C实现用ffmpeg MP3转8K PCM

可以使用ffmpeg库来实现MP3转8K PCM的功能。以下是一个简单的C语言代码示例：

#include <stdio.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>

int main(int argc, char *argv[]) {
    AVFormatContext *format_ctx = NULL;
    AVCodecContext *codec_ctx = NULL;
    AVCodec *codec = NULL;
    AVPacket *packet = av_packet_alloc();
    AVFrame *frame = av_frame_alloc();
    int stream_index = -1;
    int ret = 0;
    int got_frame = 0;
    int i, j;

    // 初始化FFmpeg库
    av_register_all();
    avformat_network_init();

    // 打开输入文件
    if ((ret = avformat_open_input(&format_ctx, argv[1], NULL, NULL)) < 0) {
        fprintf(stderr, "Could not open input file '%s'\n", argv[1]);
        goto end;
    }

    // 查找音频流
    if ((ret = avformat_find_stream_info(format_ctx, NULL)) < 0) {
        fprintf(stderr, "Could not find stream information\n");
        goto end;
    }

    for (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) {
        fprintf(stderr, "Could not find audio stream\n");
        goto end;
    }

    // 获取解码器
    codec = avcodec_find_decoder(format_ctx->streams[stream_index]->codecpar->codec_id);
    if (!codec) {
        fprintf(stderr, "Could not find decoder\n");
        goto end;
    }

    codec_ctx = avcodec_alloc_context3(codec);
    if (!codec_ctx) {
        fprintf(stderr, "Could not allocate codec context\n");
        goto end;
    }

    if ((ret = avcodec_parameters_to_context(codec_ctx, format_ctx->streams[stream_index]->codecpar)) < 0) {
        fprintf(stderr, "Could not initialize codec context\n");
        goto end;
    }

    if ((ret = avcodec_open2(codec_ctx, codec, NULL)) < 0) {
        fprintf(stderr, "Could not open codec\n");
        goto end;
    }

    // 初始化重采样上下文
    SwrContext *swr_ctx = swr_alloc_set_opts(NULL,
                                             AV_CH_LAYOUT_MONO, AV_SAMPLE_FMT_S16, 8000,
                                             av_get_default_channel_layout(codec_ctx->channels), codec_ctx->sample_fmt, codec_ctx->sample_rate,
                                             0, NULL);
    if (!swr_ctx) {
        fprintf(stderr, "Could not allocate resampling context\n");
        goto end;
    }

    if ((ret = swr_init(swr_ctx)) < 0) {
        fprintf(stderr, "Could not initialize resampling context\n");
        goto end;
    }

    // 读取并解码音频帧
    while ((ret = av_read_frame(format_ctx, packet)) == 0) {
        if (packet->stream_index == stream_index) {
            ret = avcodec_send_packet(codec_ctx, packet);
            if (ret < 0) {
                fprintf(stderr, "Error sending packet to decoder\n");
                goto end;
            }

            while (ret >= 0) {
                ret = avcodec_receive_frame(codec_ctx, frame);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                    break;
                } else if (ret < 0) {
                    fprintf(stderr, "Error receiving frame from decoder\n");
                    goto end;
                }

                // 重采样
                uint8_t *out_buf;
                int out_samples = av_rescale_rnd(swr_get_delay(swr_ctx, frame->sample_rate) + frame->nb_samples, 8000, frame->sample_rate, AV_ROUND_UP);
                av_samples_alloc(&out_buf, NULL, 1, out_samples, AV_SAMPLE_FMT_S16, 0);
                out_samples = swr_convert(swr_ctx, &out_buf, out_samples, (const uint8_t **)frame->data, frame->nb_samples);

                // 输出PCM数据
                fwrite(out_buf, sizeof(uint8_t), out_samples * 2, stdout);

                av_free(out_buf);
            }
        }
        av_packet_unref(packet);
    }

end:
    if (codec_ctx) {
        avcodec_close(codec_ctx);
        avcodec_free_context(&codec_ctx);
    }

    if (format_ctx) {
        avformat_close_input(&format_ctx);
        avformat_free_context(format_ctx);
    }

    if (packet) {
        av_packet_free(&packet);
    }

    if (frame) {
        av_frame_free(&frame);
    }

    if (swr_ctx) {
        swr_free(&swr_ctx);
    }

    return ret < 0;
}

你可以将上述代码保存为 `mp3_to_pcm.c`，然后使用以下命令编译：

gcc -o mp3_to_pcm mp3_to_pcm.c -lavformat -lavcodec -lswresample

然后执行以下命令进行转换：

./mp3_to_pcm input.mp3 > output.pcm

其中，`input.mp3` 是输入MP3文件名，`output.pcm` 是输出PCM文件名。注意，输出的PCM数据是单声道、16位、8000Hz的。