数据压缩实验六
文章目录
一、实验原理：
1.流程框图
两条线：
第一条：将输入的音频子带分解，输入的PCM码流经过一个多相滤波器组，形成32个子带。
第二条：对PCM信号进行频域分析，经过心理声学模型计算以频率为自变量的噪声掩蔽阈值。
2.心理声学模型
- 听觉系统中存在一个听觉阈值电平，低于这个电平的声音信号就听不到。听觉阈值的大小随声音频率的改变而改变。一个人是否听到声音取决于声音的频率，以及声音的幅度是否高于这种频率下的听觉阈。听觉掩蔽特性，即听觉阈值电平是自适应的，会随听到的不同频率声音而发生变化。
- 通过子带分析滤波器组使信号具有高的时间分辨率， 确保在短暂冲击信号情况下，编码的声音信号具有足够高的质量。又可以通过FFT运算使信号具有高的频率分辨率， 因为掩蔽阈值是从功率谱密度推出来的。
- 在低频子带中，为了保护音调和共振峰的结构，就要求用较小的量化阶、较多的量化级数，即分配较多的位数来表示样本值。而话音中的摩擦音和类似噪声的声音，通常出现在高频子带中，对它分配较少的位数。心理声学模型的作用即计算信号中听觉不可感知的部分。
- 临界频带：临界频带是指当某个纯音被以它为中心频率、且具有一定带宽的连续噪声所掩蔽时，如果该纯音刚好被听到时的功率等于这一频带内的噪声功率，这个带宽为临界频带带宽。
- 全局掩蔽阈值的计算：
二、实验过程及结果：
1.main函数的理解：
int main(int argc, char** argv) {
typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT];
SBS* sb_sample;
typedef double JSBS[3][SCALE_BLOCK][SBLIMIT];
JSBS* j_sample;
typedef double IN[2][HAN_SIZE];
IN* win_que;
typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT];
SUB* subband;
frame_info frame; //音频帧信息结构体：包含头信息、比特分配表、声道数、子带数等音频帧信息内容
frame_header header; //音频帧头文件结构体：包含采样频率等信息
char original_file_name[MAX_NAME_SIZE]; //输入文件名
char encoded_file_name[MAX_NAME_SIZE]; //输出文件名
short** win_buf;
static short buffer[2][1152];
static unsigned int bit_alloc[2][SBLIMIT]; //各子带比特分配
static unsigned int scfsi[2][SBLIMIT];
static unsigned int scalar[2][3][SBLIMIT]; //各子带比例因子
static unsigned int j_scale[3][SBLIMIT];
static double smr[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT];
short sam[2][1344];
int model;
int nch; //声道数
int error_protection;
static unsigned int crc;
int sb, ch;
int adb;
unsigned long frameBits, sentBits = 0;
unsigned long num_samples;
int lg_frame;
int i;
/* Used to keep the SNR values for the fast/quick psy models */
static FLOAT smrdef[2][32];
static int psycount = 0;
extern int minimum;
time_t start_time, end_time;
int total_time;
sb_sample = (SBS*)mem_alloc(sizeof(SBS), "sb_sample");
j_sample = (JSBS*)mem_alloc(sizeof(JSBS), "j_sample");
win_que = (IN*)mem_alloc(sizeof(IN), "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*)smr, 0, sizeof(smr));
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));
global_init();
header.extension = 0;
frame.header = &header;
frame.tab_num = -1; /* no table loaded */
frame.alloc = NULL;
header.version = MPEG_AUDIO_ID; /* Default: MPEG-1 */
total_time = 0;
time(&start_time);
programName = argv[0];
if (argc == 1) /* no command-line args */
short_usage();
else
parse_args(argc, argv, &frame, &model, &num_samples, original_file_name, encoded_file_name); //解析命令行参数
print_config(&frame, &model, original_file_name, encoded_file_name); // print文件参数
/* this will load the alloc tables and do some other stuff */
hdr_to_frps(&frame);
nch = frame.nch;
error_protection = header.error_protection;
/* 从数据流获取音频 */
while (get_audio(musicin, buffer, num_samples, nch, &header) > 0) {
/* 从输入文件读取数据到 buffer */
if (glopts.verbosity > 1)
if (++frameNum % 10 == 0) /* 出错 */
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;
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++) //36个样值分为3组
for (bl = 0; bl < 12; bl++) //每组做12次子带分解
for (ch = 0; ch < nch; ch++)
WindowFilterSubband(&buffer[ch][gr * 12 * 32 + 32 * bl], ch, &(*sb_sample)[ch][gr][bl][0]); //多相滤波器组
}
#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); // 计算比例因子
pick_scale(scalar, &frame, max_sc);//选择比例因子
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
/* 选择心理声学模型，计算SMR */
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); // smr为输出
break;
case 1:
psycho_1(buffer, max_sc, smr, &frame);
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); // 比特分配
if (error_protection)
CRC_calc(&frame, bit_alloc, scfsi, &crc);
encode_info(&frame, &bs); //编码
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;
}
close_bit_stream_w(&bs);
if ((glopts.verbosity > 1) && (glopts.vbr == TRUE)) {
int i;
#ifdef NEWENCODE
extern int vbrstats_new[15];
#else
extern int vbrstats[15];
#endif
fprintf(stdout, "VBR stats:\n");
for (i = 1; i < 15; i++)
fprintf(stdout, "%4i ", bitrate[header.version][i]);
fprintf(stdout, "\n");
for (i = 1; i < 15; i++)
#ifdef NEWENCODE
fprintf(stdout, "%4i ", vbrstats_new[i]);
#else
fprintf(stdout, "%4i ", vbrstats[i]);
#endif
fprintf(stdout, "\n");
}
fprintf(stderr,
"Avg slots/frame = %.3f; b/smp = %.2f; bitrate = %.3f kbps\n",
(FLOAT)sentBits / (frameNum * 8),
(FLOAT)sentBits / (frameNum * 1152),
(FLOAT)sentBits / (frameNum * 1152) *
s_freq[header.version][header.sampling_frequency]);
if (fclose(musicin) != 0) {
fprintf(stderr, "Could not close \"%s\".\n", original_file_name);
exit(2);
}
fprintf(stderr, "\nDone\n");
time(&end_time);
total_time = end_time - start_time;
printf("total time is %d\n", total_time);
exit(0);
}
在命令行输入-h，查看命令行参数：
2.对于某个数据帧，输出量化编码信息
该帧所分配的比特数
该帧的比例因子
该帧的比特分配结果
在m2aenc.c中添加如下代码：
//添加如下代码
#define OUT "out_file.txt"
#define Trace 1
FILE *fileout;
int gr;
//
//添加如下代码
#if Trace
fileout=fopen("fileout.txt","w");
if(frameNum==2)
{
fprintf(fileout,"========== 比例因子 ==========\n");
for(ch=0;ch<nch;ch++) // 每个声道单独输出,ch为当前声道,nch为总声道数
{
fprintf(fileout,"------- 声道%d -------\n", ch + 1);
for(sb=0;sb<frame.sblimit;sb++) //sb为每个子带,sblimit为总子带数
{
fprintf(fileout, "子带[%d]:\t", sb + 1);
for(gr=0;gr<3;gr++) //gr为三个分组
{
fprintf(fileout, "%d\t", scalar[ch][gr][sb]);
}
fprintf(fileout, "\n");
}
}
fprintf(fileout, "\n");
fprintf(fileout, "========== 比特分配表 ==========\n"); //输出比特分配结果
for (ch = 0; ch < nch; ch++)
{
fprintf(fileout, "------ 声道%d ------\n", ch + 1); //按声道分配
for (sb = 0; sb < frame.sblimit; sb++)
{
fprintf(fileout, "子带[%d]:\t%d\n", sb + 1, bit_alloc[ch][sb]);
}
fprintf(fileout, "\n");
}
}
#endif
用一段环境噪音environment.wav进行测试：
3.选择三个不同特性的音频文件进行分析
（1）噪声
持续噪声：采用一段嘈杂的环境声environment.wav进行测试
突发噪声：采用笔掉落在地上的声音pen.wav进行测试
（2）音乐
截取一段《英雄赞歌》的片段yxzg.wav进行测试
（3）混合
采用《英雄赞歌》和嘈杂环境声混合而成的mix.wav进行测试
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