[USRP-users] [Discuss-gnuradio] continous Tx voice transmission

Ian Buckley ianb at ionconcepts.com
Wed Mar 6 10:48:40 EST 2019

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.

> On Mar 6, 2019, at 1:47 AM, franz kurniawan via USRP-users <usrp-users at 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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