写一个ffmpeg播放MP3的demo

以下是一个使用FFmpeg播放MP3文件的示例代码，使用了libavformat和libavcodec库：

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <pthread.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libswresample/swresample.h>

#define BUFFER_SIZE 4096

int main(int argc, char *argv[]) {
    AVFormatContext *format_ctx = NULL;
    AVCodecContext *codec_ctx = NULL;
    AVCodecParameters *codec_params = NULL;
    AVCodec *codec = NULL;
    AVFrame *frame = NULL;
    AVPacket *packet = NULL;
    SwrContext *swr_ctx = NULL;
    uint8_t *buffer = NULL;
    int buffer_size = BUFFER_SIZE;
    int stream_index = -1;
    int ret;

    if (argc < 2) {
        printf("Usage: %s <input_file>\n", argv[0]);
        return 1;
    }

    av_register_all();
    avformat_network_init();

    format_ctx = avformat_alloc_context();
    if (!format_ctx) {
        printf("Error: Could not allocate format context.\n");
        return 1;
    }

    ret = avformat_open_input(&format_ctx, argv[1], NULL, NULL);
    if (ret < 0) {
        printf("Error: Could not open input file '%s'\n", argv[1]);
        return 1;
    }

    ret = avformat_find_stream_info(format_ctx, NULL);
    if (ret < 0) {
        printf("Error: Could not find stream information.\n");
        return 1;
    }

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

    for (int i = 0; i < format_ctx->nb_streams; i++) {
        if (format_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
            stream_index = i;
            break;
        }
    }

    if (stream_index == -1) {
        printf("Error: Could not find audio stream.\n");
        return 1;
    }

    codec_params = format_ctx->streams[stream_index]->codecpar;
    codec = avcodec_find_decoder(codec_params->codec_id);
    if (!codec) {
        printf("Error: Unsupported codec.\n");
        return 1;
    }

    codec_ctx = avcodec_alloc_context3(codec);
    if (!codec_ctx) {
        printf("Error: Could not allocate codec context.\n");
        return 1;
    }

    ret = avcodec_parameters_to_context(codec_ctx, codec_params);
    if (ret < 0) {
        printf("Error: Could not copy codec parameters to codec context.\n");
        return 1;
    }

    ret = avcodec_open2(codec_ctx, codec, NULL);
    if (ret < 0) {
        printf("Error: Could not open codec.\n");
        return 1;
    }

    frame = av_frame_alloc();
    packet = av_packet_alloc();
    if (!frame || !packet) {
        printf("Error: Could not allocate frame and packet.\n");
        return 1;
    }

    swr_ctx = swr_alloc();
    if (!swr_ctx) {
        printf("Error: Could not allocate resampling context.\n");
        return 1;
    }

    av_opt_set_int(swr_ctx, "in_channel_layout", codec_ctx->channel_layout, 0);
    av_opt_set_int(swr_ctx, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
    av_opt_set_int(swr_ctx, "in_sample_rate", codec_ctx->sample_rate, 0);
    av_opt_set_int(swr_ctx, "out_sample_rate", 44100, 0);
    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", codec_ctx->sample_fmt, 0);
    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);

    ret = swr_init(swr_ctx);
    if (ret < 0) {
        printf("Error: Could not initialize resampling context.\n");
        return 1;
    }

    buffer = av_malloc(buffer_size);
    if (!buffer) {
        printf("Error: Could not allocate buffer.\n");
        return 1;
    }

    while (av_read_frame(format_ctx, packet) >= 0) {
        if (packet->stream_index == stream_index) {
            ret = avcodec_send_packet(codec_ctx, packet);
            if (ret < 0) {
                printf("Error: Could not send packet to decoder.\n");
                break;
            }

            while (ret >= 0) {
                ret = avcodec_receive_frame(codec_ctx, frame);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                    break;
                else if (ret < 0) {
                    printf("Error: Could not receive frame from decoder.\n");
                    goto end;
                }

                int data_size = av_samples_get_buffer_size(NULL, codec_ctx->channels,
                                                           frame->nb_samples, AV_SAMPLE_FMT_S16, 1);
                if (data_size > buffer_size) {
                    buffer_size = data_size;
                    buffer = av_realloc(buffer, buffer_size);
                }

                swr_convert(swr_ctx, &buffer, frame->nb_samples,
                            (const uint8_t **)frame->data, frame->nb_samples);

                fwrite(buffer, 1, data_size, stdout);
            }
        }

        av_packet_unref(packet);
    }

end:
    avformat_close_input(&format_ctx);
    avcodec_free_context(&codec_ctx);
    avformat_free_context(format_ctx);
    av_frame_free(&frame);
    av_packet_free(&packet);
    swr_free(&swr_ctx);
    av_free(buffer);

    return 0;
}

这个示例代码会打开指定的MP3文件, 解码音频流, 并输出16-bit PCM音频数据到stdout。您可以将输出重定向到另一个程序或使用管道将其传递给ffplay等播放器进行播放。请注意，您需要在编译时链接FFmpeg库。