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Re: [Discuss-gnuradio] Timestamping baseband samples


From: Benny Alexandar
Subject: Re: [Discuss-gnuradio] Timestamping baseband samples
Date: Sun, 11 Feb 2018 07:02:18 +0000

Hi Marcus,

If USRP crystal has a ppm of +-25ppm, then its  RF deviation, and
the baseband decoder detects this drift and correct it by resampling
till it gets the required number of samples.

So my question is this input side drift is corrected by baseband
synchronization, and this should not be seen at audio output
side. 25ppm drift is corrected and the time stamp will be
put after that. 

For example if 10% interpolation is done on RF samples, then
the timestamp should be modified to add for 10 samples.

Is that the correct way to timestamp for audio packets ?

-ben



From: Marcus Müller <address@hidden>
Sent: Saturday, February 10, 2018 9:47 PM
To: Benny Alexandar; address@hidden
Subject: Re: [Discuss-gnuradio] Timestamping baseband samples
 
Hi ben,

the Ettus USRP series of devices (N2x0, B2xx, E3xx, X3xx, N3xx) have RX
sample time stamping that you can find as tags in your sample flow.

That means they can annotate a sample buffer with the device time of
the first contained sample. The device time can be set according to an
external standard, if helpful (e.g. a GPS clock).

You won't find anything surprising in there: The first received item
will have a tag containing the device time at which that sample was
received. From there on, you can use (sample number)/(sampling rate) to
calculate the device time corresponding to each packet. Since absolute
time doesn't matter to your application, you could just as well do
without that tag and simply count samples. With the timestamps, you
still count samples, and add the timestamp from the first sample.
There's no additional information that helps you solve the two-clock
problem to be gained here.

Best regards,
Marcus

On Sat, 2018-02-10 at 07:06 +0000, Benny Alexandar wrote:
> Hi All,
>
> In Broadcast receiver where the RF samples are received on AIR and
> channel decoder does the demodulation of digital radio symbols, and
> extracts the compressed audio data. This is then fed to an audio
> decoder to decode the audio. After decoding the uncompressed audio
> has to be played out.
> I would like to implement the jitter buffer, by having time stamps.
> So my question is where to do the timestamping, the timestamp put
> after audio decoding will be jittery because of variable audio
> content and compression effects. So the best place is to put at the
> RF input when baseband samples are buffered using DMA, this then can
> be corrected for clock recovery and use it while sending out audio.
>  
>
> Any suggestions on this approach.
>
> -ben
>
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