GstAudioAggregator with different audio rates

[Lusine Hayrapetyan]

Hi Folks,
I've a custom plugin based on GstAudioAggregator. It has 2 sink pads.
I've overwritten the aggregate base function where I output
(gst_aggregator_finish_buffer(agg, outbuf)) one of the sink pads based on
some condition - lets say I have a counter and change which sink pad to
output every 100 iteration. This is the code which I have:

static int current_id = 0;
static int count = 0;

static GstFlowReturn gst_audioagg_aggregate(GstAggregator * agg, gboolean
timeout)
{
    GstFlowReturn ret;
    gboolean pad_eos = false;
    GstBuffer *outbuf = NULL;
    GST_OBJECT_LOCK(agg);
    for (size_t i = 0; i < g_list_length(GST_ELEMENT(agg)->sinkpads); ++i)
    {
        GstAggregatorPad *aggpad = (GstAggregatorPad *)
(g_list_nth_data(GST_ELEMENT(agg)->sinkpads, i));
        pad_eos = gst_aggregator_pad_is_eos(aggpad);
        if (!pad_eos)
        {
           // Choose one pad and drop the other one.
            if (i == current_id)
                outbuf = gst_aggregator_pad_pop_buffer(aggpad);
            else
                gst_aggregator_pad_drop_buffer(aggpad);
        }
    }
    GST_OBJECT_UNLOCK(agg);
//
// Change the sink pad after 100 iteration. It means output 2 seconds audio
from first sink pad
// then 2 seconds of audio from second sink pad, then again 2 second audio
from first and etc.
//
    count ++;
    if (count == 100)
    {
        count = 0;
        current_id ++;
        if (current_id == 2)
            current_id = 0;
    }

    if (!pad_eos && outbuf)
        ret = gst_aggregator_finish_buffer(agg, outbuf);
    else
       ret = GST_FLOW_EOS;
    //
    // We only post the message here, because we can't post it while the
object
    // is locked.
    //
    return ret;
}

It  works correctly when both my inputs(sink pads) have the same audio rate
but in case of inputs have different sample rates I get the following error
message even though I do resampling(bringing to the same audio rate with
audioresample) before passing to the audio aggregator element:
ERROR                  qtmux gstqtmux.c:4535:gst_qt_mux_add_buffer:
decreasing DTS value 0:00:02.131927438 < 0:00:04.643990929

Please let me know what I'm missing. Why this code works when my input
files have the same audio rate and why I get this error when both my inputs
have different rates but I do audio resampling. Sorry for the long question.

Thanks,
Lusine
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