Audio "popping" sound during broadcast

[miken2av]

We did our first webcast using Meetinghouse Webcast Communicator. We are using the microphone, preamp, and cables all purchased from the church web site. The audio works great until music is introduced. Organ, piano, and in our case yesterday a flute solo. Once the flute solo started a clicking or popping sound started. Not real loud but definetly there and annoying.

I tried the suggestions on the wiki site titled:
I hear a soft “clicking or popping” noise in the background of the webcast. How can I fix that?

those suggestions did not help.
tried different microphone and cables.
still no joy.

any ideas appreciated.

[edwardwhite442]

Did the popping go away once the music stopped? And was the music on a different mic channel? Music and spoken voice are heard on different frequencies. Perhaps your microphone/amp was clipping (trying to operate outside it's intended range in either freq, or decibels) during the music portion only??

Just a thought...

[miken2av]

Once the popping starts it never stops. However it is not as pronounced during just speaking. Preamp is way below clipping. We are currently only using one microphone. I am actually sitting here in the broadcast right now. (Aug 21, 2011-11:07am) Most units only hear the popping during music. I am monitoring our broadcast on my laptop with a tether connection so as not to take away bandwidth from building internet. I can hear the same thing the units are reporting. Just a click click click... about every 2 seconds I guess.

[lloydburch]

We have also had the popping noise. It can be clearly heard on any constant tones such as an organ. This has happened the last two times we have done a webcast from the Stake Center to any of the buildings.

I tryed for two weeks to filter out the noise. I even setup a o-scope to check the audio signal for clipping. I setup a direct patch from the organ to a mixer that feeds the input to the encoder. I have tryed high, and low quality, test and real mode, and every thing else I can think of, all to no avail.

After checking the signal in to the encoded, and the signal comming out of the PC with an o-scope, it looks like there is a problem with the encoder unit that the church uses. I do not believe it is the decoder on the PC side because I can decode any thing the is sent from other sorces, like the general sessions. It is not a bandwidth problem, I have a 10-20 megabit link on both sides. I feel that it is a serious problem with the church provided encoder and really needs to be fixed.

The only way, that I can see, to broadcast to other building if to NOT use the church system at all, and just use a PC on both the sending and recieving side. The problem with this is that it makes it hard for others setup and use. If any wants to reproduces the problem just hold one mid-range key down on the oragan, broadcast the audio to a pc and listen on headphones or some good speakers, and you will clearly hear it. It is not a clipping problem, it is and problem encoding a contant tone. This most likely caused by a compression algorhythm.

Any help would be very useful.

[jegbert]

We've had the same issue during our last three (and only) webcast communicator stake broadcasts. The "popping" sound never ends. We've even had a satellite technician check it out from a major satellite company and he had no luck finding the issue either. Any ideas would be very appreciated. The clicking is very prominent when the organ is playing.

[pete.arnett]

A Stake had this issue when they were using a mixer for their webcasting and resolve it by opening the Audio room door and putting a fan in the room to reduce the heat and the popping stopped

[jhawkins75]

We are having this problem as well. Our setup:

- Church Webcast Communicator
- Peavey PV-6 4 channel audio mixer
- Splitter on the pulpit mic
- 2-Choir Mics
- XLR plugged directly into organ XLR output, no splitter
- Mics all go into mic in, organ goes into line in.

Sounds great off the mixing board, but after compressing turns poppy. Almost like there is some interference between the organ and the bitrate... like the organ is cycling at 60Hz and the 44kHz audio compression is causing frequency interference, just a shot in the dark.

I'm in Nashville, I should be able to half a dozen audio engineers in my Stake, I'll post if I find any answers.

[russellhltn]

There's two basic errors that might be happening. One is that the sound is too loud and being clipped or distorted. Digital encoding has an absolute hard limit on peak volume. (Which should not be confused with the average volume VU meters show.) I'd expect a piano to be a bigger issue, but the organ might have some "interesting" waveforms that cause a high peak-to-average ratio.

Another is that the sampling rate of 44KHz puts a hard limit of 22KHz as the highest frequency that can be encoded. I'd think the overtones of the highest pipes of the organ could well get into that range.

You might try this test: Play with both the level and the treble of the signal you're feeding to the encoder. See if it's more sensitive to the treble or not. That may offer a clue as to what the problem is. You may also want to try different notes on the organ since specific ones may "get through" a filter more then others.

[rpyne]

Another is that the sampling rate of 44KHz puts a hard limit of 22KHz as the highest frequency that can be encoded. I'd think the overtones of the highest pipes of the organ could well get into that range.

Unless you are running at an extreme high webcast bit rate, your sampling rate is considerably below 44kHz making the frequency limit much lower. I run an equalizer on the input to the Meetinghouse Webast appliance with a 30 dB/octave low pass rolloff at about 8 kHz.

[russellhltn]

Unless you are running at an extreme high webcast bit rate, your sampling rate is considerably below 44kHz

Good point. Don't know why I didn't think of that. Digital audio can't deal with any frequencies above half the sampling rate. It must be filtered prior to being digitized. In this day and age, I'd expect the sound card to run a much higher sampling rate and then down sample / filter prior to transmission. So it may only be necessary to filter down to the sound card's actual sampling rate. Digital filtering would have less of an effect on frequencies below the cutoff.