AMR编码基于第三方opencore-amr-0.1.5库
支持PCM2AMR，WAV2AMR
test目录下：pcm2amr.c pcm2amr.c，linux环境下 调试通过，准备移植到安卓手机
资源下载链接：
//-------------------------------------------------------------------------------//
pcm2amr.h代码如下：
#ifndef pcm2amr_h
#define pcm2amr_h
#include
#include
#include
#include
#include
//#include "wavreader.h"
#define AMR_MAGIC_NUMBER "#!AMR\n"
#define PCM_FRAME_SIZE 160 // 8khz 8000*0.02=160
#define MAX_AMR_FRAME_SIZE 32
#define AMR_FRAME_COUNT_PER_SECOND 50
//int amrEncodeMode[] = {4750, 5150, 5900, 6700, 7400, 7950, 10200, 12200}; // amr 编码方式
typedef struct
{
char chChunkID[4];
int nChunkSize;
}XCHUNKHEADER;
typedef struct
{
short nFormatTag;
short nChannels;
int nSamplesPerSec;
int nAvgBytesPerSec;
short nBlockAlign;
short nBitsPerSample;
}WAVEFORMAT;
typedef struct
{
short nFormatTag;
short nChannels;
int nSamplesPerSec;
int nAvgBytesPerSec;
short nBlockAlign;
short nBitsPerSample;
short nExSize;
}WAVEFORMATX;
typedef struct
{
char chRiffID[4];
int nRiffSize;
char chRiffFormat[4];
}RIFFHEADER;
typedef struct
{
char chFmtID[4];
int nFmtSize;
WAVEFORMAT wf;
}FMTBLOCK;
// WAVE音频采样频率是8khz
// 音频样本单元数 = 8000*0.02 = 160 (由采样频率决定)
// 声道数 1 : 160
// 2 : 160*2 = 320
// bps决定样本(sample)大小
// bps = 8 --> 8位 unsigned char
// 16 --> 16位 unsigned short
//int EncodeWAVEFileToAMRFile(const char* pchWAVEFilename, const char* pchAMRFileName, int nChannels, int nBitsPerSample);
// 将AMR文件解码成WAVE文件
//int DecodeAMRFileToWAVEFile(const char* pchAMRFileName, const char* pchWAVEFilename);
#endif
pcm2amr.c代码如下：
#include "pcm2amr.h"
// 从WAVE文件中跳过WAVE文件头，直接到PCM音频数据
void SkipToPCMAudioData(FILE* fpwave)
{
RIFFHEADER riff;
FMTBLOCK fmt;
XCHUNKHEADER chunk;
WAVEFORMATX wfx;
int bDataBlock = 0;
// 1. 读RIFF头
fread(&riff, 1, sizeof(RIFFHEADER), fpwave);
// 2. 读FMT块 - 如果 fmt.nFmtSize>16 说明需要还有一个附属大小没有读
fread(&chunk, 1, sizeof(XCHUNKHEADER), fpwave);
if ( chunk.nChunkSize>16 )
{
fread(&wfx, 1, sizeof(WAVEFORMATX), fpwave);
}
else
{
memcpy(fmt.chFmtID, chunk.chChunkID, 4);
fmt.nFmtSize = chunk.nChunkSize;
fread(&fmt.wf, 1, sizeof(WAVEFORMAT), fpwave);
}
// 3.转到data块 - 有些还有fact块等。
while(!bDataBlock)
{
fread(&chunk, 1, sizeof(XCHUNKHEADER), fpwave);
if ( !memcmp(chunk.chChunkID, "data", 4) )
{
bDataBlock = 1;
break;
}
// 因为这个不是data块,就跳过块数据
fseek(fpwave, chunk.nChunkSize, SEEK_CUR);
}
}
// 从WAVE文件读一个完整的PCM音频帧
// 返回值: 0-错误 >0: 完整帧大小
int ReadPCMFrame(short speech[], FILE* fpwave, int nChannels, int nBitsPerSample)
{
int nRead = 0;
int x = 0, y=0;
unsigned short ush1=0, ush2=0, ush=0;
// 原始PCM音频帧数据
unsigned char pcmFrame_8b1[PCM_FRAME_SIZE];
unsigned char pcmFrame_8b2[PCM_FRAME_SIZE<<1];
unsigned short pcmFrame_16b1[PCM_FRAME_SIZE];
unsigned short pcmFrame_16b2[PCM_FRAME_SIZE<<1];
if (nBitsPerSample==8 && nChannels==1)
{
nRead = fread(pcmFrame_8b1, (nBitsPerSample/8), PCM_FRAME_SIZE*nChannels, fpwave);
for(x=0; x
speech[x] =(short)((short)pcmFrame_8b1[x] << 7);
}
}
else
if (nBitsPerSample==8 && nChannels==2)
{
nRead = fread(pcmFrame_8b2, (nBitsPerSample/8), PCM_FRAME_SIZE*nChannels, fpwave);
for( x=0, y=0; y
// 1 - 取两个声道之左声道
speech[y] =(short)((short)pcmFrame_8b2[x+0] << 7);
// 2 - 取两个声道之右声道
//speech[y] =(short)((short)pcmFrame_8b2[x+1] << 7);
// 3 - 取两个声道的平均值
//ush1 = (short)pcmFrame_8b2[x+0];
//ush2 = (short)pcmFrame_8b2[x+1];
//ush = (ush1 + ush2) >> 1;
//speech[y] = (short)((short)ush << 7);
}
}
else
if (nBitsPerSample==16 && nChannels==1)
{
nRead = fread(pcmFrame_16b1, (nBitsPerSample/8), PCM_FRAME_SIZE*nChannels, fpwave);
if (nRead == PCM_FRAME_SIZE*nChannels) {
for(x=0; x
speech[x] = (short)pcmFrame_16b1[x+0];
}
}
}
else
if (nBitsPerSample==16 && nChannels==2)
{
nRead = fread(pcmFrame_16b2, (nBitsPerSample/8), PCM_FRAME_SIZE*nChannels, fpwave);
for( x=0, y=0; y
//speech[y] = (short)pcmFrame_16b2[x+0];
speech[y] = (short)((int)((int)pcmFrame_16b2[x+0] + (int)pcmFrame_16b2[x+1])) >> 1;
}
}
if (nRead != PCM_FRAME_SIZE*nChannels) return -1;
printf("ReadPCMFrame done\n");
return nRead;
}
// WAVE音频采样频率是8khz
// 音频样本单元数 = 8000*0.02 = 160 (由采样频率决定)
// 声道数 1 : 160
// 2 : 160*2 = 320
// bps决定样本(sample)大小
// bps = 8 --> 8位 unsigned char
// 16 --> 16位 unsigned short
//int EncodeWAVEFileToAMRFile(const char* pchWAVEFilename, const char* pchAMRFileName, int nChannels, int nBitsPerSample)
int main ()
{
FILE* fpwave;
FILE* fpamr;
/* input speech vector */
short speech[160];
/* counters */
int byte_counter, frames = 0, bytes = 0;
/* pointer to encoder state structure */
int *enstate;
/* requested mode */
enum Mode req_mode = MR122;
int dtx = 0;
/* bitstream filetype */
unsigned char amrFrame[MAX_AMR_FRAME_SIZE];
#if 0
//WAV和PCM数据区别仅仅是wav多了数据头
fpwave = fopen("./input.wav", "rb");
#else
fpwave = fopen("./audio.pcm", "rb");
#endif
if (fpwave == NULL)
{
return 0;
}
// 创建并初始化amr文件
fpamr = fopen("./out.amr", "wb");
if (fpamr == NULL)
{
fclose(fpwave);
return 0;
}
/* write magic number to indicate single channel AMR file storage format */
bytes = fwrite(AMR_MAGIC_NUMBER, sizeof(char), strlen(AMR_MAGIC_NUMBER), fpamr);
#if 0
/* skip to pcm audio data*/
//跳过WAV数据包头既是PCM原始数据
SkipToPCMAudioData(fpwave);
#endif
enstate = Encoder_Interface_init(dtx);
while(1)
{
// read one pcm frame
//if (!ReadPCMFrame(speech, fpwave, nChannels, nBitsPerSample)) break;
if (-1 == ReadPCMFrame(speech, fpwave, 1, 16)) {
//printf("ReadPCMFrame failed\n");
break;
}
frames++;
/* call encoder */
byte_counter = Encoder_Interface_Encode(enstate, req_mode, speech, amrFrame, 0);
printf("byte_counter == %d\n",byte_counter);
bytes += byte_counter;
fwrite(amrFrame, sizeof (unsigned char), byte_counter, fpamr );
}
Encoder_Interface_exit(enstate);
fclose(fpamr);
fclose(fpwave);
return frames;
}
//----------------------------------------------------------------------------------//
