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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