Brian, I think your idea does work. I think the tricky bit to doing this really well is having a control loop that reacts quickly enough that we don’t have to be stuck with a giant buffer that adds undesirable latency, but then conversely a control loop that adapts at a slow enough rate and with small enough steps that we don’t introduce human perceptible audio artifacts.
> On Mar 7, 2019, at 7:46 AM, Brian Padalino via USRP-users > <usrp-us...@lists.ettus.com> wrote: > > On Wed, Mar 6, 2019 at 3:12 PM Marcus Müller <marcus.muel...@ettus.com > <mailto:marcus.muel...@ettus.com>> wrote: > I've had rather longish discussions on how to solve this; essentially: > for something that actually *solves* the issue (instead of postponing > it), as Ian said, you'd need to have clock domain crossing ability. > > Could message passing from the real-time blocks solve this issue in a > flexible way? > > Imagine the following: blocks connected to actual hardware use the computer > wall clock to try to determine an average sample clock as it relates to the > CPU clock. The USRP source block would be able to determine how many > samples/(sec of CPU clock) were coming in and Audio sink blocks would be able > to determine how many samples/(sec of CPU clock) were being consumed. The > same idea for Audio sources and USRP sinks. Since the blocking calls for > USRP or Audio blocks could be wrapped in a high precision timer, once any > initial buffering had been filled up, the rate should settle out. > > The modified blocks could then send a message of actual sample rate to > whoever needed to listen, and the appropriate sample rate could be figured > out in the "resampling FIFO". > > What am I missing? Why won't this work? > > Brian > _______________________________________________ > USRP-users mailing list > usrp-us...@lists.ettus.com > http://lists.ettus.com/mailman/listinfo/usrp-users_lists.ettus.com
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