The text that got mangled:

>You could increase that to 20hz for improved quality/compression ratio.
>20hz is generally regarded as the cut off point for all audio production,
No, 50Hz is usually the default cutoff (to control power draw and excursion) unless it's an audiophile or 'classy' production, then it's less than 1Hz often as low as .1Hz, at least in the codec. The goal is only to remove DC.

>Pretty pictures might work on most people, but a page full of "hear say" does nothing to help Opus.
Hi, I'm one of the authors of the codec. I wrote detailed pages explaining how aspects of the codec work, which you could not be bothered to read before lecturing us on what's supposedly wrong with Opus. Which one of us is indulging in 'hearsay' again?

Interesting, the comment system removed several lines of text right smack in the middle of the comment.

>- What hardware playback device were you using for the tests?
>- What speakers/headphone devices were tested (DT 990 pro's / Ipod standard)
In this style of test, that's mostly irrelevant. One wants testing on a wide range of equipment, and no, the list of equipment is not documented. I can say though it encompasses the range from earbuds to electrostats.

>- Were you tests consistent across the board of devices used?
Oh right. You didn't read anything about it. You're just commenting.

>A spectrum analysis tells you everything regarding a codecs compression/quality ratio.
No, no it does not. Even if you want to be pedantic about it, the spectrum loses all phase and time alignment information. And even if it didn't (and we're _not_ being pedantic) it does not present it in a form your ears can parse or you can otherwise interpret in an appropriate way anyway.
You're saying the equivalent of 'we don't need to actually test software, we have the source code! That tells us everything about how it behaves, so actual testing is redundant'.

>If your aiming for your codec to simply "sound good" but actually "be completely false in reproduction", your codec will be ignored by professionals.
At 64kbps and below, all codecs are synthesizing most of the spectrum and have been for the past 10-20 years. You really have no idea what you're talking about.

>You could increase that to 20hz for improved quality/compression ratio.
>20hz is generally regarded as the cut off point for all audio production,
No, 50Hz is (to control power draw and excursion) unless it's an audiophile production, then it's Pretty pictures might work on most people, but a page full of "hear say" does nothing to help Opus.
Hi, I'm one of the authors of the codec. I wrote detailed pages explaining how aspects of the codec work, which you could not be bothered to read before lecturing us on what's supposedly wrong with Opus. Which one of us is indulging in 'hearsay' again?

>No offence, but this test was solely based on "user preference".
Offense taken.
That's how testing works in the audio industry. Mainly because it's the only thing that makes sense.

>- Wheres the Spectrum analysis of each codec?
A spectrum tells you almost nothing about how a codec sounds. Thus listening tests.

>- Which codec is more true to the original .wav, using the above data as comparison?
That's what a listening test answers. I know very few people who listen to music by unplugging their speakers, watching the spectrogram scroll by, and pretending to know what it sounds like.

>- Which codec cuts below 50hz?
Only telephony codecs do that; the highpass improves speech intelligibility according the studies done by Bell Labs.
Opus has a 3Hz highpass to eliminate spectral leakage in samples with a DC offset.

>- Which codec emphasizes certain frequencies (8-10khz, typical LAME mp3)?
Listening tests will tell you that. A spectrogram will not. Some amount of detailed critical band energy analysis can give you quantization bias figures, but that still won't tell you how it sounds. That's a test you run _after_ hearing a boost to determine where it's coming from.

>- I'll automatically assume
That's why I'm taking offense.

> your Opus codec (which is based on a voice codec) prioritizes bitrate quality between 500hz-4khz.
No. Opus is based on CELT, a music codec, and SILK, a speech codec. You didn't even read the demo page? Dude, tons of pretty pictures. You missed the party hat though.

We abandoned Tarkin, but not because of VP3. It was an interesting crazy idea that simply wasn't workable.

Tarkin was an attempt to replace motion compensation with a directional 3D wavelets that could predict across time. It had... disturbing... encoding artifacts.

Extensive testing reveals that no matter the wailing and gnashing of teeth claiming digital sucks, no one can actually tell the difference in a double-blind test.

Theory is indeed useless when reality doesn't agree. Every real scientist knows this. An engineer's job depends on it. I invite audiophiles to add that bit of wisdom to their own thinking.

I intentionally avoid using lavalier mics. Their amplitude and timbre are all over the place depending on the direction you're looking, they pick up clothing noise, and you're either tethered to a wire or have to deal with the complexity and limitations of a wireless system.

Headsets work better overall, but are highly visible and still add a layer of complexity. They also sound like someone talking directly into your ear. Even with additional postprod [going all the way to room modeling], I've never managed to make a headset sound totally natural. If others have, I'd appreciate some tips.

In any case, as someone who spent ten years dealing with lavs and headsets in live performance as a sound engineer, I avoid them in my own time as much as humanly possible :-)

The stereo image was intentional, it's a trick for removing perceived echo/reverb by spreading it out across a stereo image instead of it all piling up right behind the voice in mono. That said, the image was wider than I'd have liked. That was a result of mic placement and angle, another tradeoff to avoid wide amplitude changes as I moved around.

FWIW, several other video producers wrote me to ask how I got such great sound without a lav, and (like you) others wrote to mention that they found the very wide stereo distracting. I'll tinker with it more in the next vid.

On that topic, while I was filming the epilogue I started feeling ill and ignored it for a while, but eventually went to the emergency room. Three hours later I had an emergency appendectomy. Medical science also rocks. If I'd gone to a faith healer, I'd be dead. (FTR, the epilogue footage in the final vid was from three days after the appendectomy)