search for: 20hz

...opus Encoded with libopus 1.3, libopusenc 0.2.1 User comments section follows... ENCODER=opusenc from opus-tools 0.2-3-gf5f571b ENCODER_OPTIONS=--bitrate 56 --vbr --comp 5 ALBUM=Test-Sweeps ARTIST=Audacity COMMENT=60s logarithmic sweep (0-20kHz): 5s=10Hz, 10s=20Hz, 15s=39Hz, 20 s=78Hz, 25s=156Hz, 30s=312Hz, 35s=625Hz, 40s=1.25kHz, 45s=2.5kHz, 50s=5kHz, 55s= 10kHz, 60s=20kHz COMMENTS=60s logarithmisch DATE=2016 TITLE=Sweep (0-20kHz at 96kHz) log TRACKNUMBER=2 Opus stream 1: Pre-skip: 312 Playback gain: 0 dB...

...t the minimum latency of the Opus codec is 2.5ms (encoder) + 2.5ms (decoder) = 5ms? Is there any way to go down to, say 1.25ms frame size? One characteristic of wireless radio mic links, if that helps, is that they usually only need a frequency response from 50Hz to 20KHz rather than the usual 20Hz to 20KHz. Bending the code out of spec is acceptable in this particular application because it is normal for radio mic transmitters to be paired with matching receivers from the same manufacturer. I would be interested to hear the opinion of the developers to get an idea if codec could be used...

...bis without problems. Can you please post the original wav? I am not sure what Audacity means by a logarithmisch sweep. Is that a fixed number of Hertz per second (SoX calls that linear)? Or a fixed number of semitones per second (SoX calls that exponantional)? (The ogg comment says 5s=10Hz, 10s=20Hz, 15s=39Hz, 20s=78Hz, 25s=156Hz, 30s=312Hz, 35s=625Hz, 40s=1.25kHz, 45s=2.5kHz, 50s=5kHz, 55s=10kHz, 60s=20kHz so it seems the frequency rises logarithmically.) $ sox -c 1 -r 96k -b 16 -n /tmp/sweep.wav synth 60 sin create 1-20 gain -3 > With Opus I noticed that the file size for 48kHz and 48 k...

...bis without problems. Can you please post the original wav? I am not sure what Audacity means by a logarithmisch sweep. Is that a fixed number of Hertz per second (SoX calls that linear)? Or a fixed number of semitones per second (SoX calls that exponantional)? (The ogg comment says 5s=10Hz, 10s=20Hz, 15s=39Hz, 20s=78Hz, 25s=156Hz, 30s=312Hz, 35s=625Hz, 40s=1.25kHz, 45s=2.5kHz, 50s=5kHz, 55s=10kHz, 60s=20kHz so it seems the frequency rises logarithmically.) $ sox -c 1 -r 96k -b 16 -n /tmp/sweep.wav synth 60 sin create 1-20 gain -3 > With Opus I noticed that the file size for 48kHz and 48 k...

...-- Other countries snipped ---> [sg] ; Singapore section borrowed from http://csusap.csu.edu.au/~whaase01/itc308/asterisk/indications.conf description = Singapore ; Singapore ; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial = 425 ring = 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% busy = 425/750,0/750 congestion = 425/250,0/250 callwaiting = 425*24/300,0/200,425*24/300,0/3200 s...

Hello all, In changing the phone connected to my IAXy I remembered that this phone will not ring on the standard frquency (20hz?) when connected to the FXS. There is a patch, one line to change in wcfxs.c for use with a TDM400P, but obviously this doesn't change the FXS on the IAXy. Is there any possibiity of a "patch" for the provisioning data to raise the frequency to 25hz on the IAXy FXS? Maybe this is not...

...ying to get Asterisk to emulate the sounds of the earlier telephone systems, and the settings in [us-old] are pretty helpful. The only thing lacking is ringback tone, which is not quite as complex as the real phone systems of the day. For example, it is true that a ringback tone commonly used is 420Hz modulated by 40Hz. This is what shows up in [us-old]. But that modulated tone was generally overlaid on top of real ringing, i.e. 20Hz. So, using the Asterisk example of 420*40, it would seem that a decent ringback would be (420*40)*20. But, of course, that doesn't appear to exist. If it d...

...is never going to be great, but I should be able to at least make it bearable. What's the prevailing opinion on using high and low pass filters? One would assume a phone handset is expected to provide frequency response in human speech zones, and not really much outside that (certainly not the 20hz-20khz one might expect of a CD). Suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons

Shane Wegner wrote: > Hello all, > > I have two handsets connected to FXS ports on a TDM400P, > both GE models but one rings and the other does not. The > phone models are not identical. The phone which doesn't > ring on the TDM does ring when connected to a regular POTS > line and I tried connecting another phone to the port and > it rings fine. Do you have the

Hi. I gave up with the IBM NetFinity, so I'm going to buy new hardware. I'm going to install: 1-)One X100P (1 FXO module) 2-)One TDM03B (3 FXO modules) I'll have the 4 FXO channels busy almost all the time, and I would like quality to be as good as possible without going to the high-level hardware. I would like to learn of some tested configurations (I've heard of problems

I hooked up a SPA-1001 with asterisk yesterday and all works well except the phone doesn't ring. The phone I'm using has a LCD display so I can see the call come in. (with caller id info) I can answer and complete the call but it's just not ringing. The phone rings if pluged into a POTS line so it's not the phone that's the problem. I've used the SPA-1001's web

I have a normal backup phone (and an alarm panel) sharing the POTS line with the Digium X101P FXO: | | Wall |>---+------X101P FXO as Zap/5 | | | Phone & Alarm Whenever the Phone is used, Asterisk sees a 'false ring' signal immediately when the phone is hung up. The Alarm panel dials out nightly at around 1AM, and each time it completes the call, Asterisk

All- I'm attempting to set up a Sipura SPA-3000 in Sao Paolo, Brazil. Not being a portuguese speaker, I'm having a rough time of finding the relevant information on how to make the thing pick up the PSTN line and make an outbound call. The sipura in question works fine on a bench connected to a POTS line in the US, but is now plugged in in Sao Paolo. The immediate thing I notice is that

...ODER=opusenc from opus‑tools 0.2‑3‑gf5f571b >> >> > ENCODER_OPTIONS=‑‑bitrate 56 ‑‑vbr ‑‑comp 5 >> >> > ALBUM=Test‑Sweeps >> >> > ARTIST=Audacity >> >> > COMMENT=60s logarithmic sweep (0‑20kHz): 5s=10Hz, 10s=20Hz, >> > 15s=39Hz, 20 >> >> > s=78Hz, 25s=156Hz, 30s=312Hz, 35s=625Hz, 40s=1.25kHz, 45s=2.5kHz, > 50s=5kHz, >> > 55s= >> >> > 10kHz, 60s=20kHz >> >> > COMMENTS=60s logarithmisch >> >> > DATE=2016 >&g...

Just when I thought I couldn't be wrong, I was wrong. We have woodpeckers that drill into the arial telephone cables, and water seeps through the holes and partially grounds the tip and/or ring wires causing hum. I thought the hum/buz on my lines was a telco problem. The Qwest HQ noise team assures me that my lines are within spec. Sure enough, when I listen on the test set the lines

I was recently talking to a friend of mine who produces what's widely referred to as "techno", music that's heavy on electronically generated loops and effects, samples, and has plenty of bass. He doesn't like ogg for a few reasons - he sent me a rather long email, and I've stripped out the nonsense from it (he has some technically irrelevant reasons for not liking