Monty Montgomery

I love this link (also linked in the main article). https://xiph.org/~xiphmont/demo/neil-young.html Whether a person prefers 8, 16, or 24 bit is besides the point and has to do with their tastes. That article clearly debunks any arguments saying that 24 bit is superior in re-creating reality. In fact he makes some points against it. I love the analogy about a television that can display UV rays.

Monty from Xiph/Mozilla wrote a great article about why 24-bit/192kHz format (which is what Neil Young's Pono delivers) makes no sense: http://xiph.org/~xiphmont/demo/neil-young.html

It wouldn't surprise me if the hardware BOM cost of supporting these high sample rates for playback is basically zero. Any decent audio DAC will oversample internally, so if you don't actually care about fidelity at inaudibly high frequencies, you can just oversample to the same frequency but shove more bits in. Of course, if you do this, you ought to do your oversampling carefully to avoid sounding worse due to leading harmonics of your actual audible signal through your filters into your amp and tweeter. If you use a bona fide 1-bit DAC, you can sample pretty much as fast as you want if all you care about is your pretty spec sheet. 2+MHz sampling? No problem! The classic article is here: https://xiph.org/~xiphmont/demo/neil-young.html Count me out, of course, but I still think it would be amusing to find a "384 kHz" audio DAC that actually worked at high enough frequencies to transmit AM radio. You'd have to actively exploit insufficiently filtered harmonics to have a chance.

Nitpicks: "A great deal of quality is lost as those huge files are squished to the CD format, before being further squished into MP3s on your iPod." Not true, AAC files at high bitrates are proven to be indistinguishable from the original. http://xiph.org/~xiphmont/demo/neil-young.html "Every part of the signal chain—from earbuds to digital/audio converters—is improving and getting cheaper." Like other technologies, digital audio had rapid improvements in the first 10 years but has mostly leveled off as we have reached the limits of physics, signals and sampling. Good audio equipment is still expensive and power hungry in every part of the chain. Technology won't save what are fundamentally psychological issues. We respond emotionally to louder signals. We like songs that sound similar but not identical to things we've heard before. We listen to things that our friends like. All of this points to why top 40 songs sound the same, but "all records" is pretty broad. There has never been a better time to create something new in music, and yet we lean heavily on what has worked before -- in songwriting, in arrangement, in production -- so that others will like and identify with the creative output.

>"24/192 and higher playback" Utterly pointless and a waste of time and resources to implement. https://www.xiph.org/~xiphmont/demo/neil-young.html

The core is here: https://xiph.org/~xiphmont/demo/neil-young.html#toc_1ch When you play something stored at 44 khz, there are no ultrasonic sounds recorded. At 192 khz instead, there are, and the speakers may push some of the ultrasonic sounds down to the audible spectrum, causing distortion that the human ear can hear.

If your argument is along the lines of what I think it is, then this is a fallacy. See "Sampling fallacies and misconceptions" in https://xiph.org/~xiphmont/demo/neil-young.html "All signals with content entirely below the Nyquist frequency (half the sampling rate) are captured perfectly and completely by sampling; an infinite sampling rate is not required. Sampling doesn't affect frequency response or phase. The analog signal can be reconstructed losslessly, smoothly, and with the exact timing of the original analog signal."

Daala uses quite a few innovative approaches to video encoding. There's a good set of articles on some of the approaches they've tried at https://www.xiph.org/~xiphmont/demo/. Since the formation of the Alliance for Open Media, Mozilla/Xiph's work has mainly shifted towards helping them develop AV1 (for standardization by the IETF hopefully next year) which is mostly based on what would have been Google's VP10. That said, their work on Daala was not wasted as they are working on integrating the most successful parts of Daala into AV1 where it makes sense, such as their entropy coder.

Firstly, https://xiph.org/~xiphmont/demo/neil-young.html Secondly, even if you did want to process at 96kHz, you'd have plenty of CPU left to do so. It's only 2x as intensive as 48kHz (this is a 32 bit CPU so using 16 bit vs 32 bit math is mostly the same, sans DSP instructions) and that amount of headroom is likely available, for example: https://www.rockbox.org/wiki/CodecPerformanceComparison Thirdly, the article you linked talks about high end DACs but says nothing about the DSP on the card, other than that it has one, for doing... something (?)

You might be surprised to learn that "HD audio" often has worse sound quality than "CD quality". The science is here: https://xiph.org/~xiphmont/demo/neil-young.html Edit: You're being downvoted which I think is unfair. It's completely logical that higher sample rates and higher bit depths "should" sound better, and certainly the music industry wants you to believe that. Unfortunately it's simply not true above a certain threshold. "Lossless" here is defined as "stays the same once it has entered the digital domain", and not "identical to the real-life source", and so CDs are in fact lossless.

Yeah, Opus is known to compress noisy rock/metal-like audio pretty well, even at low bitrates. The stuff it finds harder to do is tonal content, though it assigns more bitrate to help itself out. See the Tonality Estimation section here: https://xiph.org/~xiphmont/demo/opus/demo3.shtml

~44kHz is actually pretty ideal, anything significantly higher can actually be harmful to the audio quality: http://xiph.org/~xiphmont/demo/neil-young.html

Monty's videos are a good start: https://www.xiph.org/video/ https://www.xiph.org/~xiphmont/demo/

I can only assume this is a joke pointed at audiophiles, the sort who promote the 192khz/24 bit music. http://xiph.org/~xiphmont/demo/neil-young.html

There is an Episode 1. http://xiph.org/video/ See also http://xiph.org/~xiphmont/demo/neil-young.html which was posted here a week ago and received 441 comments. https://news.ycombinator.com/item?id=8689231

> 96KHz sampled 24bit digital ones! Surely you mean 192KHz! No, really: http://xiph.org/~xiphmont/demo/neil-young.html

From https://xiph.org/~xiphmont/demo/daala/demo1.shtml > Our performance target is roughly a generation beyond current 'next-generation' codecs like VP9 and HEVC, making Daala a next-next-generation effort. and > The next-generation VP9 and HEVC codecs are the latest incremental refinements of a basic codec design that dates back 25 years to h.261. This conservative, linear development strategy evolving a proven design has yielded reliable improvement with relatively low risk, but the law of diminishing returns is setting in. Recent performance increases are coming at exponentially increasing computational cost. > Daala tries for a larger leap forward— by first leaping sideways— to a new codec design and numerous novel coding techniques. In addition to the technical freedom of starting fresh, this new design consciously avoids most of the patent thicket surrounding mainstream block-DCT-based codecs. At its very core, for example, Daala is based on lapped transforms, not the traditional DCT.

- 48 kHz can be more easily (and more importantly - accurately) resampled to other common sample rates (which tend to be round multiples of 8). - While 44.1 kHz can theoretically sample the entire range of human hearing there are some practical considerations that give 48 kHz an advantage. That said, there are few reasons to use sample rates above 48 kHz: https://xiph.org/~xiphmont/demo/neil-young.html - Most importantly 48 kHz is essentially "the standard" on most operating systems, hardware, hardware drivers, various audio frameworks, codecs, audio applications, audio editors, etc, etc. For example, 48 kHz is the only required sample rate in the Intel High Definition Audio Specification (section 7.3.4.7): https://www.intel.com/content/dam/www/public/us/en/documents/product-s… Section 2.2 also describes that the timing for the HDA stream format is based on 48 kHz sample rates and other sample rates require some gymnastics (section 5.4.1) to implement. - Perhaps not surprising to the HN crowd there can be various issues with other sample rates and resampling across different hardware, operating systems, etc: https://bugs.webkit.org/show_bug.cgi?id=154538 https://productforums.google.com/forum/#!topic/chromecast/k6JZRM2TUco https://bugs.maemo.org/show_bug.cgi?id=5794 https://groups.google.com/forum/#!topic/discuss-webrtc/EGvSbzQJutg https://bugs.chromium.org/p/chromium/issues/detail?id=86767 In 2018 we should be…