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FFmpeg encode_audio.c示例

FFmpeg encode_audio.c示例

作者: Goning | 来源:发表于2019-04-02 16:01 被阅读0次

    encode_audio.c

    /**
     * @file
     * audio encoding with libavcodec API example.
     *
     * @example encode_audio.c
     */
    
    #include <stdint.h>
    #include <stdio.h>
    #include <stdlib.h>
    
    #include <libavcodec/avcodec.h>
    
    #include <libavutil/channel_layout.h>
    #include <libavutil/common.h>
    #include <libavutil/frame.h>
    #include <libavutil/samplefmt.h>
    
    /* check that a given sample format is supported by the encoder */
    static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
    {
        const enum AVSampleFormat *p = codec->sample_fmts;
    
        while (*p != AV_SAMPLE_FMT_NONE) {
            if (*p == sample_fmt)
                return 1;
            p++;
        }
        return 0;
    }
    
    /* just pick the highest supported samplerate */
    static int select_sample_rate(const AVCodec *codec)
    {
        const int *p;
        int best_samplerate = 0;
    
        if (!codec->supported_samplerates)
            return 44100;
    
        p = codec->supported_samplerates;
        while (*p) {
            if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
                best_samplerate = *p;
            p++;
        }
        return best_samplerate;
    }
    
    /* select layout with the highest channel count */
    static int select_channel_layout(const AVCodec *codec)
    {
        const uint64_t *p;
        uint64_t best_ch_layout = 0;
        int best_nb_channels   = 0;
    
        if (!codec->channel_layouts)
            return AV_CH_LAYOUT_STEREO;
    
        p = codec->channel_layouts;
        while (*p) {
            int nb_channels = av_get_channel_layout_nb_channels(*p);
    
            if (nb_channels > best_nb_channels) {
                best_ch_layout    = *p;
                best_nb_channels = nb_channels;
            }
            p++;
        }
        return best_ch_layout;
    }
    
    static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
                       FILE *output)
    {
        int ret;
    
        /* send the frame for encoding */
        ret = avcodec_send_frame(ctx, frame);
        if (ret < 0) {
            fprintf(stderr, "Error sending the frame to the encoder\n");
            exit(1);
        }
    
        /* read all the available output packets (in general there may be any
         * number of them */
        while (ret >= 0) {
            ret = avcodec_receive_packet(ctx, pkt);
            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                return;
            else if (ret < 0) {
                fprintf(stderr, "Error encoding audio frame\n");
                exit(1);
            }
    
            fwrite(pkt->data, 1, pkt->size, output);
            av_packet_unref(pkt);
        }
    }
    
    int main(int argc, char **argv)
    {
        const char *filename;
        const AVCodec *codec;
        AVCodecContext *c= NULL;
        AVFrame *frame;
        AVPacket *pkt;
        int i, j, k, ret;
        FILE *f;
        uint16_t *samples;
        float t, tincr;
    
        if (argc <= 1) {
            fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
            return 0;
        }
        filename = argv[1];
    
        /* find the MP2 encoder */
        codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
        if (!codec) {
            fprintf(stderr, "Codec not found\n");
            exit(1);
        }
    
        c = avcodec_alloc_context3(codec);
        if (!c) {
            fprintf(stderr, "Could not allocate audio codec context\n");
            exit(1);
        }
    
        /* put sample parameters */
        c->bit_rate = 64000;
    
        /* check that the encoder supports s16 pcm input */
        c->sample_fmt = AV_SAMPLE_FMT_S16;
        if (!check_sample_fmt(codec, c->sample_fmt)) {
            fprintf(stderr, "Encoder does not support sample format %s",
                    av_get_sample_fmt_name(c->sample_fmt));
            exit(1);
        }
    
        /* select other audio parameters supported by the encoder */
        c->sample_rate    = select_sample_rate(codec);
        c->channel_layout = select_channel_layout(codec);
        c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
    
        /* open it */
        if (avcodec_open2(c, codec, NULL) < 0) {
            fprintf(stderr, "Could not open codec\n");
            exit(1);
        }
    
        f = fopen(filename, "wb");
        if (!f) {
            fprintf(stderr, "Could not open %s\n", filename);
            exit(1);
        }
    
        /* packet for holding encoded output */
        pkt = av_packet_alloc();
        if (!pkt) {
            fprintf(stderr, "could not allocate the packet\n");
            exit(1);
        }
    
        /* frame containing input raw audio */
        frame = av_frame_alloc();
        if (!frame) {
            fprintf(stderr, "Could not allocate audio frame\n");
            exit(1);
        }
    
        frame->nb_samples     = c->frame_size;
        frame->format         = c->sample_fmt;
        frame->channel_layout = c->channel_layout;
    
        /* allocate the data buffers */
        ret = av_frame_get_buffer(frame, 0);
        if (ret < 0) {
            fprintf(stderr, "Could not allocate audio data buffers\n");
            exit(1);
        }
    
        /* encode a single tone sound */
        t = 0;
        tincr = 2 * M_PI * 440.0 / c->sample_rate;
        for (i = 0; i < 200; i++) {
            /* make sure the frame is writable -- makes a copy if the encoder
             * kept a reference internally */
            ret = av_frame_make_writable(frame);
            if (ret < 0)
                exit(1);
            samples = (uint16_t*)frame->data[0];
    
            for (j = 0; j < c->frame_size; j++) {
                samples[2*j] = (int)(sin(t) * 10000);
    
                for (k = 1; k < c->channels; k++)
                    samples[2*j + k] = samples[2*j];
                t += tincr;
            }
            encode(c, frame, pkt, f);
        }
    
        /* flush the encoder */
        encode(c, NULL, pkt, f);
    
        fclose(f);
    
        av_frame_free(&frame);
        av_packet_free(&pkt);
        avcodec_free_context(&c);
    
        return 0;
    }
    

    1、编译示例程序

    打开msys,进入目录

    gcc -o encode_audio encode_audio.c -Iffmpeg/include -Lffmpeg/lib/ -lavformat -lavfilter -lavcodec -lswscale -lavutil -lswresample 
    

    生成可执行文件encode_audio.exe

    2、运行示例程序
    ./encode_audio.exe hello.mp3
    ./encode_audio.exe hello.wav
    ./encode_audio.exe hello.aac
    

    生成音频文件


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