MPEG-2音频的解码过程：
1、检查4个字节的音频头，过程为：
static inline int ff_mpa_check_header(uint32_t header){ /* header */ if ((header & 0xffe00000) != 0xffe00000) //即前11位为固定头信息，全为1 return -1; /* layer check */ if ((header & (3<<17)) == 0)//即第14、15两位表示layer return -1; /* bit rate */ if ((header & (0xf<<12)) == 0xf<<12)//即第17、18、19、20四位不能全为1 return -1; /* frequency */ if ((header & (3<<10)) == 3<<10)//即第21、22位不能全为1 return -1; return 0;}
2、根据header信息得到相应参数信息：
s->layer = 4 - ((header >> 17) & 3);)//即第14、15两位与0x0011相与，再被4减
sample_rate_index = (header >> 10) & 3;//获取的是采样率的索引，ff_mpa_freq_tab[3] = { 44100, 48000, 32000 };
sample_rate = ff_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);//其中，if (header & (1<<20))
//{s->lsf =(header & (1<<19)) ? 0 : 1; // mpeg25 = 0; // } else { //s->lsf = 1; //mpeg25 = 1; // }
sample_rate_index += 3 * (s->lsf + mpeg25);s->sample_rate_index = sample_rate_index;
s->sample_rate = sample_rate;
bitrate_index = (header >> 12) & 0xf;
s->mode = (header >> 6) & 3;
padding = (header >> 9) & 1;
if (s->mode == MPA_MONO) s->nb_channels = 1; else s->nb_channels = 2;
if (bitrate_index != 0) { frame_size = ff_mpa_bitrate_tab[s->lsf][s->layer - 1][bitrate_index]; s->bit_rate = frame_size * 1000; switch(s->layer) { case 1: frame_size = (frame_size * 12000) / sample_rate; frame_size = (frame_size + padding) * 4; break; case 2: frame_size = (frame_size * 144000) / sample_rate; frame_size += padding; break; default: case 3: frame_size = (frame_size * 144000) / (sample_rate << s->lsf); frame_size += padding; break; } s->frame_size = frame_size; } else { /* if no frame size computed, signal it */ return 1; }
其中，
const uint16_t ff_mpa_bitrate_tab[2][3][15] = { { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 }, {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 }, {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 } }, { {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256}, {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160} }};
根据layer数，即可知道该音频的编码类型：
layer = 1, 对应MPEG1
layer = 2, 对应MPEG2，framesize = 1152 //1152与外面开辟的保存输出数据的缓冲大小有关，如果开辟空间大小<1152*avctx- //>channels*2时，则空间太小；
layer = 3, 对应MPEG3（MP3）
通过比较送入的数据buf_size和解析header获取的数据大小 s->frame_size相比，即可得知当前送入的数据是否有问题：
如果s->frame_size > buf_size，则送入的帧不够；
如果s->frame_size < buf_size，则该帧可能有问题；
3、解码
3.1 /*先根据参数（bitrate、channel、samplerate）等信息，选取码表*/
table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels, s->sample_rate, s->lsf);
3.2 /* parse bit allocation */
j = 0; for(i=0;i<bound;i++) { bit_alloc_bits = alloc_table[j]; for(ch=0;ch<s->nb_channels;ch++) { bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits); } j += 1 << bit_alloc_bits; } for(i=bound;i<sblimit;i++) { bit_alloc_bits = alloc_table[j]; v = get_bits(&s->gb, bit_alloc_bits); bit_alloc[0][i] = v; bit_alloc[1][i] = v; j += 1 << bit_alloc_bits; }
3.3 /* scale codes */
for(i=0;i<sblimit;i++) { for(ch=0;ch<s->nb_channels;ch++) { if (bit_alloc[ch][i]) scale_code[ch][i] = get_bits(&s->gb, 2); } }
3.4 /* scale factors */
for(i=0;i<sblimit;i++) { for(ch=0;ch<s->nb_channels;ch++) { if (bit_alloc[ch][i]) { sf = scale_factors[ch][i]; switch(scale_code[ch][i]) { default: case 0: sf[0] = get_bits(&s->gb, 6); sf[1] = get_bits(&s->gb, 6); sf[2] = get_bits(&s->gb, 6); break; case 2: sf[0] = get_bits(&s->gb, 6); sf[1] = sf[0]; sf[2] = sf[0]; break; case 1: sf[0] = get_bits(&s->gb, 6); sf[2] = get_bits(&s->gb, 6); sf[1] = sf[0]; break; case 3: sf[0] = get_bits(&s->gb, 6); sf[2] = get_bits(&s->gb, 6); sf[1] = sf[2]; break; } } } }
3.5 /* samples */
for(k=0;k<3;k++) { for(l=0;l<12;l+=3) { j = 0; for(i=0;i<bound;i++) { bit_alloc_bits = alloc_table[j]; for(ch=0;ch<s->nb_channels;ch++) { b = bit_alloc[ch][i]; if (b) { scale = scale_factors[ch][i][k]; qindex = alloc_table[j+b]; bits = ff_mpa_quant_bits[qindex]; if (bits < 0) { int v2; /* 3 values at the same time */ v = get_bits(&s->gb, -bits); v2 = division_tabs[qindex][v]; steps = ff_mpa_quant_steps[qindex];
s->sb_samples[ch][k * 12 + l + 0][i] = l2_unscale_group(steps, v2 & 15, scale); s->sb_samples[ch][k * 12 + l + 1][i] = l2_unscale_group(steps, (v2 >> 4) & 15, scale); s->sb_samples[ch][k * 12 + l + 2][i] = l2_unscale_group(steps, v2 >> 8 , scale); } else { for(m=0;m<3;m++) { v = get_bits(&s->gb, bits); v = l1_unscale(bits - 1, v, scale); s->sb_samples[ch][k * 12 + l + m][i] = v; } } } else { s->sb_samples[ch][k * 12 + l + 0][i] = 0; s->sb_samples[ch][k * 12 + l + 1][i] = 0; s->sb_samples[ch][k * 12 + l + 2][i] = 0; } } /* next subband in alloc table */ j += 1 << bit_alloc_bits; } /* XXX: find a way to avoid this duplication of code */ for(i=bound;i<sblimit;i++) { bit_alloc_bits = alloc_table[j]; b = bit_alloc[0][i]; if (b) { int mant, scale0, scale1; scale0 = scale_factors[0][i][k]; scale1 = scale_factors[1][i][k]; qindex = alloc_table[j+b]; bits = ff_mpa_quant_bits[qindex]; if (bits < 0) { /* 3 values at the same time */ v = get_bits(&s->gb, -bits); steps = ff_mpa_quant_steps[qindex]; mant = v % steps; v = v / steps; s->sb_samples[0][k * 12 + l + 0][i] = l2_unscale_group(steps, mant, scale0); s->sb_samples[1][k * 12 + l + 0][i] = l2_unscale_group(steps, mant, scale1); mant = v % steps; v = v / steps; s->sb_samples[0][k * 12 + l + 1][i] = l2_unscale_group(steps, mant, scale0); s->sb_samples[1][k * 12 + l + 1][i] = l2_unscale_group(steps, mant, scale1); s->sb_samples[0][k * 12 + l + 2][i] = l2_unscale_group(steps, v, scale0); s->sb_samples[1][k * 12 + l + 2][i] = l2_unscale_group(steps, v, scale1); } else { for(m=0;m<3;m++) { mant = get_bits(&s->gb, bits); s->sb_samples[0][k * 12 + l + m][i] = l1_unscale(bits - 1, mant, scale0); s->sb_samples[1][k * 12 + l + m][i] = l1_unscale(bits - 1, mant, scale1); } } } else { s->sb_samples[0][k * 12 + l + 0][i] = 0; s->sb_samples[0][k * 12 + l + 1][i] = 0; s->sb_samples[0][k * 12 + l + 2][i] = 0; s->sb_samples[1][k * 12 + l + 0][i] = 0; s->sb_samples[1][k * 12 + l + 1][i] = 0; s->sb_samples[1][k * 12 + l + 2][i] = 0; } /* next subband in alloc table */ j += 1 << bit_alloc_bits; } /* fill remaining samples to zero */ for(i=sblimit;i<SBLIMIT;i++) { for(ch=0;ch<s->nb_channels;ch++) { s->sb_samples[ch][k * 12 + l + 0][i] = 0; s->sb_samples[ch][k * 12 + l + 1][i] = 0; s->sb_samples[ch][k * 12 + l + 2][i] = 0; } } } }
nb->frames = 36;
3.6 ff_mpa_synth_filter过程
先进行逆dct变换，dct32(synth_buf, sb_samples);
之后进行apply_window_mp3_c变换；
4.0 最后输出的PCM数据的实际大小为：
nb_frames * 32 * 2* s->nb_channels;
