FFMpeg音频混合，背景音（八）：将MP3文件解码成PCM

FFMpeg版本：4.3.1
#include<iostream>
#include<windows.h>
using namespace std;
//用到的C的头文件
extern "C"
{
    #include<libavcodec/avcodec.h>
    #include<libavfilter/avfilter.h>
    #include<libavfilter/buffersink.h>
    #include<libavfilter/buffersrc.h>
    #include<libavfilter/avfiltergraph.h>
    #include<libavformat/avformat.h>
    #include<libavutil/avutil.h>
    #include<libavutil/fifo.h>
    #include<libavutil/audio_fifo.h>
    #include<libavdevice/avdevice.h>
    #include <libavutil/frame.h>
    #include <libavutil/mem.h>
    #include <stdio.h>
    #include <stdlib.h>
    #include <string.h>
}
//对用到的预编译
#pragma comment(lib, "avcodec.lib")
#pragma comment(lib, "avfilter.lib")
#pragma comment(lib, "avutil.lib")
#pragma comment(lib, "avformat.lib")
#pragma comment(lib, "swscale.lib")
//第一个音频文件

#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static void decode(AVCodecContext* dec_ctx, AVPacket* pkt, AVFrame* frame,
    FILE* outfile)
{
    int i, ch;
    int ret, data_size;
    /* send the packet with the compressed data to the decoder */
    ret = avcodec_send_packet(dec_ctx, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error submitting the packet to the decoder\n");
        exit(1);
    }
    /* read all the output frames (in general there may be any number of them */
    while (ret >= 0) {
        ret = avcodec_receive_frame(dec_ctx, frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error during decoding\n");
            exit(1);
        }
        data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
        if (data_size < 0) {
            /* This should not occur, checking just for paranoia */
            fprintf(stderr, "Failed to calculate data size\n");
            exit(1);
        }
        for (i = 0; i < frame->nb_samples; i++)
            for (ch = 0; ch < dec_ctx->channels; ch++)
                fwrite(frame->data[ch] + data_size * i, 1, data_size, outfile);
    }
}
int main()
{
    const char* outfilename, * filename;
    const AVCodec* codec;
    AVCodecContext* c = NULL;
    AVCodecParserContext* parser = NULL;
    int len, ret;
    FILE* f, * outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    uint8_t* data;
    size_t   data_size;
    AVPacket* pkt;
    AVFrame* decoded_frame = NULL;

    filename = "D:\\FFMpeg\\project\\pcm_aac\\pcm_aac\\outdio.mp3";
    outfilename = "D:\\FFMpeg\\project\\pcm_aac\\pcm_aac\\outdio.pcm";
    /* register all the codecs */
    avcodec_register_all();
    pkt = av_packet_alloc();
    /* find the MPEG audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MP3);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }
    parser = av_parser_init(codec->id);
    if (!parser) {
        fprintf(stderr, "Parser not found\n");
        exit(1);
    }
    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }
    /* decode until eof */
    data = inbuf;
    data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
    while (data_size > 0) {
        if (!decoded_frame) {
            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
        }
        ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
            data, data_size,
            AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
        if (ret < 0) {
            fprintf(stderr, "Error while parsing\n");
            exit(1);
        }
        data += ret;
        data_size -= ret;
        cout << c->sample_rate << endl;
        cout << c->channels << endl;
        if (pkt->size)
            decode(c, pkt, decoded_frame, outfile);
        if (data_size < AUDIO_REFILL_THRESH) {
            memmove(inbuf, data, data_size);
            data = inbuf;
            len = fread(data + data_size, 1,
                AUDIO_INBUF_SIZE - data_size, f);
            if (len > 0)
                data_size += len;
        }
    }
    /* flush the decoder */
    pkt->data = NULL;
    pkt->size = 0;
    decode(c, pkt, decoded_frame, outfile);
    fclose(outfile);
    fclose(f);
    avcodec_free_context(&c);
    av_parser_close(parser);
    av_frame_free(&decoded_frame);
    av_packet_free(&pkt);
    return 0;
}