On 03/06/2019 10:48 AM, Ian Buckley via USRP-users wrote:
The elegant architectual solution (abstracted from GR) is to have a FIFO cross
those 2 real clock domains, monitor the fullness and do closed loop sample rate
adaption in reaction to the FIFO’s fullness….
Now I can also think of reasons why that would be tricky to do well in GR.
It's a tad surprising that nobody has done an elastic buffer
implementation for GR.
On Mar 6, 2019, at 1:47 AM, franz kurniawan via USRP-users
<usrp-users@lists.ettus.com> wrote:
Hi all,
i have struggling for couple of weeks in my AM transmitter project.
i use audio source (alsa) as input and USRP sink (B200mini) as output..
i know that this making a "2 clock problem" and it will create either underflow
"U" or building latency between microphone (audio source) and resulted radio transmission
after several hours of continuous running..
i have tried some solutions below but none is working well :
* set OK to Block to "No" in audio source to remove clock in audio source.
Previously i have modified the source of alsa_source.cc since in the OK to
block feature in alsa is ignored in current version of gnuradio. But the
problem still persists
* using tagged stream to enable the burst mode of USRP. But also the problem
still persists
* manipulation of resampling rate with below formula:
interpolation : int(samp_rate * 1.000)
if i make the constant < 1.000 so that the consumer rate will be slightly faster than
producer rate, it will lead to repeating "U" , but there is no latency.
else if the constant > 1.000 , there will be no "U" , but there will be
building latency..
So, is there any proper solution to this kind of problem?
Regards,
Franz
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