Default latency of interaudiosrc

[Adrian Pardini]

Hi all,

I'm working on a project and I try to mix a pair of interaudiosrc and
intervideosrc with content from a live source (alsasrc and v4l2src).
The inter elements are being fed with a decodebin.

intervideosrc reports itself as a live source with a min latency of 0
and a max of -1.
interaudiosrc on the other hand reports as being live with a min and
max latency equal to 30 * gst_util_uint64_scale_int (GST_SECOND, SIZE,
48000) and with SIZE defined as 1600 that equals to 1 second.

With that scenario both audio and video at the output play fine and
synced but there's quite a delay for the live sources.

I understand that for video the situation is different as most of the
time one buffer is actually one video frame but for audio that is not
always true. Looking at the code of gstinteraudiosrc.c  I don't really
understand how that number (30) was derived.

For my application I could set it as low as 3 and it works for me
without issues but I feel like a hack.
So, how wrong am I for doing this?

Thanks.