将 WEBM 转码为 RTMP

Question

总结

我的目标是从浏览器获取网络摄像头流并将其输入名为 Restreamer 的程序中，该程序接收 RTMP 流。

我推断浏览器Recording API生成可以保存为WEBM的Blob。

为了将 WEBM 内容流式传输到 Restreamer，我尝试使用 FFmpeg。我读过强制 FLV 格式是必需的，但除此之外所有的论点对我来说都是希腊语。我一直无法找到有人尝试从 WEBM 转到 RTMP 的类似主题。我找到了反其道而行之的例子，但事实证明反转 FFmpeg 命令没有效果。

保存到FLV效果很好。使用以下命令，我可以将 WEBM 文件转码为 FLV 并在 VLC 上播放：

ffmpeg -i ~/big-buck-bunny_trailer.webm -f flv out.flv

但是，如果不是输出到文件，而是将其传递给 RTMP，我会得到以下输出：

ffmpeg -i ~/Downloads/big-buck-bunny_trailer.webm -f flv "rtmp://example.com/live"

ffmpeg version 5.0.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 12 (GCC)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection' --extra-ldflags='-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 ' --extra-cflags=' -I/usr/include/rav1e' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --enable-chromaprint --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libaom --enable-libdav1d --enable-libass --enable-libbluray --enable-libbs2b --enable-libcdio --enable-libdrm --enable-libjack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libmysofa --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librav1e --enable-librtmp --enable-librubberband --enable-libsmbclient --enable-version3 --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-version3 --enable-vapoursynth --enable-libvpx --enable-vulkan --enable-libglslang --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libxml2 --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-avfilter --enable-libmodplug --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-lto --enable-libmfx --enable-runtime-cpudetect
  libavutil      57. 17.100 / 57. 17.100
  libavcodec     59. 18.100 / 59. 18.100
  libavformat    59. 16.100 / 59. 16.100
  libavdevice    59.  4.100 / 59.  4.100
  libavfilter     8. 24.100 /  8. 24.100
  libswscale      6.  4.100 /  6.  4.100
  libswresample   4.  3.100 /  4.  3.100
  libpostproc    56.  3.100 / 56.  3.100
Input #0, matroska,webm, from '/home/kyjus25/big-buck-bunny_trailer.webm':
  Metadata:
    encoder         : http://sourceforge.net/projects/yamka
    creation_time   : 2010-05-20T08:21:12.000000Z
  Duration: 00:00:32.48, start: 0.000000, bitrate: 533 kb/s
  Stream #0:0(eng): Video: vp8, yuv420p(progressive), 640x360, SAR 1:1 DAR 16:9, 25 fps, 25 tbr, 1k tbn (default)
  Stream #0:1(eng): Audio: vorbis, 44100 Hz, mono, fltp (default)
HandShake: client signature does not match!
Stream mapping:
  Stream #0:0 -> #0:0 (vp8 (native) -> flv1 (flv))
  Stream #0:1 -> #0:1 (vorbis (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, flv, to 'rtmp://example.com/live':
  Metadata:
    encoder         : Lavf59.16.100
  Stream #0:0(eng): Video: flv1 ([2][0][0][0] / 0x0002), yuv420p(tv, bt470bg/unknown/unknown, progressive), 640x360 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 1k tbn (default)
    Metadata:
      encoder         : Lavc59.18.100 flv
    Side data:
      cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: N/A
  Stream #0:1(eng): Audio: mp3 ([2][0][0][0] / 0x0002), 44100 Hz, mono, fltp (default)
    Metadata:
      encoder         : Lavc59.18.100 libmp3lame
WriteN, RTMP send error 32 (136 bytes)7kB time=00:00:00.39 bitrate= 136.7kbits/s speed=71.2x    
WriteN, RTMP send error 32 (35 bytes)
WriteN, RTMP send error 9 (42 bytes)
av_interleaved_write_frame(): Operation not permitted
    Last message repeated 1 times
[flv @ 0x55d0dd0af700] Failed to update header with correct duration.
[flv @ 0x55d0dd0af700] Failed to update header with correct filesize.
Error writing trailer of rtmp://example.com/live: Operation not permitted
frame=   53 fps=0.0 q=4.3 Lsize=     146kB time=00:00:02.45 bitrate= 486.8kbits/s speed=42.8x    
video:128kB audio:19kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Error closing file rtmp://example.com/live: Operation not permitted
Conversion failed!

这里有几个有趣的兔子洞可供追踪，但在追踪了所有这些之后我什么也没发现。

•

HandShake: client signature does not match!

:
我认为这更多是警告而不是错误，因为我是从“WEBM -> FLV”而不是更传统的“MP4 -> FLV”开始。

•

av_interleaved_write_frame(): Operation not permitted

:
我在这方面发现了几个问题。其中一个称其为“存储问题”，另一个称其为“文件权限问题”。我有足够的磁盘空间，并尝试将输入文件设置为 777 权限。然而，我发现的文件权限问题的示例都涉及输出到文件而不是 RTMP IP。 • Failed to update header with correct duration:

我发现的建议

是将

-flvflags no_duration_filesize
添加到命令中，这确实会抑制“更新失败...”错误，但不能解决总体问题。 我尝试过的事情• 多格式转码

MP4 到 RTMP 工作正常：

ffmpeg -i ~/Downloads/big-buck-bunny_trailer.mp4 -f flv "rtmp://example.com/live"

理论上，我可以将 WEBM 流式传输到文件，将其转码为 MP4 文件，然后将其转码为 FLV/RTMP。听起来很糟糕。

• 支付服务费用（Wowza、Flashphoner、api.video 等）

不幸的是，这正是我想要避免的。

• WebRTC 转 RTMP？

WebRTC 似乎是一种点对点连接，并且不能很好地适应服务器/客户端场景。

• WebRTC 到其他摄取格式

Restreamer 还支持 HLS、DASH、RTP、RTSP、RTMP 和 SRT 的传入流。然而，这些似乎都是通过 IP URL 公开的网络源的示例。我不确定 FFmpeg 能做到这一点。

• 使用 ffmpeg-wasm 代替 CLI

可用

here

，我想也许通过使用浏览器实现我可能会得到不同的结果。但不是。甚至控制台都没有错误。

• 来自 OBS 的流媒体 郑重声明，是的，我尝试过从 OBS 进行流式传输，而不是通过 FFmpeg，并且 Restreamer 平台本身确实可以正常使用。我经常用它。

帖子脚本

我在网上找不到任何相关的解决方案。令我震惊的是，之前 1000 次都没有解决来自浏览器网络摄像头的流传输问题。这与 7 年前提出的

问题

有关，但尚未解决，并且 Flash

不再是一个选项

。 推荐 • 添加

-c:v libx264 -flags:v +global_header -c:a aac -ac 2

:

ffmpeg version 5.0.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 12 (GCC)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection' --extra-ldflags='-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 ' --extra-cflags=' -I/usr/include/rav1e' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --enable-chromaprint --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libaom --enable-libdav1d --enable-libass --enable-libbluray --enable-libbs2b --enable-libcdio --enable-libdrm --enable-libjack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libmysofa --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librav1e --enable-librtmp --enable-librubberband --enable-libsmbclient --enable-version3 --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-version3 --enable-vapoursynth --enable-libvpx --enable-vulkan --enable-libglslang --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libxml2 --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-avfilter --enable-libmodplug --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-lto --enable-libmfx --enable-runtime-cpudetect
  libavutil      57. 17.100 / 57. 17.100
  libavcodec     59. 18.100 / 59. 18.100
  libavformat    59. 16.100 / 59. 16.100
  libavdevice    59.  4.100 / 59.  4.100
  libavfilter     8. 24.100 /  8. 24.100
  libswscale      6.  4.100 /  6.  4.100
  libswresample   4.  3.100 /  4.  3.100
  libpostproc    56.  3.100 / 56.  3.100
Input #0, matroska,webm, from '/home/kyjus25/big-buck-bunny_trailer.webm':
  Metadata:
    encoder         : http://sourceforge.net/projects/yamka
    creation_time   : 2010-05-20T08:21:12.000000Z
  Duration: 00:00:32.48, start: 0.000000, bitrate: 533 kb/s
  Stream #0:0(eng): Video: vp8, yuv420p(progressive), 640x360, SAR 1:1 DAR 16:9, 25 fps, 25 tbr, 1k tbn (default)
  Stream #0:1(eng): Audio: vorbis, 44100 Hz, mono, fltp (default)
HandShake: client signature does not match!
Stream mapping:
  Stream #0:0 -> #0:0 (vp8 (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (vorbis (native) -> aac (native))
Press [q] to stop, [?] for help
[libx264 @ 0x561564271fc0] using SAR=1/1
[libx264 @ 0x561564271fc0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0x561564271fc0] profile High, level 3.0, 4:2:0, 8-bit
[libx264 @ 0x561564271fc0] 264 - core 163 r3060 5db6aa6 - H.264/MPEG-4 AVC codec - Copyleft 2003-2021 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=11 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, flv, to 'rtmp://example.com/live':
  Metadata:
    encoder         : Lavf59.16.100
  Stream #0:0(eng): Video: h264 ([7][0][0][0] / 0x0007), yuv420p(tv, bt470bg/unknown/unknown, progressive), 640x360 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 1k tbn (default)
    Metadata:
      encoder         : Lavc59.18.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
  Stream #0:1(eng): Audio: aac (LC) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      encoder         : Lavc59.18.100 aac
Larger timestamp than 24-bit: 0xffffff77kB time=00:00:30.18 bitrate= 460.0kbits/s speed=19.7x    
[flv @ 0x56156425e440] Failed to update header with correct duration.
[flv @ 0x56156425e440] Failed to update header with correct filesize.
frame=  812 fps=475 q=-1.0 Lsize=    1901kB time=00:00:32.52 bitrate= 479.0kbits/s speed=  19x    
video:1354kB audio:508kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.123872%
[libx264 @ 0x561564271fc0] frame I:21    Avg QP:15.05  size:  8839
[libx264 @ 0x561564271fc0] frame P:293   Avg QP:20.78  size:  3087
[libx264 @ 0x561564271fc0] frame B:498   Avg QP:22.20  size:   593
[libx264 @ 0x561564271fc0] consecutive B-frames: 14.7%  7.6%  9.2% 68.5%
[libx264 @ 0x561564271fc0] mb I  I16..4: 42.6% 41.9% 15.5%
[libx264 @ 0x561564271fc0] mb P  I16..4:  4.0%  8.0%  0.9%  P16..4: 22.7%  8.0%  4.0%  0.0%  0.0%    skip:52.4%
[libx264 @ 0x561564271fc0] mb B  I16..4:  1.4%  1.5%  0.2%  B16..8: 12.0%  1.0%  0.2%  direct: 3.3%  skip:80.5%  L0:44.1% L1:47.1% BI: 8.8%
[libx264 @ 0x561564271fc0] 8x8 transform intra:53.6% inter:57.0%
[libx264 @ 0x561564271fc0] coded y,uvDC,uvAC intra: 35.1% 37.7% 11.2% inter: 7.8% 9.5% 2.4%
[libx264 @ 0x561564271fc0] i16 v,h,dc,p: 56% 21% 14%  9%
[libx264 @ 0x561564271fc0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 32% 26% 21%  3%  3%  4%  4%  3%  4%
[libx264 @ 0x561564271fc0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 27% 21% 18%  4%  7%  7%  6%  5%  4%
[libx264 @ 0x561564271fc0] i8c dc,h,v,p: 62% 21% 15%  2%
[libx264 @ 0x561564271fc0] Weighted P-Frames: Y:20.8% UV:19.5%
[libx264 @ 0x561564271fc0] ref P L0: 70.3% 17.7%  9.0%  2.9%  0.0%
[libx264 @ 0x561564271fc0] ref B L0: 89.3%  8.9%  1.9%
[libx264 @ 0x561564271fc0] ref B L1: 96.4%  3.6%
[libx264 @ 0x561564271fc0] kb/s:341.30
[aac @ 0x561564223140] Qavg: 952.636

看似成功完成，但速度相当快。日志输出一个新的

Larger timestamp than 24-bit: 0xffffff77kB

能够通过使用使其工作：

Answer 1

ffmpeg -re -i sample.webm -c:v h264 -c:a aac -f flv "rtmp://example.com/live"

-re

对于使其实时流式传输至关重要，否则@Gyan的建议也会起作用！

你好，我也尝试将网络摄像头视频发送到 rtmplink 但没有成功。尝试使用 ffmpeg 录制小视频。因此，在录制成功后，我尝试使用 webrtc 中的前端媒体录制并将其数据发送到服务器，我在其中使用 ffmpeg 来使用数据并发送到 rtmp 链接，但无法像 restreamer 那样将数据发送到 rtmp 服务器。

Answer 2

视频：真实}); const mediaRecorder = new MediaRecorder(videoStream, { mimeType: 'video/webm' });

            mediaRecorder.ondataavailable = async (event) => {
                if (event.data.size > 0) {
                    const arrayBuffer = await event.data.arrayBuffer();
                    const uint8Array = new Uint8Array(arrayBuffer);
    
                    // Convert Uint8Array to regular array of integers
                    const bufferArray = Array.from(uint8Array);
    
                    const dataToSend = {
                        buffer: bufferArray,
                        mode: "SendRtmp"
                    };
    
                    ws.send(JSON.stringify(dataToSend));
                }
            };
    
            mediaRecorder.start(1000); // Send data every 1 second
        }
    };

in server.js  

    const SendRtmpVideo = (buffer) => {
    console.log("data",buffer);
    const ffmpegCommand = [
        '-re',
        '-f', 'rawvideo',
        '-pix_fmt', 'yuv420p',
        '-s', '640x480', // Update with your video frame size
        '-i', 'pipe:0',  // Input from stdin
        '-c:v', 'libx264',
        '-loglevel', 'debug', // Set log level to debug
        '-preset', 'ultrafast',
        '-tune', 'zerolatency',
        '-b:v', '3000k',
        '-bufsize', '3000k',
        '-maxrate', '3000k',
        '-g', '60',
        '-c:a', 'aac',
        '-b:a', '128k',
        '-f', 'flv',
        RTMPLINK
    ];
    

    let ffmpeg = spawn('ffmpeg', ffmpegCommand);

    ffmpeg.on('exit', (code, signal) => {
        if (code === 0) {
            console.log('FFmpeg process exited successfully.');
        } else {
            console.error(`FFmpeg process exited with code ${code} and signal ${signal}`);
        }
    });

    ffmpeg.on('error', (error) => {
        console.error('FFmpeg error:', error);
    });

    
    ffmpeg.stdout.on('data', (data) => {
        console.log(`FFmpeg stdout: ${data}`);
    });
    
    ffmpeg.stderr.on('data', (data) => {
        console.error(`FFmpeg stderr: ${data}`);
    });
    ffmpeg.stdin.write(buffer, (err) => {
        if (err) {
            console.error('Error writing to FFmpeg stdin:', err);
        } else {
            console.log('Data written to FFmpeg stdin successfully.');
        }
    });
};

    wss.on('connection', (ws) => {
    console.log('WebSocket connection established.');

    ws.on('message', (data) => {
        try {
            const parsedData = JSON.parse(data);

        if (parsedData && typeof parsedData === 'object') {
            const { buffer, mode } = parsedData;

            if (mode === "SendRtmp" && buffer instanceof Array) {
                // Convert the buffer array to Uint8Array and then to Buffer
                const bufferArray = Buffer.from(new Uint8Array(buffer));
                SendRtmpVideo(bufferArray);
            } else {
                console.log("Received unknown or invalid mode or buffer data");
            }
        } else {
            console.log("Received unknown or invalid JSON data");
        }
    } catch (error) {
        console.error('Error parsing WebSocket message:', error);
    }
});

    ws.on('close', () => {
        console.log('WebSocket connection closed.');
    });
});

// Start the HTTP server
const PORT = 3001;
server.listen(PORT, () => {
    console.log(`WebSocket server listening on port ${PORT}`);
});