ffmpeg--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：

#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
AVFormatContext *pInFmtCtx=NULL;	//文件格式
AVCodecContext *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
AVCodec *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 AVPacket 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 = AVCODEC_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 = AVCODEC_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;
}

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：

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
AVOutputFormat *fmt;
AVFormatContext *oc;
AVStream * 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)
{
AVCodecContext *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)
{
AVCodecContext* c;
AVCodec* 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 & AVFMT_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;
}
AVCodecContext* c;
AVPacket 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 & AVFMT_NOFILE))
{
url_fclose(oc->pb);
}
av_free(oc);
fclose(fin);
}