ffmpeg中G726解码器用法 编码器用法

//解码：

本代码实现将cap1.g726文件中的g726编码帧数据进行解码，然后保存到cap1文件中

1、ffmpeg版本 ：
     0.10.2

2、源码：

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);
}