July 2020 Archives by thread

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Starting: Wed Jul 1 02:34:00 UTC 2020
Ending: Fri Jul 31 21:46:48 UTC 2020
Messages: 359

Is it possible to make videoscale dynamic? Nicolas Dufresne
How to update videomixer properties during runtime? facepalm
- How to update videomixer properties during runtime? Matthew Waters
  - How to update videomixer properties during runtime? facepalm
    - How to update videomixer properties during runtime? Josh Doe
    - How to update videomixer properties during runtime? facepalm
gst-device-monitor-1.0 on raspbian Jack
How to get pipeline from gstreamer plugin written in python? facepalm
- How to get pipeline from gstreamer plugin written in python? Josh Doe
  - How to get pipeline from gstreamer plugin written in python? Michael Gruner
Get NTP and RTP timestamp using rtspsrc Nicolas Dufresne
- Get NTP and RTP timestamp using rtspsrc pasifus
  - Get NTP and RTP timestamp using rtspsrc pasifus
save stream VP8 to webm file Trey Hutcheson
Wayland VAAPI without monitor Frederic Turmel
- Wayland VAAPI without monitor Víctor Jáquez
gst_type_find_factory_get_list() incomplete Gary Metalle
GStreamer-CRITICAL gst_buffer_resize_range: assertion 'bufmax >= bufoffs + offset + size' failed SPAM kaladin
- GStreamer-CRITICAL gst_buffer_resize_range: assertion 'bufmax >= bufoffs + offset + size' failed SPAM Nicolas Dufresne
  - GStreamer-CRITICAL gst_buffer_resize_range: assertion 'bufmax >= bufoffs + offset + size' failed SPAM kaladin
    - GStreamer-CRITICAL gst_buffer_resize_range: assertion 'bufmax >= bufoffs + offset + size' failed SPAM Nicolas Dufresne
    - GStreamer-CRITICAL gst_buffer_resize_range: assertion 'bufmax >= bufoffs + offset + size' failed SPAM kaladin
gst-rtsp-server fdsrc Android'e Doğru
<v4l2src0:pool:src> Dropping truncated buffer, this is likely a driver bug. Aniket0987
- <v4l2src0:pool:src> Dropping truncated buffer, this is likely a driver bug. Nicolas Dufresne
Fwd: wrapping my head around clocksync ruben malchow
Output WebRTCBin to GTK Window using VideoOverlay Busayo Famutimi
- Output WebRTCBin to GTK Window using VideoOverlay Matthew Waters
  - Output WebRTCBin to GTK Window using VideoOverlay Busayo Famutimi
    - Output WebRTCBin to GTK Window using VideoOverlay Matthew Waters
    - Output WebRTCBin to GTK Window using VideoOverlay Busayo Famutimi
Setting up rtsp server using gstreamer? krombopulus
- Setting up rtsp server using gstreamer? Stephenwei
gapless playback in playbin3 Seiji I.
Help on video demux plugin Myzhar
GStreamer 1.17.2 development release Tim-Philipp Müller
- GStreamer 1.17.2 development release Tim-Philipp Müller
Why the 4-second latency difference with this pipeline (udpsrc+tee+tcpserversink+hlssink)? solo_juan
Sending a 1080p stream to YouTube, and a 720p stream to Facebook? Ian
- Sending a 1080p stream to YouTube, and a 720p stream to Facebook? Mathieu Duponchelle
GStreamer Source Element for Proprietary Security Device Mandar Joshi
- GStreamer Source Element for Proprietary Security Device Nicolas Dufresne
  - GStreamer Source Element for Proprietary Security Device Mandar Joshi
    - GStreamer Source Element for Proprietary Security Device Josh Doe
    - GStreamer Source Element for Proprietary Security Device Nicolas Dufresne
    - GStreamer Source Element for Proprietary Security Device Mandar Joshi
    - GStreamer Source Element for Proprietary Security Device Mandar Joshi
Gstreamer RTSP Server program fails with "failed to create element 'rtpbin' error mg
- RE: Gstreamer RTSP Server program fails with "failed to create element 'rtpbin' error mg
Mux live video stream and live audio stream F32
RELEASE: GStreamer Rust bindings 0.16.0 Sebastian Dröge
static builds with gstreamer-rs? David Holroyd
- static builds with gstreamer-rs? Sebastian Dröge
d3d11 plugin Busayo Famutimi
- d3d11 plugin Mailing List SVR
- d3d11 plugin Busayo Famutimi
Two RTP Ports or wav files over RTSP with 2 channels of RTP ido123
Version that supports webrtc Jerry Geis
- Version that supports webrtc Matthew Waters
  - Version that supports webrtc Jerry Geis
    - Version that supports webrtc Matthew Waters
    - Version that supports webrtc Jerry Geis
decodebin3 blocks state changes when one stream ends before the other jacold
Gracefully Handle (or ignore) GstFlowReturns gotsring
- Gracefully Handle (or ignore) GstFlowReturns Tim Müller
  - Gracefully Handle (or ignore) GstFlowReturns gotsring
Problem when Stopping then Restarting Pipeline in Python (Video only) whoman at gmail.com
- Problem when Stopping then Restarting Pipeline in Python (Video only) Nicolas Dufresne
  - Problem when Stopping then Restarting Pipeline in Python (Video only) whoman at gmail.com
    - Problem when Stopping then Restarting Pipeline in Python (Video only) Nicolas Dufresne
    - Problem when Stopping then Restarting Pipeline in Python (Video only) whoman at gmail.com
vaapiencode_h264 ! vaapiparse_h264 ogurin at auritek.com
- vaapiencode_h264 ! vaapiparse_h264 David Ing
- vaapiencode_h264 ! vaapiparse_h264 Nicolas Dufresne
  - vaapiencode_h264 ! vaapiparse_h264 ogurin at auritek.com
    - vaapiencode_h264 ! vaapiparse_h264 Nicolas Dufresne
- vaapiencode_h264 ! vaapiparse_h264 Víctor Jáquez
  - vaapiencode_h264 ! vaapiparse_h264 ogurin at auritek.com
Directshow output Simon Brown
- Directshow output Nirbheek Chauhan
webrtc and qmlglsink DejanCotra
- webrtc and qmlglsink Matthew Waters
  - webrtc and qmlglsink DejanCotra
Reading RTP MJPEG stream from a file Wolfgang Grandegger
- Reading RTP MJPEG stream from a file Anton Pryima
  - Reading RTP MJPEG stream from a file Wolfgang Grandegger
ICE candidate negotiation process Neil Young
- ICE candidate negotiation process Matthew Waters
  - ICE candidate negotiation process Neil Young
    - ICE candidate negotiation process Matthew Waters
    - ICE candidate negotiation process Neil Young
    - ICE candidate negotiation process Neil Young
    - ICE candidate negotiation process Neil Young
    - ICE candidate negotiation process Olivier Crête
    - ICE candidate negotiation process Neil Young
No video displayed thegeekygeeky
http stream is stopping after a few seconds Marcel Behlau
- http stream is stopping after a few seconds gotsring
  - http stream is stopping after a few seconds Marcel Behlau
still confused about latency jim nualart
- still confused about latency Nicolas Dufresne
- still confused about latency jim nualart
regarding gst_buffer_add_parent_buffer_meta tags Naveen.C
Webrtcbin with H.264 High or H.264 Baseline Mandar Joshi
- Webrtcbin with H.264 High or H.264 Baseline Nicolas Dufresne
  - Webrtcbin with H.264 High or H.264 Baseline Mandar Joshi
    - Webrtcbin with H.264 High or H.264 Baseline Nicolas Dufresne
gst-inspect-1.0 vaapi gives 0 features on Ubuntu jeyp4
- gst-inspect-1.0 vaapi gives 0 features on Ubuntu Tyler Compton
  - gst-inspect-1.0 vaapi gives 0 features on Ubuntu Víctor Jáquez
error while building master for windows Thornton, Keith
- error while building master for windows Nirbheek Chauhan
  - AW: error while building master for windows Thornton, Keith
Recording in gstreamer-1.0 >= 1.0 Ole Aamot
- Recording in gstreamer-1.0 >= 1.0 Anton Pryima
  - Recording in gstreamer-1.0 >= 1.0 Ole Aamot
- Recording in gstreamer-1.0 >= 1.0 Ole Aamot
Playbin configuration with a sink element which has both sink pads for video and audio Wudo Balmus
gst main loop runs even when eos send to pipeline umit
- gst main loop runs even when eos send to pipeline umit
  - gst main loop runs even when eos send to pipeline Olivier Crête
gst-plugins-bad: avtp: how to notify meson to use custom kernel header John Rama
- gst-plugins-bad: avtp: how to notify meson to use custom kernel header Desouza, Ederson
  - gst-plugins-bad: avtp: how to notify meson to use custom kernel header John Rama
    - gst-plugins-bad: avtp: how to notify meson to use custom kernel header Desouza, Ederson
    - gst-plugins-bad: avtp: how to notify meson to use custom kernel header John Rama
    - gst-plugins-bad: avtp: how to notify meson to use custom kernel header John Rama
    - gst-plugins-bad: avtp: how to notify meson to use custom kernel header Nirbheek Chauhan
    - gst-plugins-bad: avtp: how to notify meson to use custom kernel header John Rama
    - gst-plugins-bad: avtp: how to notify meson to use custom kernel header Desouza, Ederson
    - gst-plugins-bad: avtp: how to notify meson to use custom kernel header John Rama
Streaming Ogg Vorbis Jonas Kvinge
Playing 6 live stream video over Qt Widgets on another thread budakf
WebRTC connection error Anton Pryima
- WebRTC connection error Nicolas Dufresne
I observe a 120ms gap in udpsrc/udpsink every 5 seconds ... why? jim nualart
GStreamer connect to Spice server facepalm
- GStreamer connect to Spice server Nicolas Dufresne
How to sync movie and audio with rtpbin in gstremer SHIOKAWA TSUYOSHI(塩川　健)
Dynamic text on video frame eyalhir74
- Dynamic text on video frame Nicolas Dufresne
  - Dynamic text on video frame eyalhir74
    - Dynamic text on video frame eyalhir74
    - Dynamic text on video frame Nicolas Dufresne
    - Dynamic text on video frame eyalhir74
    - Dynamic text on video frame eyalhir74
playback over timeline Nelu
- playback over timeline Thibault Saunier
  - playback over timeline Nelu
    - playback over timeline Thibault Saunier
    - playback over timeline Nelu
    - playback over timeline Thibault Saunier
    - playback over timeline Nelu
    - playback over timeline Thibault Saunier
    - playback over timeline Nelu
split the raw pixel stream from stereo camera with gstreamer jles
- split the raw pixel stream from stereo camera with gstreamer Jan Schmidt
  - split the raw pixel stream from stereo camera with gstreamer jles
Resolution of the output window luke
Failed to get component handle 'OMX.broadcom.egl_render' topquant
Streaming freezes (vaapi or v4l2src) Timtchenko, Michael
- Streaming freezes (vaapi or v4l2src) Víctor Jáquez
- Streaming freezes (vaapi or v4l2src) Nicolas Dufresne
  - AW: [External] Re: Streaming freezes (vaapi or v4l2src) Timtchenko, Michael
    - [External] Re: Streaming freezes (vaapi or v4l2src) Marc Leeman
    - [External] Re: Streaming freezes (vaapi or v4l2src) Marc Leeman
Range header emreoner
- Range header emreoner
Rtsp range header emreoner
- Rtsp range header Nicolas Dufresne
  - Rtsp range header emreoner
    - Rtsp range header Mailing List SVR
    - Rtsp range header emreoner
Decoding deadlock hint during seek Michael Nazzareno Trimarchi
- Decoding deadlock hint during seek David Ing
  - Decoding deadlock hint during seek Michael Nazzareno Trimarchi
    - Decoding deadlock hint during seek Michael Nazzareno Trimarchi
    - Decoding deadlock hint during seek Olivier Crête
    - Decoding deadlock hint during seek Michael Nazzareno Trimarchi
qos event does not appear nishitha
- qos event does not appear Nicolas Dufresne
webrtc Windows and Linux peers. DejanCotra
- webrtc Windows and Linux peers. Nicolas Dufresne
  - webrtc Windows and Linux peers. DejanCotra
gst-plugins-bad: avtp: error on gst_avtp_src_start:<avtpsrc0> Failed to open socket: Operation not permitted John Rama
- gst-plugins-bad: avtp: error on gst_avtp_src_start:<avtpsrc0> Failed to open socket: Operation not permitted Desouza, Ederson
- gst-plugins-bad: avtp: error on gst_avtp_src_start:<avtpsrc0> Failed to open socket: Operation not permitted John Rama
  - gst-plugins-bad: avtp: error on gst_avtp_src_start:<avtpsrc0> Failed to open socket: Operation not permitted Desouza, Ederson
error 'not-linked' for appsrc Gary Metalle
How to sanitize audio file with AAC packet corruption David Ing
Invoking "add-local-ip-address" on GstWebRTCICE Ben Rush
- Invoking "add-local-ip-address" on GstWebRTCICE Matthew Waters
  - Invoking "add-local-ip-address" on GstWebRTCICE Ben Rush
    - Invoking "add-local-ip-address" on GstWebRTCICE Matthew Waters
    - Invoking "add-local-ip-address" on GstWebRTCICE Ben Rush
    - Invoking "add-local-ip-address" on GstWebRTCICE Matthew Waters
    - Invoking "add-local-ip-address" on GstWebRTCICE Ben Rush
I want to design a source element which can share video frames. Psychesnet Hsieh
- I want to design a source element which can share video frames. Mandar Joshi
  - I want to design a source element which can share video frames. Psychesnet Hsieh
    - I want to design a source element which can share video frames. Nicolas Dufresne
    - I want to design a source element which can share video frames. Peter Maersk-Moller
    - I want to design a source element which can share video frames. Psychesnet Hsieh
- I want to design a source element which can share video frames. Nicolas Dufresne
why i never can get eos message Борис
gst-rtsp-server shared media when client is requesting separate RTSP sessions for audio and video streams Mihael Brenko
1.16.2 src install error on CentOS 8 Jerry Geis
When does souphttpsrc does not support seeking ? nishitha
WebRTC with data channel only? William Gerecke
- WebRTC with data channel only? Anton Pryima
  - WebRTC with data channel only? William Gerecke
- WebRTC with data channel only? Matthew Waters
DVR file player SerbanM
- DVR file player Nicolas Dufresne
  - DVR file player SerbanM
RTSP Stream windows Coordinates and size? Pietro_Pancino
- RTSP Stream windows Coordinates and size? Nicolas Dufresne
Souphttpsrc always shows server is not seekable nishitha
Cross compilation of GStreamer in Ubuntu docker container resulting in not all subprojects being built Ben Rush
- Cross compilation of GStreamer in Ubuntu docker container resulting in not all subprojects being built Matthew Waters
  - Cross compilation of GStreamer in Ubuntu docker container resulting in not all subprojects being built Matthew Waters
Cerbero build errors for mingw Ben Rush
- Cerbero build errors for mingw Matthew Waters
Run gst-inspect-1.0 on device, always found nothing Psychesnet Hsieh
- Run gst-inspect-1.0 on device, always found nothing Michael Gruner
  - Run gst-inspect-1.0 on device, always found nothing Psychesnet Hsieh
"Internal data stream error" when trying to restart the pipeline after EOS marko_popovic_pro at yahoo.com
no frame appearing at appsink Gary Metalle
Occasionally no audio between webrtcbin and chrome Trey Hutcheson
- Occasionally no audio between webrtcbin and chrome Olivier Crête
  - Occasionally no audio between webrtcbin and chrome Trey Hutcheson
    - Occasionally no audio between webrtcbin and chrome Anton Pryima
    - Occasionally no audio between webrtcbin and chrome Trey Hutcheson
cerbero.errors.FatalError: Fatal Error: The required packaging tool 'WiX' was not found Ben Rush
- cerbero.errors.FatalError: Fatal Error: The required packaging tool 'WiX' was not found Matthew Waters
  - cerbero.errors.FatalError: Fatal Error: The required packaging tool 'WiX' was not found Ben Rush
RTP/RTCP mux using rtpfunnel Ralf Globisch
[Git][gstreamer/meson-ports/ffmpeg] Deleted branch release/4.3 Thibault Saunier
- [Git][gstreamer/meson-ports/ffmpeg] Deleted branch release/4.3 Tim-Philipp Müller
  - [Git][gstreamer/meson-ports/ffmpeg] Deleted branch release/4.3 Thibault Saunier
channel-mask? Rand Graham
Querying framerate in Python joseph
- Querying framerate in Python Thibault Saunier
  - Querying framerate in Python joseph
  - Querying framerate in Python joseph
Are httpservers seekable ? nishitha
How to get channel setting /2 into a recvonly webrtcbin transceiver SDP Anthony Alba
- How to get channel setting /2 into a recvonly webrtcbin transceiver SDP Matthew Waters
  - How to get channel setting /2 into a recvonly webrtcbin transceiver SDP Anthony Alba
iOS / Objective C how to gracefully shutdown David Walter
No EOS message when filesrc is connected to webrtcbin Anthony Alba
RTSP latency issues, effect of sync=false Guillermo Rodriguez Garcia
- RTSP latency issues, effect of sync=false Nicolas Dufresne
  - RTSP latency issues, effect of sync=false Guillermo Rodriguez Garcia
    - RTSP latency issues, effect of sync=false Nicolas Dufresne
    - RTSP latency issues, effect of sync=false Guillermo Rodriguez Garcia
    - RTSP latency issues, effect of sync=false Nicolas Dufresne
    - RTSP latency issues, effect of sync=false Guillermo Rodriguez Garcia
Getting SIGSEGV with the sinkelement.py example Anthony Alba
Help! Could not get microphone data on android platform with GStreamer julian
WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze Jack
- WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze Jack
- WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze Vinod Kesti
  - WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze Jack
    - WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze Jack
    - WARN message convert_taps_gint16_c: can't find exact taps and pipeline freeze Jack
HLS demux with fMP4 Ciro Ferreira
Video orientation change bojan74
- Video orientation change bojan74
  - Video orientation change Nicolas Dufresne
    - Video orientation change bojan74
set property value from a string Gary Metalle
- set property value from a string Matthew Waters
  - set property value from a string Gary Metalle
    - set property value from a string Gary Metalle
webrtc and appsink freezes pipeline DejanCotra
Gstreamer: filter caps do not completely specify the output format : by flt aish
No audio for network streaming files deeps8us
Merge video and audio into one Bruno Arieira
- Merge video and audio into one parithi
  - Merge video and audio into one Bruno Arieira
gstreamer applications on VNC Bhattiprolu RaviKumar
How can I change the bitrate dinamically in a gstreamer example? peter12
- How can I change the bitrate dinamically in a gstreamer example? gotsring
ximagesrc jahn
new_caps callback bojan74
- new_caps callback bojan74
Will Nvidia NVFBC screen capture ever be supported? Davide Perini
synchronize streams from 2 appsrc to hlssink toksey
audio converter F32 unpack to F32? Rand Graham
transcoding AC3 to MP3 Rand Graham
videorate doesn't drop or duplicate frames in live stream? gotsring
- videorate doesn't drop or duplicate frames in live stream? Nicolas Dufresne
  - videorate doesn't drop or duplicate frames in live stream? gotsring
How to pass PTS/DTS from tsdemux to pipeline Michal Lazo
Gstelement refcount not descreasing to zero lead fd leak virajmistry511
NAL unit type 26, 27 not yet supported Abhinav Agrawal
How to debug/fix running error when use rtph264pay? Psychesnet Hsieh
- How to debug/fix running error when use rtph264pay? gotsring
  - How to debug/fix running error when use rtph264pay? Psychesnet Hsieh
    - AW: How to debug/fix running error when use rtph264pay? Thornton, Keith
    - How to debug/fix running error when use rtph264pay? Psychesnet Hsieh
    - How to debug/fix running error when use rtph264pay? gotsring
    - How to debug/fix running error when use rtph264pay? Psychesnet Hsieh
    - How to debug/fix running error when use rtph264pay? gotsring
    - How to debug/fix running error when use rtph264pay? Psychesnet Hsieh
    - How to debug/fix running error when use rtph264pay? Nicolas Dufresne
    - How to debug/fix running error when use rtph264pay? gotsring
    - How to debug/fix running error when use rtph264pay? Psychesnet Hsieh
    - How to debug/fix running error when use rtph264pay? Psychesnet Hsieh
    - How to debug/fix running error when use rtph264pay? Nicolas Dufresne
    - How to debug/fix running error when use rtph264pay? gotsring
    - How to debug/fix running error when use rtph264pay? Psychesnet Hsieh
How to make/fix mp4 file can be play on vlc? Psychesnet Hsieh
- How to make/fix mp4 file can be play on vlc? gotsring
  - How to make/fix mp4 file can be play on vlc? Psychesnet Hsieh
    - How to make/fix mp4 file can be play on vlc? Psychesnet Hsieh
    - AW: How to make/fix mp4 file can be play on vlc? Thornton, Keith
Gstreamer no video in output stream when using MotionCells OpenCV plugin visuatech
ges_pipeline_set_mode error killerrats
- ges_pipeline_set_mode error Thibault Saunier
  - ges_pipeline_set_mode error killerrats
Working example of RTMP stream to Gstreamer RTSP server? John Deutscher
- Working example of RTMP stream to Gstreamer RTSP server? Tim Müller
  - [EXTERNAL] Re: Working example of RTMP stream to Gstreamer RTSP server? John Deutscher
RTSP stream to webrtcbin Sumeet
Swapping Channels with Audioconvert's mix-matrix Mariano Koremblum
- Swapping Channels with Audioconvert's mix-matrix Mariano Koremblum
Building from source using Cerbero transport raj
- Building from source using Cerbero transport raj
gStreamer memory leak observed when restarting the pipeline multiple times SAMAL Suraj
- gStreamer memory leak observed when restarting the pipeline multiple times Tim Müller
I need some help. Gstreamer. H.264 encoding + decoding + displaying. In c code peter12
- I need some help. Gstreamer. H.264 encoding + decoding + displaying. In c code Marc Leeman
How to compile opencv gstreamer plugins for android? Lulek
extend GstBaseTransform ynjiun
- extend GstBaseTransform Michael Gruner
RTSP to WEBRTC without h264 transcoding pasifus
- RTSP to WEBRTC without h264 transcoding Sean DuBois
How can I set capabilities on videotestsrc? Psychesnet Hsieh
- How can I set capabilities on videotestsrc? Matthew Waters
  - How can I set capabilities on videotestsrc? Psychesnet Hsieh
4k streaming to webrtc kaladin
- 4k streaming to webrtc Patrick Cusack
  - 4k streaming to webrtc kaladin

Last message date: Fri Jul 31 21:46:48 UTC 2020
Archived on: Fri Jul 31 21:47:03 UTC 2020

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