• FFMpeg音频混合,背景音(十):将两个mp3或者wav混音成pcm


    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
    ");
            return ret;
        }
    
        if ((ret = avformat_find_stream_info(fmt_ctx1, NULL)) < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot find stream information
    ");
            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
    ");
            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
    ");
            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
    ");
            return ret;
        }
    
        if ((ret = avformat_find_stream_info(fmt_ctx2, NULL)) < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot find stream information
    ");
            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
    ");
            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
    ");
            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
    ");
            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
    ");
            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
    ");
            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
    ");
            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
    ");
            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
    ");
            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
    ",
               (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
    ");
        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
    ");
                    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
    ");
                        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
    ");
                            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
    ");
                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
    ");
                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
    ");
                    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
    ", av_err2str(ret));
            exit(1);
        }
        exit(0);
    }
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  • 原文地址:https://www.cnblogs.com/zhangxianrong/p/13900645.html
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