c++ - VLC 本地 H264 流超过 rtp 的巨大缓冲时间

Question

我正在输出一个 H264 流，由我的应用程序使用 ffmpeg 编码。我可以使用 ffplay 显示它，但是当试图在 VLC 中查看流时，我只得到第一帧，或者看起来就是这样。
消息输出显示它正在“缓冲”，当帧更新时大约需要一分钟才能达到 100%。
使用 ffplay 时，延迟最差约为 50-100ms。
我发送到 rtp://127.0.0.1:6666?pkt_size=1316格式为 rtp_mpegts .
我是新手，很可能我没有完全正确地设置框架。该过程是(减去声明和错误检查)

codec_name = "libx264";
codec = avcodec_find_encoder_by_name(codec_name.c_str());
context = avcodec_alloc_context3(codec);
pkt = av_packet_alloc();
context->bit_rate = 5 * Mega;
context->width = info.DisplayWidth;
context->height = info.DisplayHeight;
context->time_base = { 1, FPS };
context->framerate = { FPS, 1 };
context->gop_size = 100;
context->max_b_frames = 1;            
context->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec->id == AV_CODEC_ID_H264)
            {
                check_ret("set option: preset", av_opt_set(context->priv_data, "preset", "fast", 0));
                check_ret("set option: tune", av_opt_set(context->priv_data, "tune", "zerolatency", 0));
                check_ret("set option: profile", av_opt_set(context->priv_data, "profile", "baseline", 0));                
            }
check_ret("open codec", avcodec_open2(context, codec, NULL));

// setup the stream 
fmt = (AVOutputFormat*)av_guess_format("rtp_mpegts", NULL, NULL);

avformat_alloc_output_context2(&avfctx, fmt, fmt->name,
            "rtp://127.0.0.1:6666?pkt_size=1316"); 
        
avio_open(&avfctx->pb, avfctx->url, AVIO_FLAG_WRITE);
AVStream* stream = avformat_new_stream(avfctx, codec);
avcodec_parameters_from_context(stream->codecpar, context);
stream->time_base.num = 1;
stream->time_base.den = FPS;
avformat_write_header(avfctx, NULL);

// then the encoding (in an output loop)
<not shown: get frame from swapchain, sws_scale from rgba to yuv>
yuvFrame->pts = i++; // i is incremented every frame
avcodec_send_frame(enc_ctx, yuvFrame);
 while (ret >= 0) {
  ret = avcodec_receive_packet(enc_ctx, pkt);          
  //ret = av_interleaved_write_frame(avfctx, pkt); was using this, don't seem to need it
  ret = av_write_frame(avfctx, pkt);
  av_packet_unref(pkt);
}

VLC 输出如下所示:

main debug: using hw decoder module "d3d11va"
avcodec info: Using D3D11VA (NVIDIA GeForce RTX 2080 Super with Max-Q Design, vendor 10de(NVIDIA), device 1e93, revision a1) for hardware decoding
qt debug: Logical video size: 1280x720
main debug: resized to 1280x720
main debug: VoutDisplayEvent 'resize' 1280x720
main debug: Received first picture
main debug: Buffering 1%
main debug: Buffering 2%
main debug: Buffering 3%
main debug: auto hiding mouse cursor
main debug: Buffering 4%
main debug: Buffering 5%
main debug: Buffering 6%
main debug: Buffering 7%
main debug: Buffering 8%
main debug: Buffering 9%
main debug: Buffering 10%
main debug: auto hiding mouse cursor
main debug: Buffering 11%
rtp warning: 1 packet(s) lost
rtp warning: 1 packet(s) lost
rtp warning: 1 packet(s) lost
ts warning: discontinuity received 0x3 instead of 0xd (pid=256)
ts warning: discontinuity received 0x5 instead of 0xf (pid=256)
ts warning: discontinuity received 0x1 instead of 0xb (pid=256)
main debug: Buffering 12%
main debug: Buffering 13%
main debug: Buffering 14%
main debug: Buffering 15%
main debug: Buffering 16%
main debug: Buffering 17%
main debug: Buffering 18%
main debug: auto hiding mouse cursor
main debug: Buffering 19%
main debug: Buffering 20%

Answer 1

我上面的方法的问题是它基于 ffmpeg 示例 encode_video.c从谷歌借来的流输出的一些位。
感谢@rotem，我开始整理一个独立的可执行文件并偶然发现了示例 muxing.c在 ffmpeg 示例中。
这让我找到了我错过的步骤:在数据包上设置流索引，并重新调整时间:

av_packet_rescale_ts(pkt, context->time_base, stream->time_base);
pkt->stream_index = stream->index;
int ret = av_interleaved_write_frame(avfctx, pkt);