ffmpeg解码海思G726音频文件

最近需要解析海思的G726格式的音频文件，主要是AVCodecContext结构体中输入参数的设置，如果设置错误的话就不能正确的播放出声音。现写个范例如下:

#define Hisi_AUDIO_HERDER_LEN 4  //hisi 音频数据头
extern “C”
{
#include “libavcodec/avcodec.h”
#include “libavformat/avformat.h”
#include “libavutil/frame.h”
#include “libswscale/swscale.h”
#include “libavutil/imgutils.h”
}
//链接 ffmpeg lib库

AVCodec *codec;
AVCodecContext *c= NULL;
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
avcodec_register_all();
av_init_packet(&avpkt);
/* find the ffmpeg audio decoder */
/*
ffmpeg g726编码器：AV_CODEC_ID_ADPCM_G726
ffmpeg g726解码器包括：AV_CODEC_ID_ADPCM_G726、AV_CODEC_ID_ADPCM_G726LE
如果海思g726码流类型为asf时，请选择AV_CODEC_ID_ADPCM_G726解码器类型
如果海思g726码流类型为RFC3551标准时，请选择AV_CODEC_ID_ADPCM_G726LE解码器类型
*/
codec = avcodec_find_decoder(AV_CODEC_ID_ADPCM_G726);
if (!codec)
{
fprintf(stderr, “codec not found\n”);
return;
}
c = avcodec_alloc_context3(codec);
//采样率= 8000 每个采样用的bit数= 16 通道数= 1
/*
bits_per_coded_sample:表示编码压缩bit值与采样率的bit值之比。
如果为g726音频时，表示g726码流压缩与采样率比值。比如kbps码流压缩比为：40k/8k = 5，32kbps码流压缩比为32k/8k = 4, 16kbps码流压缩比为16k/8k = 2。
*/
c->bits_per_coded_sample = 4; //5;
c->channels = 1;
c->sample_fmt = AV_SAMPLE_FMT_S16;
c->sample_rate = 8000;
c->codec_type = AVMEDIA_TYPE_AUDIO;
//c->bit_rate = 16000;
int iRet = avcodec_open2(c, codec,NULL);
if ( iRet < 0 )
{
fprintf(stderr, “could not open codec\n”);
return;
}

CString filePath = “”;
CString newlFilePath = “”;
char szFilter[] = {“g726 Files (*.g726_hisi)|*.g726_hisi||”};
CFileDialog dlg(TRUE,NULL,NULL,OFN_HIDEREADONLY | OFN_OVERWRITEPROMPT,szFilter,NULL);
if(dlg.DoModal() == IDOK)
{
filePath = dlg.GetPathName();
newlFilePath = filePath;
newlFilePath.Replace(“.g726_hisi”,”.hisi2ff.pcm”);
BOOL bRet = 0;
FILE * fpSrc = fopen(filePath.GetBuffer(filePath.GetLength()),”rb”);
FILE * fpDst = fopen(newlFilePath.GetBuffer(newlFilePath.GetLength()),”wb+”);
char szData[200] = {0};
char szOutData[500] = {0};
int nDataLen = 200;
int nOutDataLen = 500;
int nReadedSize = 0;
unsigned short usHisiHeader[2] = {0};
if(fpSrc != NULL)
{
while(TRUE)
{
//读取头标记
nDataLen = Hisi_AUDIO_HERDER_LEN;
nReadedSize = fread(szData,sizeof(char),nDataLen,fpSrc);
if(nReadedSize < nDataLen)
{
break;
}
memcpy(usHisiHeader,szData,Hisi_AUDIO_HERDER_LEN);
int nAudioFrameDataLen = (usHisiHeader[1] & 0x00ff) * sizeof(unsigned short);
nDataLen = nAudioFrameDataLen;
//读取音频帧数据
nReadedSize = fread(szData,sizeof(char),nDataLen,fpSrc);
if(nReadedSize < nDataLen)
{
break;
}
avpkt.data = (uint8_t *)szData;
avpkt.size = nReadedSize;
int got_frame = 0;
if (!decoded_frame)
{
if (!(decoded_frame = avcodec_alloc_frame()))
{
return;
}
}
else
{
avcodec_get_frame_defaults(decoded_frame);
}
int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0)
{
return;
}
if (got_frame)
{
/* if a frame has been decoded, output it */
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);

fwrite(decoded_frame->data[0], 1, data_size, fpDst);

}
}

fclose(fpSrc);
fclose(fpDst);
avcodec_close(c);
av_free(c);
av_free(decoded_frame);
}

}

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