Skip to content

GitLab

  • Projects
  • Groups
  • Snippets
  • Help
    • Loading...
  • Help
    • Help
    • Support
    • Community forum
    • Submit feedback
    • Contribute to GitLab
  • Sign in / Register
gst-plugins-base
gst-plugins-base
  • Project overview
    • Project overview
    • Details
    • Activity
    • Releases
  • Repository
    • Repository
    • Files
    • Commits
    • Branches
    • Tags
    • Contributors
    • Graph
    • Compare
  • Issues 628
    • Issues 628
    • List
    • Boards
    • Labels
    • Service Desk
    • Milestones
  • Merge Requests 104
    • Merge Requests 104
  • 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-basegst-plugins-base
  • Issues
  • #134

Closed
Open
Opened Oct 09, 2014 by Bugzilla Migration User@bugzilla-migration

audiomixer: segfault with rtp and audiotestsrc mixing buffers

Submitted by Matthew Waters @ystreet

Link to original bug (#738216)

Description

git master of everything
orc enabled

To reproduce

Sending pipeline
gst-launch-1.0 -v audiotestsrc ! opusenc ! rtpopuspay ! udpsink host=127.0.0.1 port=9876

Receiving pipeline
gst-launch-1.0 udpsrc uri=udp://127.0.0.1:9876 caps="application/x-rtp, media=audio, clock-rate=48000, encoding-name=X-GST-OPUS-DRAFT-SPITTKA-00, payload=96" ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! audiomixer name=m ! alsasink audiotestsrc ! m.

Results in a nice,

(gdb) bt  
#0  0x00007ffff7fcc244 in ?? ()  
#1  0x00007ffff408e44b in audiomixer_orc_add_s16 (d1=0x7fffe000f8f0, s1=0x7fffdc031abc, n=2048) at tmp-orc.c:355  
#2  0x00007ffff408ca91 in gst_audio_mixer_mix_buffer (outmap=<optimized out>, pad=<optimized out>, audiomixer=<optimized out>) at gstaudiomixer.c:1223  
#3  gst_audiomixer_aggregate (agg=0x7fffec979910) at gstaudiomixer.c:1447  
#4  0x00007ffff3e7fdc0 in aggregate_func (self=0x968250) at gstaggregator.c:483  
#5  0x00007ffff7b58776 in gst_task_func (task=0x983050) at gsttask.c:317  
#6  0x00007ffff70a3098 in g_thread_pool_thread_proxy (data=<optimized out>) at gthreadpool.c:307  
#7  0x00007ffff70a2715 in g_thread_proxy (data=0x965e80) at gthread.c:764  
#8  0x00007ffff6c1b314 in start_thread () from /usr/lib/libpthread.so.0  
#9  0x00007ffff69593ed in clone () from /usr/lib/libc.so.6
Assignee
Assign to
None
Milestone
None
Assign milestone
Time tracking
None
Due date
None
Reference: gstreamer/gst-plugins-base#134