ffmpeg新版本库实现音频解码为wav格式的代码

以下是使用FFmpeg新版本库实现音频解码为wav格式的代码：

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <stdint.h>
#include <stdbool.h>
#include <unistd.h>
#include <libavutil/frame.h>
#include <libavutil/mem.h>
#include <libavutil/opt.h>
#include <libavutil/samplefmt.h>
#include <libavutil/channel_layout.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>

#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096

int main(int argc, char **argv)
{
    AVCodec *codec;
    AVCodecContext *c = NULL;
    int len;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;
    int got_frame;
    int ret;
    int16_t *samples;
    int sample_size;
    SwrContext *swr_ctx = NULL;
    int dst_nb_samples;
    int dst_linesize;
    uint8_t **dst_data;
    int dst_bufsize;
    int i;

    if (argc <= 2) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
        exit(0);
    }

    av_register_all();

    if (avformat_open_input(&c, argv[1], NULL, NULL) < 0) {
        fprintf(stderr, "Could not open input file '%s'\n", argv[1]);
        exit(1);
    }

    if (avformat_find_stream_info(c, NULL) < 0) {
        fprintf(stderr, "Could not find stream information\n");
        exit(1);
    }

    av_dump_format(c, 0, argv[1], 0);

    codec = avcodec_find_decoder(c->streams[0]->codecpar->codec_id);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate codec context\n");
        exit(1);
    }

    if (avcodec_parameters_to_context(c, c->streams[0]->codecpar) < 0) {
        fprintf(stderr, "Could not copy codec parameters to context\n");
        exit(1);
    }

    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }

    f = fopen(argv[2], "wb");
    if (!f) {
        fprintf(stderr, "Could not open output file '%s'\n", argv[2]);
        exit(1);
    }

    decoded_frame = av_frame_alloc();
    if (!decoded_frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
    }

    av_init_packet(&avpkt);

    while (av_read_frame(c, &avpkt) >= 0) {
        if (avpkt.stream_index == 0) {
            ret = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
            if (ret < 0) {
                fprintf(stderr, "Error decoding audio frame\n");
                exit(1);
            }

            if (got_frame) {
                sample_size = av_get_bytes_per_sample(c->sample_fmt);
                if (sample_size < 0) {
                    fprintf(stderr, "Failed to calculate sample size\n");
                    exit(1);
                }

                samples = (int16_t *)decoded_frame->data[0];
                len = decoded_frame->nb_samples * c->channels * sample_size;

                if (swr_ctx) {
                    av_freep(&dst_data[0]);
                    swr_free(&swr_ctx);
                }

                swr_ctx = swr_alloc_set_opts(NULL,
                                             av_get_default_channel_layout(c->channels),
                                             AV_SAMPLE_FMT_S16,
                                             c->sample_rate,
                                             av_get_default_channel_layout(c->channels),
                                             c->sample_fmt,
                                             c->sample_rate,
                                             0,
                                             NULL);
                if (!swr_ctx || swr_init(swr_ctx) < 0) {
                    fprintf(stderr, "Failed to initialize the resampling context\n");
                    exit(1);
                }

                dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + decoded_frame->nb_samples,
                                                c->sample_rate,
                                                c->sample_rate,
                                                AV_ROUND_UP);
                ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, c->channels, dst_nb_samples, AV_SAMPLE_FMT_S16, 0);
                if (ret < 0) {
                    fprintf(stderr, "Could not allocate destination samples\n");
                    exit(1);
                }

                dst_bufsize = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples, AV_SAMPLE_FMT_S16, 0);
                if (dst_bufsize < 0) {
                    fprintf(stderr, "Could not get sample buffer size\n");
                    exit(1);
                }

                ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)decoded_frame->data, decoded_frame->nb_samples);
                if (ret < 0) {
                    fprintf(stderr, "Error while converting\n");
                    exit(1);
                }

                fwrite(dst_data[0], 1, dst_bufsize, f);
            }
        }

        av_packet_unref(&avpkt);
    }

    fclose(f);

    avcodec_free_context(&c);
    av_frame_free(&decoded_frame);
    av_free(samples);

    if (swr_ctx) {
        av_freep(&dst_data[0]);
        swr_free(&swr_ctx);
    }

    return 0;
}

这段代码使用了FFmpeg库来解码音频文件，并将其转换为WAV格式。它可以处理多种音频格式，包括MP3、AAC、FLAC等。