MPEG原理分析及MPEG音频编码器的调试

一、程序设计的整体框架

主函数及注释：

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;                               //头信息、比特分配表、声道数、子带数等信息frame_header header;                          //头信息的内容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], 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];                    //各个子带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 ();                                 //初始化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);  //输出配置信息到窗口中/* this will load the alloc tables and do some other stuff */hdr_to_frps (&frame);                  //根据头信息来设定其他信息nch = frame.nch;error_protection = header.error_protection;while (get_audio (musicin, buffer, num_samples, nch, &header) > 0) {//获取音频信息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);        //计算可用比特数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++ )                  //每12个样点一组for ( bl = 0; bl < 12; bl++ )                   //每组12个for ( ch = 0; ch < nch; ch++ )             //声道数次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 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);}
#endif//选择合适的心理声学模型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 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 (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 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");time(&end_time);total_time = end_time - start_time;printf("total time is %d\n", total_time);exit (0);
}

二、感知音频编码的设计思想

基本思想：分析信号，去掉不能被感知的部分。

听觉阈值：

听觉掩蔽特性：

MPEG-1 Audio LayerII编码器：

该编码器的两条线： 1.码流经过滤波器组变为32个子带的频域信号，进行子带编码。
2.对码流做1024点fft变换，根据心理声学模型来分配比特数，进行编码。

时-频分析的矛盾： 时域取值间隔越短，频域带宽越宽，更难分析。

三、心理声学模型的实现过程

临界频带：

掩蔽值计算的思路：

四、码率分配的实现思路

对每个子带计算噪掩比NMR=SMR-SNR（dB）
对最高NMR的子带进行比特分配，使获益
最大的子带的量化级别增加一级，然后重新计算该子带的NMR，此时分配了更多比特的子带的信噪比（SNR）会提升，所以其NMR会下降。不断循环，直到没有比特可分配或者所有NMR都减到0。

五、输出音频的采样率和目标码率

查阅得原代码中带有输出采样频率和目标码率到屏幕的功能：

  fprintf (stderr, "Input File : '%s'   %.1f kHz\n",(strcmp (inPath, "-") ? inPath : "stdin"),s_freq[header->version][header->sampling_frequency]);            //输出采样频率fprintf (stderr, "Output File: '%s'\n",(strcmp (outPath, "-") ? outPath : "stdout"));fprintf (stderr, "%d kbps ", bitrate[header->version][header->bitrate_index]);//输出目标码率

六、输出某一数据帧的信息

为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，在主函数的while循环中添加如下代码：

FILE* frame_info_file;
frame_info_file = fopen("C:\\Users\\tonym\\Desktop\\study\\shujuyasuo\\mpeg\\实验6_MPG音频编码\\m2aenc\\frame_info_file.txt", "w");
if (frameNum == 50)//此处设定想第几帧
{fprintf(frame_info_file, "采样率：%f khz\n", s_freq[frame.header->version][frame.header->sampling_frequency]);fprintf(frame_info_file, "目标码率：%d kbps\n", bitrate[frame.header->version][frame.header->bitrate_index]);fprintf(frame_info_file, "第%d帧\n", frameNum);fprintf(frame_info_file, "所分配比特数：%d\n", adb);fprintf(frame_info_file, "比例因子：\n");for (ch = 0; ch < nch; ch++){fprintf(frame_info_file, "声道%d:\n", ch);for (sb = 0; sb < frame.sblimit; sb++){fprintf(frame_info_file, "  子带%2d:", sb);for (int gr = 0; gr < 3; gr++){fprintf(frame_info_file, "%4d", scalar[ch][gr][sb]);}fprintf(frame_info_file, "\n");}}fprintf(frame_info_file, "\n");fprintf(frame_info_file, "比特分配表：\n");for (ch = 0; ch < nch; ch++){fprintf(frame_info_file, "声道%d:\n", ch);for (sb = 0; sb < frame.sblimit; sb++){fprintf(frame_info_file, "  子带%2d:\t%d\n", sb, bit_alloc[ch][sb]);}fprintf(frame_info_file, "\n");}
}

输出样例：

输入文件为音乐（.wav）时的输出文件：

采样率：44.100000 khz
目标码率：192 kbps
第50帧
所分配比特数：5008
比例因子：
声道0:
子带 0: 10 11 11
子带 1: 18 19 23
子带 2: 17 17 21
子带 3: 22 25 27
子带 4: 30 31 33
子带 5: 24 25 29
子带 6: 22 27 30
子带 7: 19 23 26
子带 8: 42 43 45
子带 9: 29 31 35
子带10: 29 30 31
子带11: 29 29 29
子带12: 21 24 28
子带13: 24 23 27
子带14: 24 26 28
子带15: 23 23 31
子带16: 30 32 33
子带17: 28 29 32
子带18: 26 26 30
子带19: 29 31 35
子带20: 31 32 35
子带21: 29 31 36
子带22: 40 42 44
子带23: 52 54 49
子带24: 52 55 53
子带25: 52 53 55
子带26: 53 52 51
子带27: 56 52 55
子带28: 53 55 53
子带29: 54 55 51
比特分配表：
声道0:
子带 0: 9
子带 1: 8
子带 2: 8
子带 3: 8
子带 4: 6
子带 5: 7
子带 6: 7
子带 7: 7
子带 8: 3
子带 9: 6
子带10: 6
子带11: 6
子带12: 7
子带13: 6
子带14: 6
子带15: 6
子带16: 5
子带17: 5
子带18: 6
子带19: 4
子带20: 4
子带21: 4
子带22: 0
子带23: 0
子带24: 0
子带25: 0
子带26: 0
子带27: 0
子带28: 0
子带29: 0

输入文件为噪声（.wav）时的输出文件：

采样率：48.000000 khz
目标码率：192 kbps
第50帧
所分配比特数：4608
比例因子：
声道0:
子带 0: 24 23 24
子带 1: 26 26 27
子带 2: 28 25 27
子带 3: 28 28 26
子带 4: 25 24 26
子带 5: 25 26 25
子带 6: 29 27 28
子带 7: 32 31 31
子带 8: 31 31 33
子带 9: 32 32 33
子带10: 33 32 35
子带11: 37 35 38
子带12: 35 34 36
子带13: 35 34 34
子带14: 35 36 37
子带15: 39 36 37
子带16: 37 37 40
子带17: 40 40 40
子带18: 43 43 43
子带19: 45 44 43
子带20: 46 45 45
子带21: 50 48 47
子带22: 57 59 59
子带23: 60 57 60
子带24: 56 57 58
子带25: 58 57 57
子带26: 57 57 58
比特分配表：
声道0:
子带 0: 8
子带 1: 7
子带 2: 7
子带 3: 9
子带 4: 9
子带 5: 8
子带 6: 8
子带 7: 8
子带 8: 8
子带 9: 7
子带10: 7
子带11: 6
子带12: 6
子带13: 6
子带14: 6
子带15: 6
子带16: 6
子带17: 5
子带18: 5
子带19: 0
子带20: 0
子带21: 0
子带22: 0
子带23: 0
子带24: 0
子带25: 0
子带26: 0

输入为音乐与噪声混合（.wav）时的输出文件：

采样率：44.100000 khz
目标码率：192 kbps
第50帧
所分配比特数：5008
比例因子：
声道0:
子带 0: 12 11 11
子带 1: 15 16 16
子带 2: 17 15 17
子带 3: 20 22 21
子带 4: 19 20 18
子带 5: 18 18 19
子带 6: 19 20 20
子带 7: 20 21 21
子带 8: 25 26 27
子带 9: 25 26 25
子带10: 28 26 28
子带11: 26 28 29
子带12: 27 25 24
子带13: 24 26 21
子带14: 25 25 23
子带15: 26 24 24
子带16: 29 28 28
子带17: 30 29 27
子带18: 29 28 24
子带19: 29 32 32
子带20: 30 32 31
子带21: 29 30 32
子带22: 38 38 41
子带23: 58 55 57
子带24: 57 56 58
子带25: 59 60 58
子带26: 58 58 59
子带27: 57 58 57
子带28: 57 58 56
子带29: 58 54 59
声道1:
子带 0: 12 11 11
子带 1: 15 16 16
子带 2: 17 16 16
子带 3: 19 22 21
子带 4: 20 19 19
子带 5: 17 18 19
子带 6: 18 21 19
子带 7: 20 21 23
子带 8: 24 27 27
子带 9: 26 27 26
子带10: 27 27 28
子带11: 26 28 31
子带12: 27 27 24
子带13: 25 27 22
子带14: 24 24 22
子带15: 26 24 23
子带16: 29 28 29
子带17: 29 30 29
子带18: 27 28 23
子带19: 30 31 32
子带20: 30 31 32
子带21: 29 30 32
子带22: 37 39 43
子带23: 57 58 58
子带24: 59 57 57
子带25: 58 58 59
子带26: 59 58 58
子带27: 56 58 58
子带28: 57 58 56
子带29: 59 54 58
比特分配表：
声道0:
子带 0: 5
子带 1: 4
子带 2: 3
子带 3: 4
子带 4: 4
子带 5: 4
子带 6: 4
子带 7: 3
子带 8: 1
子带 9: 3
子带10: 2
子带11: 3
子带12: 3
子带13: 3
子带14: 2
子带15: 1
子带16: 1
子带17: 2
子带18: 3
子带19: 1
子带20: 1
子带21: 0
子带22: 0
子带23: 0
子带24: 0
子带25: 0
子带26: 0
子带27: 0
子带28: 0
子带29: 0
声道1:
子带 0: 5
子带 1: 4
子带 2: 3
子带 3: 4
子带 4: 5
子带 5: 3
子带 6: 4
子带 7: 3
子带 8: 1
子带 9: 3
子带10: 3
子带11: 2
子带12: 3
子带13: 2
子带14: 2
子带15: 1
子带16: 1
子带17: 1
子带18: 3
子带19: 1
子带20: 1
子带21: 0
子带22: 0
子带23: 0
子带24: 0
子带25: 0
子带26: 0
子带27: 0
子带28: 0
子带29: 0

MPEG原理分析及MPEG音频编码器的调试相关推荐

音频变速播放原理分析及实现方案
[时间:2019-05] [状态:Open] [关键词:音频,audio,倍速,变速,变调,soundtouch,sonic] 音频变调变速原理分析先来一段语音处理的理论: 变速变调可分为:变速不变 ...
sonic音频变速不变调的原理分析
1.问题背景音频的原始pcm(Pulse Code Modulation)数据是由采样频率.通道数以及采样精度(位宽)而决定.人耳能听到的频率范围是[20H~20kHz],所以常见的音频采样率是44 ...
实验五—JEPG 原理分析及 JPEG 解码器的调试
文章目录一.JPEG编码原理 1.Level Offset(水平偏移)能量的重新分配 2.DCT变换 3.Uniform scalar quantization 均匀标量量化 4.DC系数差分编码 ...
火山引擎 RTC 自研音频编码器 NICO 实践之路
本文作者:张德军 1. 前言随着互联网技术的不断发展,越来越多的人开始尝试使用或者依赖实时音视频产品解决团队沟通与协作问题.在通话过程中,我们时常会遇到因为网络波动(如拥塞.丢包.延时和抖动等)而导 ...
JPEG原理分析及 JPEG解码器的调试
文章目录数据压缩实验(五) 一.JPEG原理分析 1.概述优点缺点 2.JPEG编解码原理 (1)彩色空间 (2)Level offset--零偏置电平下移 (3)8x8 DCT--离散余弦变换 ...
bmp/gif/jpg图象最底层原理分析
bmp/gif/jpg图象最底层原理分析(1)---- JPG 转载自:http://www.cnblogs.com/glaivelee/archive/2011/10/10/2205151.html ...
一篇读懂：Android手机如何通过USB接口与外设通信（附原理分析及方案选型）
更多技术干货,欢迎扫码关注博主微信公众号:HowieXue,共同探讨软件知识经验,关注就有海量学习资料免费领哦: 目录 0背景 1.手机USB接口通信特点 1.1 使用方便 1.2 通用性强 1.3 ...
C语言C++编程学习：排序原理分析
C语言是面向过程的,而C++是面向对象的 C和C++的区别: C是一个结构化语言,它的重点在于算法和数据结构.C程序的设计首要考虑的是如何通过一个过程,对输入(或环境条件)进行运算处理得到输出(或实现 ...
数字电视电子节目指南（EPG）的实现原理分析
数字电视电子节目指南(EPG)的实现原理分析张剑武汉广电数字网络有限公司数字电视技术部副经理文章来源:局技术论文获奖文章近几年来,数字电视在我国得到了飞速发展.数字电视相对于传统的模拟电视 ...