Question

从Webrtc提供的浏览器中，我收到一个使用janus网关解密的RTP流。在仅接收到视频rtp数据包后，将其中继到本地多播组以进行测试。

因此，我们假设我在udp端口上收到vp8编码的rtp数据包。我也可以随时请求新的关键帧。

问题管道：

gst-launch-1.0 -v -v -v -v udpsrc multicast-group=224.1.1.1 auto-multicast=true port=1235 ! "application/x-rtp, payload=100, clock-rate=90000" ! rtpvp8depay ! webmmux streamable=true ! filesink location=/tmp/test.webm

产生错误

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = "application/x-rtp\,\     payload\=\(int\)100\,\ clock-rate\=\(int\)90000\,\ media\=\(string\)video\,\ encoding-name\=\(string\)VP8-DRAFT-IETF-01"
/GstPipeline:pipeline0/GstRtpVP8Depay:rtpvp8depay0.GstPad:src: caps = "video/x-vp8\,\ framerate\=\(fraction\)0/1"
/GstPipeline:pipeline0/GstRtpVP8Depay:rtpvp8depay0.GstPad:sink: caps = "application/x-rtp\,\ payload\=\(int\)100\,\ clock-rate\=\(int\)90000\,\ media\=\(string\)video\,\ encoding-name\=\(string\)VP8-DRAFT-IETF-01"
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2933): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-negotiated (-4)
Execution ended after 0:00:00.039571113
Setting pipeline to PAUSED ...
/GstPipeline:pipeline0/GstWebMMux:webmmux0.GstPad:src: caps = video/webm
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

Kurento项目提供了一个名为“vp8parse”的gstreamer插件，解决了这个问题：

gst-launch-1.0 -v -v -v -v udpsrc multicast-group=224.1.1.1 auto-multicast=true port=1235 ! "application/x-rtp, payload=100, clock-rate=90000" ! rtpvp8depay ! vp8parse ! webmmux streamable=true ! filesink location=/tmp/test.webm
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = "application/x-rtp\,\ payload\=\(int\)100\,\ clock-rate\=\(int\)90000\,\ media\=\(string\)video\,\ encoding-name\=\(string\)VP8-DRAFT-IETF-01"
/GstPipeline:pipeline0/GstRtpVP8Depay:rtpvp8depay0.GstPad:src: caps = "video/x-vp8\,\ framerate\=\(fraction\)0/1"
/GstPipeline:pipeline0/KmsVp8Parse:kmsvp8parse0.GstPad:src: caps = "video/x-vp8\,\ framerate\=\(fraction\)0/1"
/GstPipeline:pipeline0/KmsVp8Parse:kmsvp8parse0.GstPad:sink: caps = "video/x-vp8\,\ framerate\=\(fraction\)0/1"
/GstPipeline:pipeline0/GstRtpVP8Depay:rtpvp8depay0.GstPad:sink: caps = "application/x-rtp\,\ payload\=\(int\)100\,\ clock-rate\=\(int\)90000\,\ media\=\(string\)video\,\ encoding-name\=\(string\)VP8-DRAFT-IETF-01"
HERE THE PIPELINE BLOCKS UNTIL A KEYFRAME IS RECEIVED
/GstPipeline:pipeline0/KmsVp8Parse:kmsvp8parse0.GstPad:src: caps = "video/x-vp8\,\ width\=\(int\)640\,\ height\=\(int\)480\,\ framerate\=\(fraction\)10/1"
/GstPipeline:pipeline0/GstWebMMux:webmmux0.GstMatroskamuxPad:video_0: caps = "video/x-vp8\,\ width\=\(int\)640\,\ height\=\(int\)480\,\ framerate\=\(fraction\)10/1"
/GstPipeline:pipeline0/GstWebMMux:webmmux0.GstPad:src: caps = video/webm
/GstPipeline:pipeline0/GstFileSink:filesink0.GstPad:sink: caps = video/webm
/GstPipeline:pipeline0/GstWebMMux:webmmux0.GstPad:src: caps = "video/webm\,\ streamheader\=\(buffer\)\<\ 1a45dfa301000000000000104282857765626d0042878102428581021853806701ffffffffffffff1549a96601000000000000502ad7b1830f42404d809f4753747265616d657220706c7567696e2076657273696f6e20312e342e31005741994753747265616d6572204d6174726f736b61206d7578657200446188062408b80e88c4001654ae6b010000000000003cae0100000000000033d7810183810173c588786b225315e5f279536e86566964656f00e00100000000000008b0820280ba8201e08686565f56503800\ \>"
/GstPipeline:pipeline0/GstFileSink:filesink0.GstPad:sink: caps = "video/webm\,\ streamheader\=\(buffer\)\<\ 1a45dfa301000000000000104282857765626d0042878102428581021853806701ffffffffffffff1549a96601000000000000502ad7b1830f42404d809f4753747265616d657220706c7567696e2076657273696f6e20312e342e31005741994753747265616d6572204d6174726f736b61206d7578657200446188062408b80e88c4001654ae6b010000000000003cae0100000000000033d7810183810173c588786b225315e5f279536e86566964656f00e00100000000000008b0820280ba8201e08686565f56503800\ \>"

看一下vp8parse source似乎这个插件除了管道rtpvp8depay提供给它的接收器的框架之外什么都没做，但也设置了视频宽度，高度和帧速率的src大小。

另一种有效的管道是：

gst-launch-1.0 -v -v -v -v udpsrc multicast-group=224.1.1.1 auto-multicast=true port=1235 ! "application/x-rtp, payload=100, clock-rate=90000" ! rtpvp8depay ! vp8dec ! vp8enc ! webmmux streamable=true ! filesink location=/tmp/test.webm

但使用vp8dec！ vp8enc显然没有多大意义，因为我已经收到了一个vp8编码流。

现在我的问题是如何在不重新编码流而不依赖于vp8parse的情况下解决这个问题？如果没有替代品，似乎我必须使用它，但因为这是一个插件，这是通过标准gstreamer插件包无法提供，我想避免这种情况。是否可以将帽子强制设置为特定的宽度，高度和帧率，以便webmmux不会抱怨？因为我认为这就是第一个管道未经协商的原因。

我尝试使用类似rtpvp8depay的capsfilter！ capsfilter caps =“video / x-vp8，width = 640，height = 480，framerate = 10/1”！ webmmux，但它也没有谈判。

Answer 1

已经修复了上游的bug 747208，但是你系统中Gstreamer的版本可能很旧（Ubuntu 14.04中的1.2.4）仍然受到影响。

作为此类旧版本的解决方法，如果您知道视频的帧大小您可以在depayloader之后使用capssetter元素手动设置depayloader未命中的上限：

gst-launch-1.0 -v \
  udpsrc multicast-group=224.1.1.1 auto-multicast=true port=1235 \
  ! "application/x-rtp, payload=100, clock-rate=90000" \
  ! rtpvp8depay ! capsetter caps="video/x-vp8,width=640,height=480" \
  ! webmmux streamable=true ! filesink location=/tmp/test.webm

Gstreamer：如何在不重新编码的情况下将rtpvp8depay传输到webmmux？

1 个答案: