On Aug 19, 2010, at 1:31 AM, Elvis Dowson wrote:
The issue manifests itself on both a VM and with GR running natively
on my hardware (late 2009 iMac 27" i7).
My US$0.02 worth: TTBOMK&M, on the Mac OS X side you can input and
output about any sample rate & CoreAudio will handle the sample rat
Hi Eric,
On Aug 19, 2010, at 5:35 AM, Eric Blossom wrote:
> Elvis, do you understand the high level source of the problem?
> That is, that there are two hardware devices whose clocks are not
> synchronized?
Yes, I do now, and that one possible solution is to rate match or buffer the
sample data
On Wed, Aug 18, 2010 at 18:35, Eric Blossom wrote:
> The relative rates may be mismatched by less than 0.1%. Even at that
> level, on the average you'd need to be adding or dropping 1 sample
> every 1000 to match the rates between the two clock domains.
I've had (cheap) sound cards that require
On Sat, Aug 14, 2010 at 03:16:38AM +0400, Elvis Dowson wrote:
> Hi Eric,
>
> On Aug 13, 2010, at 8:38 PM, Eric Blossom wrote:
>
> > Given the interfaces exported by ALSA, you'd need to figure out how to
> > honor the condition "ok_to_block == False".
>
> Could you please tell me the intent behin
On Fri, Aug 13, 2010 at 09:59:20PM +0400, Elvis Dowson wrote:
> Hi Eric,
>Before I dive into it, I just have a quick question.
> Why is it that others are not encountering the same issue? The same
> example code was run on a Mac Pro dual quad machine belonging to
> another member
Hi Eric,
On Aug 13, 2010, at 8:38 PM, Eric Blossom wrote:
> Given the interfaces exported by ALSA, you'd need to figure out how to
> honor the condition "ok_to_block == False".
Could you please tell me the intent behind the ok_to_block flag?
By blocking, do you mean a synchronous call, which w
I guess, its perhaps an older version of GNU Radio, the one before the audio
API changed.
Best regards,
Elvis
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Hi Eric,
Before I dive into it, I just have a quick question.
Why is it that others are not encountering the same issue? The same example
code was run on a Mac Pro dual quad machine belonging to another member of this
forum, and he didn't face the issue of crackling and distorted
On Fri, Aug 13, 2010 at 03:47:12AM +0400, Elvis Dowson wrote:
> Hi Eric,
>I came across an old thread,
> http://www.ruby-forum.com/topic/107228
>
> "There is a general issue related to the fact that when using
> usrp_wfm_rcv and similar applications that there are in fact two cloc
Hi Alex,
On Aug 13, 2010, at 12:39 PM, Alexandru Csete wrote:
>
> Maybe it is a driver issue. I suppose your iMac is the latest
> generation (late 2009?) but the distributed alsa drivers are usually
> older.
Yes, my iMac 27" is a late 2009 model.
$ sudo dmidecode -s system-product-name
iMac
On 13 August 2010 01:02, Elvis Dowson wrote:
> Hi,
> Another data point, if I set the audio sink sample rate to 16kHz, there
> are no under-runs, but that obviously will cause the sound to play back at a
> lower pitch, but atleast there are no crackles and distortions. In the
> console out
On 08/12/2010 04:47 PM, Elvis Dowson wrote:
Hi Eric,
I came across an old thread,
http://www.ruby-forum.com/topic/107228
"There is a general issue related to the fact that when using
usrp_wfm_rcv and similar applications that there are in fact two clock
domains, and that they ar
Hi Eric,
I came across an old thread,
http://www.ruby-forum.com/topic/107228
"There is a general issue related to the fact that when using
usrp_wfm_rcv and similar applications that there are in fact two clock
domains, and that they are guaranteed not to match. There's an osc on
t
Hi,
Another data point, if I set the audio sink sample rate to 16kHz, there
are no under-runs, but that obviously will cause the sound to play back at a
lower pitch, but atleast there are no crackles and distortions. In the console
output window, there are no audio underruns.
When you se
Hi,
I was reading up on audio underruns, and basically its because the
application isn't being able to generate enough data for the sound-card, i.e.
GNU Radio generating data slower than the card is consuming the data.
Ubuntu Studio Upgrade from Ubuntu - Community Ubuntu Documentation
On Thu, Aug 12, 2010 at 09:29:35PM +0400, Elvis Dowson wrote:
> Hi,
> Would anyone happen to have an idea where I should start looking,
> in order to debug the audio under-runs for the audio sink?
Does the dial_tone.py example sound OK?
Eric
> Best regards,
> Elvis Dowson
___
Hi,
Would anyone happen to have an idea where I should start looking, in
order to debug the audio under-runs for the audio sink?
Best regards,
Elvis Dowson
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Hi Alex,
On Aug 12, 2010, at 1:03 PM, Alexandru Csete wrote:
> Try to resample the audio to 44.1 kHz or 48 kHz using a
> rational_resampler block. That might solve it.
Adding the Rational Resampler, with decimation = 2 and interpolation = 3, works
in terms of resampling. The two separate wave f
On 12 August 2010 09:24, Elvis Dowson wrote:
> Hi,
> I have switched my linux environment from Fedora 12 to Ubuntu 10.04.
> This gives me better network performance with the USRP2, even when running
> Ubuntu 10.04 inside a VMware environment on Mac OS X 10.4.
> However, real-time audio, e.g
Hi,
I have switched my linux environment from Fedora 12 to Ubuntu 10.04.
This gives me better network performance with the USRP2, even when running
Ubuntu 10.04 inside a VMware environment on Mac OS X 10.4.
However, real-time audio, e.g. an FM receiver application, when played back in
r
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