|Subject:||Re: [Discuss-gnuradio] Audio Source configuration|
|Date:||Tue, 17 Mar 2015 17:25:13 +0200|
if I understand correctly, then you're using the audio source as "input" to your usrp_sink-terminated flow graph.
In that case, the "U"nderruns come from UHD, not the audio source; the flow graph is not supplying samples fast enough.
The problem might be systematic, for example if the rate change in your flow graph does not reflect the actual rate ratio between your audio device and the USRP -- e.g. the your audio device generates samples at 48kS/s, and your USRP consumes them with 5MS/s, but you have an interpolation rate of 100 (leading to a rate at which samples arrive at the USRP sink of 4.8MS/s; these are all just exemplary numbers).
Also possible, though, is that this is simply an extreme case of clock mismatch: For example, assume the reference clock in your USRP is of by 2ppm, making the "sample consume rate" (1+2e-6) times higher than it should be, according to an imaginary "exact" clock.
Now, assume your sound card is a bit "slower" than it should be, so let these should-be 48kS/s be (1-500e-6) times that what it should be.
That means that for every a USRP sampling rate of let's say 5MS/s, you'd have around 2500 samples too little per second, which (there's around 250 samples transportable per USB packet) is around 10 sample packets less than the USRP consumes.
 I pulled up the first sound chip datasheet I could find (TI PCM2903), and it demanded +-500 ppm.
On 03/17/2015 03:30 PM, Larry Van Der Jagt wrote:
Can anyone point me to some "color" on how to work with the Audio Source. I am working examples to transmit FM Audio and have problems with Underruns on the UHD:USRP Sink.
The examples (from files.ettus.com/tutorials/Lab_5.grc and others) I am running select Audio Source Arch:alsa and this seems to be a problem.
I have tried to replace the Device name with plughw:0,0 as suggested in the documetnation, but this does not seem to have an impact on the choice of Audio arch: alsa
If I replace the audio source with a signal source at the same sample rate all is well, but the Audio Source implements UUUUUUUUUUUU.
Using the suggestion of having the USRP expect a bit lower rate than the audio can deliver seems to implement nearly limitless delay .... the voice is received but only after a very long ... many seconds delay ... ....
At this time I am using a B210 connected via USB3 to an I7 with plenty of memory running Ubuntu 12.04 .... I have similar issues trying to use an E310 as the transmitter ...
Any direction would be appreciated.
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