FFmpeg 代码实现流媒体推流（RTSP）

实时录屏并把视频推流到RTSP服务器，具体流程是抓取屏幕内容(bitmap)，并把bitmap转化为YUV，接着把YUV编码成H264，再把H264码流推到RTSP服务器；把采集到的PCM编码为AAC，再把AAC推流至RTSP服务器。

RTSP服务器使用的是HappyTime的免费试用版本。

1. bitmap转YUV

我抓到的bitmap是BGRA格式的，所以使用的图像格式是AV_PIX_FMT_BGRA，cropImage是含有rgba图像的数组

bool init_RGB_to_YUV(){
    
    //BGRA 转 YUV
    swrCtxBGRA2YUV = sws_getContext(
        cap_w, cap_h, AV_PIX_FMT_BGRA,
        cap_w, cap_h, AV_PIX_FMT_YUV420P,
        SWS_BICUBIC,
        NULL, NULL, NULL
        );

    //创建BGRA帧
    frame_bgra = av_frame_alloc();
    frame_bgra->format = AV_PIX_FMT_BGRA;
    frame_bgra->width = cap_w;
    frame_bgra->height = cap_h;
    if (av_frame_get_buffer(frame_bgra, 32) < 0) {
        printf("Failed: av_frame_get_buffer\n");
        return false;
    }
    frame_bgra->data[0] = cropImage;
    

    //YUV帧
    frame_yuv = av_frame_alloc();
    frame_yuv->width = cap_w;
    frame_yuv->height = cap_h;
    frame_yuv->format = AV_PIX_FMT_YUV420P;

    //
    uint8_t *picture_buf = (uint8_t *)av_malloc(cap_w * cap_h * 1.5);
    if (av_image_fill_arrays(frame_yuv->data, frame_yuv->linesize, picture_buf, AV_PIX_FMT_YUV420P, cap_w, cap_h, 1) < 0){
        printf("Failed: av_image_fill_arrays\n");
        return false;
    }
    return true;
}

调用：

//BGRA 转 YUV
    if (sws_scale(swrCtxBGRA2YUV,
        frame_bgra->data, frame_bgra->linesize,
        0, cap_h,
        frame_yuv->data, frame_yuv->linesize) < 0)
    {
        printf("失败：BGRA 转 YUV\n");
        return;
    }

    frame_yuv->pts = av_gettime();

由于我是实时抓取的屏幕，frame_yuv->pts设为当前的时间戳，以保证能正常播放。

2. H264编码

bool init_YUV_to_H264(){
    //寻找编码器
    codec_h264 = avcodec_find_encoder(AV_CODEC_ID_H264);
    if (!codec_h264){
        printf("Fail: avcodec_find_encoder\n");
        return false;
    }

    //编码器上下文
    codec_ctx_h264 = avcodec_alloc_context3(codec_h264);
    if (!codec_ctx_h264){
        printf("Fail: avcodec_alloc_context3\n");
        return false;
    }
    codec_ctx_h264->pix_fmt = AV_PIX_FMT_YUV420P;
    codec_ctx_h264->codec_type = AVMEDIA_TYPE_VIDEO;
    codec_ctx_h264->width = cap_w;
    codec_ctx_h264->height = cap_h;
    codec_ctx_h264->channels = 3;
    codec_ctx_h264->time_base = { 1, 25 };
    codec_ctx_h264->gop_size = 5;   //图像组两个关键帧（I帧）的距离
    codec_ctx_h264->max_b_frames = 0;
    //codec_ctx_h264->qcompress = 0.6;
    //codec_ctx_h264->bit_rate = 90000;
    codec_ctx_h264->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;   //添加PPS、SPS
    
    av_opt_set(codec_ctx_h264->priv_data, "preset", "ultrafast", 0);    //快速编码，但会损失质量
    //av_opt_set(codec_ctx_h264->priv_data, "tune", "zerolatency", 0);  //适用于快速编码和低延迟流式传输,但是会出现绿屏
    //av_opt_set(codec_ctx_h264->priv_data, "x264opts", "crf=26:vbv-maxrate=728:vbv-bufsize=3640:keyint=25", 0);


    //打开编码器
    if (avcodec_open2(codec_ctx_h264, codec_h264, NULL) < 0){
        printf("Fail: avcodec_open2\n");
        return false;
    }

    //用于接收编码好的H264
    pkt_h264 = av_packet_alloc();

    return true;
}

本文福利， C++音视频学习资料包、技术视频，内容包括（音视频开发，面试题，FFmpeg ，webRTC ，rtmp ，hls ，rtsp ，ffplay ，srs）↓↓↓↓↓↓见下面↓↓文章底部↓↓

调用：

    ret = avcodec_send_frame(codec_ctx_h264, frame_yuv);
    if (ret < 0){
        printf("send frame fail\n");
        return;
    }

    while (ret >= 0)        {
        ret = avcodec_receive_packet(codec_ctx_h264, pkt_h264);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF){
            break;
        }

        if (ret < 0){
            printf("Error during encoding\n");
            break;
        }

        pkt_h264->stream_index = videoindex;
        //printf("pkt_h264 timestamp = %d\n", pkt_h264->pts);

        if (av_interleaved_write_frame(fmt_ctx, pkt_h264) < 0) {
            printf("Error muxing packet\n");
        }

        av_packet_unref(pkt_h264);
    }

3. AAC编码 

bool init_PCM_to_AAC(){

    codec_aac = avcodec_find_encoder(AV_CODEC_ID_AAC);
    if (!codec_aac) {
        printf("avcodec_find_encoder fail\n");
        return false;
    }

    codec_ctx_aac = avcodec_alloc_context3(codec_aac);
    if (!codec_ctx_aac) {
        printf("avcodec_find_encoder fail\n");
        return false;
    }
    codec_ctx_aac->sample_fmt = AV_SAMPLE_FMT_FLT;
    codec_ctx_aac->codec_type = AVMEDIA_TYPE_AUDIO;
    codec_ctx_aac->channels = channels;
    codec_ctx_aac->channel_layout = av_get_default_channel_layout(channels);
    codec_ctx_aac->sample_rate = sample_rete;

    if (avcodec_open2(codec_ctx_aac, codec_aac, NULL) < 0) {
        printf("open codec fail\n");
        return false;
    }

    swrCtxS162FLT = swr_alloc_set_opts(NULL,
        codec_ctx_aac->channel_layout, codec_ctx_aac->sample_fmt, codec_ctx_aac->sample_rate,
        codec_ctx_aac->channel_layout, AV_SAMPLE_FMT_S16, codec_ctx_aac->sample_rate,
        0, 0);

    if (!swrCtxS162FLT)
    {
        printf("swr_alloc_set_opts error\n");
        return false;
    }
    if (swr_init(swrCtxS162FLT) < 0) {
        printf("open resample fail\n");
        return false;
    }

    frame_pcm = av_frame_alloc();
    frame_pcm->nb_samples = nbSamples_; //一帧音频存放的样本数量
    frame_pcm->format = codec_ctx_aac->sample_fmt;
    frame_pcm->channels = codec_ctx_aac->channels;
    frame_pcm->channel_layout = codec_ctx_aac->channel_layout;

    if (av_frame_get_buffer(frame_pcm, 0) < 0) {
        printf("av_frame_get_buffer error\n");
        return false;
    }

    pkt_aac = av_packet_alloc();

    return true;
}

调用：
其中pcm_buff是包含pcm数据的数组

const uint8_t *pcm[1];
        pcm[0] = pcm_buff;
        int len = swr_convert(swrCtxS162FLT,
            frame_pcm->data, frame_pcm->nb_samples,
            pcm, nbSamples_);

        if (len <= 0) {
            printf("---Encodec:PCM->AAC--- swr_convert fail \n");
            return;
        }

        frame_pcm->pts = av_gettime();

        //printf("channels = %d\n", frame_pcm->channels);
        //printf("framePCM->linesize = %6d %6d\n", frame_pcm->linesize[0], frame_pcm->linesize[1]);

        //AAC编码
        int ret = avcodec_send_frame(codec_ctx_aac, frame_pcm);
        if (ret < 0){
            printf("send frame fail\n");
            return;
        }

        ret = avcodec_receive_packet(codec_ctx_aac, pkt_aac);
        
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF){
            return;
        }

        if (ret < 0){
            printf("Error during encoding\n");
            return;
        }

        pkt_aac->stream_index = audioindex;
        //printf("pkt_aac timestamp = %d\n", pkt_aac->pts);

        if (av_interleaved_write_frame(fmt_ctx, pkt_aac) < 0) {
            printf("Error muxing packet\n");
        }
        av_packet_unref(pkt_aac);

4. 推流器

使用udp传输时传到1400多帧就断开链接了，原因不明，所以改用使用tcp协议传输

bool init_rtsp_pusher(){
    
    //RTSP
    if (avformat_alloc_output_context2(&fmt_ctx, NULL, "RTSP", RTSP_URL.c_str()) < 0){
        printf("Fail: avformat_alloc_output_context2\n");
        return false;
    }

    //使用tcp协议传输
    av_opt_set(fmt_ctx->priv_data, "rtsp_transport", "tcp", 0);

    //检查所有流是否都有数据，如果没有数据会等待max_interleave_delta微秒
    fmt_ctx->max_interleave_delta = 1000000;

    //输出视频流
    AVStream *video_s = avformat_new_stream(fmt_ctx, codec_h264);
    if (!video_s){
        printf("Fail: avformat_new_stream\n");
        return false;
    }
    video_s->time_base = { 1, 25 };
    videoindex = video_s->id = fmt_ctx->nb_streams - 1;  //加入到fmt_ctx流

    //复制AVCodecContext的设置
    if (avcodec_copy_context(video_s->codec, codec_ctx_h264) < 0) {
        printf("Fail: avcodec_copy_context\n");
        return false;
    }
    video_s->codec->codec_tag = 0;
    if (fmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
        video_s->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
    
    avcodec_parameters_from_context(video_s->codecpar, codec_ctx_h264);

    
        //输出音频流
        AVStream *audio_s = avformat_new_stream(fmt_ctx, codec_ctx_aac->codec);
        if (!audio_s){
            printf("Fail: avformat_new_stream\n");
            return false;
        }
        audio_s->time_base = { 1, 25 };
        audioindex = audio_s->id = fmt_ctx->nb_streams - 1;

        //复制AVCodecContext的设置
        if (avcodec_copy_context(audio_s->codec, codec_ctx_aac) < 0) {
            printf("Fail: avcodec_copy_context\n");
            return false;
        }
        audio_s->codec->codec_tag = 0;
        if (fmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
            audio_s->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
        avcodec_parameters_from_context(audio_s->codecpar, codec_ctx_aac);
    
        //printf("fmt_ctx nb_streams = %d\n", fmt_ctx->nb_streams);

    av_dump_format(fmt_ctx, 0, fmt_ctx->filename, 1);
    if (!(fmt_ctx->oformat->flags & AVFMT_NOFILE)) {    //???
        //打开输出URL（Open output URL）
        if (avio_open(&fmt_ctx->pb, fmt_ctx->filename, AVIO_FLAG_WRITE) < 0) {
            printf("Fail: avio_open('%s')\n", fmt_ctx->filename);
            return false;
        }
    }
    return true;
}

结果 

 原文链接：FFmpeg 代码实现流媒体推流（RTSP） - 资料 - 我爱音视频网 - 构建全国最权威的音视频技术交流分享论坛

本文福利， C++音视频学习资料包、技术视频，内容包括（音视频开发，面试题，FFmpeg ，webRTC ，rtmp ，hls ，rtsp ，ffplay ，srs）↓↓↓↓↓↓见下面↓↓文章底部↓↓