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)
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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: [email protected]
*/
#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;
}
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