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Comment Re:Private information leakage. (Score 1) 255

I've read their response and it is bullshit. There is a difference between trusting someone to write software for you, and trusting them to receive your private information over the internet. For example, hospitals may trust software to process patient data. They don't (and can't by law) trust that software to send patient info to some random cloud service. Home users deserve the same respect as corporate users and should be informed of any services that transmit their information so they can decide whether to use the service or not. This is basic stuff, and the fact that Canonical still doesn't get why anyone might be concerned is very troubling and a good hint to stay clear of them in the future.

Comment Private information leakage. (Score 5, Interesting) 255

Every time you search for a local file on your computer, the details of that search will be transmitted to a third party cloud service. That is a huge potential privacy issue regardless of who that service is. Worse, they they don't even make their users aware of this fact, which is completely unacceptable. That Canonical still doesn't understand this after being having it brought to their attention means they clearly cannot be trusted to assemble a secure Linux distribution.

Comment Re:More like tricked them (Score 1) 295

But the interesting part is that half of them would rather defend their accidental choice and argue against what they really believed than admit that they made a simple mistake filling out a form. It is an interesting insight into how ego can be more influential than opinion, and how that can be used against people to influence their opinions.

Comment Re:Psychoacoustics and perceptual coding (Score 1) 361

Sorry, I just now realized that I misread your post. In the example you gave, the square wave and triangle wave will contain many harmonics of the main period, many of which beyond the range of human hearing. If those three signals are low-pass filtered at 22kHz before sampling, the sampled signal will be able to completely reproduce the shape of those filtered signals. That is, it will faithfully reproduce all components of the signals that are within the range of human hearing.

Also in the last line of my post replace "dynamic range" with "frequency range".

Comment Re:Psychoacoustics and perceptual coding (Score 1) 361

If you take three electronically produced 15kHz tones, one a sine wave, one a sawtooth wave, and one a square wave, that teenager can tell the difference between them. But record those three tones at 44.1k samples per second and you have only three samples per crest; not nearly enough to discern what the shape of the wave it is.

You absolutely can discern (reproduce) the exact shape of the wave in that example. It was proven by Claude Shannon in 1949. The way that digital signals are normally drawn in examples either has stair step (zero-order hold) or jagged lines (linear interpolation) between the samples, but that isn't how actual music equipment reproduces the signal. Mathematically, if the signal is sampled with even slightly more than 2 samples per cycle (the Nyquist limit), then interpolating the samples by multiplying each with a sinc function ( sin(fc*t)/(fc*t) ) will perfectly reproduce the original sine wave(s). Equivalently, you can output a zero-order hold version of the signal and filter it with a properly designed reconstruction filter, which will give the same result. This is how real digital audio hardware works. That is also why a sampling rate of >40k was chosen for CDs; it allows you to represent the full dynamic range of human hearing.

Comment Not a step up. (Score 5, Informative) 361

No, it's not a step up. No-one has ever been able to reliably distinguish a 24/96 recording from it's downgraded 16/48 version in a properly conducted double-blind test.

It is absolutely necessary to oversample when acquiring data (since all analog filters have some roll-off), and it is good to use higher dynamic range when mixing to keep the repeated rounding errors below the noise floor. But once the final recording it is mastered, there is no benefit to distributing or listening to the result at higher than 16/48.

Comment Re:$20,000,000? (Score 4, Informative) 466

According to an article on TechCrunch, these three sites have a yearly profit (EBITDA) of $5 million. From what I've read the purchase price (3*profits + some) is typical of acquisitions of mature companies. It is neither insane dot-com buyout (expecting unrealistic growth), or clearance corner liquidation of assets (expecting to bleed it till it dies).

Comment Re:Many people missing the point: HTML5, VOIP, etc (Score 3, Informative) 327

No, he is using them correctly, referring to the application, not the medium.

Music playback doesn't require low latency; it doesn't matter if there is a 500ms delay between when you press the play button and you hear the music. Because of this data is encoded in (relatively) large blocks to allow for as much compression as possible.

VoIP on the otherhand does require low latency (100ms max). Otherwise it is very difficult to carry on a conversation because otherwise if you were to speak during a silence, it may not still be silent when the signal gets to when the other person, so you constantly talk over one another. The other potential application, live interactive music, requires even lower latency for musicians to keep in sync with each other. For this reason Opus is designed to encode in small blocks of data to obtain better latency.

Comment It fills a needed niche (Score 5, Informative) 327

To me the biggest difference is that Vorbis was competing head on with a strongly entrenched codec (MP3) and it's official successor (AAC). Opus on the other-hand fills niche in the audio encoding world that doesn't have an established winner; that is high-quality low-latency codecs. This area has largely been driven by cellphone market, and has focused on encoding voice signals at toll-quality, that is as good as an analog long-distance signal (8kHz mono). There really hasn't been much focus on creating a low-latency codec that can encode full-band (music signals), and Opus does that incredibly well. It also sounds much better encoding speech at the bitrates that are used for VoIP (rather than the lower ones used by cellphones).

The internet community has never really been happy with the performance of ITU specified codecs that have been primarily used for SIP and other VoIP applications in the past, and there is no good reason from them not to support Opus. The patent grants are there, the vender support is there, and there is no real competitor codec worth mentioning. I'm convinced this will make much deeper inroads than Vorbis did.

Comment Third way: (Score 1) 771

There is often a third way to win an argument:
You bring a larger number of your audience over to your point of view than your opponent does.

If someone insists on having an argument rather than a discussion that pretty much means that there is no way they are going to change their mind anyway, and the only reason to talk to them at all is if other people are already listening to them and falling for what they are saying.

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