新版ffmpeg编码AAC注意事项：

新版大于3.0后的版本，按雷霄骅大神的 最简单的基于FFMPEG的音频编码器(PCM编码为AAC)的例子已经无法走完流程，avcodec_open2会返回-22，其中原因是

1、新版版的ffmpeg 编码AAC只支持的AV_SAMPLE_FMT_FLTP，老版本的是AV_SAMPLE_FMT_S16，参数开之后avcodec_open2返回正确，但不要高兴太早；

2、很可能你传入的PCM数据是AV_SAMPLE_FMT_S16的，avcodec_encode_audio2还是返回-22错误，争取做法是需要AV_SAMPLE_FMT_S16的数据转为AV_SAMPLE_FMT_FLTP就可以了。下面贴一个ffmpeg4.0能编译通过执行编码的代码

/**
 *最简单的基于FFmpeg的音频编码器
 *Simplest FFmpeg Audio Encoder
 *
 *雷霄骅 Lei Xiaohua
 *leixiaohua1020@126.com
 *中国传媒大学/数字电视技术
 *Communication University of China / Digital TV Technology
 *http://blog.csdn.net/leixiaohua1020
 *
 *本程序实现了音频PCM采样数据编码为压缩码流（MP3，WMA，AAC等）。
 *是最简单的FFmpeg音频编码方面的教程。
 *通过学习本例子可以了解FFmpeg的编码流程。
 *This software encode PCM data to AAC bitstream.
 *It's the simplest audio encoding software based on FFmpeg. 
 *Suitable for beginner of FFmpeg 
 */

#include <stdio.h>

#define __STDC_CONSTANT_MACROS

#ifdef _WIN32
//Windows
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswresample/swresample.h"
#include "libavutil/opt.h"
};
#else
//Linux...
#ifdef __cplusplus
extern "C"
{
#endif
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
#ifdef __cplusplus
};
#endif
#endif


int flush_encoder(AVFormatContext *fmt_ctx,unsigned int stream_index){
	int ret;
	int got_frame;
	AVPacket enc_pkt;
	if (!(fmt_ctx->streams[stream_index]->codec->codec->capabilities &
		AV_CODEC_CAP_DELAY))
		return 0;
	while (1) {
		enc_pkt.data = NULL;
		enc_pkt.size = 0;
		av_init_packet(&enc_pkt);
		ret = avcodec_encode_audio2 (fmt_ctx->streams[stream_index]->codec, &enc_pkt,
			NULL, &got_frame);
		av_frame_free(NULL);
		if (ret < 0)
			break;
		if (!got_frame){
			ret=0;
			break;
		}
		printf("Flush Encoder: Succeed to encode 1 frame!\tsize:%5d\n",enc_pkt.size);
		/* mux encoded frame */
		ret = av_write_frame(fmt_ctx, &enc_pkt);
		if (ret < 0)
			break;
	}
	return ret;
}

int main(int argc, char* argv[])
{
	AVFormatContext* pFormatCtx;
	AVOutputFormat* fmt;
	AVStream* audio_st;
	AVCodecContext* pCodecCtx;
	AVCodec* pCodec;

	uint8_t* frame_buf;
	AVFrame* pFrame;
	AVPacket pkt;
	SwrContext *swr;


	int got_frame=0;
	int ret=0;
	int size=0;

	FILE *in_file=NULL;	                        //Raw PCM data
	int framenum=1000;                          //Audio frame number
	const char* out_file = "tdjm.aac";          //Output URL
	int i;

	in_file= fopen("tdjm.pcm", "rb");

	av_register_all();

	//Method 1.
	pFormatCtx = avformat_alloc_context();
	fmt = av_guess_format(NULL, out_file, NULL);
	pFormatCtx->oformat = fmt;


	//Method 2.
	//avformat_alloc_output_context2(&pFormatCtx, NULL, NULL, out_file);
	//fmt = pFormatCtx->oformat;

	//Open output URL
	if (avio_open(&pFormatCtx->pb,out_file, AVIO_FLAG_READ_WRITE) < 0){
		printf("Failed to open output file!\n");
		return -1;
	}

	audio_st = avformat_new_stream(pFormatCtx, 0);
	if (audio_st==NULL){
		return -1;
	}
	pCodec = avcodec_find_encoder(fmt->audio_codec);
	if (!pCodec){
		printf("Can not find encoder!\n");
		return -1;
	}

	pCodecCtx = audio_st->codec;
	pCodecCtx->codec_id = fmt->audio_codec;
	pCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
	pCodecCtx->sample_fmt = pCodec->sample_fmts[0];
	pCodecCtx->sample_rate= 44100;
	pCodecCtx->channel_layout=AV_CH_LAYOUT_STEREO;
	pCodecCtx->channels = av_get_channel_layout_nb_channels(pCodecCtx->channel_layout);
	pCodecCtx->bit_rate = 64000;  
	pCodecCtx->profile=FF_PROFILE_AAC_MAIN ;
	pCodecCtx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; 
	//Show some information
	av_dump_format(pFormatCtx, 0, out_file, 1);



	swr = swr_alloc();
	av_opt_set_int(swr, "in_channel_layout",  AV_CH_LAYOUT_STEREO, 0);
	av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO,  0);
	av_opt_set_int(swr, "in_sample_rate",     44100, 0);
	av_opt_set_int(swr, "out_sample_rate",    44100, 0);
	av_opt_set_sample_fmt(swr, "in_sample_fmt",  AV_SAMPLE_FMT_S16, 0);
	av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLTP,  0);
	swr_init(swr);
	uint8_t *outs[2];
	int len = 4096;

	outs[0]=(uint8_t *)malloc(len);//len 为4096
	outs[1]=(uint8_t *)malloc(len);


	if ((ret = avcodec_open2(pCodecCtx, pCodec,NULL)) < 0){
		printf("Failed to open encoder!\n");
		return -1;
	}
	pFrame = av_frame_alloc();
	pFrame->nb_samples= pCodecCtx->frame_size;
	pFrame->format=  pCodec->sample_fmts[0];

	size = av_samples_get_buffer_size(NULL, pCodecCtx->channels,pCodecCtx->frame_size,pCodecCtx->sample_fmt, 1);
	frame_buf = (uint8_t *)av_malloc(size);
	avcodec_fill_audio_frame(pFrame, pCodecCtx->channels, pCodecCtx->sample_fmt,(const uint8_t*)frame_buf, size, 1);

	//Write Header
	avformat_write_header(pFormatCtx,NULL);

	av_new_packet(&pkt,size);

	for (i=0; i<framenum; i++){
		//Read PCM
		if (fread(frame_buf, 1, size, in_file) <= 0){
			printf("Failed to read raw data! \n");
			return -1;
		}else if(feof(in_file)){
			break;
		}
		int count=swr_convert(swr, outs,len*4,(const uint8_t **)&frame_buf,len/4);//len 为4096
		pFrame->data[0] =(uint8_t*)outs[0];//audioFrame 是VFrame
		pFrame->data[1] =(uint8_t*)outs[1];


		//pFrame->data[0] = frame_buf;  //PCM Data
		pFrame->pts=i*100;
		got_frame=0;
		//Encode
		ret = avcodec_encode_audio2(pCodecCtx, &pkt,pFrame, &got_frame);
		if(ret < 0){
			printf("Failed to encode!\n");
			return -1;
		}
		if (got_frame==1){
			printf("Succeed to encode 1 frame! \tsize:%5d\n",pkt.size);
			pkt.stream_index = audio_st->index;
			ret = av_write_frame(pFormatCtx, &pkt);
			av_free_packet(&pkt);
		}
	}

	//Flush Encoder
	ret = flush_encoder(pFormatCtx,0);
	if (ret < 0) {
		printf("Flushing encoder failed\n");
		return -1;
	}

	//Write Trailer
	av_write_trailer(pFormatCtx);

	//Clean
	if (audio_st){
		avcodec_close(audio_st->codec);
		av_free(pFrame);
		av_free(frame_buf);
	}
	avio_close(pFormatCtx->pb);
	avformat_free_context(pFormatCtx);

	fclose(in_file);

	return 0;
}

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