本文详细介绍了MPEG-1音频编码的实验，包括多相滤波器组、心理声学模型I、Layer I和Layer II编码的过程。重点阐述了心理声学模型如何计算掩蔽阈值，并根据信号类型分配码率，以实现高效的数据压缩。实验涵盖了噪声、音乐和噪声+音乐三种类型的音频文件，通过理解这些内容，可以深入理解感知音频编码的设计原理。
数据压缩 | MPEG音频编码实验
一、内容概要
1. MPEG-1音频编码器
- 输入声音信号经过一个多相滤波器组变换到多个子带，同时经过“心理声学模型”计算以频率为自变量的噪声 掩蔽阈值。量化和编码部分用信掩比 SMR 决定分配给子带信号的量化位数，使量化噪声<掩比域值。最后通过数 据帧包装将量化的自带样本和其它数据按照规定而帧格式组装成比特数据流。
①多相滤波器组，用来分割子带
划分子带的方法有两种：线性划分和非线性
划分 线性划分可能一个子带覆盖好几个临界频带
②量化和编码——比例因子的取值和编码
对各个子带每 12 个样点金鼎一次比例因子计算。先定出 12 个样点中绝对值的最大值。查比例因子表中比这 个最大值大的最小值作为比例因子。用 6 比特表示。
③数据帧包装
2. MPEG-I 心理声学模型
- 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。
- 又可以使信号通过FFT运算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I
（1）将样本变换到频域
-
32个等分的子带信号并不能精确地反映人耳的听觉特性。引入FFT补偿频率分辨率不足的问题。
*采用Hann加权和DFT -
Hann加权减少频域中的边界效应
-
此变换不同于多相滤波器组，因为模型需要更精细的频率分辨率，而且计算掩蔽阈值也需要每个频率的幅值
-
模型1：采用512 (Layer I) 或1024 (Layers II and III)样本窗口
-
Layer I：每帧384个样本点，512个样本点足够覆盖
-
Layer II 和Layer III：每帧1152个样本点，每帧两次计算，模型1选择两个信号掩蔽比（SMR）中较小的一个
（2）确定声压级别
（3）考虑安静时阈值 -
也即绝对阈值。在标准中有根据输入PCM信号的采样率编制的“频率、临界频带率和绝对阈值”表。此表为多位科学家经多次心理声学实验所得。
（4）将音频信号分解成“乐音(tones)” 和“非乐音/噪声”部分：因为两种信号的掩蔽能力不同
同一临界频带内噪声掩蔽乐音 同一临界频带内乐音掩蔽噪声
（5）音调和非音调掩蔽成分的消除
利用标准中给出的绝对阈值消除被掩蔽成分；考虑在每个临界频带内，小于0.5Bark的距离中只保留最高功率的成分
（6）单个掩蔽阈值的计算
音调成分和非音调成分单个掩蔽阈值根据标准中给出的算法求得。
（7）全局掩蔽阈值的计算
还要考虑别的临界频带的影响。一个掩蔽信号会对其它频带上的信号产生掩蔽效应。这种掩蔽效应称为掩蔽扩散。
（8）每个子带的掩蔽阈值
选择出本子带中最小的阈值作为子带阈值
对高频不正确——高频区的临界频带很宽，可能跨越多个子带，从而导致模型1将临界带宽内所有的非音调部分集中为一个代表频率，当一个子带在很宽的频带内却远离代表频率时，无法得到准确的非音调掩蔽值。但计算量低。
（9）计算每个子带信号掩蔽比(signal-to-mask ratio, SMR)
- SMR = 信号能量 / 掩蔽阈值
- 并将SMR传递给编码单元
3. 多项滤波器组
4. Layer I 编码
4.1 码率分配
4.1.1 在调整到固定的码率之前
- 先确定可用于样值编码的有效比特数
- 这个数值取决于比例因子、比例因子选择信
息、比特分配信息以及辅助数据所需比特数
4.1.2 比特分配的过程
- 对每个子带计算掩蔽-噪声比MNR，是信噪比SNR–信掩比SMR，即：MNR = SNR–SMR
- NMR=SMR-SNR
4.1.3 算法：循环，直到没有比特可用：
- NMR = SMR– SNR (dB)
- 对最高NMR的子带分配比特，使获益最大的子带的量化级别增加一级
- 重新计算分配了更多比特子带的NMR
4.2 装帧
5. Layer II编码
5.1 量化
Layer I：每个子带从相同的量化集合中选择 每个子带取共14个量化器中的一个
Layer II： 根据采样和码率量化，不同子带可以从不同的量
化器集合中选择
- 某些（高频）子带的比特数可能为0 对量化级别在3、5、9级时，采用“颗粒” 优化
- 颗粒= 3 个样本，根据颗粒选择量化水平
5.2 装帧
二、实验要求
◼ 理解程序设计的整体框架
◼ 理解感知音频编码的设计思想
两条线
时-频分析的矛盾！
◼ 理解心理声学模型的实现过程
临界频带的概念
掩蔽值计算的思路
◼ 理解码率分配的实现思路 主页
◼ 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
噪声（持续噪声、突发噪声）
音乐
混合
◼ 某个数据帧，输出
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
三、主要程序
- main函数
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;
frame_header header;
char original_file_name[MAX_NAME_SIZE];
char encoded_file_name[MAX_NAME_SIZE];
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 smr[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT];
// FLOAT snr32[32];
short sam[2][1344]; /* was [1056]; */
int model, nch, error_protection;
static unsigned int crc;
int sb, ch, 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;
int gr;
FILE *out_txt=NULL;
unsigned char *outTXT=NULL;
out_txt=fopen("output.txt","w");
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);
print_config (&frame, &model, original_file_name, encoded_file_name);
/* this will load the alloc tables and do some other stuff */
hdr_to_frps (&frame);
nch = frame.nch;
error_protection = header.error_protection;
while (get_audio (musicin, buffer, num_samples, nch, &header) > 0) {
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;
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);
pick_scale (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 (*sb_sample, *j_sample, frame.sblimit);
scale_factor_calc (j_sample, &j_scale, 1, frame.sblimit);
}
#endif
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);
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);
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);
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");
//add
fprintf(out_txt, "该音频声道数：%d\n", nch);
fprintf(out_txt, "观测第 %d 帧\n", frameNum);
fprintf(out_txt, "本帧所分配比特：%d bits\n", adb);
fprintf(out_txt, "该帧比例因子和比特分配结果如下：\n");
for (ch = 0; ch < nch; ch++)
{
fprintf(out_txt, "--- 声道%2d ----\n", ch + 1);
for (sb = 0; sb < frame.sblimit; sb++)
{
fprintf(out_txt, "子带[%2d]比例因子:\t", sb + 1);
for (gr = 0; gr < 3; gr++)
{
fprintf(out_txt, "%2d\t", scalar[ch][gr][sb]);
}
fprintf(out_txt, "\n");
fprintf(out_txt, "子带[%2d]比特分配表:\t%2d\n", sb + 1, bit_alloc[ch][sb]);
fprintf(out_txt, "\n");
}
}
//end
time(&end_time);
total_time = end_time - start_time;
printf("total time is %d\n", total_time);
exit (0);
system("pause");
}
- 输出音频的采样率和目标码率
void print_config (frame_info * frame, int *psy, char *inPath,
char *outPath)
{
frame_header *header = frame->header;
if (glopts.verbosity == 0)
return;
fprintf (stderr, "--------------------------------------------\n");
fprintf (stderr, "Input File : '%s' %.1f kHz\n",
(strcmp (inPath, "-") ? inPath : "stdin"),
s_freq[header->version][header->sampling_frequency]);
fprintf (stderr, "Output File: '%s'\n",
(strcmp (outPath, "-") ? outPath : "stdout"));
fprintf (stderr, "%d kbps ", bitrate[header->version][header->bitrate_index]);
fprintf (stderr, "%s ", version_names[header->version]);
if (header->mode != MPG_MD_JOINT_STEREO)
fprintf (stderr, "Layer II %s Psycho model=%d (Mode_Extension=%d)\n",
mode_names[header->mode], *psy, header->mode_ext);
else
fprintf (stderr, "Layer II %s Psy model %d \n", mode_names[header->mode],
*psy);
fprintf (stderr, "[De-emph:%s\tCopyright:%s\tOriginal:%s\tCRC:%s]\n",
((header->emphasis) ? "On" : "Off"),
((header->copyright) ? "Yes" : "No"),
((header->original) ? "Yes" : "No"),
((header->error_protection) ? "On" : "Off"));
fprintf (stderr, "[Padding:%s\tByte-swap:%s\tChanswap:%s\tDAB:%s]\n",
((glopts.usepadbit) ? "Normal" : "Off"),
((glopts.byteswap) ? "On" : "Off"),
((glopts.channelswap) ? "On" : "Off"),
((glopts.dab) ? "On" : "Off"));
if (glopts.vbr == TRUE)
fprintf (stderr, "VBR Enabled. Using MNR boost of %f\n", glopts.vbrlevel);
fprintf(stderr,"ATH adjustment %f\n",glopts.athlevel);
fprintf (stderr, "--------------------------------------------\n");
}
- output
#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);
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);
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");
fprintf(out_txt, "该音频声道数：%d\n", nch);
fprintf(out_txt, "观测第 %d 帧\n", frameNum);
fprintf(out_txt, "本帧所分配比特：%d bits\n", adb);
fprintf(out_txt, "该帧比例因子和比特分配结果如下：\n");
for (ch = 0; ch < nch; ch++)
{
fprintf(out_txt, "--- 声道%2d ----\n", ch + 1);
for (sb = 0; sb < frame.sblimit; sb++)
{
fprintf(out_txt, "子带[%2d]比例因子:\t", sb + 1);
for (gr = 0; gr < 3; gr++)
{
fprintf(out_txt, "%2d\t", scalar[ch][gr][sb]);
}
fprintf(out_txt, "\n");
fprintf(out_txt, "子带[%2d]比特分配表:\t%2d\n", sb + 1, bit_alloc[ch][sb]);
fprintf(out_txt, "\n");
}
}
time(&end_time);
total_time = end_time - start_time;
printf("total time is %d\n", total_time);
exit (0);
system("pause");
}
四、实验结果
1. noise
2. music
3.noise+music
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