新版ffmpeg PCM编码到AAC，swr_convert转换采样精度，稍微修改兼容PCM编码为G711A及MP3，记录下。

#include "stdafx.h"
#include 
#include 
using namespace std;
extern "C"
{
#include "libavformat/avformat.h"  
#include "libavutil/avutil.h"  
#include "libavcodec/avcodec.h"  
#include "libswresample/swresample.h"  
#include "libavutil/frame.h"  
#include "libavutil/samplefmt.h"  
#include "libavformat/avformat.h"  
#include "libavcodec/avcodec.h"  
}

#pragma comment(lib, "avcodec.lib")  
#pragma comment(lib, "avdevice.lib")  
#pragma comment(lib, "avformat.lib")  
#pragma comment(lib, "avutil.lib")  
#pragma comment(lib, "swscale.lib")  
#pragma comment(lib, "swresample.lib")  


/* PCM转AAC */
int main()
{
	/* ADTS头 */
	char *padts = (char *)malloc(sizeof(char) * 7);
	char *mp3padts = (char *)malloc(sizeof(char) * 4);
	int profile = 2;    //AAC LC  
	int freqIdx = 4;  //44.1KHz  
	int chanCfg = 2;  //MPEG-4 Audio Channel Configuration. 1 Channel front-center,channel_layout.h  
	padts[0] = (char)0xFF; // 11111111     = syncword  
	padts[1] = (char)0xF1; // 1111 1 00 1  = syncword MPEG-2 Layer CRC  
	padts[2] = (char)(((profile - 1) << 6) + (freqIdx << 2) + (chanCfg >> 2));
	padts[6] = (char)0xFC;

	/*mp3padts[0] = (char)0xFF;
	mp3padts[1] = (char)0xFB;
	mp3padts[2] = (char)0x90;
	mp3padts[3] = (char)0x04;*/
	SwrContext *swr_ctx = NULL;
	AVCodecContext *pCodecCtx = NULL;
	AVCodec *pCodec = NULL;
	AVFrame *pFrame;
	AVPacket pkt;
	AVCodecID codec_id = AV_CODEC_ID_AAC;

	FILE *fp_in;
	FILE *fp_out;
	char filename_in[] = "tdjm.pcm";
	char filename_out[] = "audio.G711";

	uint8_t **convert_data;         //存储转换后的数据，再编码AAC  
	int i, ret, got_output;
	uint8_t* frame_buf;
	int size = 0;
	int y_size;
	int framecnt = 0;
	int framenum = 100000;


	avcodec_register_all();

	pCodec = avcodec_find_encoder(codec_id);
	if (!pCodec) {
		printf("Codec not found\n");
		return -1;
	}

	pCodecCtx = avcodec_alloc_context3(pCodec);
	if (!pCodecCtx) {
		printf("Could not allocate video codec context\n");
		return -1;
	}

	pCodecCtx->codec_id = codec_id;
	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->frame_size = 1024;


	if ((ret = avcodec_open2(pCodecCtx, pCodec, NULL)) < 0) {
		cout << "avcodec_open2 error ----> " << ret;

		printf("Could not open codec\n");
		return -1;
	}
	pCodecCtx->frame_size/* = 1024*/;
	pCodecCtx->bit_rate =128000 ;
	pFrame = av_frame_alloc();
	pFrame->nb_samples = pCodecCtx->frame_size;   //1024  
	pFrame->format = pCodecCtx->sample_fmt;
	pFrame->channels = 2;

	/* 由AV_SAMPLE_FMT_S16转为AV_SAMPLE_FMT_FLTP */
	swr_ctx = swr_alloc_set_opts(
		NULL,
		av_get_default_channel_layout(pCodecCtx->channels),
		pCodecCtx->sample_fmt,                   
		pCodecCtx->sample_rate,
		av_get_default_channel_layout(pCodecCtx->channels),
		AV_SAMPLE_FMT_S16,                      //PCM源文件的采样格式  
		/*pCodecCtx->sample_rate*/44100,
		0, NULL);

	swr_init(swr_ctx);
	/* 分配空间 */
	convert_data = (uint8_t**)calloc(pCodecCtx->channels,
		sizeof(*convert_data));
	av_samples_alloc(convert_data, NULL,
		pCodecCtx->channels, pCodecCtx->frame_size,
		pCodecCtx->sample_fmt, 0);

	size = av_samples_get_buffer_size(NULL, pCodecCtx->channels, pCodecCtx->frame_size, pCodecCtx->sample_fmt, 0);

	frame_buf = (uint8_t *)av_malloc(size);
	/* 此时data[0],data[1]分别指向frame_buf数组起始、中间地址 */
	ret = avcodec_fill_audio_frame(pFrame, pCodecCtx->channels, pCodecCtx->sample_fmt, (const uint8_t*)frame_buf, size, 0);

	if (ret < 0)
	{
		cout << "avcodec_fill_audio_frame error ";
		return 0;
	}

	//Input raw data  
	fp_in = fopen(filename_in, "rb");
	if (!fp_in) {
		printf("Could not open %s\n", filename_in);
		return -1;
	}

	//Output bitstream  
	fp_out = fopen(filename_out, "wb");
	if (!fp_out) {
		printf("Could not open %s\n", filename_out);
		return -1;
	}

	//Encode  
	for (i = 0; i < framenum; i++) {
		av_init_packet(&pkt);
		pkt.data = NULL;    
		pkt.size = 0;
		//Read raw data  
		if (fread(frame_buf, 1, 4096, fp_in) <= 0) {
			printf("Failed to read raw data! \n");
			return -1;
		}
		else if (feof(fp_in)) {
			break;
		}
		/* 转换数据，令各自声道的音频数据存储在不同的数组（分别由不同指针指向）*/
		pCodecCtx->sample_fmt;
		swr_convert(swr_ctx, convert_data, pCodecCtx->frame_size,
			(const uint8_t**)pFrame->data, pCodecCtx->frame_size);

		/* 将转换后的数据复制给pFrame */
		int length = pCodecCtx->frame_size * av_get_bytes_per_sample(pCodecCtx->sample_fmt);
	
		memcpy(pFrame->data[0], convert_data[0], length);
		memcpy(pFrame->data[1], convert_data[1], length);

		pFrame->pts = i*100;

		ret = avcodec_encode_audio2(pCodecCtx, &pkt, pFrame, &got_output);

		if (ret < 0) {
			cout << "error encoding";
			return -1;
		}

		if (pkt.data == NULL)
		{
			av_free_packet(&pkt);
			continue;
		}

		if (got_output) {
			framecnt++;
		
			padts[3] = (char)(((chanCfg & 3) << 6) + ((7 + pkt.size) >> 11));
			padts[4] = (char)(((7 + pkt.size) & 0x7FF) >> 3);
			padts[5] = (char)((((7 + pkt.size) & 7) << 5) + 0x1F);
			fwrite(padts, 7, 1, fp_out);
			fwrite(pkt.data, 1, pkt.size, fp_out);

			av_free_packet(&pkt);
		}
	}
	//Flush Encoder  
	for (got_output = 1; got_output; i++) {
		ret = avcodec_encode_audio2(pCodecCtx, &pkt, NULL, &got_output);
		if (ret < 0) {
			printf("Error encoding frame\n");
			return -1;
		}
		if (got_output) {
			
 			padts[3] = (char)(((chanCfg & 3) << 6) + ((7 + pkt.size) >> 11));
 			padts[4] = (char)(((7 + pkt.size) & 0x7FF) >> 3);
 			padts[5] = (char)((((7 + pkt.size) & 7) << 5) + 0x1F);
// 
 			fwrite(padts, 7, 1, fp_out);
//			fwrite(mp3padts, 4, 1, fp_out);
			fwrite(pkt.data, 1, pkt.size, fp_out);
			av_free_packet(&pkt);
		}
	}

	fclose(fp_out);
	avcodec_close(pCodecCtx);
	av_free(pCodecCtx);
	av_freep(&pFrame->data[0]);
	av_frame_free(&pFrame);

	av_freep(&convert_data[0]);
	free(convert_data);
	return 0;
}

有两个疑问，希望知道原理的博友能告知：

1、AV_CODEC_ID_PCM_ALAW这个为G711A编解码器，编码器的frame_size为0 ，不知道怎么回事？需要自己指定，比如和AAC的一样为1024。

2、转为MP3、G711A的时候不需要加头，但是为什么AAC的需要加头呢？