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Opus / Re: Can anyone reliably ABX OPUS at 160kbps?
Last post by jmvalin -
Opus - finally bringing you the glorious stupidity of real CBR!

Actually, this made me google for why there is a CBR mode (except for testing purposes). I found this by @polemon - is that it?
CBR is useful because not every transport mechanisms can take VBR/CVBR, especially for real-time. Keep in mind that people use Opus for a whole lot of applications that have nothing to do with files. For example, if you want to play live music across a network, you need minimal latency, which means you probably want to use the highest bitrate that'll go through without causing delay, but not a single byte more. Then there's people using Opus in real-time over a fixed-bandwidth wireless link (e.g. DECT, possibly Bluetooth too). You'd be surprised to see all the places Opus ends up in.
Audio Hardware / Re: Designing a good listening test
Last post by ajinfla -
please help me design a good listening test
If one was a undercover guerilla peddler of audio jewelry, one might suggest something like this: rather than a mentally taxing ABX.
Thereby making it impossible for the average Joe, so that casual sighted listening is reverted to as the more attenable possibility in a convoluted false equivalency.
Oh wait, did you say no instant switch? Maybe a dart board and darts?
Or there is always a good fishing rod.
Audio Hardware / Re: Designing a good listening test
Last post by DVDdoug -
How "scientific"?

See ABX test.  That's the ideal.  

For a casual non-blind A/B test, do your best to level-match.   If you have a multimeter to measure the amplifier output voltage into the speakers, that's ideal.   Or, you can use an SPL meter.   Otherwise you can set-up a laptop, or a phone with a recording application or anything that has a meter.   The voltage or SPL level isn't important as long as both amplifiers are level-matched.  If you measure acoustically, use a lower frequency for more-reliable more-consistent readings.     I was doing some high frequency measurements (~8kHz) awhile back, and just walking around behind the SPL meter changed the readings.  Audacity can generate test-tones for level matching. 

Of course, make sure tone-controls/EQ are set to flat.    With the AVR, you'll have to make sure any automatic room-adjustment or subwoofer crossover disabled.

If you want to approximate a blind ABX test, you'll need to randomize the selections (flip  a coin) and you should disconnect & re-connect the speakers even when comparing "A" to "A".     With most amplifiers you can connect/disconnect the speakers with power-on, but it's a good idea to do it with no signal and be careful not to short-out the connections.   If you can rig-up some kind of speaker connectors that you can easily un-plug and plug-in, that's better than switching wires to binding posts, especially with "live" connections. 

Get some Y-Adapters so you can feed the signal into both amps continuously without switching.

Since it won't be a double-blind it's best if the person doing the switching is hidden from sight along with the amplifiers.

There might be a pop/click when you connect the speakers and that could be a give-away and if you turn the equipment on/off there could also be give-away clicks or thumps, etc.

There will be no instant switching unfortunately.
It's easier to hear a difference with instant switching, but in the real world you wouldn't be switching amplifiers that fast.

3rd Party Plugins - (fb2k) / Re: iPod manager
Last post by TheEmpathicEar -
You could add an album remapping that checks if the album is 'Greatest Hits' and then appends the album artist if so. I'm not sure you're going to get much better than that.
What do you think about @anamorphic 's idea of Zero Width Space ? Maybe a conditional of some sort when a situation like mine with two albums with the same name in the same year? I already am using a mapping the displays my albums with the year in brackets: '[2004]'
Last post by bmcelvan -
Last question: Is there a way to get sox to create a log file so I can check for errors later in case I accidentally hit a key and close the batch script...for example when waking the computer from a screen saver if I walk away after a big batch.

I tried adding > log.txt but that yielded and empty text file and I couldn't find any options in the Sox command line help section.
Audio Hardware / Re: Audio-Technica ATW-R03 VHF radio mic receiver fault
Last post by Philip Le Riche -
Thank you, but the sort of caps used in tuning with be in pF range, probably ceramic or polyester, which don't dry out. It's electrolytics that suffer from that problem. And I've checked the on-board linear regulator (7810 - first time I've met one of those!) and it's giving a steady 10V as expected.

The frequency is not user selectable (certainly not with the pots - they'd be nothing like stable enough). It's clearly factory set.

I have another one with a lapel mic, 173.8MHz, which works. Taking another look at them both, they have different crystal frequencies - 61.5 and 61.9MHz. On a first glance I thought they were the same. Now I notice that 61.5 * 3 - 10.7 = 178.3. Also, 61.9 * 3 - 10.7 = 175. So the 3rd harmonic of the crystal is being used as the local oscillator to produce the very common IF frequency for VHF radios of 10.7MHz.

So I swapped the crystals and found that both the receivers work with the 61.5MHz crystal for the 178.3MHz mic. So my problem is actually with the 175MHz mic, not the receiver. This has a 14.573MHz crystal, which is close to 175MHz / 12. So does anyone have any experience with these?
Last post by bmcelvan -
Wow, I'm dumb. Simply changing it from
sox 01.dsf -b 16 01.wav gain 6 rate -v -s 44100
sox 01.dsf -b 16 01.wav rate -v -s 44100 gain 6
I believe did it.

Thanks again for the help
Last post by bmcelvan -
Thank you both...I have been playing around with that "other" file but wasn't sure if it was "official" or trusted. Was literally just about to write how I got it working.

Thanks for the responses.

After playing a bit, a couple more questions of advice. When doing what I'll call a standard convert:
sox 01.dsf -b 16 01.wav rate -v -s 44100

I get output files that when looked at in Audacity the peaks never go above or below 0.5 (see attached "normal").
I've read that PDM to PCM converting always loses volume and therefore it could be added back in.
Do you think it's worth trying to boost them a little bit - in theory could anything be gained other than having to turn the volume knob a little less?

I tried using both gain and volume adjustments but any positive number yields clipping. When looking at the output files in audacity...the gained and clipped files look identical to the not gained ones and doesn't show any clipping. When looking at the command prompt side of (attached "gain6") it looks like the input file was gained...not the output. I guess I'm wondering how to gain the output file.
sox 01.dsf -b 16 01.wav gain 6 rate -v -s 44100
sox 01.dsf -b 16 01.wav vol 6dB -v -s 44100

The only thing I could do that wouldn't clip was use --norm
sox --norm=-1 01.dsf -b 16 01.wav vol 1dB -v -s 44100

Am I getting the gain coding wrong?
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