Audio-resampling isn't actually performed?
lusinehayrapetyan1992 at gmail.com
Thu Nov 12 11:16:54 UTC 2020
Thanks for your feedback. My element is based on GstAudioAggregator and it's
pretty simple - I only overwrite base aggregate function in my element.
I have 2 input streams and in aggregate function "based on some condition" I
want to output only one stream. Lets say I have a counter and change which
sink pad to output every 100 iteration. This is the code:
static int current_id = 0;
static int count = 0;
static GstFlowReturn aggregate(GstAggregator * agg, gboolean timeout)
gboolean pad_eos = false;
GstBuffer *outbuf = NULL;
for (size_t i = 0; i < g_list_length(GST_ELEMENT(agg)->sinkpads); ++i)
GstAggregatorPad *aggpad = (GstAggregatorPad *)
pad_eos = gst_aggregator_pad_is_eos(aggpad);
// Choose one pad and drop the other one.
if (i == current_id)
outbuf = gst_aggregator_pad_pop_buffer(aggpad);
// 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.
if (count == 100)
count = 0;
if (current_id == 2)
current_id = 0;
if (!pad_eos && outbuf)
ret = gst_aggregator_finish_buffer(agg, outbuf);
ret = GST_FLOW_EOS;
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 0:00:08.845333332 < 0:00:18.612919076
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 even though I resample the audio streams to the same rate
with audioresample element before passing them to my custom plugin.
Sorry for the long question.
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