Re: [Flexradio] Cw Latency Problem Solved?

2006-05-25 Thread Gerald Youngblood
Hi Jimmy,

We have beaten this horse black and blue on the reflector and the prior
Forum over the last three years.  This is nothing new for discussion.  What
is new is that latency has improved and can be improved further.  Most of
the improvement can come through higher sampling rates (96k and 192k) and
proper setup of the sound card and buffers.  We intend to publish an
application note soon that details how to do the setup and optimization.
Faster PCs will help in reducing latency because they will be better able to
handle the interrupt load of smaller buffers.  This is the identical issue
faced in the professional recording industry, which has now moved mostly to
PC based systems.

Stay tuned for more on this subject from FlexRadio soon.

73,
Gerald

Gerald Youngblood, K5SDR
FlexRadio Systems
Ph: 512-535-5266
Fax: 512-233-5143
Email: [EMAIL PROTECTED]
Web: www.flex-radio.com
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Jimmy Jones
> Sent: Tuesday, May 23, 2006 7:50 PM
> To: FlexRadio@flex-radio.biz
> Subject: [Flexradio] Cw Latency Problem Solved?
> 
> Thanks Man,
> I knew it had to be a serious problem because none of the 
> gurus ever want to talk about. 
> "It's better to look good than to feel good."
> Monitoring what the circuit is actually doing is the reason I monitor.
> I still love the machine though and just can't seem to get 
> off of it and go back to my kenwood.
> How does kenwood do it in the 870? There isn't any noticeable 
> latency in that rig. That's one monitor circuit that's nearly perfect.
> Jimmy
>   - Original Message -
>   From: James Courtier-Dutton
>   To: Frank Brickle
>   Cc: Jimmy Jones ; FlexRadio@flex-radio.biz
>   Sent: Tuesday, May 23, 2006 10:24 AM
>   Subject: Re: [Flexradio] Cw Latency Problem Solved?
> 
> 
>   Frank Brickle wrote:
>   > If you want to monitor anything passing through the audio 
> subsystem --
>   > note, some sound cards provide a hard bypass, but that's 
> looping around
>   > the subsystem -- you can't ever get around a minimum of 1 
> buffer latency.
>   >
>   > 256 sample buffers / 192000 samples per sec = 1.3 ms latency.
>   >
>   > There will probably always be a way to patch signal 
> around the audio
>   > subsystem so as to eliminate latency entirely. If you 
> want to monitor
>   > what the audio subsystem and signal processing are doing, 
> however, then
>   > some latency is always going to be there.
>   >
>   > You could provide a secondary digital path through the 
> host computer
>   > that used, say, 64 sample buffers. 300 usec isn't bad. 
> But you don't
>   > want to use those same 64 sample buffers for your 
> filters, probably. So
>   > with minimum latency you won't be actually monitoring the 
> processing,
>   > just your input. FireWire systems can likely get the 
> absolute minimum
>   > down to around 10 samples. They still won't have passed 
> through the
>   > processing in that case either, however.
>   >
>   > As long as the audio systems use buffered IO, the latency 
> will never be
>   > completely eliminated. As long as you use FFT-based 
> convolution for
>   > efficient filtering, the true latency will never be less 
> than the length
>   > of the primary filters.
>   >
>   > 73
>   > Frank
>   > AB2KT
>   >   
>   Most sound PCI sound cards use a PCI transaction to 
> transfer a block of
>   bytes across the PCI bus.
>   On most sound cards this is 64bytes. So, with 16bit stereo 
> output, this
>   equals 16 samples per PCI transaction.
>   The sensible values for an application to use are some multiple of
>   64bytes for the chunk/period of the entire audio buffer. 
> For low latency
>   one would have 2 periods per buffer.
>   Your 256 samples above seems a sensible enough size for a 
> period, with
>   the entire ring buffer being 512 samples.
>   With a bit of clever tweaking, (i.e. in Linux) one could 
> pick a 64byte
>   boundary within the period and start outputting samples, 
> and they will
>   reach the speakers, thus not having to wait for the period 
> to end. It
>   would be a sort of buffer TX cut in feature.
> 
>   The FFT issue has nothing to do with sound card 
> capabilities, but there
>   is no reason why the FFT window should have to be in sync with audio
>   buffer periods.
>   I don't understand why or if FFT is used for CW TX anyway. 
> I would have
>   thought that FFT would only be needed on CW RX.
> 
>   James
> 
> 
> 
> 
>   --
>   No virus found in this incoming m

Re: [Flexradio] Cw Latency Problem Solved?

2006-05-23 Thread Jim Lux
At 08:24 AM 5/23/2006, James Courtier-Dutton wrote:
>F
>
>The FFT issue has nothing to do with sound card capabilities, but there
>is no reason why the FFT window should have to be in sync with audio
>buffer periods.
>I don't understand why or if FFT is used for CW TX anyway. I would have
>thought that FFT would only be needed on CW RX.

One does still need to do I/Q balancing on the TX side, and given that the 
mangitude and phase imbalances are not flat across the band, some sort of 
frequency domain filtering might be the best way. Or not. There's been some 
measurements of the Receive side using the onboard impulse generator, but I 
don't know that comparable measurements have been made on the transmit 
side. Since the only thing shared between sides is the "digital" local 
oscillator derived from the DDS, it's not clear that a measurement on the 
receive side can take out ALL of the imbalance on the transmit side.

As Frank has pointed out, for today, we live with the heritage of the dsp 
chain being half duplex, and using the same buffering strategy for Tx and Rx.


Jim 



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[Flexradio] Cw Latency Problem Solved?

2006-05-23 Thread Jimmy Jones
Thanks Man,
I knew it had to be a serious problem because none of the gurus ever want to 
talk about. 
"It's better to look good than to feel good."
Monitoring what the circuit is actually doing is the reason I monitor.
I still love the machine though and just can't seem to get off of it and go 
back to my kenwood.
How does kenwood do it in the 870? There isn't any noticeable latency in that 
rig. That's one monitor circuit that's nearly perfect.
Jimmy
  - Original Message - 
  From: James Courtier-Dutton 
  To: Frank Brickle 
  Cc: Jimmy Jones ; FlexRadio@flex-radio.biz 
  Sent: Tuesday, May 23, 2006 10:24 AM
  Subject: Re: [Flexradio] Cw Latency Problem Solved?


  Frank Brickle wrote:
  > If you want to monitor anything passing through the audio subsystem -- 
  > note, some sound cards provide a hard bypass, but that's looping around 
  > the subsystem -- you can't ever get around a minimum of 1 buffer latency.
  >
  > 256 sample buffers / 192000 samples per sec = 1.3 ms latency.
  >
  > There will probably always be a way to patch signal around the audio 
  > subsystem so as to eliminate latency entirely. If you want to monitor 
  > what the audio subsystem and signal processing are doing, however, then 
  > some latency is always going to be there.
  >
  > You could provide a secondary digital path through the host computer 
  > that used, say, 64 sample buffers. 300 usec isn't bad. But you don't 
  > want to use those same 64 sample buffers for your filters, probably. So 
  > with minimum latency you won't be actually monitoring the processing, 
  > just your input. FireWire systems can likely get the absolute minimum 
  > down to around 10 samples. They still won't have passed through the 
  > processing in that case either, however.
  >
  > As long as the audio systems use buffered IO, the latency will never be 
  > completely eliminated. As long as you use FFT-based convolution for 
  > efficient filtering, the true latency will never be less than the length 
  > of the primary filters.
  >
  > 73
  > Frank
  > AB2KT
  >   
  Most sound PCI sound cards use a PCI transaction to transfer a block of 
  bytes across the PCI bus.
  On most sound cards this is 64bytes. So, with 16bit stereo output, this 
  equals 16 samples per PCI transaction.
  The sensible values for an application to use are some multiple of 
  64bytes for the chunk/period of the entire audio buffer. For low latency 
  one would have 2 periods per buffer.
  Your 256 samples above seems a sensible enough size for a period, with 
  the entire ring buffer being 512 samples.
  With a bit of clever tweaking, (i.e. in Linux) one could pick a 64byte 
  boundary within the period and start outputting samples, and they will 
  reach the speakers, thus not having to wait for the period to end. It 
  would be a sort of buffer TX cut in feature.

  The FFT issue has nothing to do with sound card capabilities, but there 
  is no reason why the FFT window should have to be in sync with audio 
  buffer periods.
  I don't understand why or if FFT is used for CW TX anyway. I would have 
  thought that FFT would only be needed on CW RX.

  James




  -- 
  No virus found in this incoming message.
  Checked by AVG Free Edition.
  Version: 7.1.392 / Virus Database: 268.7.0/345 - Release Date: 5/22/2006

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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-23 Thread Frank Brickle
James Courtier-Dutton wrote:

> With a bit of clever tweaking, (i.e. in Linux) one could pick a 64byte 
> boundary within the period and start outputting samples, and they will 
> reach the speakers, thus not having to wait for the period to end. It 
> would be a sort of buffer TX cut in feature.

Again, if you have hardware loopback, you can have effectively zero 
delay. That's not the issue in general. The question is, do you want to 
monitor input to the radio, or do you want to monitor from someplace 
within the signal processing chain? If the latter, you have no choice 
but to go along with the DSP buffering size.

> The FFT issue has nothing to do with sound card capabilities, but there 
> is no reason why the FFT window should have to be in sync with audio 
> buffer periods.

It isn't, even now.

> I don't understand why or if FFT is used for CW TX anyway. I would have 
> thought that FFT would only be needed on CW RX.

As presently set up the same DSP buffer configurations are used for RX 
*and* for generation of the CW signal in the audio passband. That's 
because the audio subsystem doesn't know from TX or RX. It's just moving 
sample buffers around.

The only way that would change would be to run separate RX and TX 
processes, which is the way things are headed. For the time being, 
however, since the SDR-1000 is half-duplex, the DSP software is also.

FWIW, the latency issue is somewhat minimized under Linux since the 
keyer (and CW tone generation) live in a separate process from the DSP, 
and the sidetone is routed directly to the output without delay, while 
the actual transmitted tone is routed *through* the DSP. The buffer 
sizes are still consistent, but the sidetone arrives one buffer earlier 
than the transmitted CW.

73
Frank
AB2KT


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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-23 Thread James Courtier-Dutton
Frank Brickle wrote:
> If you want to monitor anything passing through the audio subsystem -- 
> note, some sound cards provide a hard bypass, but that's looping around 
> the subsystem -- you can't ever get around a minimum of 1 buffer latency.
>
> 256 sample buffers / 192000 samples per sec = 1.3 ms latency.
>
> There will probably always be a way to patch signal around the audio 
> subsystem so as to eliminate latency entirely. If you want to monitor 
> what the audio subsystem and signal processing are doing, however, then 
> some latency is always going to be there.
>
> You could provide a secondary digital path through the host computer 
> that used, say, 64 sample buffers. 300 usec isn't bad. But you don't 
> want to use those same 64 sample buffers for your filters, probably. So 
> with minimum latency you won't be actually monitoring the processing, 
> just your input. FireWire systems can likely get the absolute minimum 
> down to around 10 samples. They still won't have passed through the 
> processing in that case either, however.
>
> As long as the audio systems use buffered IO, the latency will never be 
> completely eliminated. As long as you use FFT-based convolution for 
> efficient filtering, the true latency will never be less than the length 
> of the primary filters.
>
> 73
> Frank
> AB2KT
>   
Most sound PCI sound cards use a PCI transaction to transfer a block of 
bytes across the PCI bus.
On most sound cards this is 64bytes. So, with 16bit stereo output, this 
equals 16 samples per PCI transaction.
The sensible values for an application to use are some multiple of 
64bytes for the chunk/period of the entire audio buffer. For low latency 
one would have 2 periods per buffer.
Your 256 samples above seems a sensible enough size for a period, with 
the entire ring buffer being 512 samples.
With a bit of clever tweaking, (i.e. in Linux) one could pick a 64byte 
boundary within the period and start outputting samples, and they will 
reach the speakers, thus not having to wait for the period to end. It 
would be a sort of buffer TX cut in feature.

The FFT issue has nothing to do with sound card capabilities, but there 
is no reason why the FFT window should have to be in sync with audio 
buffer periods.
I don't understand why or if FFT is used for CW TX anyway. I would have 
thought that FFT would only be needed on CW RX.

James


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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread root [knesbitt]
As well, if your running with an amp and have the X2-7 delay set too 
high (long delay), you will in effect, self-flagellate (is that a word?) 
yourself trying to run high-speed cw with fast break-in. Using either 
the internal or external keyer with semi-break-in enabled, the first 
character will be truncated by the amount set in x2-7. This shortening 
of the first transmitted symbol occurs between the time the radio sets 
up in transmit-mode and when rf output appears at the antenna port. The 
more t/r switching occurring, the more noticeable the shortening is at 
the higher keying speeds (>30 wpm).
Now, this is assuming your monitoring via the radio sidetone [MON] and 
have x2-7 set to some snail-like change-over duration (worst 
case/stone/age/amp). Typically here, I have x2-7 set to 5ms and its a 
non-issue running fast break-in.
Alternatively, if you are running an external keyer which generates it 
own side-tone and provides keying compensation, the x2-7 delay becomes 
completely irrelevant to this discussion.
For people forced to run with loong sequencing durations on x2-7, 
perhaps we could add a first symbol compensation feature to the 
new-keyer module, Bob?

Kirb - VE6IV
--

Robert McGwier wrote:
> I completely concur.   The external keyer, with PTT controlled by the 
> keyer and internal delay of dot/dash without delay on sidetone and 
> with PowerSDR semi-breakin turned off,  is optimal for this radio.  
> Thanks to Kirb's input on a bug which was fixed before the last 
> release,  I think we pretty much have this licked.
>
> Bob
>
>
>
> root [knesbitt] wrote:
>> Eric,
>> this is old news when running the Firebox :-)
>> Actually 0-latency operation is not dependent on running at 96 khz. I 
>> am running my system at 48 khz sampling using the stock Firebox 
>> firmware and new keyer code (external keyer not required). With the 
>> PreSonus set to 512 buffers and *2.5ms* latency there is absolutely 
>> *NO* delay! I can go as low as 2.0 ms latency while running the same 
>> buffer settings on this system without incurring ANY receive 
>> artifacts (3.2 Ghz P4-HT). The radio sidetone/monitor is perfectly in 
>> sync with the keyer sidetone and paddle actuation.
>> Saying that however, there are advantages to using an ext keyer to 
>> more precisely control the break-in action of the radio (for fast 
>> break-in operation), and first character attack and delay through the 
>> use of the PTT line.
>>
>> Kirb - VE6IV
>> ---
>>
>>
>>   
>


-- 

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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Guy Olinger, K2AV
Experience from use of existing radios...

The FT1000MP for CW has a bucket brigade delay for CW that can be set 
to "x" milliseconds between 0 and 30. When the key is hit, if assert 
transmit state to the amplifier is not already on, it is asserted 
immediately. The actual CW power is delayed for x milliseconds. The MP 
has a normal internal delay of several milliseconds from key down to 
RF out, and the bucket brigade is added to that.

The delay is intended to allow time for RX/TX transitions to complete 
so that there is no hot switching, which can destroy expensive parts 
in QRO amplifiers.

This is not clipped off the beginning of the cw, but smoothly applied 
to the entire CW transmission after the fashion of an audio delay 
line.

The monitor tone can be either set to the key input, or to the signal 
as sent, including the x delay. Listening to the signal as sent, with 
the delay set to 30ms, is for me well into unmanageable because the 
key closure does produce sound quickly enough.

My brain interprets it as malfunction of the paddle and I start 
pushing it harder to "make it work right". Trying to do 35 wpm with a 
keyer paddle and that delay sent to 30 is out of the question. This is 
even though the delay is smooth and uniform.

To have that occur *randomly* while trying to send with a paddle, at 
even half that delay, would be maddening.

We had random latency problems in a contest computer like which 
surfaced after the contest had started. I finally had to shut it down 
and trouble-shoot it. This even though most of the sent text was 
computer generated. A local op who heard that signal later said it 
sounded "like a drunk man was sending."

The trouble turned out to be a *latency problem* in the logging 
program, accidentally introduced with a quick fix of another bug 
distributed just before the contest.

We run with about 6-8 ms total fixed delay of CW text. A little more 
than needed, but QRO CW contesting is a harsh environment for amps, 
one where an amp can be run continuously at a 40 percent duty cycle 
for 48 hours straight.

A smooth 5 ms latency mentioned elsewhere, where the assert RX/TX for 
switching external devices is sent immediately via a hardware bypass, 
would work.

Some irregularity in the disappearance/reappearance of the received 
signal from the antenna would not be noticed.

Irregularity in the audio feedback to the operator from key paddle 
closure and character formation is maddening.

Irregularity in the transmitted characters will be reported by other 
operators who will question which extremity is being used to send and 
make snide remarks about apparent blood alcohol levels.

73, Guy.


- Original Message - 
From: "Jim Lux" <[EMAIL PROTECTED]>
>
> I should think it's more PowerSDR *and* Windows specific.   The 
> processing
> chains in PowerSDR (at least as of a couple months ago) assume that
> transitions between Tx and Rx occur only on buffer boundaries, so 
> the
> PowerSDR program and the dttsp code included in it are essentially 
> "half
> duplex".   Using tiny buffers, fast sample rates,  and a fast 
> computer can
> make the latency in the changeover very small.
>
> Eric's setup has buffers that are about 5 milliseconds long (512 
> samples *
> 10 microseconds/sample)
>
> A setup where you have 2048 sample buffers running at 48 ksps will 
> have
> buffers that are about 40 milliseconds long.  40 milliseconds is a 
> lot of
> delay, and if it actually stacks up with two buffers through the 
> chain,
> it's getting up towards the particularly bad delay range around 100
> milliseconds which is particularly annoying (as anyone who has had 
> poor
> echo suppression on a transoceanic phone call can attest to).
>
>
> There is also the fact that the SDR1000 hardware itself is 
> inherently half
> duplex, but the changeover speed for that should be very fast (the 
> QSD/QSE
> should enable/disable in sub-microseconds, and there's probably a 
> few
> tens/hundreds of microseconds to just talk to the parallel port). I 
> think
> there's also a relay, which is the dominant speed limeter.
>
> The actual management of the audio interface within the Windows 
> context is
> done by another program called portaudio (someone correct me if I'm 
> wrong
> here) which manages the Windows MultiMediaExtensions API calls and 
> just
> shoves/removes buffers into the dttsp input/output queues.
>
> As you say, there is more than enough hardware horsepower to do full 
> duplex
> with low latency with a much slower processor/data rate.  It's the 
> overall
> combination of PowerSDR/dttsp/portaudio/Windows that makes it a 
> coding
> challenge to get full duplex with deterministic delay times.
>
> And, for what it's worth, I suspect that from a usability standpoint 
> (not
> being a CW op myself, I can only surmise) the important thing is not 
> low
> latency, per se, but a very consistent latency.  After all, having 
> the
> speaker 2 feet away introduces a 2 millisecond

Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Robert McGwier
I completely concur.   The external keyer, with PTT controlled by the 
keyer and internal delay of dot/dash without delay on sidetone and with 
PowerSDR semi-breakin turned off,  is optimal for this radio.  Thanks to 
Kirb's input on a bug which was fixed before the last release,  I think 
we pretty much have this licked.

Bob



root [knesbitt] wrote:
> Eric,
> this is old news when running the Firebox :-)
> Actually 0-latency operation is not dependent on running at 96 khz. I am 
> running my system at 48 khz sampling using the stock Firebox firmware 
> and new keyer code (external keyer not required). With the PreSonus set 
> to 512 buffers and *2.5ms* latency there is absolutely *NO* delay! I can 
> go as low as 2.0 ms latency while running the same buffer settings on 
> this system without incurring ANY receive artifacts (3.2 Ghz P4-HT). The 
> radio sidetone/monitor is perfectly in sync with the keyer sidetone and 
> paddle actuation.
> Saying that however, there are advantages to using an ext keyer to more 
> precisely control the break-in action of the radio (for fast break-in 
> operation), and first character attack and delay through the use of the 
> PTT line.
>
> Kirb - VE6IV
> ---
>
>
>   

-- 
AMSAT VP Engineering. Member: ARRL, AMSAT-DL, TAPR, Packrats,
NJQRP/AMQRP, QRP ARCI, QCWA, FRC. ARRL SDR Wrk Grp Chairman
Laziness is the number one inspiration for ingenuity.  Guilty as charged!


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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Frank Brickle
If you want to monitor anything passing through the audio subsystem -- 
note, some sound cards provide a hard bypass, but that's looping around 
the subsystem -- you can't ever get around a minimum of 1 buffer latency.

256 sample buffers / 192000 samples per sec = 1.3 ms latency.

There will probably always be a way to patch signal around the audio 
subsystem so as to eliminate latency entirely. If you want to monitor 
what the audio subsystem and signal processing are doing, however, then 
some latency is always going to be there.

You could provide a secondary digital path through the host computer 
that used, say, 64 sample buffers. 300 usec isn't bad. But you don't 
want to use those same 64 sample buffers for your filters, probably. So 
with minimum latency you won't be actually monitoring the processing, 
just your input. FireWire systems can likely get the absolute minimum 
down to around 10 samples. They still won't have passed through the 
processing in that case either, however.

As long as the audio systems use buffered IO, the latency will never be 
completely eliminated. As long as you use FFT-based convolution for 
efficient filtering, the true latency will never be less than the length 
of the primary filters.

73
Frank
AB2KT

Jimmy Jones wrote:
> Ya Ya 
> 
>   I'm able to setup everything for speed as far as I can tell and the ssb 
> monitor is still terrible.
> 
> Firebox @ 1.5ms
> Firbox CPU on High
> Audio and dsp set at 256
> and the program runs in realtime mode.
> There are no skips on transmit or recieve.
> I know a few people that wont buy the flex just because of that. What a shame.
> I guess there will be a few cw geeks...uh freaks that won't buy the radio for 
> the same reasons.(latency)
>   From: Frank Brickle 
>   To: FlexRadio@flex-radio.biz 
>   Sent: Sunday, May 21, 2006 10:28 PM
>   Subject: Re: [Flexradio] Cw Latency Problem Solved?
> 
> 
>   Jimmy Jones wrote:
>   > I wish some of brain - o's could solve the monitor latency problem.
> 
>   Well, according to Science Today magazine this week, some string 
>   theorists are speculating that time travel may actually be feasible.
> 
>   With that solved the monitor latency problem won't be far behind. Till 
>   then, unfortunately, the only other solution is ESP.
> 
>   73
>   Frank
>   AB2KT
> 
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[Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Jimmy Jones
Ya Ya 

  I'm able to setup everything for speed as far as I can tell and the ssb 
monitor is still terrible.

Firebox @ 1.5ms
Firbox CPU on High
Audio and dsp set at 256
and the program runs in realtime mode.
There are no skips on transmit or recieve.
I know a few people that wont buy the flex just because of that. What a shame.
I guess there will be a few cw geeks...uh freaks that won't buy the radio for 
the same reasons.(latency)
  From: Frank Brickle 
  To: FlexRadio@flex-radio.biz 
  Sent: Sunday, May 21, 2006 10:28 PM
  Subject: Re: [Flexradio] Cw Latency Problem Solved?


  Jimmy Jones wrote:
  > I wish some of brain - o's could solve the monitor latency problem.

  Well, according to Science Today magazine this week, some string 
  theorists are speculating that time travel may actually be feasible.

  With that solved the monitor latency problem won't be far behind. Till 
  then, unfortunately, the only other solution is ESP.

  73
  Frank
  AB2KT

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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread James Courtier-Dutton
Alberto I2PHD wrote:
> James Courtier-Dutton wrote:
>> Just out of curiosity, what is the real reason for the latency?
>> I use Linux, and users can get high quality, reliable 2ms latency on a
>> very modest PC. That 2ms latency is with a Linux application called
>> Ardour, that users use for real time audio capture, effects and
>> playback. The same latency should be able to be achieved with any Linux
>> application.
>>
> I know almost nothing about Linux, but a friend of mine who does, once told 
> me that the Linux kernel has some long paths 
> in the disk I/O management area that run with interrupts disabled. I wonder 
> how this could allow to have a "guarantee" 
> of 2ms latency. But, as said, my knowledge of Linux is nil.
> 
> 73  Alberto  I2PHD
> 

One has to be careful in order to get the 2ms latency. For example use
"ck" kernels. These are latency tuned kernels that remove "the long
paths in the disk I/O management" that you mention. "ck" kernels also
place some restrictions on which modules one can load, so the "ck" just
don't include the modules that adversely influence latency. For example,
use ext2/ext3 and not reiserfs.

Low latency has trade offs though. In order to get low latency, one has
to slightly reduce the overall CPU and I/O processing power of the
system as a whole. E.g. One gets the low latency, but it will take just
slightly longer to compile a kernel.

James

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[Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread root [knesbitt]
Eric,
this is old news when running the Firebox :-)
Actually 0-latency operation is not dependent on running at 96 khz. I am 
running my system at 48 khz sampling using the stock Firebox firmware 
and new keyer code (external keyer not required). With the PreSonus set 
to 512 buffers and *2.5ms* latency there is absolutely *NO* delay! I can 
go as low as 2.0 ms latency while running the same buffer settings on 
this system without incurring ANY receive artifacts (3.2 Ghz P4-HT). The 
radio sidetone/monitor is perfectly in sync with the keyer sidetone and 
paddle actuation.
Saying that however, there are advantages to using an ext keyer to more 
precisely control the break-in action of the radio (for fast break-in 
operation), and first character attack and delay through the use of the 
PTT line.

Kirb - VE6IV
---


Eric Wachsmann wrote:
 >

 > Let me give you the skinny right from the horses mouth. Here's what I 
did: I was using a laptop (P4 ~2GHz) with a >FireBox. I set the FireBox 
to 6ms on the FireBox control panel. This could probably go lower on a 
more powerful >machine. I then set the PowerSDR buffers to 512 for the 
Audio and DSP. I set the FireBox and PowerSDR to use >96kHz sampling 
rate. At this speed, our prospective customer was able to send 50WPM 
using the soundcard >monitor (not an external keyer). This is the first 
time that we have tried this experiment at high speed since we >added 
96kHz support. Previous attempts at this speed all led to us directing 
our customers to use external keyers >for high speed CW. The long and 
short of this is that with the right setup (small buffers, low latency, 
fast PC, etc), >high speed semi-break in CW is possible with the 
SDR-1000 TODAY!
 >
 >
 >Eric Wachsmann FlexRadio Systems


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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Jim Lux
At 02:03 PM 5/22/2006, James Courtier-Dutton wrote:
>Eric Wachsmann - FlexRadio wrote:
> > Let me give you the skinny right from the horses mouth.  Here's what I
> > did:
> >



> > Eric Wachsmann
> > FlexRadio Systems
> >
>
>Just out of curiosity, what is the real reason for the latency?
>I use Linux, and users can get high quality, reliable 2ms latency on a
>very modest PC. That 2ms latency is with a Linux application called
>Ardour, that users use for real time audio capture, effects and
>playback. The same latency should be able to be achieved with any Linux
>application.
>
>So, is this latency problem windows specific?


I should think it's more PowerSDR *and* Windows specific.   The processing 
chains in PowerSDR (at least as of a couple months ago) assume that 
transitions between Tx and Rx occur only on buffer boundaries, so the 
PowerSDR program and the dttsp code included in it are essentially "half 
duplex".   Using tiny buffers, fast sample rates,  and a fast computer can 
make the latency in the changeover very small.

Eric's setup has buffers that are about 5 milliseconds long (512 samples * 
10 microseconds/sample)

A setup where you have 2048 sample buffers running at 48 ksps will have 
buffers that are about 40 milliseconds long.  40 milliseconds is a lot of 
delay, and if it actually stacks up with two buffers through the chain, 
it's getting up towards the particularly bad delay range around 100 
milliseconds which is particularly annoying (as anyone who has had poor 
echo suppression on a transoceanic phone call can attest to).


There is also the fact that the SDR1000 hardware itself is inherently half 
duplex, but the changeover speed for that should be very fast (the QSD/QSE 
should enable/disable in sub-microseconds, and there's probably a few 
tens/hundreds of microseconds to just talk to the parallel port). I think 
there's also a relay, which is the dominant speed limeter.

The actual management of the audio interface within the Windows context is 
done by another program called portaudio (someone correct me if I'm wrong 
here) which manages the Windows MultiMediaExtensions API calls and just 
shoves/removes buffers into the dttsp input/output queues.

As you say, there is more than enough hardware horsepower to do full duplex 
with low latency with a much slower processor/data rate.  It's the overall 
combination of PowerSDR/dttsp/portaudio/Windows that makes it a coding 
challenge to get full duplex with deterministic delay times.

And, for what it's worth, I suspect that from a usability standpoint (not 
being a CW op myself, I can only surmise) the important thing is not low 
latency, per se, but a very consistent latency.  After all, having the 
speaker 2 feet away introduces a 2 millisecond delay.  It's that rhythm 
thing that's important, and that's very sensitive to random changes in 
latencies.  I suspect that most ops could handle a fixed 50 or more 
millisecond delay, if it was truly fixed.  The problem in the PowerSDR 
software is that the changeover occurs on buffer boundaries, regardless of 
when you actually stop sending, so there's essentially a random delay of 
somewhere between one buffer and two buffer lengths introduced (depending 
on the relative phase of your QSO back and forth and the buffers going in 
and out of the dsp chain).  It's like trying to play a musical instrument 
in a band where the drummer can't keep time.  (50 milliseconds is about 
like a 1/32nd note at 120 beats/minute)

Jim




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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Alberto I2PHD
James Courtier-Dutton wrote:
> 
> Just out of curiosity, what is the real reason for the latency?
> I use Linux, and users can get high quality, reliable 2ms latency on a
> very modest PC. That 2ms latency is with a Linux application called
> Ardour, that users use for real time audio capture, effects and
> playback. The same latency should be able to be achieved with any Linux
> application.
> 
I know almost nothing about Linux, but a friend of mine who does, once told me 
that the Linux kernel has some long paths 
in the disk I/O management area that run with interrupts disabled. I wonder how 
this could allow to have a "guarantee" 
of 2ms latency. But, as said, my knowledge of Linux is nil.

73  Alberto  I2PHD

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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread James Courtier-Dutton
Eric Wachsmann - FlexRadio wrote:
> Let me give you the skinny right from the horses mouth.  Here's what I
> did:
> 
> I was using a laptop (P4 ~2GHz) with a FireBox.  I set the FireBox to
> 6ms on the FireBox control panel.  This could probably go lower on a
> more powerful machine.  I then set the PowerSDR buffers to 512 for the
> Audio and DSP.  I set the FireBox and PowerSDR to use 96kHz sampling
> rate.  At this speed, our prospective customer was able to send 50WPM
> using the soundcard monitor (not an external keyer). 
> 
> This is the first time that we have tried this experiment at high speed
> since we added 96kHz support.  Previous attempts at this speed all led
> to us directing our customers to use external keyers for high speed CW.
> The long and short of this is that with the right setup (small buffers,
> low latency, fast PC, etc), high speed semi-break in CW is possible with
> the SDR-1000 TODAY!
> 
> 
> Eric Wachsmann
> FlexRadio Systems
> 

Just out of curiosity, what is the real reason for the latency?
I use Linux, and users can get high quality, reliable 2ms latency on a
very modest PC. That 2ms latency is with a Linux application called
Ardour, that users use for real time audio capture, effects and
playback. The same latency should be able to be achieved with any Linux
application.

So, is this latency problem windows specific?

Kind Regards

James



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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Eric Wachsmann - FlexRadio
Let me give you the skinny right from the horses mouth.  Here's what I
did:

I was using a laptop (P4 ~2GHz) with a FireBox.  I set the FireBox to
6ms on the FireBox control panel.  This could probably go lower on a
more powerful machine.  I then set the PowerSDR buffers to 512 for the
Audio and DSP.  I set the FireBox and PowerSDR to use 96kHz sampling
rate.  At this speed, our prospective customer was able to send 50WPM
using the soundcard monitor (not an external keyer). 

This is the first time that we have tried this experiment at high speed
since we added 96kHz support.  Previous attempts at this speed all led
to us directing our customers to use external keyers for high speed CW.
The long and short of this is that with the right setup (small buffers,
low latency, fast PC, etc), high speed semi-break in CW is possible with
the SDR-1000 TODAY!


Eric Wachsmann
FlexRadio Systems

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> radio.biz] On Behalf Of Toby Pennington
> Sent: Sunday, May 21, 2006 2:02 PM
> To: Flex
> Cc: Ron Hinton
> Subject: [Flexradio] Cw Latency Problem Solved?
> 
> I was just told by someone that Gerald made the comment on Teamspeak
last
> night that the cw latency problem had been solved.  Someone is already
> able to send cw at 60 wpm QSK with no latency and has found a way to
do
> this.
> 
> Does anyone else know about this and in particular how is this
> achieved. by a software change or hardware addition.  Also does
anyone
> know about the timeline for this improvement to the SDR 1K.
> 
> Toby  W4CAK
> -- next part --
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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Eric Wachsmann - FlexRadio
Let me qualify that by saying there is no latency in the monitor audio
when you use and external keyer.  What goes out over the air is still a
few milliseconds behind what you hear.

It's also worth noting that the truth is that the SDR-1000 is not (and
never will be without hardware changes) a QSK radio.  Having said that,
it can make one darn good semi break in rig.  :)


Eric Wachsmann
FlexRadio Systems

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> radio.biz] On Behalf Of Gerald Youngblood
> Sent: Monday, May 22, 2006 11:06 AM
> To: 'N3EVL'; 'Ken N9VV'; 'Christopher T. Day'
> Cc: 'Ron Hinton'; 'Flex'
> Subject: Re: [Flexradio] Cw Latency Problem Solved?
> 
--snip--
> 
> Let me reiterate that there is NO latency today on CW when you use an
> external keyer with built in monitor.
> 
> 73,
> Gerald
> 
> Gerald Youngblood, K5SDR
> FlexRadio Systems
//www.flex-radio.com


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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-22 Thread Gerald Youngblood
Hello Flexers,

Let me clarify.  Eric1 set up our laptop demo in the Dayton booth for NA9F,
who sent at up to 50 WPM per his email to me this morning.  Eric set the
Firebox to 96 KHz sampling and 6 ms latency.  The report from Ron was that
he "noticed some latency but not a showstopper for me."  Running this
configuration on a high performance desktop should give better results than
the laptop demo did. 

With 192 KHz sound cards, latency will be cut in half again. PowerSDR
software already supports 192 KHz sampling rates.  We expect to qualify
sound cards or sound card replacements such as "Janus" over the coming
months that will support 192 KHz.  Further, FlexRadio may choose to
manufacture and market Janus or its equivalent as an add on option for the
SDR-1000.  Janus is essentially conceived to be a high performance sound
card with a built in keyer.  

Let me reiterate that there is NO latency today on CW when you use an
external keyer with built in monitor.

73,
Gerald

Gerald Youngblood, K5SDR
FlexRadio Systems
Ph: 512-535-5266
Fax: 512-233-5143
Email: [EMAIL PROTECTED]
Web: www.flex-radio.com
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of N3EVL
> Sent: Sunday, May 21, 2006 6:47 PM
> To: 'Ken N9VV'; 'Christopher T. Day'
> Cc: 'Flex'; 'Ron Hinton'
> Subject: Re: [Flexradio] Cw Latency Problem Solved?
> 
> I seem to remember mention of a solution found right there at 
> Dayton - some tweak to the soundcard config, perhaps, 
> resulting in successful semi-breakin capability at speeds 
> around 60wpm, but not full QSK.  I guess we need Bob or 
> Gerald to spell out the details.
> 
> Pete 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of Ken N9VV
> > Sent: Sunday, May 21, 2006 11:15 PM
> > To: Christopher T. Day
> > Cc: Ron Hinton; Flex
> > Subject: Re: [Flexradio] Cw Latency Problem Solved?
> > 
> > I have listened to that part of the recording again today (now on
> > teamspeak) and he clearly said that "the current problem and 
> > limitation is the SOUNDCARD and not the SDR-1000 hardware".
> > So I guess that means that the HPSDR Janus board is a LIFE 
> SAVER and 
> > may mean some super CW performance from the current SDR-1000 hw?
> > bk de ken
> > 
> > 
> > Christopher T. Day wrote:
> > > Toby,
> > > 
> > > I'm not certain, but I suspect this was a comment about the
> > SDR-X, not
> > > the SDR-1000. I doubt that there is a simple full-QSK 
> solution with 
> > > the latter given its half-duplex design.
> > > 
> > > 
> > >   Chris - AE6VK
> > > 
> > > 
> > > -Original Message-
> > > From: Toby Pennington [mailto:[EMAIL PROTECTED]
> > > Sent: Sunday, May 21, 2006 12:02 PM
> > > To: Flex
> > > Cc: Ron Hinton
> > > Subject: [Flexradio] Cw Latency Problem Solved?
> > > 
> > > I was just told by someone that Gerald made the comment on
> > Teamspeak
> > > last night that the cw latency problem had been solved.  
> Someone is 
> > > already able to send cw at 60 wpm QSK with no latency and
> > has found a
> > > way to do this.
> > > 
> > > Does anyone else know about this and in particular how is this 
> > > achieved. by a software change or hardware addition.  
> Also does 
> > > anyone know about the timeline for this improvement to the
> > SDR 1K.   
> > > 
> > > Toby  W4CAK
> > > -- next part -- An HTML attachment was 
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> > 
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> > > s/20060521/d293d1ed/attachment.html
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> &

Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Frank Brickle
Jimmy Jones wrote:
> I wish some of brain - o's could solve the monitor latency problem.

Well, according to Science Today magazine this week, some string 
theorists are speculating that time travel may actually be feasible.

With that solved the monitor latency problem won't be far behind. Till 
then, unfortunately, the only other solution is ESP.

73
Frank
AB2KT

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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Jimmy Jones
I wish some of brain - o's could solve the monitor latency problem.
  - Original Message - 
  From: Christopher T. Day 
  To: Toby Pennington ; Flex 
  Cc: Ron Hinton 
  Sent: Sunday, May 21, 2006 4:42 PM
  Subject: Re: [Flexradio] Cw Latency Problem Solved?


  Toby,

  I'm not certain, but I suspect this was a comment about the SDR-X, not
  the SDR-1000. I doubt that there is a simple full-QSK solution with the
  latter given its half-duplex design.


  Chris - AE6VK


  -Original Message-
  From: Toby Pennington [mailto:[EMAIL PROTECTED] 
  Sent: Sunday, May 21, 2006 12:02 PM
  To: Flex
  Cc: Ron Hinton
  Subject: [Flexradio] Cw Latency Problem Solved?

  I was just told by someone that Gerald made the comment on Teamspeak
  last night that the cw latency problem had been solved.  Someone is
  already able to send cw at 60 wpm QSK with no latency and has found a
  way to do this.  

  Does anyone else know about this and in particular how is this
  achieved. by a software change or hardware addition.  Also does
  anyone know about the timeline for this improvement to the SDR 1K.   

  Toby  W4CAK
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[Flexradio] CW Latency Problem Solved?

2006-05-21 Thread Toby Pennington
Is the Firefox card on the approved lost to enable warranty guarantees?  Or
are you talking about the Firebox Firewire which is approved?   Toby

- Original Message - 
From: "Mel Whitten" <[EMAIL PROTECTED]>
To: "Toby Pennington" <[EMAIL PROTECTED]>
Sent: Sunday, May 21, 2006 7:03 PM
Subject: Re: [Flexradio] Cw Latency Problem Solved?


> HI Tony
> I was there when the "annoucement" was made.  Seems they (not sure besides
> Eric who found this out) were able to remove the latency in the Firefox
> Soundcard by make some adjustment the card has available.  I dont own the
> card, but that is "all they did" becuase Bob N4HY says the latency problem
> had already been confirmed was caused by the sound card, not the PC/sdr
> console.  So if you have a Firefox card (evidently this is the only card
> that this adjustment can be made on) then this may be the answer.  You may
> want to "get more info on this" to confirm what I am saying.  :-)
> Mel
> k0pfx
>
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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Jim Lux
At 08:15 PM 5/21/2006, Ken N9VV wrote:
>I have listened to that part of the recording again today (now on
>teamspeak) and he clearly said that "the current problem and
>limitation is the SOUNDCARD and not the SDR-1000 hardware".

This is incorrect.  Many, many high performance gaming applications run on 
the Windows platform and achieve sub=millisecond timing. There are also 
realtime conferencing type applications that run in the Windows environment 
with full duplex audio processing.  For that matter, successfully playing 
high rate streaming video/audio requires careful alignment of the streams.

It's more a matter of the huge software development effort required to do 
this within the Windows framework.  Devoting a work-year of effort 
($100K-250K) to learning to effectively work in the Windows environment may 
be a reasonable matter to a game developer expecting million unit sales or 
to a high end video conferencing application developer.  However, that 
scale of labor is not generally available to the folks at Flex.

I would imagine that, by and large, the PowerSDR development folk are not 
Windows multimedia extension devotees, nor do they wish to devote their 
lives to such an activity.  In fact, PowerSDR depends on portaudio to 
encapsulate and hide most of the mm ickyness, and it would appear that the 
portaudio developers aren't interested in this either.

Or, if some ham who also happens to be a game developer gets interested, we 
could be in luck.


>  So I
>guess that means that the HPSDR Janus board is a LIFE SAVER and may
>mean some super CW performance from the current SDR-1000 hw?
>bk de ken

or, just buy a second SDR as a receiver.

Jim 



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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Jim Lux
At 03:22 PM 5/21/2006, Mike Naruta wrote:
>And if you gotta have QSK, when the new software
>comes out, just buy another SDR-1000 receiver
>and you can run full duplex.:)


That is a fairly straightforward approach to the "full duplex" need.  And, 
it pushes the cost onto just those who want the capability (you could 
clearly use a 1Watt version for the "receiver" side, too).

However, it should be pointed out that there is no inherent reason why the 
existing SDR1000 hardware and a PC cannot do microsecond scale QSK 
(depending on how fast the relays are, more than anything).  It's mostly a 
matter of spending the (huge) amount of work involved in making a full 
duplex windows audio application that interacts with the hardware. 
Considering the amount of work involved in doing the latter, and the 
limited software development resources available, it's probably not a good 
investment of those resources.  Personally, I'd rather see a push to a 
better separation between DSP and user interface in the Windows environment 
(as in running as separate tasks/processes), which would be of more general 
applicability than high performance QSK. For example, having a control and 
audio interface that was sufficiently decoupled from the PowerSDR UI would 
allow third party digital modes software to work without things like VAC, 
the use of dual sound cards, or serial port emulators. That was one of the 
high priority things discussed a year or so ago, but which has languished 
as other issues have been attacked.


Jim 



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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread N3EVL
I seem to remember mention of a solution found right there at Dayton - some
tweak to the soundcard config, perhaps, resulting in successful semi-breakin
capability at speeds around 60wpm, but not full QSK.  I guess we need Bob or
Gerald to spell out the details.

Pete 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Ken N9VV
> Sent: Sunday, May 21, 2006 11:15 PM
> To: Christopher T. Day
> Cc: Ron Hinton; Flex
> Subject: Re: [Flexradio] Cw Latency Problem Solved?
> 
> I have listened to that part of the recording again today (now on
> teamspeak) and he clearly said that "the current problem and 
> limitation is the SOUNDCARD and not the SDR-1000 hardware". 
> So I guess that means that the HPSDR Janus board is a LIFE 
> SAVER and may mean some super CW performance from the current 
> SDR-1000 hw?
> bk de ken
> 
> 
> Christopher T. Day wrote:
> > Toby,
> > 
> > I'm not certain, but I suspect this was a comment about the 
> SDR-X, not 
> > the SDR-1000. I doubt that there is a simple full-QSK solution with 
> > the latter given its half-duplex design.
> > 
> > 
> > Chris - AE6VK
> > 
> > 
> > -Original Message-
> > From: Toby Pennington [mailto:[EMAIL PROTECTED]
> > Sent: Sunday, May 21, 2006 12:02 PM
> > To: Flex
> > Cc: Ron Hinton
> > Subject: [Flexradio] Cw Latency Problem Solved?
> > 
> > I was just told by someone that Gerald made the comment on 
> Teamspeak 
> > last night that the cw latency problem had been solved.  Someone is 
> > already able to send cw at 60 wpm QSK with no latency and 
> has found a 
> > way to do this.
> > 
> > Does anyone else know about this and in particular how is this 
> > achieved. by a software change or hardware addition.  Also does
> > anyone know about the timeline for this improvement to the 
> SDR 1K.   
> > 
> > Toby  W4CAK
> > -- next part -- An HTML attachment was 
> > scrubbed...
> > URL:
> > 
> http://mail.flex-radio.biz/pipermail/flexradio_flex-radio.biz/attachme
> > nt
> > s/20060521/d293d1ed/attachment.html
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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Mike Naruta
And if you gotta have QSK, when the new software
comes out, just buy another SDR-1000 receiver
and you can run full duplex.:)


Mike - AA8K


Ken N9VV wrote:
> I have listened to that part of the recording again today (now on 
> teamspeak) and he clearly said that "the current problem and 
> limitation is the SOUNDCARD and not the SDR-1000 hardware". So I 
> guess that means that the HPSDR Janus board is a LIFE SAVER and may 
> mean some super CW performance from the current SDR-1000 hw?
> bk de ken
> 
> 
> Christopher T. Day wrote:
>> Toby,
>>
>> I'm not certain, but I suspect this was a comment about the SDR-X, not
>> the SDR-1000. I doubt that there is a simple full-QSK solution with the
>> latter given its half-duplex design.
>>
>>
>>  Chris - AE6VK

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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Christopher T. Day
Ok, I misremembered. I'll have to review the recording.


Chris - AE6VK


-Original Message-
From: Ken N9VV [mailto:[EMAIL PROTECTED] 
Sent: Sunday, May 21, 2006 8:15 PM
To: Christopher T. Day
Cc: Toby Pennington; Flex; Ron Hinton
Subject: Re: [Flexradio] Cw Latency Problem Solved?

I have listened to that part of the recording again today (now on 
teamspeak) and he clearly said that "the current problem and 
limitation is the SOUNDCARD and not the SDR-1000 hardware". So I 
guess that means that the HPSDR Janus board is a LIFE SAVER and may 
mean some super CW performance from the current SDR-1000 hw?
bk de ken


Christopher T. Day wrote:
> Toby,
> 
> I'm not certain, but I suspect this was a comment about the SDR-X, not
> the SDR-1000. I doubt that there is a simple full-QSK solution with
the
> latter given its half-duplex design.
> 
> 
>   Chris - AE6VK
> 
> 
> -Original Message-
> From: Toby Pennington [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, May 21, 2006 12:02 PM
> To: Flex
> Cc: Ron Hinton
> Subject: [Flexradio] Cw Latency Problem Solved?
> 
> I was just told by someone that Gerald made the comment on Teamspeak
> last night that the cw latency problem had been solved.  Someone is
> already able to send cw at 60 wpm QSK with no latency and has found a
> way to do this.  
> 
> Does anyone else know about this and in particular how is this
> achieved. by a software change or hardware addition.  Also does
> anyone know about the timeline for this improvement to the SDR 1K.   
> 
> Toby  W4CAK
> -- next part --
> An HTML attachment was scrubbed...
> URL:
>
http://mail.flex-radio.biz/pipermail/flexradio_flex-radio.biz/attachment
> s/20060521/d293d1ed/attachment.html
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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Ken N9VV
I have listened to that part of the recording again today (now on 
teamspeak) and he clearly said that "the current problem and 
limitation is the SOUNDCARD and not the SDR-1000 hardware". So I 
guess that means that the HPSDR Janus board is a LIFE SAVER and may 
mean some super CW performance from the current SDR-1000 hw?
bk de ken


Christopher T. Day wrote:
> Toby,
> 
> I'm not certain, but I suspect this was a comment about the SDR-X, not
> the SDR-1000. I doubt that there is a simple full-QSK solution with the
> latter given its half-duplex design.
> 
> 
>   Chris - AE6VK
> 
> 
> -Original Message-
> From: Toby Pennington [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, May 21, 2006 12:02 PM
> To: Flex
> Cc: Ron Hinton
> Subject: [Flexradio] Cw Latency Problem Solved?
> 
> I was just told by someone that Gerald made the comment on Teamspeak
> last night that the cw latency problem had been solved.  Someone is
> already able to send cw at 60 wpm QSK with no latency and has found a
> way to do this.  
> 
> Does anyone else know about this and in particular how is this
> achieved. by a software change or hardware addition.  Also does
> anyone know about the timeline for this improvement to the SDR 1K.   
> 
> Toby  W4CAK
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> http://mail.flex-radio.biz/pipermail/flexradio_flex-radio.biz/attachment
> s/20060521/d293d1ed/attachment.html
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Re: [Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Christopher T. Day
Toby,

I'm not certain, but I suspect this was a comment about the SDR-X, not
the SDR-1000. I doubt that there is a simple full-QSK solution with the
latter given its half-duplex design.


Chris - AE6VK


-Original Message-
From: Toby Pennington [mailto:[EMAIL PROTECTED] 
Sent: Sunday, May 21, 2006 12:02 PM
To: Flex
Cc: Ron Hinton
Subject: [Flexradio] Cw Latency Problem Solved?

I was just told by someone that Gerald made the comment on Teamspeak
last night that the cw latency problem had been solved.  Someone is
already able to send cw at 60 wpm QSK with no latency and has found a
way to do this.  

Does anyone else know about this and in particular how is this
achieved. by a software change or hardware addition.  Also does
anyone know about the timeline for this improvement to the SDR 1K.   

Toby  W4CAK
-- next part --
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[Flexradio] Cw Latency Problem Solved?

2006-05-21 Thread Toby Pennington
I was just told by someone that Gerald made the comment on Teamspeak last night 
that the cw latency problem had been solved.  Someone is already able to send 
cw at 60 wpm QSK with no latency and has found a way to do this.  

Does anyone else know about this and in particular how is this achieved. by 
a software change or hardware addition.  Also does anyone know about the 
timeline for this improvement to the SDR 1K.   

Toby  W4CAK
-- next part --
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