Ffmpeg ffserver保留原始流大小

希望你们能帮助我，因为我陷入了困境，自己找不到解决这个问题的办法。 我正在尝试使用ffmpeg+ffserver将视频从网络摄像头传输给用户。但我面临一个问题：

ffmpeg gets stream from camera and pushes it to feed of ffserver:
ffmpeg -rtsp_transport tcp -i rtsp://admin:admin@192.168.10.76:80 -y -vcodec libvpx http://127.0.0.1:8090/1.ffm

ffserver流选项：

<Stream 1.webm>
Feed 1.ffm
Format webm
NoAudio
#VideoCodec libvpx
#VideoSize 480x320
VideoFrameRate 24
AVOptionVideo flags +global_header
AVOptionVideo cpu-used 0
AVOptionVideo qmin 1
AVOptionVideo qmax 31
AVOptionVideo quality good
PreRoll 0
StartSendOnKey
VideoBitRate 128
</Stream>

但有时视频内容不会被发送

如果我取消注释VideoSize选项-同样的问题，但与默认视频大小相比，成功的请求要少得多

ffserver日志看起来正常，没有错误。但正如您所看到的，有时它只向客户端发送头：

Thu Nov 27 12:49:11 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
Thu Nov 27 12:49:25 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
Thu Nov 27 12:49:36 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
Thu Nov 27 12:50:52 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
Thu Nov 27 12:53:54 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
Thu Nov 27 13:30:19 2014 127.0.0.1 - - [GET] "/1.ffm HTTP/1.1" 200 4175
Thu Nov 27 13:30:34 2014 127.0.0.1 - - [GET] "/1.webm HTTP/1.1" 200 385731
Thu Nov 27 13:30:34 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 458752
Thu Nov 27 13:30:36 2014 127.0.0.1 - - [GET] "/1.ffm HTTP/1.1" 200 4175
Thu Nov 27 13:30:58 2014 127.0.0.1 - - [GET] "/1.webm HTTP/1.1" 200 493
Thu Nov 27 13:30:58 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 622592

有人知道会是什么吗？实际上，我需要保存流的原始视频大小。我尝试使用命令（传递与ffserver流中相同的参数）用ffmpeg覆盖ffserver流选项：

但在上午我仍然有错误消息“输出文件是空的，没有编码”。以下是ffmpeg的输出：

ffmpeg version 2.4.2 Copyright (c) 2000-2014 the FFmpeg developers
  built on Oct  6 2014 17:33:05 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  configuration: --prefix=/opt/ffmpeg --libdir=/opt/ffmpeg/lib/ --enable-shared --enable-avresample --disable-stripping --enable-gpl --enable-version3 --enable-runtime-cpudetect --build-suffix=.ffmpeg --enable-postproc --enable-x11grab --enable-libcdio --enable-vaapi --enable-vdpau --enable-bzlib --enable-gnutls --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libfaac --enable-libvo-aacenc --enable-nonfree --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfdk_aac --enable-libopus --enable-pthreads --enable-zlib --enable-libvpx --enable-libfreetype --enable-libpulse --enable-debug=3
  libavutil      54.  7.100 / 54.  7.100
  libavcodec     56.  1.100 / 56.  1.100
  libavformat    56.  4.101 / 56.  4.101
  libavdevice    56.  0.100 / 56.  0.100
  libavfilter     5.  1.100 /  5.  1.100
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  0.100 /  3.  0.100
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  0.100 / 53.  0.100
Guessed Channel Layout for  Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://admin:admin@192.168.10.76:80':
  Metadata:
    title           : RTSP Session/2.0
  Duration: N/A, start: 0.000000, bitrate: 128 kb/s
    Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709), 1280x720 [SAR 1:1 DAR 16:9], 25 fps, 100 tbr, 90k tbn, 50 tbc
    Stream #0:1: Audio: pcm_alaw, 16000 Hz, 1 channels, s16, 128 kb/s
[swscaler @ 0x197f7a0] deprecated pixel format used, make sure you did set range correctly
[libvpx @ 0x1a0c080] Bitrate 128 is extremely low, maybe you mean 128k
[libvpx @ 0x1a0c080] v1.3.0
The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s
Output #0, webm, to 'http://127.0.0.1:8090/1.ffm':
  Metadata:
    title           : RTSP Session/2.0
    encoder         : Lavf56.4.101
    Stream #0:0: Video: vp8 (libvpx), yuv420p, 480x320 [SAR 32:27 DAR 16:9], q=1-31, 0 kb/s, 24 fps, 1k tbn, 24 tbc
    Metadata:
      encoder         : Lavc56.1.100 libvpx
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> vp8 (libvpx))
Press [q] to stop, [?] for help
frame=   33 fps= 22 q=0.0 size=       0kB time=00:00:00.00 bitrate=N/A dup=0 droframe=   43 fps= 22 q=0.0 Lsize=       0kB time=00:00:00.00 bitrate=N/A dup=0 drop=1    
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
Received signal 2: terminating.

---

提前谢谢。

据我所知，您想：

从IP摄像机获取流

使用ffmpeg和保留原始视频分辨率的vp8编解码器对流进行编码

将其发送到ffserver

再流一点

为此，您需要：

创建ffserver.config文件

启动ffserver

启动ffmpeg控制台脚本

你的情况需要：

在配置文件中取消对视频编解码器的注释并删除比特率

启动服务器后，在ffmpeg命令中键入：

ffmpeg-irtsp://admin:admin@192.168.10.76:80-c:v libvpx-b:v 800-an htt\u p://xxx.xx.xxx.xxx:SERVER\u HTTP\u PORT/feed1.ffm

[从http中删除“\u1”）

请注意，如果您的相机支持一种以上的分辨率，如果您不指定需要哪种分辨率，ffmpeg将选择最差的分辨率。您可以通过键入

ffmpeg-f v4l2-list\u formats all-i/dev/video0

来检查网络摄像头中支持的分辨率。如果将

/dev/video0

替换为

rtsp://admin:admin@192.168.10.76:80

---

还要记住，视频比特率（-b:v）取决于帧大小。

您想要精确实现什么？保留原始编解码器、分辨率、视频比特率？我想用vp8编码到webm容器，但也必须保存原始分辨率。希望它将是相同的ffserver流，但它应该流原始视频大小（不是160x128）。

ffmpeg -re -override_ffserver -rtsp_transport tcp -i rtsp://admin:admin@192.168.10.76:80 -an -r 24 -qmin 1 -qmax 31 -cpu-used 0 -quality good -flags:v +global_header -b:v 128 -vcodec libvpx -f webm -y http://127.0.0.1:8090/1.ffm

ffmpeg version 2.4.2 Copyright (c) 2000-2014 the FFmpeg developers
  built on Oct  6 2014 17:33:05 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  configuration: --prefix=/opt/ffmpeg --libdir=/opt/ffmpeg/lib/ --enable-shared --enable-avresample --disable-stripping --enable-gpl --enable-version3 --enable-runtime-cpudetect --build-suffix=.ffmpeg --enable-postproc --enable-x11grab --enable-libcdio --enable-vaapi --enable-vdpau --enable-bzlib --enable-gnutls --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libfaac --enable-libvo-aacenc --enable-nonfree --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfdk_aac --enable-libopus --enable-pthreads --enable-zlib --enable-libvpx --enable-libfreetype --enable-libpulse --enable-debug=3
  libavutil      54.  7.100 / 54.  7.100
  libavcodec     56.  1.100 / 56.  1.100
  libavformat    56.  4.101 / 56.  4.101
  libavdevice    56.  0.100 / 56.  0.100
  libavfilter     5.  1.100 /  5.  1.100
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  0.100 /  3.  0.100
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  0.100 / 53.  0.100
Guessed Channel Layout for  Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://admin:admin@192.168.10.76:80':
  Metadata:
    title           : RTSP Session/2.0
  Duration: N/A, start: 0.000000, bitrate: 128 kb/s
    Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709), 1280x720 [SAR 1:1 DAR 16:9], 25 fps, 100 tbr, 90k tbn, 50 tbc
    Stream #0:1: Audio: pcm_alaw, 16000 Hz, 1 channels, s16, 128 kb/s
[swscaler @ 0x197f7a0] deprecated pixel format used, make sure you did set range correctly
[libvpx @ 0x1a0c080] Bitrate 128 is extremely low, maybe you mean 128k
[libvpx @ 0x1a0c080] v1.3.0
The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s
Output #0, webm, to 'http://127.0.0.1:8090/1.ffm':
  Metadata:
    title           : RTSP Session/2.0
    encoder         : Lavf56.4.101
    Stream #0:0: Video: vp8 (libvpx), yuv420p, 480x320 [SAR 32:27 DAR 16:9], q=1-31, 0 kb/s, 24 fps, 1k tbn, 24 tbc
    Metadata:
      encoder         : Lavc56.1.100 libvpx
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> vp8 (libvpx))
Press [q] to stop, [?] for help
frame=   33 fps= 22 q=0.0 size=       0kB time=00:00:00.00 bitrate=N/A dup=0 droframe=   43 fps= 22 q=0.0 Lsize=       0kB time=00:00:00.00 bitrate=N/A dup=0 drop=1    
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
Received signal 2: terminating.