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PostPosted: Tue May 27, 2014 8:17 pm 
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At the new Normal output settings there is a problem with "Input without processing" option . Sound is gagged/strange


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PostPosted: Tue May 27, 2014 8:31 pm 
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I listened to the new AM setting and having listened to the pre-emphasized output of some hardware processors, this one sounds quite nice. The harsh ringing sound isn't there. A couple of thoughts :

1) Part 15 stations aren't limited by the NRSC filter that requires a sharp cut at 10khz (probably also true for full power stations in other countries). Would it be possible to have a 10khz NRSC filter switch that would allow licensed US stations to comply with the NRSC standard yet allow Part 15 stations to operate with full bandwidth pre-emphasis?

2) I find it interesting that when I run the low pass filter setting from 16000 (where I run it on FM) down to 10000, I hear a difference. It sounds notably better at 16000. Does it meet the NRSC 10khz standard with the low pass filter set above 10khz? If so, WOW!
Hi Bob,

Unfortunately no, you have to set it to 10. For the stations that you mentioned, they might then be able to set it higher.. In most countries that I'm aware off the cutoff frequency is 4.5 kHz, which probably explains why AM is nearly dead everywhere else... This also means that all the AM stations that I can receive here stop at 4.5 kHz so I don't know how my processing compares to others... I also don't know the ringing that you mentioned (although I can imagine that bad steep lowpass filter designs can cause bad things).


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PostPosted: Tue May 27, 2014 8:56 pm 

Joined: Mon Feb 04, 2013 11:49 pm
Posts: 187
Quote:
Quote:
I listened to the new AM setting and having listened to the pre-emphasized output of some hardware processors, this one sounds quite nice. The harsh ringing sound isn't there. A couple of thoughts :

1) Part 15 stations aren't limited by the NRSC filter that requires a sharp cut at 10khz (probably also true for full power stations in other countries). Would it be possible to have a 10khz NRSC filter switch that would allow licensed US stations to comply with the NRSC standard yet allow Part 15 stations to operate with full bandwidth pre-emphasis?

2) I find it interesting that when I run the low pass filter setting from 16000 (where I run it on FM) down to 10000, I hear a difference. It sounds notably better at 16000. Does it meet the NRSC 10khz standard with the low pass filter set above 10khz? If so, WOW!
Hi Bob,

Unfortunately no, you have to set it to 10. For the stations that you mentioned, they might then be able to set it higher.. In most countries that I'm aware off the cutoff frequency is 4.5 kHz, which probably explains why AM is nearly dead everywhere else... This also means that all the AM stations that I can receive here stop at 4.5 kHz so I don't know how my processing compares to others... I also don't know the ringing that you mentioned (although I can imagine that bad steep lowpass filter designs can cause bad things).
4.5khz....YUK. That's not far above a land line telephone. AM here sounded better before we were saddled with the 10khz limit. It could sound better than mono FM as there was no pre-emphasis/de-emphasis to deal with. Still, on almost radios, a 10khz limit is adequate....especially without the ringing that was so noticeable on the Optimod 9100. The 9100 sounded great on a typical radio but you didn't want to listen to it on an air monitor without any de-emphasis. The ST version, on the other hand, doesn't sound obnoxious at all...even without de-emphasis. Thanks for adding this...I think it may be time to play with an AM part 15 station again :)


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PostPosted: Wed May 28, 2014 12:10 am 
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4.5khz....YUK. That's not far above a land line telephone.
Actually, it sounds so bad that I remember when I was little and was scanning the FM and also AM dial, I had difficulty understanding speech on AM due to the lack of highs...

Stations are placed 9 kHz apart, without bigger holes in between them (as is apparently the case in the US, the same is true for FM by the way, stations here are 100 kHz apart instead of 200). So, in order to have good reception without catching disturbances of other channels, not only the transmitter but also the receiver must filter out everything above 4.5 kHz. Older analog radio's (like the one I had as a kid) probably didn't have very steep filters, so I guess they would be filtering out a lot below 4.5 kHz already...


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PostPosted: Wed May 28, 2014 12:13 am 
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Joined: Tue Mar 17, 2009 2:56 pm
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Quote:
Actually, it sounds so bad that I remember when I was little and was scanning the FM and also AM dial, I had difficulty understanding speech on AM due to the lack of highs...
That's why you keep treble on Your amplifier at +10 :)


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PostPosted: Wed May 28, 2014 12:20 am 
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Quote:
Quote:
Actually, it sounds so bad that I remember when I was little and was scanning the FM and also AM dial, I had difficulty understanding speech on AM due to the lack of highs...
That's why you keep treble on Your amplifier at +10 :)
Naah. I do that always. Also with FM, or CD's. In fact, years ago I had an equalizer between my other equipment and my amplifier, and I would boost the treble to the max both on the equalizer and on the amplifier. What can I say... I just love highs :)


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PostPosted: Wed May 28, 2014 3:07 am 
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BETA743-003
NOTE: I have NOT fixed any of the reported bugs yet, so you have to assume at this point that this version is very buggy.
I did add a new feature though for which I would like to get some feedback: Output streaming via VLC.

To use this, you must have installed the 32 bit version of VLC Media Player: http://www.videolan.org/vlc/

Stereo Tool Windows 32 bit stand alone version: http://www.stereotool.com/download/ster ... 43-003.exe

How to use it:
- Generate a 'sout' string in VLC (you can also modify the string that's set as default)
- Paste it in Stereo Tool (see Configuration -> Sound Cards -> Normal Output panel)
- Select VLC as output sound card (currently only works for Normal Output)

With the default string
Code:
--sout=#transcode{vcodec=none,acodec=mp3,ab=200,channels=2,samplerate=44100}:std{access=http,mux=raw,dst=127.0.0.1:8080}
you can connect with a separate VLC player by going to Open Network Stream, and then typing http://127.0.0.1:8080/ . The codec in this case is mp3, 200 kbit/s, 2 channels, 44100 Hz.
It should be possible to stream to Shoutcast etc. as well but I have not tested this (among others because I have no experience with it and no server to stream to) - so I hope someone who has more experience can check if it works. See http://shootingstarbbs.blogspot.nl/2010 ... io-to.html for a description on how to do it. They came up with the following sout string:
Code:
sout=#transcode{vcodec= none,acodec=mp3,ab=128,channels=2,samplerate=44100}:std{access=shout,mux=ogg,dst=icandy:password@lema-awards.ipr365.com:1234/icandy}
--sout-keep
I don't know if the --sout-keep is needed, there's currently no way to fill it in in Stereo Tool (I can of course add it if needed).

Known bug: If you close Stereo Tool while output streaming is active, it hangs. Disable the output first to avoid this.


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PostPosted: Wed May 28, 2014 7:42 am 

Joined: Mon Mar 17, 2014 6:42 pm
Posts: 49
Hi Hans,

here are my testresults:

- when using this new version, the stream input vlc doesn't work anymore :(
- when resampling in the VLC streaming output (e.g. 48000 to 44100) it sounds terrible
- local streaming works (127.0.0.1)
- streaming to icecast doesn't work *). Doing it directly from VLC, it works with this SOUT Line:
Code:
:sout=#transcode{vcodec=none,acodec=mp3,ab=256,channels=2,samplerate=48000}:std{access=shout,mux=ogg,dst=source:xxxxxxxxxxx@172.20.11.120:8000/test} :sout-keep
*) Update: It connects to Icecast, the mountpoint is available but there is no data flowing...

Could you imagine to do such an output for the FM also? We could pack it into an 128kHz Mono FLAC Stream, send it to icecast and we have > tadaaaa < a full MPX Stream incl. Stereo and RDS. So at the remote end we may use small Raspberry PIs with USB soundcards for playout.


Christian


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PostPosted: Wed May 28, 2014 9:46 am 
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Joined: Wed Nov 19, 2008 7:44 pm
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I`ve never seen such a plugin or stream solution for 192/128 KHz stream nevermind if it`s Flac or Wave. It can be fitted in 128 KHz because RDS carrier is 57KHz and is with bandwidth +-3Khz , so the last frequency is arround 60KHz , maximum 60.5 . With 128Khz rate we have real 64Khz bandwidth..
This will be a good thing to have an ability to stream the whole MPX signal. If someone have a solution till this moment, let me know please.


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PostPosted: Wed May 28, 2014 11:46 am 
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Quote:
- when using this new version, the stream input vlc doesn't work anymore :(
- when resampling in the VLC streaming output (e.g. 48000 to 44100) it sounds terrible
- local streaming works (127.0.0.1)
- streaming to icecast doesn't work *). Doing it directly from VLC, it works with this SOUT Line:
Code:
:sout=#transcode{vcodec=none,acodec=mp3,ab=256,channels=2,samplerate=48000}:std{access=shout,mux=ogg,dst=source:xxxxxxxxxxx@172.20.11.120:8000/test} :sout-keep
*) Update: It connects to Icecast, the mountpoint is available but there is no data flowing...
Strange, input still works fine here. And I don't hear anything bad when resampling in VLC.

Can you paste the *exact* sout line that you pasted in Stereo Tool? (Control+C works in the ST tekst-fields).
Quote:
Could you imagine to do such an output for the FM also? We could pack it into an 128kHz Mono FLAC Stream, send it to icecast and we have > tadaaaa < a full MPX Stream incl. Stereo and RDS. So at the remote end we may use small Raspberry PIs with USB soundcards for playout.
Definitely worth a test. I only have to add VLC SOUT field for that, except for that everything is already there.


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