[arm-allstar] Audio processing, was Re: cdm1250 transmit audio
Chris Smart
csmart8 at cogeco.ca
Tue Mar 3 04:18:35 EST 2020
Thanks David!
I am keeping a copy of this! 😀
Keeping in mind I am a newbie to all this, how many places in the various .conf files do I need to comment out the lesser codecs? (GSM) and leave .sln uncommented? I gather that doing so will force SLN to be used?
Thanks again and 73!
Chris, ve3rwj
Sent from Chris's iPhone.
> On Mar 3, 2020, at 04:26, David McGough via ARM-allstar <arm-allstar at hamvoip.org> wrote:
>
>
> Patrick,
>
> There seems to be some confusion here.
>
> As you comment, ALL audio for Asterisk/AllStar/HamVoIP is sampled at 8KHz.
> Physics (google the Nyquist-Shannon sampling theorem) states that there is
> a hard limit where frequency response can be no more than 1/2 the sampling
> rate. So, the theoretical frequency limit is 4 KHz.
>
> The only audio Low or High pass filtering that occurs in AllStar is in the
> the channel drivers which directly interface with analog audio. Two of
> these drivers are chan_simpleusb and chan_usbradio. Most other channel
> drivers which deal only with digital audio pass-thru have no filters
> (there is no need). Some of these examples are: chan_echolink, chan_irlp,
> chan_usrp, chan_iax2, chan_sip.
>
> There is NO low/high pass filtering in the app_Rpt module itself. There is
> a notch filter capability, that's all. The app_Rpt notch filter code
> implements an on-demand configured IIR type notch filter. These filters
> can and will RING, so you must be VERY careful how you define them. They
> can generate all kinds of odd sounding audio artifacts.
>
> In the HamVoIP chan_simpleusb driver, all the high-pass filters can be
> disabled (turn off the PL, DCS and pre-emphasis filters). The low-pass
> de-emphasis filter can also be turned off.
>
> There is only a single always-on, brick-wall FIR low-pass filter. It is
> used for anti-aliasing and is not adjustable, with a corner frequency at
> 3.5KHz, and ultimate attenuation of -50dB at 4KHz. I carefully
> implemented this filter (it is unique to HamVoIP) over weeks of testing
> and it provides excellent performance. The actual filter is part of the
> interpolator and decimator DSP routines in simpleusb. This code is
> responsible for up and down sampling between the 8KHz sampling rate used
> by AllStar and the 48KHz native sampling rate used by the CM108/CM119
> audio chips. Here is a graphic showing the filter response:
>
> https://hamvoip.org/docs/FIR-LP-simpleusb-2018-05-27-1.png
>
>
> Finally, the asterisk CODEC you choose will significantly impact frequency
> response and dynamic range. If you want the VERY BEST, use the
> signed-linear (SLN) CODEC. This does no compression and sends full 16-bit,
> 8KHz samples between systems, using significant network bandwidth, of
> course. This should provide dynamic range and low-end frequency response
> only limited by the CM108/CM119 sound chips---which is probably somewhere
> above 70dB and below 30Hz. This is way, way more performance than even
> available with broadcast FM radio, which is "only" in the 50dB range.
>
>
> AllStar is truly a STAR performer, when it comes to audio quality.
>
>
>
> 73, David KB4FXC
>
>
>
>
>> On Tue, 3 Mar 2020, "Patrick Perdue via ARM-allstar" wrote:
>>
>> The entire system runs at 8 kHz sampling rate. Even if the rate was
> increased, there is filtering happening in app.rpt. Doug could give you
> more details about exactly how that low pass filter works.
>
> As you're going through two radios to get to the network, if you're not
> using a radio-less node or something like iaxrpt, which has it's own
> non-defeatable high pass filter, each of those radios will apply their
> own characteristics.
>
> Using the G7RPG microhub and it's Baofeng node vs. my Motorola SM50, I
> can say that the Motorola does let some more highs through in both
> directions, but not all that much, even on simplex when monitored
> through discriminator output with an SDR. It is definitely better on the
> low end for both receiving and transmitting, and has a much, much lower
> noise floor on the audio path. Re the Motorola, I wish I could turn off
> all the software filtering of SimpleUSB, notch out my PL tone frequency,
> and let it roar, but when I tried that, DTMF detection totally failed.
> Just using an old Wouxun HT, people did notice a difference when I took
> the PL filter away, especially those with good speakers, but most people
> probably wouldn't care.
>
> I have also plugged an Alinco EMS-57 clone microphone into my URI,
> unfortunately without the added op-amp of the URI or another preamp to
> get it closer to line level, so I had to turn RXBoost on and menu 2 to
> 999 just to be heard, and the mic doesn't sound that great anyway, but
> you do get the idea that it's a fuller sound.
>
> I'd like to use a better mic and some processing, but that's a project
> for when I actually have a permanent place to live, I.E. not now.
>
>
>> On 3/2/2020 7:18 PM, "Chris Smart via ARM-allstar" wrote:
>>
>> I have a much more basic audio question.
>>
>> what in an Allstar node chops things off around 3K? Is it the Baofeng
> radio module in most nodes, or is it something to do withthe fact that
> we're using pbx sofftware, so it treats everything as telephone audio?
>>
>>
>>
>> Sent from Chris's iPhone.
>>
>>
>>>> On Mar 2, 2020, at 16:51, Patrick Perdue via ARM-allstar <arm-allstar at hamvoip.org> wrote:
>>>
>>> I've gotten quite good results using hardware vocoders. DMR is far more robust than many give it credit for, especially if you don't use the software md380-emu implementation. Even some background is tolerated, which is pretty awesome when you consider the limited bandwidth and complexity of the DV modes. I'm always interested in improving audio wherever it can be improved, however.
>>>
>>>
>>>> On 3/2/2020 3:12 PM, "Doug Crompton via ARM-allstar" wrote:
>>>> Patrick,
>>>>
>>>> First of all this is probably not happening soon. There are many other
>>>> more important things on the list. Also it is almost impossible to improve
>>>> something like crappy digital audio. If there was it would already be done.
>>>> My suggestion is don't use it! But many people put up with it as they think
>>>> it is some kind of advance in technology. Funny what you can seel people!
>>>>
>>>>
>>>> *73 Doug*
>>>>
>>>> *WA3DSP*
>>>>
>>>> *http://www.crompton.com/hamradio <http://www.crompton.com/hamradio>*
>>>>
>>>> On Mon, Mar 2, 2020 at 3:01 PM "Patrick Perdue via ARM-allstar" <
>>>> arm-allstar at hamvoip.org> wrote:
>>>>
>>>>> I'm definitely looking forward to any future audio processing support,
>>>>> especially if SoX is involved, since we could then use ladspa. I would
>>>>> especially like to process analog input to clean things up for digital
>>>>> bridges. I'm thinking downward expander plus gate, so the digital
>>>>> vocoders have less noise to process, especially with strong to moderate
>>>>> analog signals, EQ and limiter. This could also help with the ever
>>>>> present echo link 8-bit dithering. This is something I've wanted to do
>>>>> for a while. Having a moderate downward expander and EQ or light
>>>>> multi-band compression on my transmitted audio to Allstar would be nice
>>>>> as well. Just something subtle to keep levels and equalization in line
>>>>> between radios, but not to be obvious that there is processing.
>>>>>
>>>>>
>>>>> On 3/1/2020 10:23 PM, "Doug Crompton via ARM-allstar" wrote:
>>>>>> Jed,
>>>>>>
>>>>>> Pre and de-emphasis are not meant to be used when they are applied to
>>>>> the
>>>>>> audio to/from the transceiver. Using them to make things sound more or
>>>>> less
>>>>>> bright when they have already been applied is not a good idea especially
>>>>> on
>>>>>> audio going out to the network. Pre-emphasis increases the high end at a
>>>>>> prescribed rate for audio FROM your node, what you hear. De-emphasis
>>>>>> decreases high end on audio TO your node, what others hear. Most of us
>>>>> are
>>>>>> using radios that do this internally and leave these settings OFF. If you
>>>>>> were using a radio that did not apply one or the other or both you would
>>>>>> turn the appropriate one on. The bottom line is that they are not meant
>>>>> for
>>>>>> casual audio equalization but rather for a defined method every FM signal
>>>>>> uses.
>>>>>>
>>>>>> It is in our plans and sometime in the future you will be able to apply
>>>>>> software equalization using an external program such as sox but for ow if
>>>>>> you must change equalization your best bet is feeding the analog audio
>>>>> out
>>>>>> of the FOB through a hardware equalizer.
>>>>>>
>>>>>>
>>>>>> *73 Doug*
>>>>>>
>>>>>> *WA3DSP*
>>>>>>
>>>>>> *http://www.crompton.com/hamradio <http://www.crompton.com/hamradio>*
>>>>>>
>>>>>>
>>>>>> On Sun, Mar 1, 2020 at 5:00 PM "Jed Barton via ARM-allstar" <
>>>>>> arm-allstar at hamvoip.org> wrote:
>>>>>>
>>>>>>> Hey guys,
>>>>>>>
>>>>>>> So i'm trying to do some audio adjustment on my cdm1250 to make it
>>>>>>> sound better. I'm trying to figure out the pre-emphasis and
>>>>>>> deemphasis on the radio. I wanna add a little more high end to the
>>>>>>> transmit audio. Can i try this in simple USB, or where should i go?
>>>>>>>
>>>>>>> Cheers,
>>>>>>>
>>>>>>> Jed
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> ARM-allstar mailing list
>>>>>>> ARM-allstar at hamvoip.org
>>>>>>> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar
>>>>>>>
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>>>>>>>
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>>>>>>
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>>>>>>
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>>>>> _______________________________________________
>>>>>
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>>>>>
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