Hi,

I'm new here and I have found a problem with the default android audio sink.

I'm using gst_parse_launch("playbin") to create my pipeline.
If I use appsrc:\\ as url and and configure the appsrc within the source-setup callback with: format=GST_FORMAT_TIME, is-live=false, do-timestamp=false, block=true and stream-type=GST_APP_STREAM_TYPE_SEEKABLE the current position of the pipeline freezes after seeking.
The Timestamps of the pushed buffers are generated as followed:
GST_BUFFER_PTS(buffer) = audiotimestamp;
GST_BUFFER_DURATION(buffer) = getFrameTime(audio_caps, gst_buffer_get_size(buffer));
audiotimestamp += GST_BUFFER_DURATION(buffer);
getFrameTime calculates the duration of the current byte array by its size and the caps.
After seeking the "seek-data" callback of the appsrc will be called where I set the audiotimestamp to the given offset.

1) If I use the openslessink with its default configuration the playback works great but after seeking the first time the current position "gst_element_query_position(pipeline, ..." doesn't change anymore but the sound keeps playing.  

2) If I set "gst_base_sink_set_async_enabled" to false for the audio sink the position will update after seeking but the sound stops playing and I get the following warnings "0x5cad0600 gstaudiobasesink.c:1507:gst_audio_base_sink_skew_slaving:<audiosink> correct clock skew -24378431 < -20000000"


Is this a bug or do I have to change someting?