ffmpeg中G726解码器用法 编码器用法

//解码：

extern "C"
{
#include "./h/libavcodec/avcodec.h"
};#pragma comment(lib, "avutil.lib")  //ffmpeg
#pragma comment(lib, "avformat.lib")
#pragma comment(lib, "avcodec.lib")
AVCodec *codec;
AVCodecContext *c= NULL;
AVPacket avpkt;
AVFrame *decoded_frame = NULL;

//编码后一帧大小为 84字节
#define framesize 84
int main(int argc, char* argv[])
{
printf("Hello World!\n");
unsigned char inbuffer[framesize] = {0};

/* register all the codecs */
avcodec_register_all();
av_init_packet(&avpkt);

/* find the mpeg audio decoder */
codec = avcodec_find_decoder(CODEC_ID_ADPCM_G726);
if (!codec)
{
fprintf(stderr, "codec not found\n");
return 0;
}

c = avcodec_alloc_context3(codec);
//没有如下两句 avcodec_open会返回 -22 错误
//采样率 = 8000 每个采样用的bit数 = 16 通道数 = 1
c->bits_per_coded_sample = 2;   //g726压缩比为 8:1 编码前采样用bit数为16 那么编码后应该占16/8 = 2 这是我的理解
c->channels = 1;

/* open it */
int iRet = avcodec_open(c, codec);
if ( iRet < 0 )
{
fprintf(stderr, "could not open codec\n");
return 0;
}

FILE *f, *outfile;    //打开存放g726音频帧的文件
f = fopen( "cap1.g726", "rb" );
if (!f)
{
return 0;
}    //打开要存放解码后音频帧的文件
outfile = fopen("cap1", "wb");
if (!outfile)
{
av_free(c);
return 0;
}

avpkt.data = inbuffer;

while ( (avpkt.size = fread(inbuffer, 1, framesize, f) ) > 0 )
{
int got_frame = 0;
if (!decoded_frame)
{
if (!(decoded_frame = avcodec_alloc_frame()))
{
return 0;
}
}
else
{
avcodec_get_frame_defaults(decoded_frame);
}

int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0)
{
return 0;
}

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, outfile);
}
}

fclose(outfile);
fclose(f);

avcodec_close(c);
av_free(c);
av_free(decoded_frame);

return 0;
}

//编码  
ffmpeg g726 编码  
2012-10-11 10:19:21|  分类： ffmpeg|举报|字号 订阅一、将文件source中原始音频数据进行编码，并保存到dest中
采样率 = 8000
通道数 = 1
采样位数 = 16
波特率 = 16000
二、源码
extern "C"
{
#include "./h/libavcodec/avcodec.h"
};

//视频相关库
#pragma comment(lib, "avutil.lib")  //ffmpeg
#pragma comment(lib, "avformat.lib")
#pragma comment(lib, "avcodec.lib")

AVCodec *codec;
AVCodecContext *c= NULL;
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
#define framesize 80
#define buffersize 640

int main(int argc, char* argv[])
{
/* register all the codecs */
avcodec_register_all();
av_init_packet(&avpkt);    //如下两句在编码中是必须的，要不avcodec_encode_audio2会失败
avpkt.data = NULL;    avpkt.size = 0;

/* find the mpeg audio decoder */
codec = avcodec_find_encoder(CODEC_ID_ADPCM_G726);
if (!codec)
{
fprintf(stderr, "codec not found\n");
return 0;
}

c = avcodec_alloc_context3(codec);
//put sample parameters
c->bits_per_coded_sample = 2;
c->bit_rate = 16000;    //
c->sample_rate = 8000;    //采样率
c->channels = 1;     //通道数
c->sample_fmt = AV_SAMPLE_FMT_S16;                 //应该是采样位数

/* open it */
int iRet = avcodec_open(c, codec);
if ( iRet < 0 )
{
fprintf(stderr, "could not open codec\n");
return 0;
}
int buf_size = buffersize;
char szBuffer[buffersize] = {0};
int ret = 0;
int got_packet = 0;

FILE *f;
FILE *output;    //打开存放编码前音频文件
f = fopen("source", "rb");    //打开存放编码后音频文件
output = fopen( "dest", "wb" );
while( fread( szBuffer, 1, buffersize, f ) )
{

if (!decoded_frame)
{
if (!(decoded_frame = avcodec_alloc_frame()))
{
return 0;
}
}
else
{
avcodec_get_frame_defaults(decoded_frame);
}

decoded_frame->nb_samples = buf_size/(c->channels*av_get_bytes_per_sample(c->sample_fmt));

ret = avcodec_fill_audio_frame(decoded_frame, c->channels, c->sample_fmt, (unsigned char *)szBuffer,
buf_size, 1);
if (ret < 0)
{
//av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
return 0;
}

if (avcodec_encode_audio2(c, &avpkt, decoded_frame, &got_packet) < 0)
{
return 0;
}

if (got_packet)
{
int iiiii = fwrite( avpkt.data, 1, avpkt.size, output );
}
}

fclose(f);
fclose(output);}