[Bug 761008] New: tcpserversink buffers audio until first client connects
GStreamer (GNOME Bugzilla)
bugzilla at gnome.org
Fri Jan 22 14:37:42 PST 2016
https://bugzilla.gnome.org/show_bug.cgi?id=761008
Bug ID: 761008
Summary: tcpserversink buffers audio until first client
connects
Classification: Platform
Product: GStreamer
Version: 1.4.5
OS: Linux
Status: NEW
Severity: normal
Priority: Normal
Component: gst-plugins-base
Assignee: gstreamer-bugs at lists.freedesktop.org
Reporter: roysjosh at gmail.com
QA Contact: gstreamer-bugs at lists.freedesktop.org
GNOME version: ---
When doing the following, tcpserversink buffers audio until the first client
connects, forever maintaining the latency between starting the server and the
first client connecting:
server:
gst-launch-1.0 -vmt pulsesrc device=alsa_output...stereo.monitor !
audio/x-raw,format=S16LE,rate=48000,channels=1 ! opusenc bitrate=8000 ! gdppay
! tcpserversink host=0.0.0.0
client:
gst-launch-1.0 tcpclientsrc host=... ! gdpdepay ! opusdec ! pulsesink
Steps to reproduce:
1. start server (note the use of the pulsesrc monitor device to serve a remote
server's audio over the network)
2. wait 15 seconds, connect a client
3. immediately play some audio on the server
4. observe a 15 second delay before audio is played on the client
Things I have tried:
server: ... ! tcpserversink unit-format=time units-max=2000000
server: inserting a leaky=downstream queue before the tcpserversink with
max-size-time=1000
server: ... ! tcpserversink unit-format=buffers units-max=1 buffers-max=1
server: leaky queue with max-size-buffers=1
Things that work:
A RTP/UDP stream from the server to the client works with <1 second latency
(pulsesrc ! opusenc ! rtpopuspay ! udpsink host=$client).
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