Return-Path: Received: from post.thorcom.com (post.thorcom.com [195.171.43.25]) by mtain-mc04.r1000.mx.aol.com (Internet Inbound) with ESMTP id 2046B380000A0; Sun, 6 Jan 2013 08:07:29 -0500 (EST) Received: from majordom by post.thorcom.com with local (Exim 4.14) id 1Trpvt-0005kk-4n for rs_out_1@blacksheep.org; Sun, 06 Jan 2013 13:06:33 +0000 Received: from [195.171.43.32] (helo=relay1.thorcom.net) by post.thorcom.com with esmtp (Exim 4.14) id 1Trpvs-0005kb-8K for rsgb_lf_group@blacksheep.org; Sun, 06 Jan 2013 13:06:32 +0000 Received: from vsmtp21.tin.it ([212.216.176.109]) by relay1.thorcom.net with esmtp (Exim 4.77) (envelope-from ) id 1Trpvn-0000nB-0q for rsgb_lf_group@blacksheep.org; Sun, 06 Jan 2013 13:06:31 +0000 Received: from marco09cqcdi12 (95.244.206.44) by vsmtp21.tin.it (8.6.023.02) id 50DAB36A01923115 for rsgb_lf_group@blacksheep.org; Sun, 6 Jan 2013 14:06:06 +0100 Message-ID: <004a01cdec0e$965b1020$0201a8c0@marco09cqcdi12> From: "Marco Cadeddu" To: References: <871180718.807959.1357202851097.JavaMail.open-xchange@email.1und1.de> <8CFB7BD49F0E58C-6E8-60E@webmail-d164.sysops.aol.com> <4D45055C381D4BECA4B1BEF1254E1059@Black> Date: Sun, 6 Jan 2013 13:06:05 -0000 MIME-Version: 1.0 X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2800.1106 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2800.1106 X-Spam-Score: 2.4 (++) X-Spam-Report: Spam detection software, running on the system "relay1.thorcom.net", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hi Markus, stucked of bad cnd on 472 (Stefan arrives irregularly with -30 dB S/N....) I'm now in rx usb on 136 GL 73 Marco, IK1HSS ----- Original Message ----- From: Markus Vester To: Sunday, January 06, 2013 12:50 PM Subject: LF: Bandlimited SSB test on 136 kHz [...] Content analysis details: (2.4 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 RCVD_IN_DNSWL_NONE RBL: Sender listed at http://www.dnswl.org/, no trust [212.216.176.109 listed in list.dnswl.org] 0.0 FREEMAIL_FROM Sender email is commonly abused enduser mail provider (marcocadeddu[at]tin.it) 0.0 RP_MATCHES_RCVD Envelope sender domain matches handover relay domain 0.0 HTML_MESSAGE BODY: HTML included in message 2.4 AXB_XMAILER_MIMEOLE_OL_A7B9C AXB_XMAILER_MIMEOLE_OL_A7B9C X-Scan-Signature: ad4be66560ed407055baa7738753e104 Subject: LF: Re: Bandlimited SSB test on 136 kHz Content-Type: multipart/alternative; boundary="----=_NextPart_000_0047_01CDEC0E.95D7FD70" X-Spam-Checker-Version: SpamAssassin 2.63 (2004-01-11) on post.thorcom.com X-Spam-Level: X-Spam-Status: No, hits=0.6 required=5.0 tests=HTML_20_30, HTML_FONTCOLOR_UNKNOWN,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 x-aol-global-disposition: G x-aol-sid: 3039ac1d604c50e977117d76 X-AOL-IP: 195.171.43.25 X-AOL-SPF: domain : blacksheep.org SPF : none This is a multi-part message in MIME format. ------=_NextPart_000_0047_01CDEC0E.95D7FD70 Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: quoted-printable Hi Markus, stucked of bad cnd on 472 (Stefan arrives irregularly with -30 dB = S/N....) I'm now in rx usb on 136=20 GL 73 Marco, IK1HSS ----- Original Message -----=20 From: Markus Vester=20 To: rsgb_lf_group@blacksheep.org=20 Sent: Sunday, January 06, 2013 12:50 PM Subject: LF: Bandlimited SSB test on 136 kHz Dear LF, around midday I have been conducting some test transmissions using = 136.0 kHz USB voice. To comply with German regulations, the audio = spectrum has been cut off sharply below 300 Hz and above 1100 kHz, = occupying 136.3 to 137.1 kHz RF. Despite this rather mediocre speech = quality, it is still possible to hear the voice and copy a simple = message.=20 To my surprise, the speech was quite audible on the Twente WebSDR (434 = km), squeezed between the bursts of DCF and HGA. There's a recording and = screenshot at http://www.df6nm.bplaced.net/LF/ssb/ Of course it does get boring speaking only to myself ;-). If you'd = like to listen yourself, I intend to do another test transmission = starting 13:30 this afternoon. As the audio is realtime and standard SSB = (apart from the filtering), no software postprocessing is needed.. Slow voice transmission (ie the audio deceleration/acceleration method = originally used by DK8KW and myself) would be nicer as it can fit a full = SSB channel into 800 Hz. I have been working on a semi-automatic = control, with a fixed one minute raster similar to JT9-1. This will = hopefully allow us to comfortably exchange one 20 second voice message = per time slot (speak during seconds 0 to 20, concurrent transmit and = receive at 1/3 speed from 0 to 60, replay starting 40 to 60). Anyone = interested? Best 73, Markus (DF6NM) =20 ----- Original Message -----=20 From: Markus Vester=20 To: rsgb_lf_group@blacksheep.org=20 Sent: Thursday, January 03, 2013 12:33 PM Subject: Re: LF: SSB - why not go digital?=20 Hi Geri, all, I like the idea of trying voice on MF or even LF, and I was = fascinated by actually hearing Gus talking on 507 kHz, and Geri on LF = slow voice. I admit that I even tried (low-power daytime) SSB above = 135.7 kHz once, and Stefan was actually able to hear and record my = voice, albeit at marginal SNR at 180 km. As my LF TX antenna is fairly = narrowband (Q ~ 200), I had inserted a phase shifter to transform the = low output impedance of the PA into a current source at the feedpoint. The good thing about slow voice is that we also get the SNR gain = associated with the narrower bandwidth. But the non-realtime operating = procedure is sort of difficult. We did use "roger beeps" to mark the end = of message, but there is always an inconvenient delay for recording and = replaying the messages. So some sort of realtime narrowband voice = transmission would indeed be desirable. However I would NOT fancy digital voice modes at all. For one, like = all digital modes it's all-or-nothing. Either the message decodes well, = or you get garbage, and you don't know what has been going on in the = channel. In linear analog SSB, you hear if someone else is calling on = frequency, or what type of interference came up, or whether there is = selective fading. The other thing I don't like about digital modes is = that they tend to occupy the whole channel permanently with high average = power - just look at the spectra of DRM vs AM modulated BC = transmissions. It may well turn out that a 2 kHz linear SSB transmission = is much more friendly than a 1 kHz digital channel.=20 I have been making some attempts on analog quasi-realtime narrowband = voice transmissions. The principle was is much the same that has long = been used for changing audio speed without changing pitch, ie cut the = audio into ~ 20 ms grains. To accellerate or reduce bandwidth, you = either leave out or average several grains, using a sliding window. On = replay, each grain is repeated to extend it in time. This is not = difficult to implement. In my own trials, I could maintain speech = readability with a factor four bandwidth reduction, but it did sound = very "robotic" because the fixed timing for the grains impressed itself = onto the voice pitch. I think that there are better ways of adapting the = timing by tracking the fundamental frequency to preserve the pitch = modulation. Using Windows media player for accelerating and decelerating = seemed to provide quite natural speech quality at a reduction factor of = four. Regarding the frequency allocations, like Graham I would think that = on MF the narrowband beacons should best be placed in narrow slots near = the band edges. To avoid blocking, we would preferably again use split = band for west-east vs east-west intercontinental work, as we do on LF. = Parts of these slots should also be utilized for the slow versions of = Opera or WSPR. The wider modes would better be placed in the in the = middle, perhaps with 2 kHz each for CW, middle-range digital modes, and = one full-width SSB channel. Best 73, Markus (DF6NM) ... ------=_NextPart_000_0047_01CDEC0E.95D7FD70 Content-Type: text/html; charset="utf-8" Content-Transfer-Encoding: quoted-printable =EF=BB=BF
Hi Markus,
 
stucked of bad cnd on 472 (Stefan = arrives=20 irregularly with -30 dB S/N....) I'm now in rx usb on 136
 
GL
73 Marco, IK1HSS
----- Original Message -----
From:=20 Markus=20 Vester
Sent: Sunday, January 06, 2013 = 12:50=20 PM
Subject: LF: Bandlimited SSB = test on 136=20 kHz

Dear LF,
 
around midday I have=20 been conducting some test transmissions using 136.0 kHz = USB=20 voice. To comply with German regulations, the audio = spectrum has=20 been cut off sharply below 300 Hz and above 1100 kHz, = occupying=20 136.3 to 137.1 kHz RF. Despite this rather mediocre speech = quality,=20 it is still possible to hear the voice and copy a simple=20 message. 
 
To my surprise, the speech was = quite=20 audible on the Twente WebSDR (434 km), squeezed between the = bursts=20 of DCF and HGA. There's a recording and screenshot at
http://www.df6nm.bplaced.ne= t/LF/ssb/
Of course it does get boring speaking = only to=20 myself ;-). If you'd like to listen yourself, I intend to do another = test=20 transmission starting 13:30 this afternoon. As the audio is = realtime and=20 standard SSB (apart from the filtering), no software postprocessing is = needed..
 
Slow voice transmission (ie = the audio=20 deceleration/acceleration method originally used by DK8KW and myself) = would=20 be nicer as it can fit a full SSB channel into 800=20 Hz. I have been working on a semi-automatic control,=20 with a fixed one minute raster similar to JT9-1. This will = hopefully=20 allow us to comfortably exchange one 20 second voice message per time = slot=20 (speak during seconds 0 to 20, concurrent = transmit and receive=20 at 1/3 speed from 0 to 60, replay starting 40 to 60). Anyone=20 interested?
 
Best 73,
Markus (DF6NM)
   
----- Original Message -----
From:=20 Markus=20 Vester
To: rsgb_lf_group@blacksheep.org= =20
Sent: Thursday, January 03, = 2013 12:33=20 PM
Subject: Re: LF: SSB - why = not go=20 digital?

Hi Geri, all,
 
I = like the=20 idea of trying voice on MF or even LF, and I was fascinated by = actually=20 hearing Gus talking on 507 kHz, and Geri on LF slow=20 voice. I admit that I even tried (low-power daytime)=20 SSB above 135.7 kHz once, and Stefan was actually able to hear = and=20 record my voice, albeit at marginal SNR at 180 km. As my LF TX = antenna is=20 fairly narrowband (Q ~ 200), I had inserted a phase shifter to=20 transform the low output impedance of the PA into a current source = at the=20 feedpoint.
 
The = good=20 thing about slow voice is that we also get the SNR gain associated = with the=20 narrower bandwidth. But the non-realtime operating procedure = is sort=20 of difficult. We did use "roger beeps" to mark the = end of=20 message, but there is always an inconvenient delay for = recording=20 and replaying the messages. So some sort of realtime narrowband = voice=20 transmission would indeed be desirable.
 
However I=20 would NOT fancy digital voice modes at all. For one, like all = digital=20 modes it's all-or-nothing. Either the message decodes = well, or you get garbage, and you don't know what has been = going on in=20 the channel. In linear analog SSB, you hear if someone else is = calling=20 on frequency, or what type of interference came up, or = whether=20 there is selective fading. The other thing I don't like about = digital=20 modes is that they tend to occupy the whole channel permanently with = high=20 average power - just look at the spectra of DRM vs AM modulated = BC=20 transmissions. It may well turn out that a 2 kHz linear = SSB=20 transmission is much more friendly than a 1 kHz digital = channel. 
 
I = have been=20 making some attempts on analog quasi-realtime narrowband voice=20 transmissions. The principle was is much the same that has long been = used=20 for changing audio speed without changing pitch, ie cut = the audio=20 into ~ 20 ms grains. To accellerate or reduce = bandwidth, you=20 either leave out or average several grains, using a = sliding=20 window. On replay, each grain is repeated to extend it in = time.=20 This is not difficult to implement. In my own trials, I could = maintain=20 speech readability with a factor four bandwidth reduction, but it = did sound=20 very "robotic" because the fixed timing for the grains = impressed=20 itself onto the voice pitch. I think that there are better ways of = adapting=20 the timing by tracking the fundamental frequency to = preserve=20 the pitch modulation. Using Windows media player for=20 accelerating and decelerating seemed to = provide quite natural=20 speech quality at a reduction factor of four.
 
Regarding the=20 frequency allocations, like Graham I would think that on MF the = narrowband=20 beacons should best be placed in narrow slots near the band edges. = To avoid=20 blocking, we would preferably again use split band for = west-east vs=20 east-west intercontinental work, as we do on LF. Parts of these = slots should=20 also be utilized for the slow versions of Opera or WSPR. The = wider=20 modes would better be placed in the in the middle, perhaps = with 2 kHz=20 each for CW, middle-range digital = modes, and one=20 full-width SSB channel.
 
Best=20 73,
Markus=20 (DF6NM)
 ...
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