After upgrade gstreamer 1.20.0 by brew, recently. gst does not work.

path\pkg\mod\github.com\notedit\[email protected]/gst.c:248: undefined reference to `g_memdup2'
collect2.exe: error: ld returned 1 exit status
# github.com/notedit/gst
gst.c: In function 'X_gst_app_src_push_buffer':
gst.c:248:16: warning: implicit declaration of function 'g_memdup2'; did you mean 'g_memdup'? [-Wimplicit-function-declaration]
248 | gpointer p = g_memdup2(buffer, len);
| ^~~~~~~~~
| g_memdup
==========================================================================
github.com\notedit\gst\element_other.go:13:96: cannot use _Ctype_long(duration.Nanoseconds()) (type _Ctype_long) as type _Ctype_longlong in assignmen

It would be extremely helpful to enable support for dumping dot files for pipeline debugging.

您好，我是这样写的，但是不行，麻烦您帮忙看一下

appsrc, _ := gst.ElementFactoryMake("appsrc", "src")
appsrc.SetObject("format", gst.FormatTime)

After finally getting things compiling.... It won't find appsrc etc...

0:00:03.567196000 60583    0x15460be40 WARN     GST_ELEMENT_FACTORY gstelementfactory.c:701:gst_element_factory_make_with_properties: no such element factory "appsrc"!
0:00:03.567229000 60583    0x15460be40 ERROR           GST_PIPELINE gst/parse/grammar.y:851:priv_gst_parse_yyparse: no element "appsrc"
0:00:03.567245000 60583    0x15460be40 WARN     GST_ELEMENT_FACTORY gstelementfactory.c:701:gst_element_factory_make_with_properties: no such element factory "videoconvert"!
0:00:03.567248000 60583    0x15460be40 ERROR           GST_PIPELINE gst/parse/grammar.y:851:priv_gst_parse_yyparse: no element "videoconvert"
0:00:03.567253000 60583    0x15460be40 ERROR           GST_PIPELINE gst/parse/grammar.y:939:priv_gst_parse_yyparse: link has no source [sink=@0x0]
0:00:03.567256000 60583    0x15460be40 WARN     GST_ELEMENT_FACTORY gstelementfactory.c:701:gst_element_factory_make_with_properties: no such element factory "autovideosink"!
0:00:03.567317000 60583    0x15460be40 ERROR           GST_PIPELINE gst/parse/grammar.y:851:priv_gst_parse_yyparse: no element "autovideosink"
0:00:03.567322000 60583    0x15460be40 ERROR           GST_PIPELINE gst/parse/grammar.y:939:priv_gst_parse_yyparse: link has no source [sink=@0x0]
2022/05/24 18:56:44 error was: create pipeline error
panic: pipeline error

But gstreamer can see it normally....

❯ gst-inspect-1.0 --print-plugin-auto-install-info /opt/homebrew/lib/gstreamer-1.0/libgstapp.dylib
element-appsrc
urisource-appsrc
element-appsink
urisink-appsink

❯ pkg-config --cflags gstreamer-plugins-base-1.0
-I/opt/homebrew/Cellar/libffi/3.4.2/include -I/opt/homebrew/Cellar/gst-plugins-base/1.20.2/include/gstreamer-1.0 -I/opt/homebrew/Cellar/gstreamer/1.20.2/include/gstreamer-1.0 -I/opt/homebrew/Cellar/glib/2.72.1/include -I/opt/homebrew/Cellar/glib/2.72.1/include/glib-2.0 -I/opt/homebrew/Cellar/glib/2.72.1/lib/glib-2.0/include -I/opt/homebrew/opt/gettext/include -I/opt/homebrew/Cellar/pcre/8.45/include

I'm not really sure what's up.... any ideas?

Hello,

What is the correct procedure to get one of the examples working on Windows?

I probably missed something silly, but I tried downloading and installing https://gstreamer.freedesktop.org/data/pkg/windows/1.18.2/mingw/gstreamer-1.0-devel-mingw-x86_64-1.18.2.msi then I tried to go build one of the examples:

package main

import (
	"fmt"
	"log"
	"time"

	"github.com/notedit/gst"
)

func main() {
	log.Printf("HELLO")

	pipeline, err := gst.ParseLaunch("appsrc name=mysource format=time is-live=true do-timestamp=true ! videoconvert ! autovideosink")

	if err != nil {
		panic("pipeline error")
	}
	log.Printf("pipe: %v", pipeline)
	videoCap := gst.CapsFromString("video/x-raw,format=RGB,width=320,height=240,bpp=24,depth=24")

	element := pipeline.GetByName("mysource")

	element.SetObject("caps", videoCap)

	pipeline.SetState(gst.StatePlaying)

	i := 0
	for {

		if i > 100 {
			break
		}

		data := make([]byte, 320*240*3)

		err := element.PushBuffer(data)

		if err != nil {
			fmt.Println("push buffer error")
			break
		}

		fmt.Println("push one")
		i++
		time.Sleep(50000000)
	}

	pipeline.SetState(gst.StateNull)

	pipeline = nil
	element = nil
	videoCap = nil
}

After an error about PKG_CONFIG_PATH not set properly, I notice there is a folder with a bunch of .pc files, including C:\gstreamer\1.0\mingw_x86_64\lib\pkgconfig\gstreamer-1.0.pc

So I set my PKG_CONFIG_PATH env var to C:/gstreamer/1.0/mingw_x86_64/lib/pkgconfig and finally go build silently builds an .exe, yet when I run it nothing happens. No error, no console output, it just quits immediately.

Any ideas?

My gstreamer-1.0.pc looks like this:

prefix=C:/gstreamer/1.0/mingw_x86_64
exec_prefix=${prefix}
libexecdir=${prefix}/libexec
libdir=${prefix}/lib
includedir=${prefix}/include/gstreamer-1.0
toolsdir=${exec_prefix}/bin
pluginsdir=${prefix}/lib/gstreamer-1.0
datarootdir=${prefix}/share
datadir=${datarootdir}
girdir=${datadir}/gir-1.0
typelibdir=${libdir}/girepository-1.0
pluginscannerdir=${libexecdir}/gstreamer-1.0

Name: GStreamer
Description: Streaming media framework
Version: 1.18.2
Requires: glib-2.0, gobject-2.0
Requires.private: gmodule-no-export-2.0  
Libs: -L${libdir} -lgstreamer-1.0
Cflags: -I${includedir}

First, thanks a lot for this project, highly appreciated!

I'm building a pipeline which uses an appsrc, is fed raw PCM data and encodes to webm/opus. The result is read back using an appsink.

gst.ParseLaunch("appsrc name=appsrc format=time is-live=true do-timestamp=true block=true max-bytes=128000 ! audio/x-raw,format=S16LE,channels=1,rate=48000,layout=interleaved ! audioconvert ! audioresample ! opusenc bitrate=128000 bitrate-type=vbr ! webmmux ! appsink name=appsink")

As per the GST docs, one should call gst_app_src_end_of_stream() once one is done feeding data:

https://gstreamer.freedesktop.org/documentation/application-development/advanced/pipeline-manipulation.html?gi-language=c

When the last byte is pushed into appsrc, you must call gst_app_src_end_of_stream () to make it send an EOS downstream.

When not sending EOS, the pipeline will hang as appsink.PullSample will block forever.

Something similar has been discussed in issue #11 and the related PR #12 added the required code to make it possible to send an EOS event from Go's gst. The issue cited the code which says that either the above mentioned function or a normal EOS signal could be used.

I tried using the newly possible code (appsrc.SendEvent(gst.NewEosEvent())) and the pipeline would no longer hang. However, it would randomly lose the last samples. As I am working with rather short fragments (~200ms chunks of raw PCM), I'm sometimes losing all the samples except for some 150 bytes of header information. This seems to depend on timing. When adding a short sleep (10ms were sufficient in my tests) between the .PushBuffer() call and the .SendEvent(gst.NewEosEvent()) call, the problem vanished. However, this feels wrong and as I am working on something timing-critical, I want to avoid that sleep call.

After lots of research, I have found that others struggled with such problems as well when using a signal instead of the documented function call:
http://gstreamer-devel.966125.n4.nabble.com/How-to-handle-EOS-from-an-appsrc-to-ensure-all-frames-are-played-out-before-pipeline-closes-td4671225.html#a4671257

Therefore, I assume that the function call and the signal do not behave exactly the same. I suspect that the signal may either propagate through the pipeline immediately after being sent without the pipeline having actually finished all processing. I'm not familiar enough with C or GST to find the actual spot which would explain the difference.

However, a quick test shows that using the function call works. I am no longer losing any samples even when running without the sleep call.

I therefore propose exposing the required function gst_app_src_end_of_stream() to Go. I will submit a PR in a minute.

❯ go run .
# github.com/notedit/gst
../../go/pkg/mod/github.com/notedit/[email protected]/bus.go:33:42: cannot use _Ciconst_GST_CLOCK_TIME_NONE (untyped int constant -1) as _Ctype_ulonglong value in variable declaration (overflows)
../../go/pkg/mod/github.com/notedit/[email protected]/element_other.go:13:96: cannot use _Ctype_long(duration.Nanoseconds()) (value of type _Ctype_long) as type _Ctype_longlong in variable declaration

Based on another comment in another ticket it looks like this is an arm issue....

In python it would be something like:

pipeline.send_event(Gst.Event.new_eos())

i am trying to use it from an appsrc and found this
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/blob/1.16.2/gst-libs/gst/app/gstappsrc.c

 * When the application has finished pushing data into appsrc, it should call
 * gst_app_src_end_of_stream() or emit the end-of-stream action signal. After
 * this call, no more buffers can be pushed into appsrc until a flushing seek
 * occurs or the state of the appsrc has gone through READY.

@notedit any help is highly appreciated

I was going to give this library a try and implement this in Go:
http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html

But quickly ran into the limited callback functions and private callbackID on Element. I'm experimenting with adding things like SetOnNegotiationNeededCallback and SetOnICECandidateCallback, or something a little more general allowing arbitrary callbacks, but I thought I'd register this issue to see if adding something like this upstream would be possible.

I couldnt find any examples of adding caps.

I wish to do something like this

gst-launch-1.0 appsrc name=src ! application/x-rtp, payload=96, encoding-name=OPUS ! rtpopusdepay ! decodebin ! appsink name=sink

How do I add application/x-rtp, payload=96, encoding-name=OPUS to dynamic pipelines?

I have been trying to mix audio and vide using compositor and audio mixer. Here is the situation.
I have pointer for gst elements [appsrc, rtph264depay, avdec_h264, videoconvert]. I have added them to the pipeline. When i try to link them (in order) src.Link(dest) throws me the error gst_element_link_pads_full: assertion 'GST_IS_ELEMENT (dest)' failed. I checked the source type its gst.Element. The errors are follow:
(__debug_bin2560998154:2129271): GStreamer-CRITICAL **: 12:23:28.168: gst_element_set_state: assertion 'GST_IS_ELEMENT (element)' failed

(__debug_bin2560998154:2129271): GStreamer-CRITICAL **: 12:23:46.738: gst_element_link_pads_full: assertion 'GST_IS_ELEMENT (dest)' failed

2023/09/12 12:23:46 Linking: [rtph264depay] => [avdec_h264]

(__debug_bin2560998154:2129271): GStreamer-CRITICAL **: 12:23:46.738: gst_element_link_pads_full: assertion 'GST_IS_ELEMENT (src)' failed

(__debug_bin2560998154:2129271): GStreamer-CRITICAL **: 12:23:46.738: gst_element_link_pads_full: assertion 'GST_IS_ELEMENT (src)' failed

(__debug_bin2560998154:2129271): GStreamer-CRITICAL **: 12:23:46.738: gst_element_request_pad: assertion 'templ != NULL' failed'

(__debug_bin2560998154:2129271): GStreamer-CRITICAL **: 12:23:46.739: gst_pad_link_full: assertion 'GST_IS_PAD (sinkpad)' failed

Anything would help.

Noticed today that a number of SeekFlags in element.go (introduced by: 06dccea) are not compatible with older versions of Gstreamer (<1.18).

We need to decide which versions we are going to support.

For example, i needed to comment lines 59:62 in element.go to get it working on Gstreamer 1.14.

What do you guys think?

Add support for adding probe's to pads.