一、ffmpeg版本说明

ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-39)
  configuration: --disable-debug --enable-libx264 --enable-libx265 --enable-gpl --enable-shared --prefix=/usr/local/ffmpeg
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...

二、代码使用说明

输入：3.mp3,4.mp3  或者3.wav，4.wav

输出：temp.pcm

本代码是在官方文档中的例子中filtering_audio.c修改而成。

三、混音代码

/**
 * @file
 * API example for audio decoding and filtering
 * @example filtering_audio.c
 */

#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#define ENABLE_FILTERS 1
static const char *filter_descr = "[in0][in1]amix=inputs=2[out]";
static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

static AVFormatContext *fmt_ctx1;
static AVFormatContext *fmt_ctx2;

static AVCodecContext *dec_ctx1;
static AVCodecContext *dec_ctx2;

AVFilterContext *buffersink_ctx;

AVFilterContext *buffersrc_ctx1;
AVFilterContext *buffersrc_ctx2;

AVFilterGraph *filter_graph;

static int audio_stream_index_1 = -1;
static int audio_stream_index_2 = -1;

static int open_input_file_1(const char *filename)
{
    int ret;
    AVCodec *dec;

    if ((ret = avformat_open_input(&fmt_ctx1, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
        return ret;
    }

    if ((ret = avformat_find_stream_info(fmt_ctx1, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
        return ret;
    }

    /* select the audio stream */
    ret = av_find_best_stream(fmt_ctx1, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
        return ret;
    }
    audio_stream_index_1 = ret;

    /* create decoding context */
    dec_ctx1 = avcodec_alloc_context3(dec);
    if (!dec_ctx1)
        return AVERROR(ENOMEM);
    avcodec_parameters_to_context(dec_ctx1, fmt_ctx1->streams[audio_stream_index_1]->codecpar);

    /* init the audio decoder */
    if ((ret = avcodec_open2(dec_ctx1, dec, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
        return ret;
    }

    return 0;
}



static int open_input_file_2(const char *filename)
{
    int ret;
    AVCodec *dec;

    if ((ret = avformat_open_input(&fmt_ctx2, filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
        return ret;
    }

    if ((ret = avformat_find_stream_info(fmt_ctx2, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
        return ret;
    }

    /* select the audio stream */
    ret = av_find_best_stream(fmt_ctx2, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
        return ret;
    }
    audio_stream_index_2 = ret;

    /* create decoding context */
    dec_ctx2 = avcodec_alloc_context3(dec);
    if (!dec_ctx2)
        return AVERROR(ENOMEM);
    avcodec_parameters_to_context(dec_ctx2, fmt_ctx2->streams[audio_stream_index_2]->codecpar);

    /* init the audio decoder */
    if ((ret = avcodec_open2(dec_ctx2, dec, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
        return ret;
    }

    return 0;
}







static int init_filters(const char *filters_descr)
{
    char args1[512];
    char args2[512];
    int ret = 0;
    const AVFilter *abuffersrc1  = avfilter_get_by_name("abuffer");
    const AVFilter *abuffersrc2  = avfilter_get_by_name("abuffer");
    const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");

    AVFilterInOut *outputs1 = avfilter_inout_alloc();
    AVFilterInOut *outputs2 = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();

    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
    static const int out_sample_rates[] = { 8000, -1 };
    const AVFilterLink *outlink;

    AVRational time_base_1 = fmt_ctx1->streams[audio_stream_index_1]->time_base;
    AVRational time_base_2 = fmt_ctx2->streams[audio_stream_index_2]->time_base;

    filter_graph = avfilter_graph_alloc();

    if (!outputs1 || !inputs || !filter_graph) {
        ret = AVERROR(ENOMEM);
        goto end;
    }

    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
    if (!dec_ctx1->channel_layout)
        dec_ctx1->channel_layout = av_get_default_channel_layout(dec_ctx1->channels);
    snprintf(args1, sizeof(args1),
            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
             time_base_1.num, time_base_1.den, dec_ctx1->sample_rate,
             av_get_sample_fmt_name(dec_ctx1->sample_fmt), dec_ctx1->channel_layout);
    ret = avfilter_graph_create_filter(&buffersrc_ctx1, abuffersrc1, "in1",
                                       args1, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
        goto end;
    }

#if (ENABLE_FILTERS)
    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
    if (!dec_ctx2->channel_layout)
    dec_ctx2->channel_layout = av_get_default_channel_layout(dec_ctx2->channels);
    snprintf(args2, sizeof(args2),
        "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
    time_base_2.num, time_base_2.den, dec_ctx2->sample_rate,
    av_get_sample_fmt_name(dec_ctx2->sample_fmt), dec_ctx2->channel_layout);
    ret = avfilter_graph_create_filter(&buffersrc_ctx2, abuffersrc1, "in2",
                            args2, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
        goto end;
    }
#endif

    /* buffer audio sink: to terminate the filter chain. */
    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
        goto end;
    }

    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
        goto end;
    }

    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
        goto end;
    }

    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
        goto end;
    }

    /*
     * Set the endpoints for the filter graph. The filter_graph will
     * be linked to the graph described by filters_descr.
     */

    /*
     * The buffer source output must be connected to the input pad of
     * the first filter described by filters_descr; since the first
     * filter input label is not specified, it is set to "in" by
     * default.
     */
    outputs1->name       = av_strdup("in0");
    outputs1->filter_ctx = buffersrc_ctx1;
    outputs1->pad_idx    = 0;
#if (ENABLE_FILTERS)
    outputs1->next       = outputs2;
    outputs2->name       = av_strdup("in1");
    outputs2->filter_ctx = buffersrc_ctx2;
    outputs2->pad_idx    = 0;
    outputs2->next       = NULL;
#else
    outputs1->next       = NULL;
#endif
    /*
     * The buffer sink input must be connected to the output pad of
     * the last filter described by filters_descr; since the last
     * filter output label is not specified, it is set to "out" by
     * default.
     */
    inputs->name       = av_strdup("out");
    inputs->filter_ctx = buffersink_ctx;
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

    AVFilterInOut* filter_outputs[2];
    filter_outputs[0] = outputs1;
#if (ENABLE_FILTERS)
    filter_outputs[1] = outputs2;
#endif

    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                        &inputs, &outputs1, NULL)) < 0)
        goto end;

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
        goto end;

    /* Print summary of the sink buffer
     * Note: args buffer is reused to store channel layout string */
    outlink = buffersink_ctx->inputs[0];
    av_get_channel_layout_string(args1, sizeof(args1), -1, outlink->channel_layout);
    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
           (int)outlink->sample_rate,
           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
           args1);

end:
    avfilter_inout_free(&inputs);
    avfilter_inout_free(&outputs1);

    return ret;
}

static void print_frame(const AVFrame *frame)
#if 1
{
    FILE *file = NULL;
    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
    const uint16_t *p     = (uint16_t*)frame->data[0];
    const uint16_t *p_end = p + n;

    file = fopen("tmp.pcm", "ab+");
    if (NULL == file){
        perror("fopen tmp.mp3 error\n");
    return;
    }
    
    fwrite(frame->data[0], n * 2, 1, file);
    fclose(file);
    file = NULL;
}
#else
{
    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
    const uint16_t *p     = (uint16_t*)frame->data[0];
    const uint16_t *p_end = p + n;

    while (p < p_end) {
        fputc(*p    & 0xff, stdout);
        fputc(*p>>8 & 0xff, stdout);
        p++;
    }
    fflush(stdout);
}
#endif

int main(int argc, char **argv)
{
    int ret;
    AVPacket packet1;
    AVPacket packet2;
    AVFrame *frame = av_frame_alloc();
    AVFrame *filt_frame = av_frame_alloc();

    if (!frame || !filt_frame) {
        perror("Could not allocate frame");
        exit(1);
    }

    if ((ret = open_input_file_1("3_0.1.wav")) < 0)
        goto end;
    if ((ret = open_input_file_2("4.wav")) < 0)
    goto end;

    if ((ret = init_filters(filter_descr)) < 0)
        goto end;
    /* read all packets */
    while (1) {
        if ((ret = av_read_frame(fmt_ctx1, &packet1)) < 0)
                break;
        if (packet1.stream_index == audio_stream_index_1) {
            ret = avcodec_send_packet(dec_ctx1, &packet1);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
                break;
            }
            while (ret >= 0) {
                ret = avcodec_receive_frame(dec_ctx1, frame);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                    break;
                } else if (ret < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
                    goto end;
                }
                if (ret >= 0) {
                    /* push the audio data from decoded frame into the filtergraph */
                    if (av_buffersrc_add_frame_flags(buffersrc_ctx1, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
                        av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                        break;
                    }

                    /* pull filtered audio from the filtergraph */
                    while (1) {
                        ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                            break;
                        if (ret < 0)
                            goto end;
                        print_frame(filt_frame);
                        av_frame_unref(filt_frame);
                    }
                    av_frame_unref(frame);
                }
            }
        }
    if ((ret = av_read_frame(fmt_ctx2, &packet2)) < 0)
        break;
    if (packet2.stream_index == audio_stream_index_2) {
        ret = avcodec_send_packet(dec_ctx2, &packet2);
        if (ret < 0) {  
            av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
            break;
            }
        while (ret >= 0) {
            ret = avcodec_receive_frame(dec_ctx2, frame);
            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                break;
                }else if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
            goto end;
                }
            if (ret >= 0) {
            if (av_buffersrc_add_frame_flags(buffersrc_ctx2, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                break;
            }
            while (1) {
                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                break;
                if (ret < 0)
                goto end;
                print_frame(filt_frame);
                av_frame_unref(filt_frame);
            }
            av_frame_unref(frame);
                }
        }
    }
        av_packet_unref(&packet1);
    av_packet_unref(&packet2);
    }
end:
    avfilter_graph_free(&filter_graph);
    avcodec_free_context(&dec_ctx1);
    avcodec_free_context(&dec_ctx2);
    avformat_close_input(&fmt_ctx2);
    avformat_close_input(&fmt_ctx1);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
        exit(1);
    }
    exit(0);
}