On Tue, Oct 25, 2016 at 2:21 PM, Fons Adriaensen
wrote:
> Between the two incoherent domains there is a buffer and a resampler. The
> resampler is adjusted so that the average number of samples 'in the
> pipeline' is constant. The problem is finding a reliable and noise-free
Hi Marcus,
> sorry, please believe me, no-one meant to upset you. I'm a bit surprised
> you're taking this so personally!
I'm pretty sure nobody wanted to upset me ! But apparently nobody is
interested to make the Jack interface work as intended, not even its
author...
> Anyway, how do you
Hi Fons,
sorry, please believe me, no-one meant to upset you. I'm a bit surprised
you're taking this so personally!
This is Free Software - since none of us felt confident in their Jack
skills, I guess no-one answered, and we can't blame you for not
contributing your time, but, of course, we
On Tue, Oct 25, 2016 at 09:07:57PM +0200, Marcus Müller wrote:
> So, yeah, again, two-clock problem:
> If your red pitaya samples at, let's say, 4.8 MHz, and your sound card
> samples at 48kHz, then you need to decimate by a factor of hundred.
> Problem is that the sampling rate of the Sound card
Hi Markus,
> The only blocking element is the [audio sink] in "blocking mode" (of
> course).
yep. It can only consume as fast as the sound hardware consumes audio
samples.
> If i start an audio flow graph without [throttle] and increase some additive
> noise or change the volume with a multiplier
> If i start an audio flow graph without [throttle] and increase some additive
> noise or change the volume with a multiplier in the stream the audible
> signal gets effected way too slow (action-to-effect-time approximately 10 s
> or more). If I use a [throttle] after the [wav file source] i can
Dear Marcus,
thanks for your response.
Marcus Müller-3 wrote
> That sounds like a unrelated thing; aU's typically happen when you have
> two clocks in your system, for example, one SDR device's sampling clock
> and a soundcard sampling clock, as those never align perfectly, and one
> can drift
Dear Mark,
On 09.10.2016 14:54, MarkO wrote:
>
> For this issue I've tried [Packet Encoder] and [Packet Decoder].
I'd say: don't. These blocks are pretty deprecated by now, being rather
inflexible, slow python blocks. For now, try
http://gnuradio.org/doc/doxygen/page_packet_data.html .
>
> With
Hi Sylyain,
thank you for your quick response.
Sylvain Munaut-2 wrote
> Demodulation is not an exact process and the various loops in there
> will take some time to lock and so the beginning will be corrupted and
> your output won't be synced on the bytes boundaries of your input.
Ok, this is
Hi,
> Maybe I am wrong, but i expect an
> lossless connection in GRC if I directly connect an modulator to its
> appropriate demodulator - particularly when i use the default values.
Yes you are wrong.
Demodulation is not an exact process and the various loops in there
will take some time to
Hi everyone,
I currently use GRC 3.7.9.1 and only its In-Tree-Modules.
PC: Lenovo Laptop W530
CPU: Intel i7-3610QM with 8GB RAM
OS: Ubuntu 16.04 64 bit
I tried to create some easy examples of different modems. My first goal is
to modulate ASCII- or WAV-files, add some noise, demodulate the
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