Skip to content

GitLab

  • Projects
  • Groups
  • Snippets
  • Help
    • Loading...
  • Help
    • Help
    • Support
    • Community forum
    • Submit feedback
    • Contribute to GitLab
  • Sign in / Register
gst-plugins-bad
gst-plugins-bad
  • Project overview
    • Project overview
    • Details
    • Activity
    • Releases
  • Repository
    • Repository
    • Files
    • Commits
    • Branches
    • Tags
    • Contributors
    • Graph
    • Compare
  • Issues 994
    • Issues 994
    • List
    • Boards
    • Labels
    • Service Desk
    • Milestones
  • Merge Requests 192
    • Merge Requests 192
  • CI / CD
    • CI / CD
    • Pipelines
    • Jobs
    • Schedules
  • Operations
    • Operations
    • Incidents
    • Environments
  • Packages & Registries
    • Packages & Registries
    • Container Registry
  • Analytics
    • Analytics
    • CI / CD
    • Repository
    • Value Stream
  • Snippets
    • Snippets
  • Members
    • Members
  • Collapse sidebar
  • Activity
  • Graph
  • Create a new issue
  • Jobs
  • Commits
  • Issue Boards
  • GStreamer
  • gst-plugins-badgst-plugins-bad
  • Issues
  • #973

Closed
Open
Opened May 17, 2019 by Neil Dwyer@neildwyer1991

Webrtc 1.16.0 Blocked in the GstOutputSelector (Regression from 1.14.4)

Hey Team, I recently upgraded my gstreamer version to 1.16.0. I used to have a working pipeline where I transcode an RTMP stream to vp8/opus to send out to a website. Since upgrading to 1.16.0, the pipeline appears to do everything correctly except actually send the media. I see a blocked pad in the outputselector which I'm assuming is the culprit but might be a red herring.

Here's the pipeline for reference: pipeline

In chrome://webrtc-internals I have both a video and audio candidate pair that send and receive requests.

Any Ideas?

Assignee
Assign to
None
Milestone
None
Assign milestone
Time tracking
None
Due date
None
Reference: gstreamer/gst-plugins-bad#973