音频编码流程 ----- PCM编码为AAC格式

1.音频编码流程

2.编码函数API含义解释

av_frame_make_writable 确保帧数据可写，尽可能避免数据复制.如果帧可写，则不执行任何操作，如果不可写，则分配新缓冲区并复制数据.返回: 成功时为 0，错误时为负 AVERROR.

av_samples_fill_arrays 这个函数就是将你输入的一帧数据写入AVFrame格式输出,便于后面编码send,recvice使用

3.音频编码实战Demo PCM转AAC格式

编码注意点:

1. 编码器相关配置得设置正确,比如码率,采样率,声道,采样格式等
2. 也要设置AVFrame的格式,主要是采样率,声道,采样格式
3. 如果数据不是FLTP格式的花,需要转化为FLTP格式.
4. PCM编码后的数据想要编码为AAC格式,得在前面加上adtsheader.

#ifndef MAINBACK_C
#define MAINBACK_C
#endif // MAINBACK_C
#include 
#include 
#include 
#include 
#include 
#include 
#include 
#include 
#include 
void f32le_convert_to_fltp(float *f32le, float *fltp, int nb_samples) {
    float *fltp_l = fltp;   // 左通道
    float *fltp_r = fltp + nb_samples;   // 右通道
    for(int i = 0; i < nb_samples; i++) {
        fltp_l[i] = f32le[i*2];     // 0 1   - 2 3
        fltp_r[i] = f32le[i*2+1];   // 可以尝试注释左声道或者右声道听听声音
    }
}
static void get_adts_header(AVCodecContext *ctx, uint8_t *adts_header, int aac_length)
{
    uint8_t freq_idx = 0;    //0: 96000 Hz  3: 48000 Hz 4: 44100 Hz
    switch (ctx->sample_rate) {
        case 96000: freq_idx = 0; break;
        case 88200: freq_idx = 1; break;
        case 64000: freq_idx = 2; break;
        case 48000: freq_idx = 3; break;
        case 44100: freq_idx = 4; break;
        case 32000: freq_idx = 5; break;
        case 24000: freq_idx = 6; break;
        case 22050: freq_idx = 7; break;
        case 16000: freq_idx = 8; break;
        case 12000: freq_idx = 9; break;
        case 11025: freq_idx = 10; break;
        case 8000: freq_idx = 11; break;
        case 7350: freq_idx = 12; break;
        default: freq_idx = 4; break;
    }
    uint8_t chanCfg = ctx->channels;
    uint32_t frame_length = aac_length + 7;
    adts_header[0] = 0xFF;
    adts_header[1] = 0xF1;
    adts_header[2] = ((ctx->profile) << 6) + (freq_idx << 2) + (chanCfg >> 2);
    adts_header[3] = (((chanCfg & 3) << 6) + (frame_length  >> 11));
    adts_header[4] = ((frame_length & 0x7FF) >> 3);
    adts_header[5] = (((frame_length & 7) << 5) + 0x1F);
    adts_header[6] = 0xFC;
}
int encode(AVCodecContext* codec_ctx,AVFrame* frame,AVPacket* pkt,FILE* out)
{
    int ret=0;
    ret = avcodec_send_frame(codec_ctx,frame);
    if(ret<0){
        printf("avcodec_send_frame failed\n");
        return ret;
    }

    while(1)
    {
        ret=avcodec_receive_packet(codec_ctx,pkt);
        if(ret==AVERROR(EAGAIN)||ret==AVERROR(AVERROR_EOF))
            break;
        if(ret<0)return ret;
        if(codec_ctx->flags&AV_CODEC_FLAG_GLOBAL_HEADER)
        {
            char header[7]={0};
            get_adts_header(codec_ctx,header,pkt->size);
            int size=fwrite(header,1,7,out);
            if(size!=7)
            {
                printf("fwrite error\n");
                return 0;
            }
        }

        fwrite(pkt->data,1,pkt->size,out);
    }

}
//编码流程  先初始化编码器，读取原始文件，send，recvice，写入
int main(int argc,char* argv[])
{
    if(argv<3)
    {
        printf("argv<3\n");
        return -1;
    }
    char* pcm_path=argv[1];
    char* out_path=argv[2];
    FILE* pcm_file=fopen(pcm_path,"rb");
    FILE* out_file=fopen(out_path,"wb");
    AVCodec* codec=avcodec_find_encoder(AV_CODEC_ID_AAC);
    AVCodecContext* codec_ctx=avcodec_alloc_context3(codec);

    codec_ctx->bit_rate=100000;
    codec_ctx->channel_layout=AV_CH_LAYOUT_STEREO;
    codec_ctx->channels=av_get_channel_layout_nb_channels(codec_ctx->channel_layout);
    codec_ctx->sample_fmt=AV_SAMPLE_FMT_FLTP;
//    codec_ctx->codec_id=AV_CODEC_ID_AAC;
//    codec_ctx->codec_type=AVMEDIA_TYPE_AUDIO;
    codec_ctx->sample_rate=48000;
    codec_ctx->profile=FF_PROFILE_AAC_LOW;
    codec_ctx->flags|=AV_CODEC_FLAG_GLOBAL_HEADER;
    avcodec_open2(codec_ctx,codec,NULL);

    AVFrame* frame=av_frame_alloc();
    AVPacket* pkt=av_packet_alloc();
//    frame->channel_layout=codec_ctx->channel_layout;
    frame->channels=codec_ctx->channels;
//    frame->sample_rate=codec_ctx->sample_rate;
    frame->format=codec_ctx->sample_fmt;
    frame->nb_samples=codec_ctx->frame_size;
    av_frame_get_buffer(frame,0);//分配frame的buff
    int frame_bytes=av_get_bytes_per_sample(frame->format)*frame->channels*frame->nb_samples;
    uint8_t* pcm_buff=av_malloc(frame_bytes);
    memset(pcm_buff,0,frame_bytes);
    uint8_t *pcm_temp_buf = (uint8_t *)malloc(frame_bytes);
    memset(pcm_temp_buf,0,frame_bytes);
    int pts=0;
    int ret=0;
    while(1)
    {
        int rsize=fread(pcm_buff,1,frame_bytes,pcm_file);
        ret = av_frame_make_writable(frame);
        if(ret != 0)
            printf("av_frame_make_writable failed, ret = %d\n", ret);
        f32le_convert_to_fltp(pcm_buff,pcm_temp_buf,frame->nb_samples);
        int need_size=av_samples_fill_arrays(frame->data,frame->linesize,pcm_temp_buf,frame->channels,frame->nb_samples,frame->format,0);
        if(rsize!=need_size)break;
        frame->pts=pts;
        pts+=1;
        printf("frame pts:%d\n",pts);
        encode(codec_ctx,frame,pkt,out_file);
    }
    encode(codec_ctx,NULL,pkt,out_file);
    return 0;
}