GStreamer issueshttps://gitlab.freedesktop.org/groups/gstreamer/-/issues2021-09-24T14:38:47Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1423wasapi2src: Could not open resource for reading2021-09-24T14:38:47ZRoman Shpuntovwasapi2src: Could not open resource for readingI tried to capture audio from device, but when I provide `device` property for `wasapi2src` I have:
```
Could not open resource for reading., debugging information: ../sys/wasapi2/gstwasapi2src.c(351): gst_wasapi2_src_open (): /GstPipeli...I tried to capture audio from device, but when I provide `device` property for `wasapi2src` I have:
```
Could not open resource for reading., debugging information: ../sys/wasapi2/gstwasapi2src.c(351): gst_wasapi2_src_open (): /GstPipeline:pipeline/GstWasapi2Src:audio_source_capture:
Failed to open
```
I have device id (field `device.id`) value extracted from (`GstStructure`) properties on enum devices:
```
"\\\\\?\\SWD\#MMDEVAPI\#\{0.0.1.00000000\}.\{dfac857a-c87d-4e0a-84b1-03c250a88014\}\#\{2eef81be-33fa-4800-9670-1cd474972c3f\}"
```
I tried also:
```
"{0.0.1.00000000}.{dfac857a-c87d-4e0a-84b1-03c250a88014}"
```
and
```
"{dfac857a-c87d-4e0a-84b1-03c250a88014}"
```
and
```
"\\\?\SWD#MMDEVAPI#{0.0.1.00000000}.{dfac857a-c87d-4e0a-84b1-03c250a88014}#{2eef81be-33fa-4800-9670-1cd474972c3f}"
```
but every time I have `Could not open resource for reading`. With default device (not set `device` property) everything is fine.
Tested on desktop UWP app, gstreamer 1.18.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1425rtmp2src: handle leak when connection is unsuccessful2020-12-14T09:10:42ZYury Shatzrtmp2src: handle leak when connection is unsuccessfulWe have the live input that is not guaranteed to be always on. So, we create rtmp2src, connect and wait for data. If we get an error we destroy it and retry after a while. This causes program to leak file descriptors, and eventually cras...We have the live input that is not guaranteed to be always on. So, we create rtmp2src, connect and wait for data. If we get an error we destroy it and retry after a while. This causes program to leak file descriptors, and eventually crash (because linux has file descriptors limit of 1024 per process).
File descriptors belong to RtmpConnection object, which is not always destroyed. I tried to debug but I can't understand the root cause.
To repeat the error: simply point rtmp2src to a valid server with non-existing stream. When you get an error on the bus, destroy the element.
I found at least two scenarios where/how it happens
1) Client connection is successful, we get a connection in client_connect_done, but an error happens after that. For instance, if we get "NetStream.Play.StreamNotFound" error. In connect_task_done we call g_task_propagate_pointer, but it does not return a connection, self->connection is not set and we never call gst_rtmp_connection_close_and_unref. I was able to patch it in my copy (by saving pointer to connection in client_connect_done) but this is an ugly hack.
2) Client connection is unsuccessful, for example we get an error "failed to parse auth rejection" (or, I guess, any other error from send_connect_done). In this case, plugin does not get a reference to connection anywhere and no one else seems to close it.
Either plugin should die cleanly without a leak, or it needs "retry" mode where it retries failed connection.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1426Undeclared dependency on rtpmanager in tests2021-09-24T14:38:47ZMarius BakkeUndeclared dependency on rtpmanager in testsHi,
`meson test` fails on `elements_rtpsrc` and `elements_rtpsink` when gst-plugins-good is installed to a non-standard directory (but available on GST_SYSTEM_PLUGIN_PATH, etc), because `tests/check/meson.build` does not make rtpmanager...Hi,
`meson test` fails on `elements_rtpsrc` and `elements_rtpsink` when gst-plugins-good is installed to a non-standard directory (but available on GST_SYSTEM_PLUGIN_PATH, etc), because `tests/check/meson.build` does not make rtpmanager available.
The meson.build checks for `gstreamer` and `gst-plugins-base` with pkg-config and adds them on `GST_PLUGIN_PATH_1_0`. Something similar is needed for `gst-plugins-good`, but it currently lacks a pkg-config file.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1427webrtcdsp not found in Windows complete install2021-09-24T14:38:48ZAndreswebrtcdsp not found in Windows complete installI want to use webrtcdsp to carry on echo cancellation, but it is not installed in Windows. I am using last MSVC x64 msi files.
How can I install this plugin?I want to use webrtcdsp to carry on echo cancellation, but it is not installed in Windows. I am using last MSVC x64 msi files.
How can I install this plugin?https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1428srt: Weird double-up in pbkeylen enum2021-09-24T14:38:48ZJan Schmidtsrt: Weird double-up in pbkeylen enumThere's a duplicate 0 entry in the GstSRTKeyLength enum that seems unnecessary and is publically exposed in the pbkeylen property of the SRT elements:
```
/**
* GstSRTKeyLengthBits:
* @GST_SRT_KEY_LENGTH_NO_KEY: no encryption
* @GST_...There's a duplicate 0 entry in the GstSRTKeyLength enum that seems unnecessary and is publically exposed in the pbkeylen property of the SRT elements:
```
/**
* GstSRTKeyLengthBits:
* @GST_SRT_KEY_LENGTH_NO_KEY: no encryption
* @GST_SRT_KEY_LENGTH_0: no encryption
* @GST_SRT_KEY_LENGTH_16: 16 bytes (128-bit) length
* @GST_SRT_KEY_LENGTH_24: 24 bytes (192-bit) length
* @GST_SRT_KEY_LENGTH_32: 32 bytes (256-bit) length
*
* Crypto key length in bits
*/
typedef enum
{
GST_SRT_KEY_LENGTH_NO_KEY = 0,
GST_SRT_KEY_LENGTH_0 = GST_SRT_KEY_LENGTH_NO_KEY,
GST_SRT_KEY_LENGTH_16 = 16,
GST_SRT_KEY_LENGTH_24 = 24,
GST_SRT_KEY_LENGTH_32 = 32,
} GstSRTKeyLength;
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1429cuda: Add to documentation2021-11-05T13:24:45ZNicolas Dufresnecuda: Add to documentationAs noted in comment https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1633#note_655985 the new cuda work got merged without a documentation complaint, a clear sign it's not hooked into it. Would be nice to get tha...As noted in comment https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1633#note_655985 the new cuda work got merged without a documentation complaint, a clear sign it's not hooked into it. Would be nice to get that done by 1.20.
cc @julianhttps://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/870cuda: Add to documentation2021-11-05T14:07:00ZNicolas Dufresnecuda: Add to documentationAs noted in comment https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1633#note_655985 the new cuda work got merged without a documentation complaint, a clear sign it's not hooked into it. Would be nice to get tha...As noted in comment https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1633#note_655985 the new cuda work got merged without a documentation complaint, a clear sign it's not hooked into it. Would be nice to get that done by 1.20.
cc @julianhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1430gstspanplc.c FTBFS on Mageia: error: invalid use of incomplete typedef ‘plc_s...2021-09-24T14:38:49ZRémi Verscheldegstspanplc.c FTBFS on Mageia: error: invalid use of incomplete typedef ‘plc_state_t’Hi,
When enabling spandsp support, gstreamer-plugins-bad 1.18.0 fails building on Mageia Cauldron with the following error:
```
[791/911] cc -Iext/spandsp/libgstspandsp.so.p -Iext/spandsp -I../ext/spandsp -I. -I.. -I/usr/include/gstream...Hi,
When enabling spandsp support, gstreamer-plugins-bad 1.18.0 fails building on Mageia Cauldron with the following error:
```
[791/911] cc -Iext/spandsp/libgstspandsp.so.p -Iext/spandsp -I../ext/spandsp -I. -I.. -I/usr/include/gstreamer-1.0 -I/usr/include/glib-2.0 -I/usr/lib64/glib-2.0/include -I/usr/include/orc-0.4 -fdiagnostics-color=always -pipe -D_FILE_OFFSET_BITS=64 -fvisibility=hidden -fno-strict-aliasing -Wmissing-prototypes -Wdeclaration-after-statement -Wold-style-definition -Wmissing-declarations -Wredundant-decls -Wwrite-strings -Wformat -Wformat-security -Winit-self -Wmissing-include-dirs -Waddress -Wno-multichar -Wvla -Wpointer-arith -O2 -g -Wformat -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -fcommon -fstack-protector --param=ssp-buffer-size=4 -fasynchronous-unwind-tables -fPIC -pthread -DHAVE_CONFIG_H -MD -MQ ext/spandsp/libgstspandsp.so.p/gstspanplc.c.o -MF ext/spandsp/libgstspandsp.so.p/gstspanplc.c.o.d -o ext/spandsp/libgstspandsp.so.p/gstspanplc.c.o -c ../ext/spandsp/gstspanplc.c
FAILED: ext/spandsp/libgstspandsp.so.p/gstspanplc.c.o
cc -Iext/spandsp/libgstspandsp.so.p -Iext/spandsp -I../ext/spandsp -I. -I.. -I/usr/include/gstreamer-1.0 -I/usr/include/glib-2.0 -I/usr/lib64/glib-2.0/include -I/usr/include/orc-0.4 -fdiagnostics-color=always -pipe -D_FILE_OFFSET_BITS=64 -fvisibility=hidden -fno-strict-aliasing -Wmissing-prototypes -Wdeclaration-after-statement -Wold-style-definition -Wmissing-declarations -Wredundant-decls -Wwrite-strings -Wformat -Wformat-security -Winit-self -Wmissing-include-dirs -Waddress -Wno-multichar -Wvla -Wpointer-arith -O2 -g -Wformat -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -fcommon -fstack-protector --param=ssp-buffer-size=4 -fasynchronous-unwind-tables -fPIC -pthread -DHAVE_CONFIG_H -MD -MQ ext/spandsp/libgstspandsp.so.p/gstspanplc.c.o -MF ext/spandsp/libgstspandsp.so.p/gstspanplc.c.o.d -o ext/spandsp/libgstspandsp.so.p/gstspanplc.c.o -c ../ext/spandsp/gstspanplc.c
../ext/spandsp/gstspanplc.c: In function ‘gst_span_plc_create_stats’:
../ext/spandsp/gstspanplc.c:87:45: error: invalid use of incomplete typedef ‘plc_state_t’ {aka ‘struct plc_state_s’}
87 | "pitch", G_TYPE_INT, self->plc_state->pitch,
| ^~
../ext/spandsp/gstspanplc.c:88:52: error: invalid use of incomplete typedef ‘plc_state_t’ {aka ‘struct plc_state_s’}
88 | "pitch-offset", G_TYPE_INT, self->plc_state->pitch_offset, NULL);
| ^~
```
Full build log: [build.log](/uploads/dce95aa2a9b0dcce61eb3487d6b4c73a/build.log)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1431webpenc: creation of animated images is not supported2021-09-24T14:38:49ZNazar Mokrynskyiwebpenc: creation of animated images is not supportedCurrently it produces a sequence of separate images instead.
Property `animated=bool` would be great to have to control this behavior, can default to `false` to match current behavior.Currently it produces a sequence of separate images instead.
Property `animated=bool` would be great to have to control this behavior, can default to `false` to match current behavior.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1432https://webrtc.nirbheek.in/ Too many connection attempts2020-10-21T15:55:38ZRaj123456788https://webrtc.nirbheek.in/ Too many connection attempts![Screen_Shot_2020-10-20_at_9.31.22_AM](/uploads/95eff8930cae25af364ba78c41463e08/Screen_Shot_2020-10-20_at_9.31.22_AM.png)
@nirbheek Please fix this. Thanks!![Screen_Shot_2020-10-20_at_9.31.22_AM](/uploads/95eff8930cae25af364ba78c41463e08/Screen_Shot_2020-10-20_at_9.31.22_AM.png)
@nirbheek Please fix this. Thanks!https://gitlab.freedesktop.org/gstreamer/gst-examples/-/issues/21https://webrtc.nirbheek.in/ Too many connection attempts2020-10-21T06:47:06ZRaj123456788https://webrtc.nirbheek.in/ Too many connection attempts![Screen_Shot_2020-10-20_at_9.31.22_AM](/uploads/65a2ba659c0f613695fc696beb67c1cc/Screen_Shot_2020-10-20_at_9.31.22_AM.png)
@nirbheek Please fix this. Thanks!![Screen_Shot_2020-10-20_at_9.31.22_AM](/uploads/65a2ba659c0f613695fc696beb67c1cc/Screen_Shot_2020-10-20_at_9.31.22_AM.png)
@nirbheek Please fix this. Thanks!https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1433gst-player: Handle image-rotation tags2021-09-24T14:38:51ZPhilippe Normandgst-player: Handle image-rotation tagsThe player could set the playbin::video-filter either to videoflip or glvideoflip (with video-direction=automatic), depending on the video caps features.The player could set the playbin::video-filter either to videoflip or glvideoflip (with video-direction=automatic), depending on the video caps features.https://gitlab.freedesktop.org/gstreamer/gst-examples/-/issues/22janusvideoroom.py example doesn't work for H.2642020-10-23T07:25:44ZAlexandru Olteanujanusvideoroom.py example doesn't work for H.264If I try to use **H.264** as encoding in [janusvideoroom.py](/webrtc/janus/janusvideoroom.py), by making line 19 to be False, I always get **HangUp(sender=7135993183297719, reason='ICE failed')** in the console output.
Errors in Janus lo...If I try to use **H.264** as encoding in [janusvideoroom.py](/webrtc/janus/janusvideoroom.py), by making line 19 to be False, I always get **HangUp(sender=7135993183297719, reason='ICE failed')** in the console output.
Errors in Janus log:
- **[ERR]** [ice.c:janus_ice_check_failed:1760] [1786197890426270] ICE fa- iled for component 1 in stream 1...
- **[ERR]** [sdp-utils.c:janus_sdp_get_codec_pt_full:693] Unsupported codec 'none'
Gstreamer 1.18.0 is installed in a Ubuntu 20.10(yes I just downloaded the Beta :) ) docker container and janus gateway is installed in another docker container.
**The example works smoothly using VP8.**
I also attached a log file for the janus gateway,a capture file taken from Wireshark and the output for janusvideoroom.py
[gstwebrtc-to-janus.pcapng](/uploads/6fbf745325e3f0d88408e56ac3105e01/gstwebrtc-to-janus.pcapng)
[janus-log-stdout.txt](/uploads/320f5e708e181242a049c1c670abf601/janus-log-stdout.txt)
[janusvideroom-output.txt](/uploads/1a1db2107271788da4b6a1434dad1317/janusvideroom-output.txt)
P.S: I am sorry if this is not the right place to ask, but I tried all day long things that did not workhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1434AMC: fail to switch audio track when using AAC audio decoder2021-09-24T14:38:51ZXavier Claessensxclaesse@gmail.comAMC: fail to switch audio track when using AAC audio decoderI wrote a small Android app using GstPlayer. When playing an mp4 file with h264 video and 2 AAC audio tracks, it plays fine until I select the 2nd audio track. I see those errors:
```
10-21 10:42:58.072 4732 4807 E GStreamer+amcaudiod...I wrote a small Android app using GstPlayer. When playing an mp4 file with h264 video and 2 AAC audio tracks, it plays fine until I select the 2nd audio track. I see those errors:
```
10-21 10:42:58.072 4732 4807 E GStreamer+amcaudiodec: 0:00:13.585458281 0x7363f11f60 ../subprojects/gst-plugins-bad/sys/androidmedia/gstamcaudiodec.c:478:gst_amc_audio_dec_loop:<amcaudiodec-omxgoogleaacdecoder1> Failure dequeueing output buffer
10-21 10:42:58.072 4732 4807 W GStreamer+amcaudiodec: 0:00:13.585567239 0x7363f11f60 ../subprojects/gst-plugins-bad/sys/androidmedia/gstamcaudiodec.c:620:gst_amc_audio_dec_loop:<amcaudiodec-omxgoogleaacdecoder1> error: Failed to call Java method: java.lang.IllegalStateException
10-21 10:42:58.072 4732 4807 W GStreamer+amcaudiodec: java.lang.IllegalStateException
10-21 10:42:58.072 4732 4807 W GStreamer+amcaudiodec: at android.media.MediaCodec.native_dequeueOutputBuffer(Native Method)
10-21 10:42:58.072 4732 4807 W GStreamer+amcaudiodec: at android.media.MediaCodec.dequeueOutputBuffer(MediaCodec.java:3452)
10-21 10:42:58.072 4732 4807 W GStreamer+amcaudiodec:
```
If I rank google audio decoder SECONDARY then it picks libav decoder and it works.https://gitlab.freedesktop.org/gstreamer/gst-examples/-/issues/23webrtc-unidirectional-h264.c in VS 2017 crashing when closing the browser(bot...2021-09-24T22:43:56ZAshok Mishrawebrtc-unidirectional-h264.c in VS 2017 crashing when closing the browser(both Mozilla and Chrome)I'm using GStreamer 1.18 version. I used the following pipeline in webrtc-unidirectional-h264.c
receiver_entry->pipeline =
gst_parse_launch("webrtcbin name=webrtcbin stun-server=stun://"
STUN_SERVER " "
"videotestsrc is-live=tr...I'm using GStreamer 1.18 version. I used the following pipeline in webrtc-unidirectional-h264.c
receiver_entry->pipeline =
gst_parse_launch("webrtcbin name=webrtcbin stun-server=stun://"
STUN_SERVER " "
"videotestsrc is-live=true pattern=ball ! video/x-raw,width=640,height=360,framerate=15/1,format=I420 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max=15 ! video/x-h264,profile=constrained-baseline ! queue max-size-time=100000000 ! h264parse ! "
"rtph264pay config-interval=-1 name=payloader ! "
"application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin. ", &error);
It's working fine. When I'm closing the browser, the program is crashing when it is freeing the connection.
if (receiver_entry->connection != NULL)
g_object_unref(G_OBJECT(receiver_entry->connection));https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1435wpe: Audio support2021-08-29T10:56:36ZPhilippe Normandwpe: Audio supportI will try to work on this "soon"... The wpesrc should become a bin, exposing "sometimes" audio pads.I will try to work on this "soon"... The wpesrc should become a bin, exposing "sometimes" audio pads.https://gitlab.freedesktop.org/gstreamer/gst-examples/-/issues/24Webrtcbin - streaming not working for chrome.2020-11-02T03:25:35ZJaskarn KalsiWebrtcbin - streaming not working for chrome.We have streaming pipeline, and use webrtcbin to stream video from a decklink card to the browser.
Gstreamer version is 1.16.2
python 3.8
OS Ubuntu 20
signaling server is tornado. we are not using secure connection.
pipeline is decklink...We have streaming pipeline, and use webrtcbin to stream video from a decklink card to the browser.
Gstreamer version is 1.16.2
python 3.8
OS Ubuntu 20
signaling server is tornado. we are not using secure connection.
pipeline is decklink -> nvh264enc -> rtph264pay -> webrtcbin
We can play the video on the streaming server, but cant play the video in chrome on any of the connected devices on the local LAN.
Stream is going to be in the internal network so we didn't use any turn servers.
We can play the same stream in an Edge browser on any machine in the network but chrome and firefox will not play the video.
We wana stream video to chrome without changing any chrome flags. Are we missing something that is leading to this behavior.
Thanks
Jaskarn Kalsihttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1436curlhttpsink content-type ignored2022-09-12T00:06:09ZMichael Schmitcurlhttpsink content-type ignoredI have a problem, i can not solve it and you are my last option.
with gstreamer version 1.14.4 i could transmit audio data to my doorbird device.
now with gstreamer version 1.16.2 it does not work.
the httpcurlsink ignores the defined ...I have a problem, i can not solve it and you are my last option.
with gstreamer version 1.14.4 i could transmit audio data to my doorbird device.
now with gstreamer version 1.16.2 it does not work.
the httpcurlsink ignores the defined content-type and because of that the doorbird device does not allow to send data. see attachment.
is there a option to set the header content-type?
I hope you can help me.
![image](/uploads/7d4b6142b13e7fcec5b8a3806b7a06c5/image.png)![image__1_](/uploads/ae4050c786dcf662404e3e3ae96b8926/image__1_.png)https://gitlab.freedesktop.org/gstreamer/gst-examples/-/issues/25webrtc: `webrtc-unidirectional-h264.c` sample can't work without STUN server2020-12-09T13:21:02ZAdam Hlavatovicwebrtc: `webrtc-unidirectional-h264.c` sample can't work without STUN serverHello I'm trying to get `gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c` sample working without STUN server so I've modified GST pipeline and removed `stun-server` webrtcbin property and then also modified JS part of the solut...Hello I'm trying to get `gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c` sample working without STUN server so I've modified GST pipeline and removed `stun-server` webrtcbin property and then also modified JS part of the solution this way
```js
const config = { 'iceServers': [] };
playStream(vidstream, null, null, null, config, function (errmsg) { console.error(errmsg); });
```
but the sample is not working even both unidirectional and the page runs on the same machine. Can unidirectional sample work without STUN server ?
I'm using 1.16.2 libgstreamer1.0-0 package from Ubuntu 20.04 LTS.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1437d3d11: Add support for video processing filter2021-04-29T16:02:53ZSeungha Yangseungha@centricular.comd3d11: Add support for video processing filter[ID3D11VideoContext](https://docs.microsoft.com/en-us/windows/win32/api/d3d11/nn-d3d11-id3d11videocontext), [ID3D11VideoContext1](https://docs.microsoft.com/en-us/windows/win32/api/d3d11_1/nn-d3d11_1-id3d11videocontext1) and [ID3D11Video...[ID3D11VideoContext](https://docs.microsoft.com/en-us/windows/win32/api/d3d11/nn-d3d11-id3d11videocontext), [ID3D11VideoContext1](https://docs.microsoft.com/en-us/windows/win32/api/d3d11_1/nn-d3d11_1-id3d11videocontext1) and [ID3D11VideoContext2](https://docs.microsoft.com/en-us/windows/win32/api/d3d11_4/nn-d3d11_4-id3d11videocontext2) support VA-VPP equivalent video processing.
Implementation shouldn't be hard but some code for device compatibility check and per-device element registration would be required