<div dir="auto">Hi,<div dir="auto"> </div><div dir="auto">The latest development binary release of gstreamer ships the Rust plugins. You don't need to build anything:</div><div dir="auto"> </div><div dir="auto"><a href="https://gstreamer.freedesktop.org/data/pkg/windows/1.21.2/">https://gstreamer.freedesktop.org/data/pkg/windows/1.21.2/</a></div><div dir="auto"> </div><div dir="auto">I'd recommend the MSVC binaries unless your app uses MinGW.</div></div> <div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, 15 Nov, 2022, 11:33 Jun Xiao via gstreamer-devel, <<a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>> wrote: </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div dir="ltr">Hi Mathieu,<div> </div><div>Thanks for your help that I can run webrtcsink successfully on ubuntu 22.04.</div><div>Now when I try to build gstreamer/gst-plugins-rs on windows 10, say I use the command `meson -Dbad=enabled -Dgst-plugins-bad:openh264=disabled -Dgst-plugins-bad:webrtc=enabled -Dgst-plugins-bad:srtp=enabled -Dgst-plugins-ugly:x264=enabled -Dgpl=enabled builddir`, I met a couple of problems:</div><div>- If I build in x86 Native Tools Command Prompt for VS 2019, I cannot install some lib dependencies, e.g. libsrtp. I guess it's not feasible to install those libs in the Command Prompt.</div><div>- If I build in mingw64, still cannot resolve libsrtp issue.</div><div> </div><div>Do you have any suggestion to build gstreamer/gst-plugins-rs on windows? which build environment should i go after?</div><div> </div><div>Thanks,</div><div>Jun</div></div></div> <div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, Nov 10, 2022 at 11:15 PM Mathieu Duponchelle <<a href="mailto:mathieu@centricular.com" target="_blank" rel="noreferrer">mathieu@centricular.com</a>> wrote: </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">Good :) On Thu, 2022-11-10 at 14:12 +0800, Jun Xiao via gstreamer-devel wrote: > Thanks Mathieu! It works after I built a gstreamer with x264enc. > > On Thu, Nov 10, 2022 at 1:07 AM Mathieu Duponchelle > <<a href="mailto:mathieu@centricular.com" target="_blank" rel="noreferrer">mathieu@centricular.com</a>> wrote: > > Thanks. It looks like the only encoder available on your system is > > opusenc. This is not enough, make sure to at least have x264enc / > > vp8enc / x265enc / vp9enc. > > > > On Wed, 2022-11-09 at 20:52 +0800, Jun Xiao via gstreamer-devel > > wrote: > > > Hi Mathieu, > > > > > > Following is the log with the patch, if you need more debug > > > patches > > > let me know. > > > > > > gst-plugins-rs$ GST_DEBUG=webrtcsink:6 gst-launch-1.0 webrtcsink > > > name=ws videotestsrc ! ws. audiotestsrc ! ws. > > > Setting pipeline to PAUSED ... > > > 0:00:00.124050809 10519 0x55cdb072a190 DEBUG > > > webrtcsink > > > net/webrtc/src/webrtcsink/imp.rs:1192:gstrswebrtc::webrtcsink::im > > > p:<w > > > s> > > > preparing > > > Pipeline is live and does not need PREROLL ... > > > Pipeline is PREROLLED ... > > > Setting pipeline to PLAYING ... > > > 0:00:00.125241154 10519 0x55cdb0973180 INFO > > > webrtcsink > > > net/webrtc/src/webrtcsink/imp.rs:2281:gstrswebrtc::webrtcsink::im > > > p:<w > > > s:video_0> > > > Received caps event Caps(Event { ptr: 0x55cdb0979560, type: > > > "caps", > > > seqnum: Seqnum(45), structure: Some(GstEventCaps { caps: > > > (GstCaps) > > > video/x-raw, format=(string)ABGR64_LE, width=(int)320, > > > height=(int)240, framerate=(fraction)30/1, > > > multiview-mode=(string)mono, pixel-aspect-ratio=(fraction)1/1, > > > interlace-mode=(string)progressive }) }) > > > New clock: GstSystemClock > > > 0:00:00.125881460 10519 0x55cdb09731e0 INFO > > > webrtcsink > > > net/webrtc/src/webrtcsink/imp.rs:2281:gstrswebrtc::webrtcsink::im > > > p:<w > > > s:audio_0> > > > Received caps event Caps(Event { ptr: 0x7fb74c003470, type: > > > "caps", > > > seqnum: Seqnum(55), structure: Some(GstEventCaps { caps: > > > (GstCaps) > > > audio/x-raw, format=(string)S16LE, layout=(string)interleaved, > > > rate=(int)44100, channels=(int)1 }) }) > > > Redistribute latency... > > > 0:00:00.128845016 10519 0x7fb744024290 DEBUG > > > webrtcsink > > > net/webrtc/src/webrtcsink/imp.rs:2218:gstrswebrtc::webrtcsink::im > > > p:<w > > > s> > > > Looked up codecs {96: Codec { encoder: ElementFactory { inner: > > > TypedObjectRef { inner: 0x55cdb059e2a0, type: GstElementFactory } > > > }, > > > payloader: ElementFactory { inner: TypedObjectRef { inner: > > > 0x55cdb06cab40, type: GstElementFactory } }, caps: > > > Caps(audio/x-opus(memory:SystemMemory)), payload: 96 }} > > > 0:00:00.129043362 10519 0x7fb744024290 DEBUG > > > webrtcsink > > > net/webrtc/src/webrtcsink/imp.rs:2086:gstrswebrtc::webrtcsink::im > > > p:<w > > > s> > > > Running discovery pipeline for caps audio/x-raw, > > > format=(string)S16LE, > > > layout=(string)interleaved, rate=(int)44100, channels=(int)1 with > > > codec Codec { encoder: ElementFactory { inner: TypedObjectRef { > > > inner: > > > 0x55cdb059e2a0, type: GstElementFactory } }, payloader: > > > ElementFactory > > > { inner: TypedObjectRef { inner: 0x55cdb06cab40, type: > > > GstElementFactory } }, caps: Caps(audio/x- > > > opus(memory:SystemMemory)), > > > payload: 96 } > > > Redistribute latency... > > > 0:00:00.137439122 10519 0x7fb744024290 DEBUG > > > webrtcsink > > > net/webrtc/src/webrtcsink/imp.rs:2149:gstrswebrtc::webrtcsink::im > > > p:<w > > > s> > > > Codec discovery pipeline for caps audio/x-raw, > > > format=(string)S16LE, > > > layout=(string)interleaved, rate=(int)44100, channels=(int)1 with > > > codec Codec { encoder: ElementFactory { inner: TypedObjectRef { > > > inner: > > > 0x55cdb059e2a0, type: GstElementFactory } }, payloader: > > > ElementFactory > > > { inner: TypedObjectRef { inner: 0x55cdb06cab40, type: > > > GstElementFactory } }, caps: Caps(audio/x- > > > opus(memory:SystemMemory)), > > > payload: 96 } succeeded: application/x-rtp, media=(string)audio, > > > clock-rate=(int)48000, encoding-name=(string)OPUS, > > > sprop-stereo=(string)0, encoding-params=(string)2, > > > sprop-maxcapturerate=(string)48000, payload=(int)96, > > > extmap- > > > 1=(string)<a href="http://www.ietf.org/id/draft-holmer-rmcat-transport-" rel="noreferrer noreferrer" target="_blank">http://www.ietf.org/id/draft-holmer-rmcat-transport-</a> > > > wide-cc-extensions-01; > > > 0:00:00.138044111 10519 0x7fb744024290 ERROR > > > webrtcsink > > > net/webrtc/src/webrtcsink/imp.rs:2319:gstrswebrtc::webrtcsink::im > > > p:<w > > > s> > > > error: No caps found for stream video_0 > > > ERROR: from element /GstPipeline:pipeline0/GstWebRTCSink:ws: > > > There is > > > no codec present that can handle the stream's type. > > > Additional debug info: > > > net/webrtc/src/webrtcsink/<a href="http://imp.rs" rel="noreferrer noreferrer" target="_blank">imp.rs</a>(2320): > > > gstrswebrtc::webrtcsink::imp > > > (): /GstPipeline:pipeline0/GstWebRTCSink:ws: > > > Failed to look up output caps: No caps found for stream video_0 > > > Execution ended after 0:00:00.013297770 > > > Setting pipeline to NULL ... > > > 0:00:00.138701195 10519 0x55cdb072a190 INFO > > > webrtcsink > > > net/webrtc/src/webrtcsink/imp.rs:1207:gstrswebrtc::webrtcsink::im > > > p:<w > > > s> > > > unpreparing > > > Freeing pipeline ... > > > > > > Thanks, > > > Jun > > > > > > On Wed, Nov 9, 2022 at 8:05 PM Mathieu Duponchelle > > > <<a href="mailto:mathieu@centricular.com" target="_blank" rel="noreferrer">mathieu@centricular.com</a>> wrote: > > > > > > > > Hey, unfortunately webrtcsink needs a few more logs on the > > > > codec > > > > discovery code path to be useful her, can you apply > > > > <a href="https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/972/" rel="noreferrer noreferrer" target="_blank">https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/972/</a> > > > > and try again? > > > > > > > > On Wed, 2022-11-09 at 00:11 +0800, Jun Xiao via gstreamer-devel > > > > wrote: > > > > > Thanks Mathieu, > > > > > > > > > > Following is the message with debug log: > > > > > gst-plugins-rs$ GST_DEBUG=webrtcsink:6 gst-launch-1.0 > > > > > webrtcsink > > > > > name=ws videotestsrc ! ws. audiotestsrc ! ws. > > > > > Setting pipeline to PAUSED ... > > > > > 0:00:00.017416246 7466 0x55655b56c2d0 DEBUG > > > > > webrtcsink > > > > > net/webrtc/src/webrtcsink/imp.rs:1192:gstrswebrtc::webrtcsink > > > > > ::im > > > > > p:<w > > > > > s> > > > > > preparing > > > > > Pipeline is live and does not need PREROLL ... > > > > > Pipeline is PREROLLED ... > > > > > Setting pipeline to PLAYING ... > > > > > New clock: GstSystemClock > > > > > 0:00:00.018513516 7466 0x55655b5fc580 INFO > > > > > webrtcsink > > > > > net/webrtc/src/webrtcsink/imp.rs:2260:gstrswebrtc::webrtcsink > > > > > ::im > > > > > p:<w > > > > > s:video_0> > > > > > Received caps event Caps(Event { ptr: 0x55655b603150, type: > > > > > "caps", > > > > > seqnum: Seqnum(50), structure: Some(GstEventCaps { caps: > > > > > (GstCaps) > > > > > video/x-raw, format=(string)ABGR64_LE, width=(int)320, > > > > > height=(int)240, framerate=(fraction)30/1, > > > > > multiview-mode=(string)mono, pixel-aspect- > > > > > ratio=(fraction)1/1, > > > > > interlace-mode=(string)progressive }) }) > > > > > Redistribute latency... > > > > > 0:00:00.028336475 7466 0x55655b5fc5e0 INFO > > > > > webrtcsink > > > > > net/webrtc/src/webrtcsink/imp.rs:2260:gstrswebrtc::webrtcsink > > > > > ::im > > > > > p:<w > > > > > s:audio_0> > > > > > Received caps event Caps(Event { ptr: 0x7f77680034a0, type: > > > > > "caps", > > > > > seqnum: Seqnum(61), structure: Some(GstEventCaps { caps: > > > > > (GstCaps) > > > > > audio/x-raw, format=(string)S16LE, > > > > > layout=(string)interleaved, > > > > > rate=(int)44100, channels=(int)1 }) }) > > > > > Redistribute latency... > > > > > 0:00:00.095827060 7466 0x7f776c05aea0 ERROR > > > > > webrtcsink > > > > > net/webrtc/src/webrtcsink/imp.rs:2298:gstrswebrtc::webrtcsink > > > > > ::im > > > > > p:<w > > > > > s> > > > > > error: No caps found for stream video_0 > > > > > ERROR: from element /GstPipeline:pipeline0/GstWebRTCSink:ws: > > > > > There is > > > > > no codec present that can handle the stream's type. > > > > > Additional debug info: > > > > > net/webrtc/src/webrtcsink/<a href="http://imp.rs" rel="noreferrer noreferrer" target="_blank">imp.rs</a>(2299): > > > > > gstrswebrtc::webrtcsink::imp > > > > > (): /GstPipeline:pipeline0/GstWebRTCSink:ws: > > > > > Failed to look up output caps: No caps found for stream > > > > > video_0 > > > > > Execution ended after 0:00:00.077864232 > > > > > Setting pipeline to NULL ... > > > > > 0:00:00.101370102 7466 0x55655b56c2d0 INFO > > > > > webrtcsink > > > > > net/webrtc/src/webrtcsink/imp.rs:1207:gstrswebrtc::webrtcsink > > > > > ::im > > > > > p:<w > > > > > s> > > > > > unpreparing > > > > > Freeing pipeline ... > > > > > > > > > > On Tue, Nov 8, 2022 at 11:59 PM Mathieu Duponchelle > > > > > <<a href="mailto:mathieu@centricular.com" target="_blank" rel="noreferrer">mathieu@centricular.com</a>> wrote: > > > > > > > > > > > > Do you have video encoders available on your system? > > > > > > Perhaps > > > > > > try to > > > > > > run > > > > > > with GST_DEBUG=webrtcsink:6 to get more information. > > > > > > > > > > > > On Tue, 2022-11-08 at 23:54 +0800, Jun Xiao via gstreamer- > > > > > > devel > > > > > > wrote: > > > > > > > Hi, > > > > > > > > > > > > > > When I tried to run webrtc rust sample in ubuntu 22.04 VM > > > > > > > by > > > > > > > following > > > > > > > <a href="https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/tree/main/net/webrtc" rel="noreferrer noreferrer" target="_blank">https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/tree/main/net/webrtc</a> > > > > > > > , > > > > > > > I met following problem: > > > > > > > > > > > > > > $gst-launch-1.0 webrtcsink name=ws videotestsrc ! ws. > > > > > > > audiotestsrc ! > > > > > > > ws. > > > > > > > Setting pipeline to PAUSED ... > > > > > > > Pipeline is live and does not need PREROLL ... > > > > > > > Pipeline is PREROLLED ... > > > > > > > Setting pipeline to PLAYING ... > > > > > > > New clock: GstSystemClock > > > > > > > Redistribute latency... > > > > > > > Redistribute latency... > > > > > > > 0:00:00.052590872 7339 0x7fb29006b8a0 ERROR > > > > > > > webrtcsink > > > > > > > net/webrtc/src/webrtcsink/imp.rs:2298:gstrswebrtc::webrtc > > > > > > > sink > > > > > > > ::im > > > > > > > p:<w > > > > > > > s> > > > > > > > error: No caps found for stream video_0 > > > > > > > ERROR: from element > > > > > > > /GstPipeline:pipeline0/GstWebRTCSink:ws: > > > > > > > There is > > > > > > > no codec present that can handle the stream's type. > > > > > > > Additional debug info: > > > > > > > net/webrtc/src/webrtcsink/<a href="http://imp.rs" rel="noreferrer noreferrer" target="_blank">imp.rs</a>(2299): > > > > > > > gstrswebrtc::webrtcsink::imp > > > > > > > (): /GstPipeline:pipeline0/GstWebRTCSink:ws: > > > > > > > Failed to look up output caps: No caps found for stream > > > > > > > video_0 > > > > > > > Execution ended after 0:00:00.036349323 > > > > > > > Setting pipeline to NULL ... > > > > > > > Freeing pipeline ... > > > > > > > > > > > > > > Could anyone share any suggestions? > > > > > > > Thanks, > > > > > > > Jun </blockquote></div> </blockquote></div>