[Bug 761008] New: tcpserversink buffers audio until first client connects

[GStreamer (GNOME Bugzilla)]

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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