 http://www.etsi.org /
/ see bitstream.c */
if (frameNum == 1)
minimum = lg_frame + MINIMUM;
adb -= header.dab_extension * 8 + header.dab_length * 8 + 16;
}

{
  int gr, bl, ch;
  /* New polyphase filter
 Combines windowing and filtering. Ricardo Feb'03 */
  for( gr = 0; gr < 3; gr++ )
for ( bl = 0; bl < 12; bl++ )
  for ( ch = 0; ch < nch; ch++ )
    WindowFilterSubband( &buffer[ch][gr * 12 * 32 + 32 * bl], ch,
			 &(*sb_sample)[ch][gr][bl][0] );
}

#ifdef REFERENCECODE
{
/* Old code. left here for reference */
int gr, bl, ch;
for (gr = 0; gr < 3; gr++)
for (bl = 0; bl < SCALE_BLOCK; bl++)
for (ch = 0; ch < nch; ch++) {
window_subband (&win_buf[ch], &(*win_que)[ch][0], ch);
filter_subband (&(*win_que)[ch][0], &(*sb_sample)[ch][gr][bl][0]);
}
}
#endif

#ifdef NEWENCODE
scalefactor_calc_new(sb_sample, scalar, nch, frame.sblimit);
find_sf_max (scalar, &frame, max_sc);
if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
/ this way we calculate more mono than we need /
/ but it is cheap */
combine_LR_new (*sb_sample, *j_sample, frame.sblimit);
scalefactor_calc_new (j_sample, &j_scale, 1, frame.sblimit);
}
#else
scale_factor_calc (sb_sample, scalar, nch, frame.sblimit);//比例因子提取
pick_scale (scalar, &frame, max_sc);//比例因子选择
if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
/ this way we calculate more mono than we need /
/ but it is cheap */
combine_LR (*sb_sample, *j_sample, frame.sblimit);
scale_factor_calc (j_sample, &j_scale, 1, frame.sblimit);
}
#endif

//心理声学模型
if ((glopts.quickmode == TRUE) && (++psycount % glopts.quickcount != 0)) {
  /* We're using quick mode, so we're only calculating the model every
     'quickcount' frames. Otherwise, just copy the old ones across */
  for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++)
  smr[ch][sb] = smrdef[ch][sb];
  }
} else {
  /* calculate the psymodel */
  switch (model) {
  case -1:
psycho_n1 (smr, nch);
break;
  case 0:	/* Psy Model A */
psycho_0 (smr, nch, scalar, (FLOAT) s_freq[header.version][header.sampling_frequency] * 1000);	
break;
  case 1:
psycho_1 (buffer, max_sc, smr, &frame);//进入这个模型
break;
  case 2:
for (ch = 0; ch < nch; ch++) {
  psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
	     (FLOAT) s_freq[header.version][header.sampling_frequency] *
	     1000, &glopts);
}
break;
  case 3:
/* Modified psy model 1 */
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
break;
  case 4:
/* Modified Psycho Model 2 */
for (ch = 0; ch < nch; ch++) {
  psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
	     (FLOAT) s_freq[header.version][header.sampling_frequency] *
	     1000, &glopts);
}
break;	
  case 5:
/* Model 5 comparse model 1 and 3 */
psycho_1 (buffer, max_sc, smr, &frame);
fprintf(stdout,"1 ");
smr_dump(smr,nch);
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
fprintf(stdout,"3 ");
smr_dump(smr,nch);
break;
  case 6:
/* Model 6 compares model 2 and 4 */
for (ch = 0; ch < nch; ch++) 
  psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
	    (FLOAT) s_freq[header.version][header.sampling_frequency] *
	    1000, &glopts);
fprintf(stdout,"2 ");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++) 
  psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
	     (FLOAT) s_freq[header.version][header.sampling_frequency] *
	     1000, &glopts);
fprintf(stdout,"4 ");
smr_dump(smr,nch);
break;
  case 7:
fprintf(stdout,"Frame: %i\n",frameNum);
/* Dump the SMRs for all models */	
psycho_1 (buffer, max_sc, smr, &frame);
fprintf(stdout,"1");
smr_dump(smr, nch);
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
fprintf(stdout,"3");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++) 
  psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
	    (FLOAT) s_freq[header.version][header.sampling_frequency] *
	    1000, &glopts);
fprintf(stdout,"2");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++) 
  psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
	     (FLOAT) s_freq[header.version][header.sampling_frequency] *
	     1000, &glopts);
fprintf(stdout,"4");
smr_dump(smr,nch);
break;
  case 8:
/* Compare 0 and 4 */	
psycho_n1 (smr, nch);
fprintf(stdout,"0");
smr_dump(smr,nch);

for (ch = 0; ch < nch; ch++) 
  psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
	     (FLOAT) s_freq[header.version][header.sampling_frequency] *
	     1000, &glopts);
fprintf(stdout,"4");
smr_dump(smr,nch);
break;
  default:
fprintf (stderr, "Invalid psy model specification: %i\n", model);
exit (0);
  }

  if (glopts.quickmode == TRUE)
/* copy the smr values and reuse them later */
for (ch = 0; ch < nch; ch++) {
  for (sb = 0; sb < SBLIMIT; sb++)
    smrdef[ch][sb] = smr[ch][sb];
}

  if (glopts.verbosity > 4) 
smr_dump(smr, nch);
 
  


}

#ifdef NEWENCODE
sf_transmission_pattern (scalar, scfsi, &frame);
main_bit_allocation_new (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
//main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);

if (error_protection)
  CRC_calc (&frame, bit_alloc, scfsi, &crc);

write_header (&frame, &bs);
//encode_info (&frame, &bs);
if (error_protection)
  putbits (&bs, crc, 16);
write_bit_alloc (bit_alloc, &frame, &bs);
//encode_bit_alloc (bit_alloc, &frame, &bs);
write_scalefactors(bit_alloc, scfsi, scalar, &frame, &bs);
//encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
subband_quantization_new (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
			  *subband, &frame);
//subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
//	  *subband, &frame);
write_samples_new(*subband, bit_alloc, &frame, &bs);
//sample_encoding (*subband, bit_alloc, &frame, &bs);

#else
transmission_pattern (scalar, scfsi, &frame);
main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);//动态比特分配，分配的bit数存储在bit_alloc中
if (error_protection)
CRC_calc (&frame, bit_alloc, scfsi, &crc);
encode_info (&frame, &bs);
if (error_protection)
encode_CRC (crc, &bs);
encode_bit_alloc (bit_alloc, &frame, &bs);
encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
*subband, &frame);//量化
sample_encoding (*subband, bit_alloc, &frame, &bs);
#endif

/* If not all the bits were used, write out a stack of zeros */
for (i = 0; i < adb; i++)
  put1bit (&bs, 0);
if (header.dab_extension) {
  /* Reserve some bytes for X-PAD in DAB mode */
  putbits (&bs, 0, header.dab_length * 8);
  
  for (i = header.dab_extension - 1; i >= 0; i--) {
CRC_calcDAB (&frame, bit_alloc, scfsi, scalar, &crc, i);
/* this crc is for the previous frame in DAB mode  */
if (bs.buf_byte_idx + lg_frame < bs.buf_size)
  bs.buf[bs.buf_byte_idx + lg_frame] = crc;
/* reserved 2 bytes for F-PAD in DAB mode  */
putbits (&bs, crc, 8);
  }
  putbits (&bs, 0, 16);
}

frameBits = sstell (&bs) - sentBits;

if (frameBits % 8) {	/* a program failure */
  fprintf (stderr, "Sent %ld bits = %ld slots plus %ld\n", frameBits,
       frameBits / 8, frameBits % 8);
  fprintf (stderr, "If you are reading this, the program is broken\n");
  fprintf (stderr, "email [mfc at NOTplanckenerg.com] without the NOT\n");
  fprintf (stderr, "with the command line arguments and other info\n");
  exit (0);
}

sentBits += frameBits;

}//while循环结束

main函数的主体为该while循环，每次循环操作一个帧，get_audio的作用为从码流中读出一个帧的数据，判断码流为单声道还是双声道，将1声道读入buffer[0][]中，将2声道读入buffer[1][]。scale_factor_calc函数为比例因子提取函数，pick_scale函数为比例因子选择函数。从心理声学模型开始主要为流程图的下分枝的实现。main_bit_allocation函数为实现动态比特分配的函数，将分配的比特数存储在bit_alloc中。

2.scale_factor_calc函数

void scale_factor_calc (double sb_sample[][3][SCALE_BLOCK][SBLIMIT],
unsigned int scalar[][3][SBLIMIT], int nch,
int sblimit)
{
/* Optimized to use binary search instead of linear scan through the
scalefactor table; guarantees to find scalefactor in only 5
jumps/comparisons and not in {0 (lin. best) to 63 (lin. worst)}.
Scalefactors for subbands > sblimit are no longer computed.
Uses a single sblimit-loop.
Patrick De Smet Oct 1999.
/
int k, t;
/ Using ‘–’ loops to avoid possible “cmp value + bne/beq” compiler /
/ inefficiencies. Below loops should compile to “bne/beq” only code /
for (k = nch; k–😉
for (t = 3; t–😉 {
int i;
for (i = sblimit; i–😉 {
int j;
unsigned int l;
register double temp;
unsigned int scale_fac;
/ Determination of max. over each set of 12 subband samples: /
/ PDS TODO: maybe this could/should ??!! be integrated into /
/ the subband filtering routines? /
register double cur_max = fabs (sb_sample[k][t][SCALE_BLOCK - 1][i]);//fabs()取绝对值
for (j = SCALE_BLOCK - 1; j–😉 {
if ((temp = fabs (sb_sample[k][t][j][i])) > cur_max)
cur_max = temp;
}//找出每个子带中的最大值
/ PDS: binary search in the scalefactor table: /
/ This is the real speed up: */
for (l = 16, scale_fac = 32; l; l >>= 1) {//16转换为二进制为10000，移位判断可进行五次判断，multiple共64个值，进行五次判断刚好可以找到multiple表中与cur_max最接近的值
if (cur_max <= multiple[scale_fac])
scale_fac += l;
else
scale_fac -= l;
}
if (cur_max > multiple[scale_fac])//在multiple表中寻找的是最接近且不小于cur_max的值
scale_fac–;
scalar[k][t][i] = scale_fac;//scalar中存储的是在multiple表中的索引号，而非实际比例因子
}
}
}