FFmpeg浅尝辄止（四）——音频的解码和编码

音频和视频其实是一样的，在文件中寻找音频流，然后解压出来，得到音频帧的数据，同样也可以按照设定的编码格式进行压缩，我这里把音频的解码和编码做成了两个工程，也是直接上代码：

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#include <stdio.h>

#include <stdlib.h>



extern "C"

{

#include <libavcodec\avcodec.h>

#include <libavformat\avformat.h>

#include <libswscale\swscale.h>

}



int main(char arg,char *argv[])

{

char *filename = argv[1];



av_register_all(); //注册所有可解码类型

***FormatContext *pInFmtCtx=NULL; //文件格式

***CodecContext *pInCodecCtx=NULL; //编码格式

if (av_open_input_file(&pInFmtCtx, filename, NULL, 0, NULL)!=0) //获取文件格式

printf("av_open_input_file error\n");

if (av_find_stream_info(pInFmtCtx) < 0) //获取文件内音视频流的信息

printf("av_find_stream_info error\n");



unsigned int j;

// Find the first audio stream



int audioStream = -1;

for (j=0; j<pInFmtCtx->nb_streams; j++) //找到音频对应的stream

{

if (pInFmtCtx->streams[j]->codec->codec_type == CODEC_TYPE_AUDIO)

{

audioStream = j;

break;

}

}

if (audioStream == -1)

{

printf("input file has no audio stream\n");

return 0; // Didn't find a audio stream

}

printf("audio stream num: %d\n",audioStream);

pInCodecCtx = pInFmtCtx->streams[audioStream]->codec; //音频的编码上下文

***Codec *pInCodec = NULL;



pInCodec = avcodec_find_decoder(pInCodecCtx->codec_id); //根据编码ID找到用于解码的结构体

if (pInCodec == NULL)

{

printf("error no Codec found\n");

return -1 ; // Codec not found

}



if(avcodec_open(pInCodecCtx, pInCodec)<0)//将两者结合以便在下面的解码函数中调用pInCodec中的对应解码函数

{

printf("error avcodec_open failed.\n");

return -1; // Could not open codec



}



static ***Packet packet;



printf(" bit_rate = %d \r\n", pInCodecCtx->bit_rate);

printf(" sample_rate = %d \r\n", pInCodecCtx->sample_rate);

printf(" channels = %d \r\n", pInCodecCtx->channels);

printf(" code_name = %s \r\n", pInCodecCtx->codec->name);

printf(" block_align = %d\n",pInCodecCtx->block_align);



uint8_t *pktdata;

int pktsize;

int out_size = ***CODEC_MAX_AUDIO_FRAME_SIZE*100;

uint8_t * inbuf = (uint8_t *)malloc(out_size);

FILE* pcm;

pcm = fopen("result.pcm","wb");

long start = clock();

while (av_read_frame(pInFmtCtx, &packet) >= 0)//pInFmtCtx中调用对应格式的packet获取函数

{

if(packet.stream_index==audioStream)//如果是音频

{

pktdata = packet.data;

pktsize = packet.size;

while(pktsize>0)

{

out_size = ***CODEC_MAX_AUDIO_FRAME_SIZE*100;

//解码

int len = avcodec_decode_audio2(pInCodecCtx, (short*)inbuf, &out_size, pktdata, pktsize);

if (len < 0)

{

printf("Error while decoding.\n");

break;

}

if(out_size > 0)

{

fwrite(inbuf,1,out_size,pcm);//pcm记录

fflush(pcm);

}

pktsize -= len;

pktdata += len;

}

}

av_free_packet(&packet);

}

long end = clock();

printf("cost time :%f\n",double(end-start)/(double)CLOCKS_PER_SEC);

free(inbuf);

fclose(pcm);

if (pInCodecCtx!=NULL)

{

avcodec_close(pInCodecCtx);

}

av_close_input_file(pInFmtCtx);



return 0;

}

解码后的文件保存为result.pcm中，现在用这个文件压缩出新的音频文件，代码如下：

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void main()

{

int16_t *samples;

uint8_t *audio_outbuf;

int audio_outbuf_size;

int audio_input_frame_size;

double audio_pts;



const char* filename = "test.wav";

FILE *fin = fopen("result.pcm", "rb"); //音频源文件

***OutputFormat *fmt;

***FormatContext *oc;

***Stream * audio_st;

av_register_all();

fmt = guess_format(NULL, filename, NULL);

oc = av_alloc_format_context();

oc->oformat = fmt;

snprintf(oc->filename, sizeof(oc->filename), "%s", filename);

audio_st = NULL;



if (fmt->audio_codec != CODEC_ID_NONE)

{

***CodecContext *c;

audio_st = av_new_stream(oc, 1);

c = audio_st->codec;

c->codec_id = fmt->audio_codec;

c->codec_type = CODEC_TYPE_AUDIO;

c->bit_rate = 128000;

c->sample_rate = 44100;

c->channels = 2;

}

if (av_set_parameters(oc, NULL) < 0)

{

return;

}

dump_format(oc, 0, filename, 1);

if (audio_st)

{

***CodecContext* c;

***Codec* codec;

c = audio_st->codec;

codec = avcodec_find_encoder(c->codec_id);

avcodec_open(c, codec);

audio_outbuf_size = 10000;

audio_outbuf = (uint8_t*)av_malloc(audio_outbuf_size);

if (c->frame_size <= 1)

{

audio_input_frame_size = audio_outbuf_size / c->channels;

switch (audio_st->codec->codec_id)

{

case CODEC_ID_PCM_S16LE:

case CODEC_ID_PCM_S16BE:

case CODEC_ID_PCM_U16LE:

case CODEC_ID_PCM_U16BE:

audio_input_frame_size >>= 1;

break;

default:

break;

}

}

else

{

audio_input_frame_size = c->frame_size;

}

samples = (int16_t*)av_malloc(audio_input_frame_size*2*c->channels);

}

if (!fmt->flags & ***FMT_NOFILE)

{

if (url_fopen(&oc->pb, filename, URL_WRONLY) < 0)

{

return;

}

}

av_write_header(oc);

for (;;)

{

if (audio_st)

{

audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;

}

else

{

audio_pts = 0.0;

}

if (!audio_st || audio_pts >= 360.0)

{

break;

}

if (fread(samples, 1, audio_input_frame_size*2*audio_st->codec->channels, fin) <= 0)

{

break;

}

***CodecContext* c;

***Packet pkt;

av_init_packet(&pkt);

c = audio_st->codec;

pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples);

pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, audio_st->time_base);

pkt.flags |= PKT_FLAG_KEY;

pkt.stream_index = audio_st->index;

pkt.data = audio_outbuf;

if (av_write_frame(oc, &pkt) != 0)

{

return;

}

}

if (audio_st)

{

avcodec_close(audio_st->codec);

av_free(samples);

av_free(audio_outbuf);

}

av_write_trailer(oc);

for (int i=0; i<oc->nb_streams; i++)

{

av_freep(&oc->streams[i]->codec);

av_freep(&oc->streams[i]);

}

if (!(fmt->flags & ***FMT_NOFILE))

{

url_fclose(oc->pb);

}

av_free(oc);

fclose(fin);

}

对应的下载链接：

音频解码：http://download.csdn.net/detail/yang_xian521/4399219

音频编码：http://download.csdn.net/detail/yang_xian521/4399293

至此，我已经实现了视频的解码编码，音频的解码编码，相信有心人肯定可以在此基础上实现视频音频的同步压缩，不过要再看一些网上的资料，也是很轻松的。至于视频的显示播放，大多数是要结合SDL实现，由于我只是浅尝辄止，只实现了图片的显示和简单的视频播放，比起我之前推荐的几个链接是弱爆了的，就不写下去了，这个系列就到这里吧，希望能给像我一样的一些入门级选手点帮助。也欢迎大神来多多指教。