综合编码——MPEG音频编码实验
作者：互联网
目录
一、程序设计整体框架
1、MPEG-1 Audio LayerII编码器原理
将信源输出分解为不同频率的子带，然后对不同频率的子带进行编码
2、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
（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。
（2）确定声压级别
（3）考虑安静时阈值
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。
（4）音频信号分解
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
（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）音调和非音调掩蔽成分的消除
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）音调和非音调掩蔽成分的消除
还要考虑掩蔽效应的影响。
（8）音调和非音调掩蔽成分的消除
- 选择出本子带中最小的阈值作为子带阈值
- 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。
- 选择出本子带中最小的阈值作为子带阈值
（9）计算掩蔽比SMR
SMR = 信号能量 / 掩蔽阈值
计算每个子带信号掩蔽比，并将SMR传递给编码单元
3、多相滤波器组设计
- 将PCM样本变换到32个子带的频域信号
（1）量化和编码
① 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。
第2层的一帧对应36个子带样值，是第1层的三倍，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。
使用这一算法后，和第1层相比，第2层传输的比例因 子平均减少了2个，即传输码率由22.5Kb/s降低到了 7.5Kb/s。
② 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，是信噪比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。
第1层一帧用4比特给每个子带的比特分配信 息编码；而第2层只在低频段用4比特，高频段则用2比特。
③ 子带样值的量化和编码
输入以12个样本为一组，每组样本经过时间-频率变换 之后进行一次比特分配并记录一个比例因子(scale factor)
比特分配信息告诉解码器每个样本由几位表示，比例 因子用6比特表示，解码器使用这个6比特的比例因子 乘逆量化器的每个输出样本值，以恢复被量化的子带 值。比例因子的作用是充分利用量化器的量化范围， 通过比特分配和比例因子相配合，可以表示动态范围 超过120dB的样本 。
第2层中，量化级别的数目随子带的不同而不同，但量 化等级仍然覆盖了3~65535的范围，同时子带不被分 配给比特的概率增加了，没有分配给比特的子带就不 被量化。低频段的量化等级有15级，中频段7级，高频段只有3级。
二、具体程序实现
m2aenc.c
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <time.h>
#include "common.h"
#include "encoder.h"
#include "musicin.h"
#include "options.h"
#include "audio_read.h"
#include "bitstream.h"
#include "mem.h"
#include "crc.h"
#include "psycho_n1.h"
#include "psycho_0.h"
#include "psycho_1.h"
#include "psycho_2.h"
#include "psycho_3.h"
#include "psycho_4.h"
#include "encode.h"
#include "availbits.h"
#include "subband.h"
#include "encode_new.h"
#include "m2aenc.h"
#include <assert.h>
FILE *musicin;
Bit_stream_struc bs;
char *programName;
char toolameversion[10] = "0.2l";
void global_init (void)
{
glopts.usepsy = TRUE;
glopts.usepadbit = TRUE;
glopts.quickmode = FALSE;
glopts.quickcount = 10;
glopts.downmix = FALSE;
glopts.byteswap = FALSE;
glopts.channelswap = FALSE;
glopts.vbr = FALSE;
glopts.vbrlevel = 0;
glopts.athlevel = 0;
glopts.verbosity = 2;
}
/************************************************************************
*
* main
*
* PURPOSE: MPEG II Encoder with
* psychoacoustic models 1 (MUSICAM) and 2 (AT&T)
*
* SEMANTICS: One overlapping frame of audio of up to 2 channels are
* processed at a time in the following order:
* (associated routines are in parentheses)
*
* 1. Filter sliding window of data to get 32 subband
* samples per channel.
* (window_subband,filter_subband)
*
* 2. If joint stereo mode, combine left and right channels
* for subbands above #jsbound#.
* (combine_LR)
*
* 3. Calculate scalefactors for the frame, and
* also calculate scalefactor select information.
* (*_scale_factor_calc)
*
* 4. Calculate psychoacoustic masking levels using selected
* psychoacoustic model.
* (psycho_i, psycho_ii)
*
* 5. Perform iterative bit allocation for subbands with low
* mask_to_noise ratios using masking levels from step 4.
* (*_main_bit_allocation)
*
* 6. If error protection flag is active, add redundancy for
* error protection.
* (*_CRC_calc)
*
* 7. Pack bit allocation, scalefactors, and scalefactor select
*headerrmation onto bitstream.
* (*_encode_bit_alloc,*_encode_scale,transmission_pattern)
*
* 8. Quantize subbands and pack them into bitstream
* (*_subband_quantization, *_sample_encoding)
*
************************************************************************/
int frameNum = 0;
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;
/*-------------------------*/
FILE* Trace = NULL;
fopen_s(&Trace, "trace.txt", "w");
fprintf(Trace, "该帧比例因子和比特分配表如下：\n");
/*------------------------*/
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 (frameNum == 1)
{
fprintf(Trace, "bit allocation:\n");
for (int i = 0; i < frame.sblimit; i++)
fprintf(Trace, "subband[%d]:%d bits\n", i, bit_alloc[0][i]);
}
/*---------------------*/
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);
/*-------------------------*/
if (frameNum == 1)
{
fprintf(Trace, "scalefactors:\n");
for (int i = 0; i < frame.sblimit; i++)
fprintf(Trace, "subband[%d] scalefactors:%d %d %d\n", i, scalar[0][0][i], scalar[0][1][i], scalar[0][2][i]);
}
/*-----------------------*/
#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(Trace, "该音频声道数：%d\n", nch);
fprintf(Trace, "观测第 %d 帧\n", frameNum);
fprintf(Trace, "本帧比特预算：%d bits\n", adb);
time(&end_time);
total_time = end_time - start_time;
printf("total time is %d\n", total_time);
exit (0);
}
/************************************************************************
*
* print_config
*
* PURPOSE: Prints the encoding parameters used
*
************************************************************************/
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");
}
/************************************************************************
*
* usage
*
* PURPOSE: Writes command line syntax to the file specified by #stderr#
*
************************************************************************/
void usage (void)
{ /* print syntax & exit */
/* FIXME: maybe have an option to display better definitions of help codes, and
long equivalents of the flags */
fprintf (stdout, "\ntooLAME version %s (http://toolame.sourceforge.net)\n",
toolameversion);
fprintf (stdout, "MPEG Audio Layer II encoder\n\n");
fprintf (stdout, "usage: \n");
fprintf (stdout, "\t%s [options] <input> <output>\n\n", programName);
fprintf (stdout, "Options:\n");
fprintf (stdout, "Input\n");
fprintf (stdout, "\t-s sfrq input smpl rate in kHz (dflt %4.1f)\n",
DFLT_SFQ);
fprintf (stdout, "\t-a downmix from stereo to mono\n");
fprintf (stdout, "\t-x force byte-swapping of input\n");
fprintf (stdout, "\t-g swap channels of input file\n");
fprintf (stdout, "Output\n");
fprintf (stdout, "\t-m mode channel mode : s/d/j/m (dflt %4c)\n",
DFLT_MOD);
fprintf (stdout, "\t-p psy psychoacoustic model 0/1/2/3 (dflt %4u)\n",
DFLT_PSY);
fprintf (stdout, "\t-b br total bitrate in kbps (dflt 192)\n");
fprintf (stdout, "\t-v lev vbr mode\n");
fprintf (stdout, "\t-l lev ATH level (dflt 0)\n");
fprintf (stdout, "Operation\n");
// fprintf (stdout, "\t-f fast mode (turns off psy model)\n");
// deprecate the -f switch. use "-p 0" instead.
fprintf (stdout,
"\t-q num quick mode. only calculate psy model every num frames\n");
fprintf (stdout, "Misc\n");
fprintf (stdout, "\t-d emp de-emphasis n/5/c (dflt %4c)\n",
DFLT_EMP);
fprintf (stdout, "\t-c mark as copyright\n");
fprintf (stdout, "\t-o mark as original\n");
fprintf (stdout, "\t-e add error protection\n");
fprintf (stdout, "\t-r force padding bit/frame off\n");
fprintf (stdout, "\t-D len add DAB extensions of length [len]\n");
fprintf (stdout, "\t-t talkativity 0=no messages (dflt 2)");
fprintf (stdout, "Files\n");
fprintf (stdout,
"\tinput input sound file. (WAV,AIFF,PCM or use '/dev/stdin')\n");
fprintf (stdout, "\toutput output bit stream of encoded audio\n");
fprintf (stdout,
"\n\tAllowable bitrates for 16, 22.05 and 24kHz sample input\n");
fprintf (stdout,
"\t8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160\n");
fprintf (stdout,
"\n\tAllowable bitrates for 32, 44.1 and 48kHz sample input\n");
fprintf (stdout,
"\t32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384\n");
exit (1);
}
/*********************************************
* void short_usage(void)
********************************************/
void short_usage (void)
{
/* print a bit of info about the program */
fprintf (stderr, "tooLAME version %s\n (http://toolame.sourceforge.net)\n",
toolameversion);
fprintf (stderr, "MPEG Audio Layer II encoder\n\n");
fprintf (stderr, "USAGE: %s [options] <infile> [outfile]\n\n", programName);
fprintf (stderr, "Try \"%s -h\" for more information.\n", programName);
exit (0);
}
/*********************************************
* void proginfo(void)
********************************************/
void proginfo (void)
{
/* print a bit of info about the program */
fprintf (stderr,
"\ntooLAME version 0.2g (http://toolame.sourceforge.net)\n");
fprintf (stderr, "MPEG Audio Layer II encoder\n\n");
}
/************************************************************************
*
* parse_args
*
* PURPOSE: Sets encoding parameters to the specifications of the
* command line. Default settings are used for parameters
* not specified in the command line.
*
* SEMANTICS: The command line is parsed according to the following
* syntax:
*
* -m is followed by the mode
* -p is followed by the psychoacoustic model number
* -s is followed by the sampling rate
* -b is followed by the total bitrate, irrespective of the mode
* -d is followed by the emphasis flag
* -c is followed by the copyright/no_copyright flag
* -o is followed by the original/not_original flag
* -e is followed by the error_protection on/off flag
* -f turns off psy model (fast mode)
* -q <i> only calculate psy model every ith frame
* -a downmix from stereo to mono
* -r turn off padding bits in frames.
* -x force byte swapping of input
* -g swap the channels on an input file
* -t talkativity. how verbose should the program be. 0 = no messages.
*
* If the input file is in AIFF format, the sampling frequency is read
* from the AIFF header.
*
* The input and output filenames are read into #inpath# and #outpath#.
*
************************************************************************/
void parse_args (int argc, char **argv, frame_info * frame, int *psy,
unsigned long *num_samples, char inPath[MAX_NAME_SIZE],
char outPath[MAX_NAME_SIZE])
{
FLOAT srate;
int brate;
frame_header *header = frame->header;
int err = 0, i = 0;
long samplerate;
/* preset defaults */
inPath[0] = '\0';
outPath[0] = '\0';
header->lay = DFLT_LAY;
switch (DFLT_MOD) {
case 's':
header->mode = MPG_MD_STEREO;
header->mode_ext = 0;
break;
case 'd':
header->mode = MPG_MD_DUAL_CHANNEL;
header->mode_ext = 0;
break;
/* in j-stereo mode, no default header->mode_ext was defined, gave error..
now default = 2 added by MFC 14 Dec 1999. */
case 'j':
header->mode = MPG_MD_JOINT_STEREO;
header->mode_ext = 2;
break;
case 'm':
header->mode = MPG_MD_MONO;
header->mode_ext = 0;
break;
default:
fprintf (stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD);
abort ();
}
*psy = DFLT_PSY;
if ((header->sampling_frequency =
SmpFrqIndex ((long) (1000 * DFLT_SFQ), &header->version)) < 0) {
fprintf (stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ);
abort ();
}
header->bitrate_index = 14;
brate = 0;
switch (DFLT_EMP) {
case 'n':
header->emphasis = 0;
break;
case '5':
header->emphasis = 1;
break;
case 'c':
header->emphasis = 3;
break;
default:
fprintf (stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP);
abort ();
}
header->copyright = 0;
header->original = 0;
header->error_protection = FALSE;
header->dab_extension = 0;
/* process args */
while (++i < argc && err == 0) {
char c, *token, *arg, *nextArg;
int argUsed;
token = argv[i];
if (*token++ == '-') {
if (i + 1 < argc)
nextArg = argv[i + 1];
else
nextArg = "";
argUsed = 0;
if (!*token) {
/* The user wants to use stdin and/or stdout. */
if (inPath[0] == '\0')
strncpy (inPath, argv[i], MAX_NAME_SIZE);
else if (outPath[0] == '\0')
strncpy (outPath, argv[i], MAX_NAME_SIZE);
}
while ((c = *token++)) {
if (*token /* NumericQ(token) */ )
arg = token;
else
arg = nextArg;
switch (c) {
case 'm':
argUsed = 1;
if (*arg == 's') {
header->mode = MPG_MD_STEREO;
header->mode_ext = 0;
} else if (*arg == 'd') {
header->mode = MPG_MD_DUAL_CHANNEL;
header->mode_ext = 0;
} else if (*arg == 'j') {
header->mode = MPG_MD_JOINT_STEREO;
} else if (*arg == 'm') {
header->mode = MPG_MD_MONO;
header->mode_ext = 0;
} else {
fprintf (stderr, "%s: -m mode must be s/d/j/m not %s\n",
programName, arg);
err = 1;
}
break;
case 'p':
*psy = atoi (arg);
argUsed = 1;
break;
case 's':
argUsed = 1;
srate = atof (arg);
/* samplerate = rint( 1000.0 * srate ); $A */
samplerate = (long) ((1000.0 * srate) + 0.5);
if ((header->sampling_frequency =
SmpFrqIndex ((long) samplerate, &header->version)) < 0)
err = 1;
break;
case 'b':
argUsed = 1;
brate = atoi (arg);
break;
case 'd':
argUsed = 1;
if (*arg == 'n')
header->emphasis = 0;
else if (*arg == '5')
header->emphasis = 1;
else if (*arg == 'c')
header->emphasis = 3;
else {
fprintf (stderr, "%s: -d emp must be n/5/c not %s\n", programName,
arg);
err = 1;
}
break;
case 'D':
argUsed = 1;
header->dab_length = atoi (arg);
header->error_protection = TRUE;
header->dab_extension = 2;
glopts.dab = TRUE;
break;
case 'c':
header->copyright = 1;
break;
case 'o':
header->original = 1;
break;
case 'e':
header->error_protection = TRUE;
break;
case 'f':
*psy = 0;
/* this switch is deprecated? FIXME get rid of glopts.usepsy
instead us psymodel 0, i.e. "-p 0" */
glopts.usepsy = FALSE;
break;
case 'r':
glopts.usepadbit = FALSE;
header->padding = 0;
break;
case 'q':
argUsed = 1;
glopts.quickmode = TRUE;
glopts.usepsy = TRUE;
glopts.quickcount = atoi (arg);
if (glopts.quickcount == 0) {
/* just don't use psy model */
glopts.usepsy = FALSE;
glopts.quickcount = FALSE;
}
break;
case 'a':
glopts.downmix = TRUE;
header->mode = MPG_MD_MONO;
header->mode_ext = 0;
break;
case 'x':
glopts.byteswap = TRUE;
break;
case 'v':
argUsed = 1;
glopts.vbr = TRUE;
glopts.vbrlevel = atof (arg);
glopts.usepadbit = FALSE; /* don't use padding for VBR */
header->padding = 0;
/* MFC Feb 2003: in VBR mode, joint stereo doesn't make
any sense at the moment, as there are no noisy subbands
according to bits_for_nonoise in vbr mode */
header->mode = MPG_MD_STEREO; /* force stereo mode */
header->mode_ext = 0;
break;
case 'l':
argUsed = 1;
glopts.athlevel = atof(arg);
break;
case 'h':
usage ();
break;
case 'g':
glopts.channelswap = TRUE;
break;
case 't':
argUsed = 1;
glopts.verbosity = atoi (arg);
break;
default:
fprintf (stderr, "%s: unrec option %c\n", programName, c);
err = 1;
break;
}
if (argUsed) {
if (arg == token)
token = ""; /* no more from token */
else
++i; /* skip arg we used */
arg = "";
argUsed = 0;
}
}
} else {
if (inPath[0] == '\0')
strcpy (inPath, argv[i]);
else if (outPath[0] == '\0')
strcpy (outPath, argv[i]);
else {
fprintf (stderr, "%s: excess arg %s\n", programName, argv[i]);
err = 1;
}
}
}
if (header->dab_extension) {
/* in 48 kHz */
/* if the bit rate per channel is less then 56 kbit/s, we have 2 scf-crc */
/* else we have 4 scf-crc */
/* in 24 kHz, we have 4 scf-crc, see main loop */
if (brate / (header->mode == MPG_MD_MONO ? 1 : 2) >= 56)
header->dab_extension = 4;
}
if (err || inPath[0] == '\0')
usage (); /* If no infile defined, or err has occured, then call usage() */
if (outPath[0] == '\0') {
/* replace old extension with new one, 1992-08-19, 1995-06-12 shn */
new_ext (inPath, DFLT_EXT, outPath);
}
if (!strcmp (inPath, "-")) {
musicin = stdin; /* read from stdin */
*num_samples = MAX_U_32_NUM;
} else {
if ((musicin = fopen (inPath, "rb")) == NULL) {
fprintf (stderr, "Could not find \"%s\".\n", inPath);
exit (1);
}
parse_input_file (musicin, inPath, header, num_samples);
}
/* check for a valid bitrate */
if (brate == 0)
brate = bitrate[header->version][10];
/* Check to see we have a sane value for the bitrate for this version */
if ((header->bitrate_index = BitrateIndex (brate, header->version)) < 0)
err = 1;
/* All options are hunky dory, open the input audio file and
return to the main drag */
open_bit_stream_w (&bs, outPath, BUFFER_SIZE);
}
void smr_dump(double smr[2][SBLIMIT], int nch) {
int ch, sb;
fprintf(stdout,"SMR:");
for (ch = 0;ch<nch; ch++) {
if (ch==1)
fprintf(stdout," ");
for (sb=0;sb<SBLIMIT;sb++)
fprintf(stdout,"%3.0f ",smr[ch][sb]);
fprintf(stdout,"\n");
}
}
三、实验结果
输出音频的采样率和目标码率以及某个数据帧分配的比特数、该帧的比例因子和比特分配结果
- 音乐
噪声
标签：编码,ch,stdout,音频,header,MPEG,fprintf,glopts,frame 来源： https://blog.csdn.net/weixin_44221452/article/details/117827535