Return-Path: Received: from post.thorcom.com (post.thorcom.com [195.171.43.25]) by klubnl.pl (8.14.4/8.14.4/Debian-8+deb8u2) with ESMTP id w2O8c6Ph011223 for ; Sat, 24 Mar 2018 09:38:09 +0100 Received: from majordom by post.thorcom.com with local (Exim 4.14) id 1ezeaN-00044V-OL for rs_out_1@blacksheep.org; Sat, 24 Mar 2018 08:31:35 +0000 Received: from [195.171.43.32] (helo=relay1.thorcom.net) by post.thorcom.com with esmtp (Exim 4.14) id 1ezeaB-00044M-KZ for rsgb_lf_group@blacksheep.org; Sat, 24 Mar 2018 08:31:23 +0000 Received: from mail-wm0-f50.google.com ([74.125.82.50]) by relay1.thorcom.net with esmtps (TLSv1.2:ECDHE-RSA-AES256-GCM-SHA384:256) (Exim 4.89) (envelope-from ) id 1ezea7-0004DU-8b for rsgb_lf_group@blacksheep.org; Sat, 24 Mar 2018 08:31:22 +0000 Received: by mail-wm0-f50.google.com with SMTP id f125so7234198wme.4 for ; Sat, 24 Mar 2018 01:31:18 -0700 (PDT) X-DKIM-Result: Domain=gmail.com Result=Good and Known Domain DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20161025; h=mime-version:in-reply-to:references:from:date:message-id:subject:to :cc; bh=VWnhDtMqVX3azPSpYi1vI9xKW56jOohysNhsTIg4fRA=; b=StPr1FjaZXsU+u6rTpK3rkMGW5k53zkhlw6FQNygjrtQObZIbjvpW9bDIumhbyYKeg 5fpUtZLMVpfvb+/MwuXxxOKuSHDhblXqcYc0HCljr8hYprkHhcVyJXykcCcT5ZVkl5mg SmFWN0d2xTNnW6rgVFW3jfDfL3ZTzidlIghfULiJuEKoDW0VKjgeci71Bz+3EuGMM4MK ntgQ1PHbuZE4Xq/mHuUv4ntIO3yebqyRygA3sw49aIz5OsqTutCkHYzz5fyc5CgTcs6s tnBPkYX00VKXLMqjETMU1F6iBpdznlr/3NJJtwNYHdqQxMF0i0nyhiRHPuTmc38lJRep WciQ== X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=1e100.net; s=20161025; h=x-gm-message-state:mime-version:in-reply-to:references:from:date :message-id:subject:to:cc; bh=VWnhDtMqVX3azPSpYi1vI9xKW56jOohysNhsTIg4fRA=; b=NdG3T2R2tm+Jl1lhXtwiusUVTESoA9qU79vGbf9NoM+nJtgFg6S+lgMU/PSJrw8jvU FwsJDKjBctTUhOLs0n9A2mhUQMTeVMQL48e6+KwCjNW1JoTpW+ouN1c7ohqCAONnhuPp oACfhiyKBSlnDcTSf9YTcUidL4nij8FsSz0DUB387+5GLUU5RhN4BQv8KQxE8nvqIR4i q/fjyRqucOPoKAQMg0y+ZVOSTVAN7cP0HI+4FE1wyFwdIcthLoRDhZx2w+zE6DxNf2S/ DOzbuDQ2TuB/3JKjORs65fjZy/Iz0g5CNBufsUr9NMHeC9028pdFb/rHZDJWpBPI8Q7Q afQw== X-Gm-Message-State: AElRT7GmOMfFiPsZss6mQMwRhg+T5Df2Qhs9g85Fl9S6rUjVRg6uK/xE JwEpDOQDmEmOjDC5ytAq9VjcG3M4JQJ5cWSGWySMcv2G X-Google-Smtp-Source: AG47ELv7sEX7B/BarSCcjA0ygwjzp/JjKm/BRcWAT/N+u9/2XK1ZxfDzHELLOJ3i6BkaD+PrviFvV9JYBT3MpegIQN8= X-Received: by 10.80.144.54 with SMTP id b51mr29479611eda.194.1521880218010; Sat, 24 Mar 2018 01:30:18 -0700 (PDT) MIME-Version: 1.0 Received: by 10.80.154.195 with HTTP; Sat, 24 Mar 2018 01:30:17 -0700 (PDT) In-Reply-To: <702478716.20180324081432@gmail.com> References: <702478716.20180324081432@gmail.com> From: Andy Talbot Date: Sat, 24 Mar 2018 08:30:17 +0000 Message-ID: To: LineOne Cc: Chris Wilson X-Spam-Score: 1.0 (+) X-Spam-Report: Spam detection software, running on the system "relay1.thorcom.net", has NOT identified this incoming email as spam. The original message has been attached to this so you can view it or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I made a custom LF receiver and digitiser, then send the I/Q sampled at 1kHz to the PC on an RS422 interface. It's described here http://www.g4jnt.com/Coherent_LF_Receiver.pdf As the sampling is at 1kHz I had to write my own software to decimate and generate the .WAV files that Ebnaut requires. But do remember I successfully decoded some Ebnaut tests on 137kHz. The A/D converter is only 10 bits, but the way the I/Q sampling is done itself elevates that to 12 bits then decimation further down the line does the rest. [...] Content analysis details: (1.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.0 FREEMAIL_FROM Sender email is commonly abused enduser mail provider (andy.g4jnt[at]gmail.com) 0.0 HTML_MESSAGE BODY: HTML included in message 0.0 T_DKIM_INVALID DKIM-Signature header exists but is not valid 1.0 FREEMAIL_REPLY From and body contain different freemails X-Scan-Signature: 80dfd4418e586cee7526d670ae10acbc Subject: Re: LF: Adaptive variable resampling Content-Type: multipart/alternative; boundary="f403045c2584c908dd0568245936" X-Spam-Checker-Version: SpamAssassin 2.63 (2004-01-11) on post.thorcom.com X-Spam-Level: X-Spam-Status: No, hits=0.1 required=5.0 tests=HTML_FONTCOLOR_UNSAFE, HTML_MESSAGE autolearn=no version=2.63 X-SA-Exim-Scanned: Yes Sender: owner-rsgb_lf_group@blacksheep.org Precedence: bulk Reply-To: rsgb_lf_group@blacksheep.org X-Listname: rsgb_lf_group X-SA-Exim-Rcpt-To: rs_out_1@blacksheep.org X-SA-Exim-Scanned: No; SAEximRunCond expanded to false --f403045c2584c908dd0568245936 Content-Type: text/plain; charset="UTF-8" Content-Transfer-Encoding: quoted-printable I made a custom LF receiver and digitiser, then send the I/Q sampled at 1kHz to the PC on an RS422 interface. It's described here http://www.g4jnt.com/Coherent_LF_Receiver.pdf As the sampling is at 1kHz I had to write my own software to decimate and generate the .WAV files that Ebnaut requires. But do remember I successfully decoded some Ebnaut tests on 137kHz. The A/D converter is only 10 bits, but the way the I/Q sampling is done itself elevates that to 12 bits then decimation further down the line does the rest. Like you, I distrust sound cards, and all this over-complex injection of calibration signals to make a silk purse out of a sow's ear seems the wrong route. Other people have made digitisers, and I believe G3PLX or some associates of his did manage to come up with a proper coherent locked digitiser that emulated a soundcard. Andy G4JNT On 24 March 2018 at 08:14, Chris Wilson wrote: > > > Hello LF'ers, > > I am taking more of an interest in Ebnauat, and was wondering about > hardware versus software locking of the PC clock to the sound card > clock. Firstly, are there motherboards and sound cards available that > will take a common external reference? Are they very expensive if > available? > > Secondly, does Spectrum Lab allow software "locking" and is it similar > to this latest addition to HPSDR that's just been announced? the > explanation just posted on the Apache radios forum states: > > "Monitoring and optimization is a nice benefit; however, it=E2=80=99s not= the > primary function of this feature. > > > > Digital transfer of data between radio and computer is desirable > because it introduces NO noise or distortion (unlike analog > interfaces); it=E2=80=99s also simpler hardware-wise. However, there=E2= =80=99s a > problem. The SDR hardware has a clock and the computer/sound-card also > has a clock. Even if both those clocks are supposedly creating, for > example, a 48Khz sample-rate to interchange data between radio and > computer, they are not locked together. Therefore, for example, the > radio might be generating samples at 48.0001Khz and the sound-card > might be absorbing them at 47.9999Khz, or something like that. With > such scenarios, there will eventually be buffer under-runs or > over-runs because samples are not being generated at the same rate > they are being consumed. When such an under-run/over-run occurs, there > will be a discontinuity/glitch/drop-out in the data stream. If those > don=E2=80=99t happen too often, they aren=E2=80=99t terribly noticeable. = However, for > those who expect pure audio and digital emissions, they aren=E2=80=99t ve= ry > acceptable. There is a hardware solution: One of our developers has > purchased a computer sound card that can be locked to an external > clock reference, as can the radio. So, when both are locked to the > same reference, the problem goes away. However, that turns out to > require additional hardware and to be a rather expensive solution. > > > > The Adaptive Resampler continually compares the rates at which samples > are being produced and consumed and it calculates a resampling ratio. > The incoming samples are then, mathematically, used to recreate a > continuous waveform and then samples with a slightly different > spacing, corresponding to the output rate, are taken from that > continuous waveform. Thus, the waveform has been resampled. This > matches the two different sample rates. > > > > There are some open-source software solutions out there for variable > or adaptive resampling. However, our solution has an EXTREMELY low > spur level, making it suitable for I-Q data as well as audio. For > audio, most would say if the spurs are 60dB+ down, that=E2=80=99s plenty = good > enough. We do MUCH better than that, perhaps a couple hundred dB down, > more than you need. There are also some other somewhat unique features > of this implementation. > > > > Hope that helps. > > > > 73, > > Warren NR0V" > > > Thanks! > > > > > > -- > Best regards, > Chris mailto:dead.fets@gmail.com > > > --f403045c2584c908dd0568245936 Content-Type: text/html; charset="UTF-8" Content-Transfer-Encoding: quoted-printable
I made a custom LF receiver and digitiser, then send the I= /Q sampled at 1kHz to the PC on an RS422 interface.

As the = sampling is at 1kHz I had to write my own software to decimate and generate= the .WAV files that Ebnaut requires.=C2=A0 But do remember I successfully = decoded some Ebnaut tests on 137kHz.=C2=A0 =C2=A0The A/D converter is only = 10 bits, but the way the I/Q sampling is done itself elevates that to 12 bi= ts then decimation further down the line does the rest.

Like you, I distrust sound cards, and all this over-complex injection= of calibration signals to make a silk purse out of a sow's ear seems t= he wrong route.=C2=A0 =C2=A0 Other people have made digitisers, and I belie= ve G3PLX or some associates of his did manage to come up with a proper cohe= rent locked digitiser that emulated a soundcard.

A= ndy=C2=A0 G4JNT=C2=A0

On 24 March 2018 at 08:14, Chris Wilson = <dead.fets@gmai= l.com> wrote:


Hello=C2=A0 LF'ers,

I=C2=A0 am=C2=A0 taking=C2=A0 more of an interest in Ebnauat, and was wonde= ring about
hardware=C2=A0 versus=C2=A0 software=C2=A0 locking=C2=A0 of the PC clock to= the sound card
clock.=C2=A0 Firstly, are there motherboards and sound cards available that=
will=C2=A0 take=C2=A0 a=C2=A0 common=C2=A0 external=C2=A0 reference? Are th= ey very expensive if
available?

Secondly, does Spectrum Lab allow software "locking" and is it si= milar
to=C2=A0 this=C2=A0 latest=C2=A0 addition=C2=A0 to=C2=A0 HPSDR=C2=A0 that&#= 39;s just been announced? the
explanation just posted on the Apache radios forum states:

"Monitoring and optimization is a nice benefit; however, it=E2=80=99s = not the
primary function of this feature.



Digital transfer of data between radio and computer is desirable
because it introduces NO noise or distortion (unlike analog
interfaces); it=E2=80=99s also simpler hardware-wise. However, there=E2=80= =99s a
problem. The SDR hardware has a clock and the computer/sound-card also
has a clock. Even if both those clocks are supposedly creating, for
example, a 48Khz sample-rate to interchange data between radio and
computer, they are not locked together. Therefore, for example, the
radio might be generating samples at 48.0001Khz and the sound-card
might be absorbing them at 47.9999Khz, or something like that. With
such scenarios, there will eventually be buffer under-runs or
over-runs because samples are not being generated at the same rate
they are being consumed. When such an under-run/over-run occurs, there
will be a discontinuity/glitch/drop-out in the data stream. If those
don=E2=80=99t happen too often, they aren=E2=80=99t terribly noticeable. Ho= wever, for
those who expect pure audio and digital emissions, they aren=E2=80=99t very=
acceptable. There is a hardware solution: One of our developers has
purchased a computer sound card that can be locked to an external
clock reference, as can the radio. So, when both are locked to the
same reference, the problem goes away. However, that turns out to
require additional hardware and to be a rather expensive solution.



The Adaptive Resampler continually compares the rates at which samples
are being produced and consumed and it calculates a resampling ratio.
The incoming samples are then, mathematically, used to recreate a
continuous waveform and then samples with a slightly different
spacing, corresponding to the output rate, are taken from that
continuous waveform. Thus, the waveform has been resampled. This
matches the two different sample rates.



There are some open-source software solutions out there for variable
or adaptive resampling. However, our solution has an EXTREMELY low
spur level, making it suitable for I-Q data as well as audio. For
audio, most would say if the spurs are 60dB+ down, that=E2=80=99s plenty go= od
enough. We do MUCH better than that, perhaps a couple hundred dB down,
more than you need. There are also some other somewhat unique features
of this implementation.



Hope that helps.



73,

Warren=C2=A0 NR0V"


Thanks!





--
Best regards,
=C2=A0Chris=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 = =C2=A0 =C2=A0 =C2=A0 =C2=A0 mailto:d= ead.fets@gmail.com



--f403045c2584c908dd0568245936--