[Bug 759227] New: ring buffer state is still GST_AUDIO_RING_BUFFER_STATE_ERROR after changing element state from null to ready failed.

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=759227

            Bug ID: 759227
           Summary: ring buffer state is still
                    GST_AUDIO_RING_BUFFER_STATE_ERROR after changing
                    element state from null to ready failed.
    Classification: Platform
           Product: GStreamer
           Version: 1.0.0
                OS: Linux
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: 27210403 at qq.com
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

try to setup a pipeline of which the source is alsasrc.
when set pipeline state to playing, the alsasrc device is not ready and we got
a resource error, the state change failed.
after this the state of audiosrcringbuffer is
GST_AUDIO_RING_BUFFER_STATE_ERROR.
then we set pipeline state to null and try to set the pipeline state to play
again, this time the alsa src device is ready and open is success. but we still
get an internal flow error as the ring buffer is not created again, the state
is still GST_AUDIO_RING_BUFFER_STATE_ERROR.
simply use following modification to help understanding this problem:
===========================code===============================
static GstStateChangeReturn
gst_audio_base_src_change_state (GstElement * element,
    GstStateChange transition)
{
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
  GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (element);

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      GST_DEBUG_OBJECT (src, "NULL->READY");
      GST_OBJECT_LOCK (src);
+     if (src->ringbuffer != NULL) {
+       gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
+       src->ringbuffer = NULL;
+     }
      if (src->ringbuffer == NULL) {
        gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
        src->ringbuffer = gst_audio_base_src_create_ringbuffer (src);
      }

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