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);
}