一、程序设计整体框架
- 1、MPEG-1 Audio LayerII编码器原理
- 2、心理声学模型
- - （1）将样本变换到频域
  - （2）确定声压级别
  - （3）考虑安静时阈值
  - （4）音频信号分解
  - （5）音调和非音调掩蔽成分的消除
  - （6）音调和非音调掩蔽成分的消除
  - （7）音调和非音调掩蔽成分的消除
  - （8）音调和非音调掩蔽成分的消除
  - （9）计算掩蔽比SMR
- 3、多相滤波器组设计
- - （1）量化和编码
  - - ① 比例因子的取值和编码
    - ② 比特分配及编码
    - ③ 子带样值的量化和编码
二、具体程序实现
三、实验结果

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 ();elseparse_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 filterCombines 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 NEWENCODEscalefactor_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);}
#elsescale_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);}
#endifif ((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 NEWENCODEsf_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);
#elsetransmission_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 NEWENCODEextern int vbrstats_new[15];
#elseextern int vbrstats[15];
#endiffprintf (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 NEWENCODEfprintf (stdout,"%4i ",vbrstats_new[i]);
#elsefprintf (stdout, "%4i ", vbrstats[i]);
#endiffprintf (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);elsefprintf (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, andlong 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];elsenextArg = "";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;elsearg = 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.usepsyinstead 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 makeany 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 andreturn 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

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目录