ffmpeg 实现音频aac编码

1、编译ffmepg

./configure --disable-yasm --enable-nonfree --enable-libfaac --prefix=/home/ffmpeg/1_ffmpeg-2.1.1/install

2、编译audio_enc.c

makefile：

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#!/bin/sh INCLUDE = .. /include LIB_DIR = .. /lib LDFLAGS =  -lfaac -lavcodec -lavformat -lavdevice -lavfilter -lavutil -lswresample -pthread  -ldl -lswscale -lasound -lz -lm -lbz2   SRC=audio_enc.c all:$(SRC)      gcc -g -Wall $(SRC) -o target -I $(INCLUDE) -L $(LIB_DIR) $(LDFLAGS)

注意：在这里编译的时候需要加上aac库，可能会找不到库函数undefined reference to `faacEncEncode'等

程序运行时，需要提供一个和程序参数一致的wav音频文件：

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/* * Copyright(C), 2013, Ubuntu Inc. * File name:        audio_enc.c * Author:           xubinbin 徐彬彬 (Beijing China) * Version:          1.0 * Date:             2013.12.23 * Description:      Use ffmpeg achieve aac audio coding. * Function List:    * Email:            xubbwd@gmail.com */   #include <stdlib.h> #include <stdio.h> #include <string.h> #include <math.h>   #include <libavutil/opt.h> #include <libavutil/mathematics.h> #include <libavformat/avformat.h>   FILE  * fp_in = NULL; FILE  * fp_out = NULL;   static  int  frame_count;   int  main( int  argc,  char  **argv) {      int  ret;      AVCodec *audio_codec;      AVCodecContext *c;      AVFrame *frame;      AVPacket pkt;      int  got_output;        /* Initialize libavcodec, and register all codecs and formats. */      av_register_all();      avcodec_register_all();      //avdevice_register_all();        audio_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);      c = avcodec_alloc_context3(audio_codec);        c->codec_id = AV_CODEC_ID_AAC;      c->sample_fmt = AV_SAMPLE_FMT_S16;      c->sample_rate = 44100;      c->channels = 2;      c->channel_layout = AV_CH_LAYOUT_STEREO;      c->bit_rate = 64000;        /* open the codec */      ret = avcodec_open2(c, audio_codec, NULL);      if  (ret < 0) {          fprintf (stderr,  "Could not open video codec: %s\n" , av_err2str(ret));          exit (1);      }        /* allocate and init a re-usable frame */      frame = avcodec_alloc_frame();      if  (!frame) {          fprintf (stderr,  "Could not allocate video frame\n" );          exit (1);      }          frame->nb_samples = c->frame_size;      frame->format = c->sample_fmt;      frame->channels = c->channels;      frame->channel_layout = c->channel_layout;      frame->linesize[0] = 4096;      frame->extended_data = frame->data[0] = av_malloc(( size_t )frame->linesize[0]);        av_init_packet(&pkt);        fp_in =  fopen ( "in.wav" , "rb" );      fp_out=  fopen ( "out.aac" , "wb" );        //printf("frame->nb_samples = %d\n",frame->nb_samples);            while (1)      {          frame_count++;          bzero(frame->data[0],frame->linesize[0]);          ret =  fread (frame->data[0],frame->linesize[0],1,fp_in);          if (ret <= 0)          {              printf ( "read over !\n" );              break ;          }          ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);          if  (ret < 0) {              fprintf (stderr,  "Error encoding audio frame: %s\n" , av_err2str(ret));              exit (1);          }                if (got_output > 0)          {              //printf("pkt.size = %d\n",pkt.size);              fwrite (pkt.data,pkt.size,1,fp_out);              av_free_packet(&pkt);          }            #if 0          if (frame_count > 10)          {              printf ( "break @@@@@@@@@@@@\n" );              break ;          }          #endif      }        avcodec_close(c);      av_free(c);      avcodec_free_frame(&frame);        fclose (fp_in);      fclose (fp_out);        return  0; }

转载自：http://hi.baidu.com/285988185/item/6864b8b0c9640445ba0e12a6