C语言:wav to amr 后播放出来声音的变的很尖，可能是代码出现问题，求大神帮看看，急，在线等！！！

欧阳_7777 2014-05-20 08:56:41

#include <stdio.h>

#include "interf_enc.h"
#include <memory.h >
#include <string.h >
//#pragma comment(lib,"AMRNB.lib")
#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);

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 )
{
int a = sizeof(WAVEFORMATX);//结构体是有对齐的啊
fread(&wfx, 1, chunk.nChunkSize, 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<PCM_FRAME_SIZE; 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<PCM_FRAME_SIZE; y++,x+=2 )
{
// 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);
for(x=0; x<PCM_FRAME_SIZE; 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<PCM_FRAME_SIZE; y++,x+=2 )
{
speech[y] = (short)pcmFrame_16b2[x+0];
//speech[y] = (short)((int)((int)pcmFrame_16b2[x+0] + (int)pcmFrame_16b2[x+1])) >> 1;
}
}

// 如果读到的数据不是一个完整的PCM帧, 就返回0
if (nRead<PCM_FRAME_SIZE*nChannels) return 0;

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)
{
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];
fpwave = fopen(pchWAVEFilename, "rb");
if (fpwave == NULL)
{
return 0;
}
// 创建并初始化amr文件
fpamr = fopen(pchAMRFileName, "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);
/* skip to pcm audio data*/
SkipToPCMAudioData(fpwave);
enstate = (int *)Encoder_Interface_init(dtx);
while(1)
{
// read one pcm frame
if (!ReadPCMFrame(speech, fpwave, nChannels, nBitsPerSample)) break;
frames++; /* call encoder */
byte_counter = Encoder_Interface_Encode(enstate, req_mode, speech, amrFrame, 0);
bytes += byte_counter;
fwrite(amrFrame, sizeof (unsigned char), byte_counter, fpamr );
}
Encoder_Interface_exit(enstate);
fclose(fpamr);
fclose(fpwave);
return frames;
}

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