gst-plugins-bad issueshttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues2021-09-24T14:36:31Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/742player: Clear external subtitle uri before selecting in-band text track2021-09-24T14:36:31ZBugzilla Migration Userplayer: Clear external subtitle uri before selecting in-band text track## Submitted by Philippe Normand `@philn`
**[Link to original bug (#796725)](https://bugzilla.gnome.org/show_bug.cgi?id=796725)**
## Description
See commit message.## Submitted by Philippe Normand `@philn`
**[Link to original bug (#796725)](https://bugzilla.gnome.org/show_bug.cgi?id=796725)**
## Description
See commit message.Philippe NormandPhilippe Normandhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/741srt: Add a property to set stream id for retrieving client socket2021-09-24T14:36:31ZBugzilla Migration Usersrt: Add a property to set stream id for retrieving client socket## Submitted by Yeongjin Jeong
**[Link to original bug (#796719)](https://bugzilla.gnome.org/show_bug.cgi?id=796719)**
## Description
Created attachment 372878
srt: Add a property to set stream id for client socket
srt: Add a...## Submitted by Yeongjin Jeong
**[Link to original bug (#796719)](https://bugzilla.gnome.org/show_bug.cgi?id=796719)**
## Description
Created attachment 372878
srt: Add a property to set stream id for client socket
srt: Add a property to set stream id for client socket
This patch adds a support to get/set stream id of socket prior to connecting.
Stream ID will be able to be retrieved by the listener side from the socket
that is returned from srt_accept and was connected by a socket with that set stream ID.
This option in pre-1.3.0 SRT version is unavailable.
**Patch 372878**, "srt: Add a property to set stream id for client socket":
[0002-srt-Add-a-property-to-set-stream-id-for-client-socke.patch](/uploads/8c72c6490257c2e75a69ceec207cadda/0002-srt-Add-a-property-to-set-stream-id-for-client-socke.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/739shmsrc: is-live=true leads to segfault during state transition2021-09-24T14:36:31ZBugzilla Migration Usershmsrc: is-live=true leads to segfault during state transition## Submitted by Alexander Duda
**[Link to original bug (#796658)](https://bugzilla.gnome.org/show_bug.cgi?id=796658)**
## Description
Conditions:
* property is-live=true
* pipeline goes from playing to paused
Cause:
* gst...## Submitted by Alexander Duda
**[Link to original bug (#796658)](https://bugzilla.gnome.org/show_bug.cgi?id=796658)**
## Description
Conditions:
* property is-live=true
* pipeline goes from playing to paused
Cause:
* gst_shm_src_stop_reading deletes the underlying pipe without locking the object
* gstshmsrc.c#L357: rv = sp_client_recv (self->pipe->pipe, &buf) access invalid pipehttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/738gst-player: API for when the stream is interrupted or ended.2021-09-24T14:36:30ZBugzilla Migration Usergst-player: API for when the stream is interrupted or ended.## Submitted by Jordan Petridis
**[Link to original bug (#796630)](https://bugzilla.gnome.org/show_bug.cgi?id=796630)**
## Description
Use case, a media player wants to track the progress of a stream and save the last known position...## Submitted by Jordan Petridis
**[Link to original bug (#796630)](https://bugzilla.gnome.org/show_bug.cgi?id=796630)**
## Description
Use case, a media player wants to track the progress of a stream and save the last known position before the app was closed or the stream swapped out for another. So if it's an hour long audio file for example next time you hit play it will start from where you left of.
When I asked around in irc, I was told that `connect_end_of_stream` does not do that but it could. (I think in the sense that it's was not meant for that usecase and thus not reliable).https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/736ivfparse: Add vp9 header parsing to support dynamic resolution change notific...2021-09-24T14:36:30ZBugzilla Migration Userivfparse: Add vp9 header parsing to support dynamic resolution change notification## Submitted by Sreerenj Balachandran `@sree`
**[Link to original bug (#796599)](https://bugzilla.gnome.org/show_bug.cgi?id=796599)**
## Description
IVF parser can detect the vp9 streams, but it only extracts the IVF header for noti...## Submitted by Sreerenj Balachandran `@sree`
**[Link to original bug (#796599)](https://bugzilla.gnome.org/show_bug.cgi?id=796599)**
## Description
IVF parser can detect the vp9 streams, but it only extracts the IVF header for notifying the resolution. This won't work for dynamic resolution change, especially vp9 supports to have inter prediction from varying resolution frames.
For VP8, the ivfparser has uncompressed header parsing and it announces any possible resolution change.
Adding support for VP9 FrameHeader parsing in ivfparse is more complicated than vp8.
The easy option could be to add a dependency to the codecparser library so that we can use vp9 codecparsing apis directly.
The second option is to add a dependency to the bitreader api from GStreamer and implement the parsing support in ivfparse.
I would like to know what everybody thinks about this.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/733waylandsink: Error when trying to playback 348x240 in I4202021-09-24T14:36:28ZBugzilla Migration Userwaylandsink: Error when trying to playback 348x240 in I420## Submitted by Nicolas Dufresne `@ndufresne`
**[Link to original bug (#796565)](https://bugzilla.gnome.org/show_bug.cgi?id=796565)**
## Description
There is an error when trying to playback 348x240. This was reported in a comment h...## Submitted by Nicolas Dufresne `@ndufresne`
**[Link to original bug (#796565)](https://bugzilla.gnome.org/show_bug.cgi?id=796565)**
## Description
There is an error when trying to playback 348x240. This was reported in a comment here:
https://bugzilla.gnome.org/show_bug.cgi?id=790057#c28
The issue is that the code currently assumes that the GStreamer default strides will match the validation code. But this is not true for I420, since GStreamer will round up by 4 the U and the V stride. So we endup with strides:
Gst: 348 176 176
Validate: 348 174 174
I've looked into Weston code, it seems to expect the validated value. Which we means the failure is correct, we do have an incompatible SHM based wl_buffer. The following is an attempt to try and allocate the SHM based wl_buffer with the following data. Though, it does not render properly in Weston. I have spent quite some time to figure-out why, but could not. Here's the WIP branch:
https://gitlab.collabora.com/nicolas/gst-plugins-bad/commits/wayland-pool-fixhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/732waylandsink: support video crop using viewporter protocol2021-09-24T14:36:28ZBugzilla Migration Userwaylandsink: support video crop using viewporter protocol## Submitted by Haihua Hu `@JaredHu`
**[Link to original bug (#796541)](https://bugzilla.gnome.org/show_bug.cgi?id=796541)**
## Description
waylandsink: support video crop using viewporter protocol
use API wp_viewport_set_s...## Submitted by Haihua Hu `@JaredHu`
**[Link to original bug (#796541)](https://bugzilla.gnome.org/show_bug.cgi?id=796541)**
## Description
waylandsink: support video crop using viewporter protocol
use API wp_viewport_set_source() to support video crop handlehttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/731pcapparse: ts-offset does not work2021-09-24T14:36:28ZBugzilla Migration Userpcapparse: ts-offset does not work## Submitted by Daniel F
**[Link to original bug (#796529)](https://bugzilla.gnome.org/show_bug.cgi?id=796529)**
## Description
I have pcap file with multiple RTP audio streams. One of them starts about 2 secs after beginning of pac...## Submitted by Daniel F
**[Link to original bug (#796529)](https://bugzilla.gnome.org/show_bug.cgi?id=796529)**
## Description
I have pcap file with multiple RTP audio streams. One of them starts about 2 secs after beginning of packet capture, 2nd one about 3 secs. I wanted to extract them and save to one wav file, and have them synchronized in the same way as in pcap file. To do this, I have tried to use following pipeline:
gst-launch-1.0 audiointerleave name=int start-time-selection=first \
filesrc location=file.pcap do-timestamp=true ! tee name=tee \
tee. ! pcapparse src-ip=192.168.100.20 src-port=5002 ts-offset=-1 caps="application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,payload=(int)8,clock-rate=(int)8000,encoding-params=(string)1" ! queue ! rtpjitterbuffer ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! capsfilter caps="audio/x-raw,channels=(int)1,channel-mask=(bitmask)0x1,rate=(int)8000" ! queue ! int.sink_0 \
tee. ! pcapparse dst-ip=192.168.100.20 dst-port=5004 ts-offset=-1 caps="application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,payload=(int)8,clock-rate=(int)8000,encoding-params=(string)1" ! queue ! rtpjitterbuffer ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! capsfilter caps="audio/x-raw,channels=(int)1,channel-mask=(bitmask)0x2,rate=(int)8000" ! queue ! int.sink_1 \
int.src ! audioconvert ! audioresample ! wavenc ! filesink location=file.wav
However it turned out that original stream synchronization is lost - both streams starts at timestamp 0 in output wav file. I also tried to set ts-offset to 0 for 1st pcapparse, and to 1000000000 (1sec) for 2nd but this had no effect - audio in both channels also started at timestamp 0 as previously.
Version: 1.14.1https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/730msdk: "msdkmpeg2dec ! glimagesink" is giving distorted output.2021-09-24T14:36:27ZBugzilla Migration Usermsdk: "msdkmpeg2dec ! glimagesink" is giving distorted output.## Submitted by wangfei `@wangfei`
**[Link to original bug (#796522)](https://bugzilla.gnome.org/show_bug.cgi?id=796522)**
## Description
Reproduce Steps:
============================================
$gst-launch-1.0 filesrc loca...## Submitted by wangfei `@wangfei`
**[Link to original bug (#796522)](https://bugzilla.gnome.org/show_bug.cgi?id=796522)**
## Description
Reproduce Steps:
============================================
$gst-launch-1.0 filesrc location=/media/ts/Sally.ts '!' tsdemux '!' mpegvideoparse '!' msdkmpeg2dec '!' glimagesink
Video rendered with distorted image. If export GST_GL_PLATFORM=egl, then it looks good.
### Blocking
* [Bug 789886](https://bugzilla.gnome.org/show_bug.cgi?id=789886)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/729msdk: Playback not smooth by using ximagesink.2021-09-24T14:36:27ZBugzilla Migration Usermsdk: Playback not smooth by using ximagesink.## Submitted by wangfei `@wangfei`
**[Link to original bug (#796521)](https://bugzilla.gnome.org/show_bug.cgi?id=796521)**
## Description
Reproduce Steps:
============================================
$gst-launch-1.0 filesrc loca...## Submitted by wangfei `@wangfei`
**[Link to original bug (#796521)](https://bugzilla.gnome.org/show_bug.cgi?id=796521)**
## Description
Reproduce Steps:
============================================
$gst-launch-1.0 filesrc location=/media/ts/Sally.ts '!' tsdemux '!' mpegvideoparse '!' msdkmpeg2dec '!' msdkvpp '!' video/x-raw,format=NV12 '!' videoconvert '!' ximagesink
Video rendered not smooth. And if remove video/x-raw,format=NV12:
$ gst-launch-1.0 filesrc location=/media/ts/Sally.ts '!' tsdemux '!' mpegvideoparse '!' msdkmpeg2dec '!' msdkvpp '!' videoconvert '!' ximagesink
Video rendered looks good.
### Depends on
* [Bug 796699](https://bugzilla.gnome.org/show_bug.cgi?id=796699)
### Blocking
* [Bug 789886](https://bugzilla.gnome.org/show_bug.cgi?id=789886)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/727rfbsrc doesn't work with UltraVNC server2021-09-24T14:36:26ZBugzilla Migration Userrfbsrc doesn't work with UltraVNC server## Submitted by Mikhaylo
**[Link to original bug (#796517)](https://bugzilla.gnome.org/show_bug.cgi?id=796517)**
## Description
Follow pipeline works fine with TigerVNC and TightVNC, but generates errors with UltraVNC
>gst-lau...## Submitted by Mikhaylo
**[Link to original bug (#796517)](https://bugzilla.gnome.org/show_bug.cgi?id=796517)**
## Description
Follow pipeline works fine with TigerVNC and TightVNC, but generates errors with UltraVNC
>gst-launch-1.0 rfbsrc host="192.168.10.97" port=5900 password=pass view-only=True ! fakesink
The error is "CRITICAL **: gst_video_info_set_format: assertion 'format != GST_VIDEO_FORMAT_UNKNOWN' failed"
full log is below
D:\gstreamer\1.0\x86_64\bin>gst-launch-1.0 rfbsrc host="192.168.10.97" port=5900 password=pass view-only=True ! fakesink
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
0:00:00.088776Se92tting pipeline to PLAYING ...
1 7828New c 00000000031E41C0 lock: GFIXME stSystemCl ock
default gstutils.c:3963:gst_pad_create_stream_id_internal:<rfbsrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:00:00.924737239 7828 00000000031E41C0 FIXME default gstutils.c:3963:gst_pad_create_stream_id_internal:<rfbsrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:00:00.930134156 7828 00000000031E41C0 FIXME basesink gstbasesink.c:3145:gst_base_sink_default_event:`<fakesink0>` stream-start event without group-id. Consider implementing group-id handling in the upstream elements
** (gst-launch-1.0:7828): CRITICAL **: gst_video_info_set_format: assertion 'format != GST_VIDEO_FORMAT_UNKNOWN' failed
** (gst-launch-1.0:7828): CRITICAL **: gst_video_info_to_caps: assertion 'info->finfo->format != GST_VIDEO_FORMAT_UNKNOWN' failed
(gst-launch-1.0:7828): GStreamer-CRITICAL **: gst_event_new_caps: assertion 'caps != NULL' failed
(gst-launch-1.0:7828): GStreamer-CRITICAL **: gst_pad_push_event: assertion 'GST_IS_EVENT (event)' failed
(gst-launch-1.0:7828): GStreamer-CRITICAL **: gst_mini_object_unref: assertion 'mini_object != NULL' failed
0:00:00.947142432 7828 00000000031E41C0 WARN basesrc gstbasesrc.c:3275:gst_base_src_prepare_allocation:`<rfbsrc0>` Subclass failed to decide allocation
0:00:00.950423690 7828 00000000031E41C0 WARN basesrc gstbasesrc.c:3055:gst_base_src_loop:`<rfbsrc0>` error: Internal data stream error.
0:00:00.953940566 7828 00000000031E41C0 WARN basesrc gstbasesrc.c:3055:gst_base_src_loop:`<rfbsrc0>` error: streaming stopped, reason not-negotiated (-4)
ERROR: from element /GstPipeline:pipeline0/GstRfbSrc:rfbsrc0: Internal data stream error.
Additional debug info:
gstbasesrc.c(3055): gst_base_src_loop (): /GstPipeline:pipeline0/GstRfbSrc:rfbsrc0:
streaming stopped, reason not-negotiated (-4)
Execution ended after 0:00:00.875125228
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
Version: 1.14.1https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/726kms: add support for kms atomic api2021-09-24T14:36:25ZBugzilla Migration Userkms: add support for kms atomic api## Submitted by Randy Li (ayaka)
**[Link to original bug (#796516)](https://bugzilla.gnome.org/show_bug.cgi?id=796516)**
## Description
Created attachment 372577
A early version
This patch only add a basic flow for atomic API...## Submitted by Randy Li (ayaka)
**[Link to original bug (#796516)](https://bugzilla.gnome.org/show_bug.cgi?id=796516)**
## Description
Created attachment 372577
A early version
This patch only add a basic flow for atomic API.
I have not done yet.
Except removing kms sync mechanism there is not much different to legacy API.
I am planning to install some kms properties to gobject like v4l2 does. Also GstVideoAggregator is a important feature that would be introduced, as kms atomic api supports commit multiple plane at the same time. With the helper of that, it is possible to show both video and video subtitle with this plugin.
~~**Patch 372577**~~, "A early version":
[0003-WIP-kms-support-atomic-api.patch](/uploads/5fe31d7411a33ff3aa92e8372d399f75/0003-WIP-kms-support-atomic-api.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/725kmssink: add support for GBM allocator2021-09-24T14:36:24ZBugzilla Migration Userkmssink: add support for GBM allocator## Submitted by Randy Li (ayaka)
**[Link to original bug (#796493)](https://bugzilla.gnome.org/show_bug.cgi?id=796493)**
## Description
Created attachment 372542
allocator
In the Android, there is a similiar usuage with GPU m...## Submitted by Randy Li (ayaka)
**[Link to original bug (#796493)](https://bugzilla.gnome.org/show_bug.cgi?id=796493)**
## Description
Created attachment 372542
allocator
In the Android, there is a similiar usuage with GPU memory, it use galloc() to allocate and access a memory from GPU.
There is a GBM support in -base, it doesn't include any memory operation.
This patch is verified with ARM MALI GPU library.
~~**Patch 372542**~~, "allocator":
[0003-kms-gbm-add-a-new-gbm-allocator.patch](/uploads/75848bf3817c5c9472e34daa84a30136/0003-kms-gbm-add-a-new-gbm-allocator.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/724avfvideosrc Consuming Too much CPU2021-09-24T14:36:24ZBugzilla Migration Useravfvideosrc Consuming Too much CPU## Submitted by Prafulla Kumar Singh
**[Link to original bug (#796478)](https://bugzilla.gnome.org/show_bug.cgi?id=796478)**
## Description
Compare to videotestsrc, avfvideosrc taking too much CPU.
pipeline with videotestsrc CPU u...## Submitted by Prafulla Kumar Singh
**[Link to original bug (#796478)](https://bugzilla.gnome.org/show_bug.cgi?id=796478)**
## Description
Compare to videotestsrc, avfvideosrc taking too much CPU.
pipeline with videotestsrc CPU use : ~ 28.2%
pipeline with avfvideosrc CPU use : ~ 207.2%
Following are terminal log for reference:
ADFGTECHs-MacBook-Pro-2:iOSGstreamerClient prafullasingh$ ps
PID TTY TIME CMD
1111 ttys000 0:00.49 -bash
12116 ttys000 0:20.14 ./objs/srs -c ./conf/srs.conf
12196 ttys000 0:00.06 tail -f objs/srs.log
14198 ttys002 0:00.46 -bash
22944 ttys002 0:03.17 gst-launch-1.0 videotestsrc is-live=true ! videoconvert ! x264enc tune=zerolatency ! video/x-h264 ! mpegtsmux name=mux ! queu
22733 ttys003 0:00.15 -bash
ADFGTECHs-MacBook-Pro-2:iOSGstreamerClient prafullasingh$ ps -p 22944 -o %cpu,%mem
%CPU %MEM
25.6 0.4
ADFGTECHs-MacBook-Pro-2:iOSGstreamerClient prafullasingh$ ps -p 22944 -o %cpu,%mem
%CPU %MEM
28.2 0.4
ADFGTECHs-MacBook-Pro-2:iOSGstreamerClient prafullasingh$ ps -p 22944 -o %cpu,%mem
%CPU %MEM
28.5 0.4
ADFGTECHs-MacBook-Pro-2:iOSGstreamerClient prafullasingh$ ps
PID TTY TIME CMD
1111 ttys000 0:00.49 -bash
12116 ttys000 0:20.15 ./objs/srs -c ./conf/srs.conf
12196 ttys000 0:00.06 tail -f objs/srs.log
14198 ttys002 0:00.47 -bash
22951 ttys002 0:11.46 gst-launch-1.0 avfvideosrc ! videoconvert ! x264enc tune=zerolatency ! video/x-h264 ! mpegtsmux name=mux ! queue ! udpsink ho
22733 ttys003 0:00.17 -bash
ADFGTECHs-MacBook-Pro-2:iOSGstreamerClient prafullasingh$ ps -p 22951 -o %cpu,%mem
%CPU %MEM
211.1 0.7
ADFGTECHs-MacBook-Pro-2:iOSGstreamerClient prafullasingh$ ps -p 22951 -o %cpu,%mem
%CPU %MEM
207.2 0.7
ADFGTECHs-MacBook-Pro-2:iOSGstreamerClient prafullasingh$ gst-inspect-1.0 --version
gst-inspect-1.0 version 1.14.0
GStreamer 1.14.0
Version: 1.xhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/722msdk: Add MediaSDK versioned feature support and dynamic element registration2021-09-24T14:36:24ZBugzilla Migration Usermsdk: Add MediaSDK versioned feature support and dynamic element registration## Submitted by Sreerenj Balachandran `@sree`
**[Link to original bug (#796461)](https://bugzilla.gnome.org/show_bug.cgi?id=796461)**
## Description
We haven't added the code to guard against different MediaSDK versions.
The easy ...## Submitted by Sreerenj Balachandran `@sree`
**[Link to original bug (#796461)](https://bugzilla.gnome.org/show_bug.cgi?id=796461)**
## Description
We haven't added the code to guard against different MediaSDK versions.
The easy way is to add a hard-limit by raising the minimum required version to the recently released MediaStudio or only support the open source version.
But this may make many people unhappy :)
Otherwise, we should check the msdk spec and guard many of the features based on version (For eg: MFX_FOURCC_P010 was introduced by SDK API 1.9).
Dynamic element registration should also be handled.
Unfortunately, there is no MediaSDK API to retrieve the list of all supported decoders (AFAIK). Either we need to query the msdk with MFXVideoDECODE_Query
n times (n = number of all known codecs), or should keep some static platform specific pre-defined tables.
### Blocking
* [Bug 789886](https://bugzilla.gnome.org/show_bug.cgi?id=789886)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/718hls: adaptivedemux: Add support for EXT-X-START tag2021-09-24T14:36:23ZBugzilla Migration Userhls: adaptivedemux: Add support for EXT-X-START tag## Submitted by Hosang Lee
**[Link to original bug (#796434)](https://bugzilla.gnome.org/show_bug.cgi?id=796434)**
## Description
Support EXT-X-START tag to start playing from a preferred point indicated in the playlist.
For EX...## Submitted by Hosang Lee
**[Link to original bug (#796434)](https://bugzilla.gnome.org/show_bug.cgi?id=796434)**
## Description
Support EXT-X-START tag to start playing from a preferred point indicated in the playlist.
For EXT-X-START tag information refer to:
https://tools.ietf.org/html/draft-pantos-http-live-streaming-23#section-4.3.5.2https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/715wasapi: add a new volume property and implement volume/mute using ISimpleAudi...2021-09-24T14:36:22ZBugzilla Migration Userwasapi: add a new volume property and implement volume/mute using ISimpleAudioVolume## Submitted by Christoph Reiter (lazka)
**[Link to original bug (#796386)](https://bugzilla.gnome.org/show_bug.cgi?id=796386)**
## Description
Created attachment 372385
wasapi: add a new volume property and implement volume/mute ...## Submitted by Christoph Reiter (lazka)
**[Link to original bug (#796386)](https://bugzilla.gnome.org/show_bug.cgi?id=796386)**
## Description
Created attachment 372385
wasapi: add a new volume property and implement volume/mute using ISimpleAudioVolume
Implement mute/volume getters setters using ISimpleAudioVolume. This allows setting
those properties without delay and volume/mute changes will show up in sndvol.exe.
ISimpleAudioVolume only works in shared mode, so keep the old way of muting around
by setting the buffer to silent, to not break any existing code. Volume changes in
exclusive mode have no effect atm. Also missing is event handling of external
volume/mute changes through IAudioSessionEvents.
~~**Patch 372385**~~, "wasapi: add a new volume property and implement volume/mute using ISimpleAudioVolume":
[0001-wasapi-add-a-new-volume-property-and-implement-volum.patch](/uploads/94e6e41b4315bf2e21dd3c671e29fcc9/0001-wasapi-add-a-new-volume-property-and-implement-volum.patch)
Version: 1.14.0https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/713audiolatency does not work well with encoders/decoders and queues2021-09-24T14:36:22ZBugzilla Migration Useraudiolatency does not work well with encoders/decoders and queues## Submitted by Daniel F
**[Link to original bug (#796352)](https://bugzilla.gnome.org/show_bug.cgi?id=796352)**
## Description
New audiolatency element has some problem if both encoders/decoders and queue are used.
This works ...## Submitted by Daniel F
**[Link to original bug (#796352)](https://bugzilla.gnome.org/show_bug.cgi?id=796352)**
## Description
New audiolatency element has some problem if both encoders/decoders and queue are used.
This works as expected:
gst-launch-1.0 audiolatency name=a print-latency=true ! audioconvert ! audioresample ! opusenc ! opusdec ! audioconvert ! audioresample ! a.
After inserting queue between encoder and decoder it stops printing results periodically:
st-launch-1.0 audiolatency name=a print-latency=true ! audioconvert ! audioresample ! opusenc ! queue ! opusdec ! audioconvert ! audioresample ! a.
However when I tried to use queue without encoder/decoder, it started working again:
gst-launch-1.0 audiolatency name=a print-latency=true ! audioconvert ! audioresample ! queue ! audioconvert ! audioresample ! a.
In second case audiolatency most probably does not collect measurements at all. I have more complex pipeline in my app, which sends and receives RTP streams. I tried to use audiolatency there with print-latency=true, and also it does not work. I also periodically print values of last and avg latency, and I always get zeroes there.
Version: 1.14.0https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/710Wrong image size on full width or height2021-09-24T14:36:22ZBugzilla Migration UserWrong image size on full width or height## Submitted by Roland Peffer
**[Link to original bug (#796198)](https://bugzilla.gnome.org/show_bug.cgi?id=796198)**
## Description
If x + width or y + height matches the maximum screen width or height, always a fullscreen image is...## Submitted by Roland Peffer
**[Link to original bug (#796198)](https://bugzilla.gnome.org/show_bug.cgi?id=796198)**
## Description
If x + width or y + height matches the maximum screen width or height, always a fullscreen image is returned.
Example:
Screen size is 1920 x 1080
setting x = y =0, width 1920 and height 700 returns an image of size 1920 x 1080 instead of 1920 x 720.
Version: 1.12.4https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/709audiolatency cannot be used with rtpbin because this creates loop in pipeline2021-09-24T14:36:21ZBugzilla Migration Useraudiolatency cannot be used with rtpbin because this creates loop in pipeline## Submitted by Daniel F
**[Link to original bug (#796172)](https://bugzilla.gnome.org/show_bug.cgi?id=796172)**
## Description
I have app which uses rtpbin to both send and receive RTP streams. I wanted to use new audiolatency elem...## Submitted by Daniel F
**[Link to original bug (#796172)](https://bugzilla.gnome.org/show_bug.cgi?id=796172)**
## Description
I have app which uses rtpbin to both send and receive RTP streams. I wanted to use new audiolatency element to measure delay of external VoIP server. However it turned out that I cannot use it together with rtpbin, because this creates loop in pipeline. Here are errors reported by my app:
1526484976.964529 1003: Pipeline error: Internal data stream error. (gstbasesrc.c(3055): gst_base_src_loop (): /GstPipeline:pipeline0/GstAudioLatency:audiolatency0/GstAudioTestSrc:audiotestsrc0:
streaming stopped, reason not-linked (-1))
1526484976.964638 1003: Pipeline error: Internal data stream error. (gstqueue.c(988): gst_queue_handle_sink_event (): /GstPipeline:pipeline0/GstQueue:queue2:
streaming stopped, reason not-linked (-1))
Please add new elements audiolatencysrc and audiolatencysink, which would communicate together "behind the scenes" and appear to GStreamer as a two separate elements. This will allow to use audiolatency together with rtpbin.
Version: 1.14.0