MPEG音频编码
文章目录
MPEG-1 Audio Layer II编码器原理
编码器主要分为两条线，上方为子带编码部分，下方为比特分配部分。
- 上方滤波器组将PCM码流分割为多个子带，可使声音信号具有高的时间分辨率，确保在短暂冲击信号情况下，编码的信号具有足够高的质量。
- 下方1024点FFT运算将PCM码流转换到频域，使声音信号具有高的频率分辨率，再经过“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＜掩蔽域值。在低频子带中，为了保护音调和共振峰的结构，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。
- 最后通过数据帧包装将量化的子带样本和其它数据按照规定的帧格式组装成比特数据流。
多相滤波器组
多相滤波器组将PCM样本变换到32个等宽频带子带。
缺点如下：
- 等带宽的滤波器组与人类听觉系统的临界频带不对应，在低频区域，单个子带会覆盖多个临界频带。在这种情况下，量化比特数不能兼每个临界频带。
- 滤波器组与其逆过程不是无失真的，但滤波器组引入的误差差很小，且听不到。
- 子带间频率有混叠，滤波后的相邻子带有频率混叠现象，一个子带中的信号可以影响相邻子带的输出。
**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，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宽度。
心理声学模型
听觉系统中存在一个听觉阈值电平，听不到低于这个电平的声音信号。听觉阈值的大小随声音频率的改变而改变，这就是听觉掩蔽特性。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值。心理声学模型用于计算信号中不可听觉感知的部分。
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。Layer II采用1024点FFT，将信号变换到频域。
确定声压级别
考虑安静时阈值
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。
音频信号分解
将音频信号分解成“乐音(tones)” 和“非乐音/噪声”两部分，这两种信号的掩蔽能力不同。根据音频频谱的局部功率最大值确定乐音成分局部峰值为乐音，然后将本临界频带内的剩余频谱合在一起，组成一个代表噪声频率成分。
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。
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。
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。
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。
计算每个子带信号掩蔽比（SMR）
SMR = 信号能量 / 掩蔽阈值
将SMR传递给编码单元。
量化和编码
比例因子的取值和编码
对各个子带每12个样点进行一次比例因子计算。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。
Layer II的一帧对应36个子带样值，是Layer I的三倍，原则上要传三个比例因子。为了降低比例因子的传输码率，采用了利用人耳时域掩蔽特性的编码策略。每帧中每个子带的三个比例因子被一起考虑，划分成特定的几种模式。根据这些模式，1个、2个或3个比例因子和比例因子选择信息（每子带2比特）一起被传送。如果一个比例因子和下一个只有很小的差别，就只传送大的一个，这种情况对于稳态信号经常出现。
比特分配及编码
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。
比特分配的过程：
- 对每个子带计算掩蔽-噪声比MNR（信噪比SNR–信掩比SMR），即MNR = SNR–SMR。
- 使整个一帧和每个子带的总噪声-掩蔽比最小。这是一个循环过程，每一次循环使获益最大的子带的量化级别增加一级，当然所用比特数不能超过一帧所能提供的最大数目。
- Layer I一帧用4比特给每个子带的比特分配信息编码；而Layer II只在低频段用4比特，高频段则用2比特。
子带样值的量化和编码
每个子带都以比例系数归一化的值X进行量化计算：AxX+B，A和B：量化系数。根据比特分配信息得到的量化级数，查量化表得A、B。
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。量化级数随子带的不同而不同，量化等级覆盖了3~65535的范围，没有分配给比特的子带就不被量化。
程序实现
命令参数
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;
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);
/*将结果输出到txt文件*/
FILE* f;
f = fopen("output.txt", "wb");
print_config (&frame, &model, original_file_name, encoded_file_name, f);
/* 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);
/*添加部分*/
if (frameNum == 4)
{
fprintf(f, "\n");
fprintf(f, "第 %d 帧\n", frameNum);
fprintf(f, "该帧分配比特数：%d bits\n", adb);
fprintf(f, "\n");
fprintf(f, "————比例因子————\n");
for (ch = 0; ch < nch; ch++)
{
fprintf(f, "------ 声道%2d ------\n", ch + 1);
for (sb = 0; sb < frame.sblimit; sb++)
{
fprintf(f, "子带 %2d:\t", sb + 1);
for (int gr = 0; gr < 3; gr++) {
fprintf(f, "%2d\t", scalar[ch][gr][sb]); //输出比例因子
}
fprintf(f, "\n");
}
}
fprintf(f, "\n");
fprintf(f, " ————比特分配————\n");
for (ch = 0; ch < nch; ch++)
{
fprintf(f, "------ 声道%2d ------\n", ch + 1);
for (sb = 0; sb < frame.sblimit; sb++)
{
fprintf(f, "子带 %2d:\t%2d\n", sb + 1, bit_alloc[ch][sb]); //输出比特分配结果
}
}
}
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");
time(&end_time);
total_time = end_time - start_time;
printf("total time is %d\n", total_time);
exit (0);
}
输出音频的采样率和目标码率
main函数与print_config函数更改部分
int main (int argc, char **argv)
{
//......
/*将结果输出到txt文件*/
FILE* f;
f = fopen("output.txt", "wb");
print_config (&frame, &model, original_file_name, encoded_file_name, f);
//......
}
void print_config (frame_info * frame, int *psy, char *inPath,
char *outPath, FILE *f)
{
//......
fprintf(f, "输入文件：%s\n", inPath);
fprintf(f, "输出文件：%s\n", outPath);
fprintf(f, "采样频率：%.1f kHz\n", s_freq[header->version][header->sampling_frequency]);
fprintf(f, "目标码率：%d kbps\n", bitrate[header->version][header->bitrate_index]);
//......
}
输出某个数据帧所分配的比特数、比例因子和比特分配结果
main函数更改部分
int main (int argc, char **argv)
{
//......
if (frameNum == 4) //选择第4帧
{
fprintf(f, "\n");
fprintf(f, "第 %d 帧\n", frameNum);
fprintf(f, "该帧分配比特数：%d bits\n", adb); //输出该帧所分配的比特数
fprintf(f, "\n");
fprintf(f, "————比例因子————\n");
for (ch = 0; ch < nch; ch++)
{
fprintf(f, "------ 声道%2d ------\n", ch + 1);
for (sb = 0; sb < frame.sblimit; sb++)
{
fprintf(f, "子带 %2d:\t", sb + 1);
for (int gr = 0; gr < 3; gr++) {
fprintf(f, "%2d\t", scalar[ch][gr][sb]); //输出该帧的比例因子
}
fprintf(f, "\n");
}
}
fprintf(f, "\n");
fprintf(f, " ————比特分配————\n");
for (ch = 0; ch < nch; ch++)
{
fprintf(f, "------ 声道%2d ------\n", ch + 1);
for (sb = 0; sb < frame.sblimit; sb++)
{
fprintf(f, "子带 %2d:\t%2d\n", sb + 1, bit_alloc[ch][sb]); //输出该帧的比特分配结果
}
}
}
//......
}
实验结果
白噪声
歌曲片段
白噪声叠加歌曲
扫一扫
5860
被折叠的 条评论
为什么被折叠?
到【灌水乐园】发言
