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increasing audio bandwidth on ASL3
Greetings:
I've just discovered that, since ASL3 runs modern Asterisk, I can connect ASL3 nodes together using slin16 (raw 16 kHz linear PCM), and both the usbradio and simpleusb drivers do, in fact, allow for that little extra bit of high end to come through the connection. HamVoIP, while it supports G722, which is a 16 kHz codec, hard filters the highs on it's simpleusb channel driver, so there isn't much point other than for compatibility to do this, but ASL3's drivers are capable of passing more highs. This would be especially great for repeater controllers, as with a properly calibrated system, you could achieve audio that sounds more like simplex than it would at 8 kHz. I tried this test with a SHARI connected to a radio-less node. To do this, I put allow => slin16 allow => slin at the top of the codec priority lists in /etc/asterisk/iax.conf under the general and radio stanzas of both nodes. This won't work with HamVoIP or anything older than ASL3. Just FYI. I am way more excited about this than I probably should be, considering it probably won't be widely deployed. 73 N2DYI |
¿ªÔÆÌåÓýI'd love to hear/use an active system consisting of all SLIN16-connected nodes. Please post when this is a thing.
On 7/7/2024 4:21 PM, Carl Swanson via
groups.io wrote:
This is the #1 reason we are overhauling two of our hubs from v2 and installing v3. |
¿ªÔÆÌåÓýInterestingly, that was already enabled on my node. Maybe I did that and forgot to mention it? Not sure.
On 7/8/2024 8:13 AM, Danny Lloyd via
groups.io wrote:
In order for me to use slin16, I had to update modules.conf with the following entry. |
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