解码aac-1

标注：
本文代码直接修改了ffmpeg的example,可直接编译使用

/**
 * @file
 * audio decoding with libavcodec API example
 *
 * @example decode_audio.c
 */
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
extern "C"
{
#include <libavutil/frame.h>
#include <libavutil/mem.h>
#include <libavcodec/avcodec.h>
}
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static int get_format_from_sample_fmt(const char** fmt,
    enum AVSampleFormat sample_fmt)
{
    int i;
    struct sample_fmt_entry {
        enum AVSampleFormat sample_fmt; const char* fmt_be, * fmt_le;
    } sample_fmt_entries[] = {
        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
    };
    *fmt = NULL;
    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
        struct sample_fmt_entry* entry = &sample_fmt_entries[i];
        if (sample_fmt == entry->sample_fmt) {
            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
            return 0;
        }
    }
    fprintf(stderr,
        "sample format %s is not supported as output format\n",
        av_get_sample_fmt_name(sample_fmt));
    return -1;
}
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 = "out.pcm";
    const char* filename = "test.aac";
    const AVCodec* codec;
    AVCodecContext* codec_ctx = 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;
    enum AVSampleFormat sfmt;
    int n_channels = 0;
    const char* fmt;
    pkt = av_packet_alloc();
    /* find the MPEG audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
    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);
    }
    codec_ctx = avcodec_alloc_context3(codec);
    if (!codec_ctx) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    /* open it */
    if (avcodec_open2(codec_ctx, 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(codec_ctx);
        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, codec_ctx, &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;
        if (pkt->size)
            decode(codec_ctx, 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(codec_ctx, pkt, decoded_frame, outfile);
    /* print output pcm infomations, because there have no metadata of pcm */
    sfmt = codec_ctx->sample_fmt;
    if (av_sample_fmt_is_planar(sfmt)) {
        const char* packed = av_get_sample_fmt_name(sfmt);
        printf("Warning: the sample format the decoder produced is planar "
            "(%s). This example will output the first channel only.\n",
            packed ? packed : "?");
        sfmt = av_get_packed_sample_fmt(sfmt);
    }
    n_channels = codec_ctx->channels;
    if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
        goto end;
    printf("Play the output audio file with the command:\n"
        "ffplay -f %s -ac %d -ar %d %s\n",
        fmt, n_channels, codec_ctx->sample_rate,
        outfilename);
end:
    fclose(outfile);
    fclose(f);
    avcodec_free_context(&codec_ctx);
    av_parser_close(parser);
    av_frame_free(&decoded_frame);
    av_packet_free(&pkt);
    return 0;
}