[Bug 720199] New: basesrc: DTS becomes 0 in gst_base_src_do_sync() for first buffer

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=720199
  GStreamer | gstreamer (core) | git

           Summary: basesrc: DTS becomes 0 in gst_base_src_do_sync() for
                    first buffer
    Classification: Platform
           Product: GStreamer
           Version: git
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gstreamer (core)
        AssignedTo: gstreamer-bugs at lists.freedesktop.org
        ReportedBy: davidsf at axis.com
         QAContact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---


Created an attachment (id=263926)
 View: https://bugzilla.gnome.org/attachment.cgi?id=263926
 Review: https://bugzilla.gnome.org/review?bug=720199&attachment=263926

basesrc: use segment start if DTS for first buffer is unset

We run into a problem with a small audio pipeline with one src and one sink.
The first buffer is dropped by the sink, because the src sets its DTS to 0 in
gst_base_src_do_sync(), and that is outside of the segment.

The attached patch is a suggested fix that solves the problem in our case. It
sets DTS to segment start instead of 0, if DTS is unset for the first buffer in
gst_base_src_do_sync().

It would be nice with feedback on whether this is the correct thing to do in
general.

Part of the background is that we use timestamps that don't start on zero,
which is why the DTS of zero ends up outside of the segment. (These timestamps
originate from alsasrc, monotonic clock).

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