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ADPCM压缩算法

2024-06-29 来源:客趣旅游网
ADPCM压缩算法

ADPCM(Adaptive Differential Pulse Code Modulation),是一种针对 16bits( 或8bits或者更高) 声音波形数据的一种有损压缩算法,它将声音流中每次采样的 16bit 数据以 4bit 存储,所以压缩比 1:4. 而且压缩/解压缩算法非常简单,所以是一种低空间消耗,高质量高效率声音获得的好途径。保存声音的数据文件后缀名为 .AUD 的大多用ADPCM 压缩。 ADPCM 主要是针对连续的波形数据的,保存的是波形的变化情况,以达到描述整个波形的目的,由于它的编码和解码的过程却很简洁,列在后面,相信大家能够看懂。 8bits采样的声音人耳是可以勉强接受的,而 16bit 采样的声音可以算是高音质了。ADPCM 算法却可以将每次采样得到的 16bit 数据压缩到 4bit 。需要注意的是,如果要压缩/解压缩得是立体声信号,采样时,声音信号是放在一起的,需要将两个声道分别处理。 ADPCM 压缩过程

首先我们认为声音信号都是从零开始的,那么需要初始化两个变量 int index=0,prev_sample=0;

下面的循环将依次处理声音数据流,注意其中的 getnextsample() 应该得到一个 16bit 的采样数据,而 outputdata() 可以将计算出来的数据保存起来,程序中用到的 step_table[],index_adjust[] 附在后面: int index=0,prev_sample:=0; while (还有数据要处理) {

cur_sample=getnextsample(); // 得到当前的采样数据 delta=cur_sample-prev_sample; // 计算出和上一个的增量 if (delta<0) delta=-delta,sb=8; // 取绝对值

else sb = 0 ; // sb 保存的是符号位 code = 4*delta / step_table[index]; (取余运算) // 根据 steptable[]得到一个 0-7 的值

if (code>7) code=7; // 它描述了声音强度的变化量

index += index_adjust[code] ; // 根据声音强度调整下次取steptable 的序号

if (index<0) index=0; // 便于下次得到更精确的变

化量的描述

else if (index>88) index=88; prev_sample=cur_sample;

outputode(code|sb); // 加上符号位保存起来 }

ADPCM 解压缩过程

接压缩实际是压缩的一个逆过程,同样其中的 getnextcode() 应该得到一个编码,,而 outputsample() 可以将解码出来的声音信号保存起来。这段代码同样使用了同一个的 setp_table[] 和 index_adjust() 附在后面: int index=0,cur_sample=0; while (还有数据要处理) {

code=getnextcode(); // 得到下一个数据

if ((code & 8) != 0) sb=1 else sb=0;

code&=7; // 将 code 分离为数据和符号

delta = (step_table[index]*code)/4+step_table[index]/8; // 后面加的一项是为了减少误差

if (sb==1) delta=-delta;

cur_sample+=delta; // 计算出当前的波形数据

if (cur_sample>32767) output_sample(32767); else if (cur_sample<-32768) output_sample(-32768); else output_sample(cur_sample); index+=index_adjust[code]; if (index<0) index=0; if (index>88) index=88; } 附表

int index_adjust[8] = {-1,-1,-1,-1,2,4,6,8}; int step_table[89] = { 7,8,9,10,11,12,13,14, 16,17,19,21,23,25, 28,31,34,37 ,41,45,50,55,60,

66,73,80,88,97,107,118,130,143,

157,173,190,209,230,253,279,307,337,371, 408,449,494,544,598,658,

724,796,876,963,1060,1166,1282,1411,1552,1707,1878,2066, 2272,2499,2749,3024, 3327,3660,4026,4428,4871,5358,5894,6484,7132,7845,8630,9493,10442,11487,12635,13899,

15289,16818,18500,20350,22385,24623,27086,29794,32767 } TCPMP原代码赏析

/***************************************************************************** *

* This program is free software ; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. *

* This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. *

* You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software

* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA

*

* $Id: adpcm.h 271 2005-08-09 08:31:35Z picard $ *

* The Core Pocket Media Player * Copyright (c) 2004-2005 Gabor Kovacs *

****************************************************************************/ #ifndef __ADPCM_H #define __ADPCM_H

#define ADPCM_CLASS FOURCC('A','D','P','C') #define ADPCM_MS_ID FOURCC('A','D','M','S') #define ADPCM_IMA_ID FOURCC('A','D','I','M') #define ADPCM_IMA_QT_ID FOURCC('A','D','I','Q') #define ADPCM_G726_ID FOURCC('G','7','2','6') extern void ADPCM_Init(); extern void ADPCM_Done(); #endif

/***************************************************************************** *

* This program is free software ; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. *

* This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. *

* You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software

* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA

*

* $Id: adpcm.c 565 2006-01-12 14:11:44Z picard $ *

* The Core Pocket Media Player * Copyright (c) 2004-2005 Gabor Kovacs *

****************************************************************************/ #include \"../common/common.h\" #include \"adpcm.h\" #include \"g726/g72x.h\" typedef struct state {

int Predictor; int StepIndex; int Step; int Sample1; int Sample2; int CoEff1; int CoEff2; int IDelta; } state;

typedef struct adpcm {

codec Codec; buffer Data;

int Channel; //IMA_QT int16_t* Buffer; state State[2]; g726_state G726[2]; } adpcm;

static const int IndexTable[16] = {

-1, -1, -1, -1, 2, 4, 6, 8, -1, -1, -1, -1, 2, 4, 6, 8, };

static const int StepTable[89] = {

7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 };

// AdaptationTable[], AdaptCoeff1[], and AdaptCoeff2[] are from libsndfile static const int AdaptationTable[] = {

230, 230, 230, 230, 307, 409, 512, 614, 768, 614, 512, 409, 307, 230, 230, 230 };

static const int AdaptCoeff1[] = {

256, 512, 0, 192, 240, 460, 392 };

static const int AdaptCoeff2[] = {

0, -256, 0, 64, 0, -208, -232 };

static _INLINE int IMA_Calc(state* s, int v) {

int StepIndex; int Predictor; int Diff,Step;

Step = StepTable[s->StepIndex];

StepIndex = s->StepIndex + IndexTable[v]; if (StepIndex < 0) StepIndex = 0;

else if (StepIndex > 88) StepIndex = 88;

Diff = ((2 * (v & 7) + 1) * Step) >> 3; Predictor = s->Predictor; if (v & 8) Predictor -= Diff; else

Predictor += Diff; if (Predictor > 32767) Predictor = 32767;

else if (Predictor < -32768) Predictor = -32768;

s->Predictor = Predictor; s->StepIndex = StepIndex; return Predictor; }

static _INLINE int MS_Calc(state* s, int v) {

int Predictor;

Predictor = ((s->Sample1 * s->CoEff1) + (s->Sample2 * s->CoEff2)) >> 8; Predictor += ((v & 0x08) ? v-0x10:v) * s->IDelta;

if (Predictor > 32767) Predictor = 32767;

else if (Predictor < -32768) Predictor = -32768;

s->Sample2 = s->Sample1; s->Sample1 = Predictor;

s->IDelta = (AdaptationTable[v] * s->IDelta) >> 8; if (s->IDelta < 16) s->IDelta = 16; return Predictor; }

static int Process(adpcm* p, const packet* Packet, const flowstate* State) { int i; int Predictor; const uint8_t* In; const uint8_t* InEnd; int16_t* Out = p->Buffer; if (Packet) {

if (Packet->RefTime >= 0)

p->Codec.Packet.RefTime = Packet->RefTime; BufferPack(&p->Data,0);

BufferWrite(&p->Data,Packet->Data[0],Packet->Length,1024); } else

p->Codec.Packet.RefTime = TIME_UNKNOWN;

if (!BufferRead(&p->Data,&In,p->Codec.In.Format.Format.Audio.BlockAlign)) return ERR_NEED_MORE_DATA;

InEnd = In + p->Codec.In.Format.Format.Audio.BlockAlign;

switch (p->Codec.Node.Class) {

case ADPCM_G726_ID: {

g726_state *g1,*g2; g1 = g2 = &p->G726[0];

if (p->Codec.In.Format.Format.Audio.Channels==2) ++g2;

switch (p->Codec.In.Format.Format.Audio.Bits) { case 2:

for (;InOut[0] = (int16_t)g726_16_decoder(In[0] >> 6,g1); Out[1] = (int16_t)g726_16_decoder(In[0] >> 4,g2); Out[2] = (int16_t)g726_16_decoder(In[0] >> 2,g1); Out[3] = (int16_t)g726_16_decoder(In[0],g2); } break; case 3: InEnd -= 2;

for (;InOut[0] = (int16_t)g726_24_decoder(In[0] >> 5,g1); Out[1] = (int16_t)g726_24_decoder(In[0] >> 2,g2);

Out[2] = (int16_t)g726_24_decoder((In[0] << 1) | (In[1] >> 7),g1); Out[3] = (int16_t)g726_24_decoder(In[1] >> 4,g2); Out[4] = (int16_t)g726_24_decoder(In[1] >> 1,g1);

Out[5] = (int16_t)g726_24_decoder((In[1] << 2) | (In[2] >> 6),g2); Out[6] = (int16_t)g726_24_decoder(In[2] >> 3,g1); Out[7] = (int16_t)g726_24_decoder(In[2] >> 0,g2); } break; case 4:

for (;In{

Out[0] = (int16_t)g726_32_decoder(In[0] >> 4,g1); Out[1] = (int16_t)g726_32_decoder(In[0],g2); } break; case 5: InEnd -= 4;

for (;InOut[0] = (int16_t)g726_40_decoder(In[0] >> 3,g1);

Out[1] = (int16_t)g726_40_decoder((In[0] << 2) | (In[1] >> 6),g2); Out[2] = (int16_t)g726_40_decoder(In[1] >> 1,g1);

Out[3] = (int16_t)g726_40_decoder((In[1] << 4) | (In[2] >> 4),g2); Out[4] = (int16_t)g726_40_decoder((In[2] << 1) | (In[3] >> 7),g1); Out[5] = (int16_t)g726_40_decoder(In[3] >> 2,g2);

Out[6] = (int16_t)g726_40_decoder((In[3] << 3) | (In[4] >> 5),g1); Out[7] = (int16_t)g726_40_decoder(In[4] >> 0,g2); } break; } break; }

case ADPCM_IMA_QT_ID: {

int No,Ch;

Ch = p->Codec.In.Format.Format.Audio.Channels; for (No=0;Nostate *s; s = &p->State[0];

s->Predictor = (int16_t)((In[1] & 0x80) | (In[0] << 8)); s->StepIndex = In[1] & 0x7F; if (s->StepIndex > 88) s->StepIndex = 88;

In+=2; InEnd=In+32; Out = p->Buffer+No; for (;In*Out = (int16_t)IMA_Calc(s, In[0] & 0x0F); Out+=Ch;

*Out = (int16_t)IMA_Calc(s, In[0] >> 4); Out+=Ch; } }

Out = p->Buffer+Ch*64; break; }

case ADPCM_IMA_ID: {

state *s1,*s2; s1 = &p->State[0];

s1->Predictor = (int16_t)(In[0] | (In[1] << 8)); In+=2;

s1->StepIndex = *In++; if (s1->StepIndex > 88) s1->StepIndex = 88; ++In;

if (p->Codec.In.Format.Format.Audio.Channels == 2) {

s2 = &p->State[1];

s2->Predictor = (int16_t)(In[0] | (In[1] << 8)); In+=2;

s2->StepIndex = *In++; if (s2->StepIndex > 88)

s2->StepIndex = 88; ++In;

for (i=4;InOut[0] = (int16_t)IMA_Calc(s1, In[0] & 0x0F); Out[1] = (int16_t)IMA_Calc(s2, In[4] & 0x0F); Out[2] = (int16_t)IMA_Calc(s1, In[0] >> 4); Out[3] = (int16_t)IMA_Calc(s2, In[4] >> 4); if (--i==0) { i=4; In+=4; } } } else {

for (;InOut[0] = (int16_t)IMA_Calc(s1, In[0] & 0x0F); Out[1] = (int16_t)IMA_Calc(s1, In[0] >> 4); } } break; }

case ADPCM_MS_ID: {

state *s1,*s2; s1 = &p->State[0];

s2 = p->Codec.In.Format.Format.Audio.Channels==2 ? &p->State[1] : s1; Predictor = *In++; if (Predictor > 7) Predictor = 7;

s1->CoEff1 = AdaptCoeff1[Predictor]; s1->CoEff2 = AdaptCoeff2[Predictor]; if (s2 != s1) {

Predictor = *In++; if (Predictor > 7) Predictor = 7;

s2->CoEff1 = AdaptCoeff1[Predictor]; s2->CoEff2 = AdaptCoeff2[Predictor]; }

s1->IDelta = (int16_t)(In[0] | (In[1] << 8)); In+=2; if (s2 != s1) {

s2->IDelta = (int16_t)(In[0] | (In[1] << 8)); In+=2; }

s1->Sample1 = (int16_t)(In[0] | (In[1] << 8)); In+=2; if (s2 != s1) {

s2->Sample1 = (int16_t)(In[0] | (In[1] << 8)); In+=2; }

s1->Sample2 = (int16_t)(In[0] | (In[1] << 8)); In+=2; if (s2 != s1) {

s2->Sample2 = (int16_t)(In[0] | (In[1] << 8)); In+=2; }

*Out++ = (int16_t)s1->Sample1;

if (s2 != s1) *Out++ = (int16_t)s2->Sample1; *Out++ = (int16_t)s1->Sample2;

if (s2 != s1) *Out++ = (int16_t)s2->Sample2; for (;InOut[0] = (int16_t)MS_Calc(s1, In[0] >> 4); Out[1] = (int16_t)MS_Calc(s2, In[0] & 0x0F); } break; } }

p->Codec.Packet.Length = (uint8_t*)Out - (uint8_t*)p->Buffer; return ERR_NONE; }

static int UpdateInput(adpcm* p) {

BufferClear(&p->Data); free(p->Buffer); p->Buffer = NULL;

if (p->Codec.In.Format.Type == PACKET_AUDIO) {

PacketFormatPCM(&p->Codec.Out.Format,&p->Codec.In.Format,16); if (!p->Codec.In.Format.Format.Audio.BlockAlign) p->Codec.In.Format.Format.Audio.BlockAlign = 1024; if (p->Codec.Node.Class == ADPCM_IMA_QT_ID) p->Codec.In.Format.Format.Audio.BlockAlign = (32+2)*p->Codec.In.Format.Format.Audio.Channels; if (p->Codec.Node.Class == ADPCM_G726_ID) {

p->Codec.In.Format.Format.Audio.BlockAlign = 120; g726_init_state(&p->G726[0]); g726_init_state(&p->G726[1]); }

p->Buffer = (int16_t*)

malloc(sizeof(int16_t)*4*p->Codec.In.Format.Format.Audio.BlockAlign); if (!p->Buffer)

return ERR_OUT_OF_MEMORY;

p->Codec.Packet.Data[0] = p->Buffer; }

return ERR_NONE; }

static int Flush(adpcm* p) {

if (p->Codec.Node.Class == ADPCM_G726_ID) {

g726_init_state(&p->G726[0]); g726_init_state(&p->G726[1]); }

BufferDrop(&p->Data); p->Channel = 0; return ERR_NONE; }

static int Create(adpcm* p) {

p->Codec.Process = (packetprocess)Process; p->Codec.UpdateInput = (nodefunc)UpdateInput; p->Codec.Flush = (nodefunc)Flush; return ERR_NONE; }

static const nodedef ADPCM = {

sizeof(adpcm)|CF_ABSTRACT, ADPCM_CLASS, CODEC_CLASS, PRI_DEFAULT, (nodecreate)Create, NULL, };

static const nodedef ADPCM_MS = {

0, //parent size ADPCM_MS_ID, ADPCM_CLASS, PRI_DEFAULT, NULL, NULL, };

static const nodedef ADPCM_IMA = {

0, //parent size ADPCM_IMA_ID, ADPCM_CLASS, PRI_DEFAULT, NULL, NULL, };

static const nodedef ADPCM_IMA_QT = {

0, //parent size ADPCM_IMA_QT_ID, ADPCM_CLASS, PRI_DEFAULT, NULL,

NULL, };

static const nodedef ADPCM_G726 = {

0, //parent size ADPCM_G726_ID, ADPCM_CLASS, PRI_DEFAULT, NULL, NULL, };

void ADPCM_Init() {

NodeRegisterClass(&ADPCM); NodeRegisterClass(&ADPCM_MS); NodeRegisterClass(&ADPCM_IMA); NodeRegisterClass(&ADPCM_IMA_QT); NodeRegisterClass(&ADPCM_G726); }

void ADPCM_Done() {

NodeUnRegisterClass(ADPCM_MS_ID); NodeUnRegisterClass(ADPCM_IMA_ID); NodeUnRegisterClass(ADPCM_IMA_QT_ID); NodeUnRegisterClass(ADPCM_G726_ID); NodeUnRegisterClass(ADPCM_CLASS); }

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