科学与数学xiph.org编辑精选
24/192 音乐下载非常愚蠢（2012）
24/192 Music Downloads Are Very Silly Indeed (2012)
2014 — 2017，它一次次回到首页
黑点是上首页（≥ 50 分）的那次投稿，空心点是没冲上去的投稿，点越大分数越高。
导读
这是 Xiph.org 的 Monty 写的一篇经典技术长文，用采样定理和听感实验论证 24bit/192kHz 音乐下载纯属营销噱头：16bit/44.1kHz 已能完整覆盖人耳听域，更高规格不会带来可闻的音质提升。文章还批评了高规格音频背后的商业动机和常见误解。
为什么经久不衰：它把数字音频采样理论讲得深入浅出，多年来反复被引用，是理解'高解析度音频是否值得'这一争论的最佳入门读物。
HN 评论精选
高规格文件的实际价值在于它们通常保留了被响度战争毁掉的动态范围
也不全然是噱头。高保真音频宣称的好处确实是想象的，但如今流行音乐的'重制版'大多为了耳机听歌的路人把动态压扁到'响度战争'水平，24/192 下载或 SACD 往往是唯一能听到真实动态范围的途径。上一代愿意投资好音箱、追求安静听音环境的人，如今都屈服于手机和廉价电脑音箱，这是巨大的转变。
Mediterraneo10查看英文原文
Not entirely silly. Yes, the purported benefits of this high-fidelity audio are imaginary or even include undesirable traits. However, when most "remasters" of pop music today involve the dynamics being boosted to "loudness wars" standards for a target audience of people listening to the music through earbuds in the street, the 24/192 downloads or SACD releases are often the only way to hear the album with real dynamic range. I’m not sure how big that market is these days, though. I recently decided to move from a block in the city to a house in the surrounding countryside purely to get a quieter listening environment and let me really enjoy the high-dynamic-range recordings I have – I listen to a lot of classical music, especially the avant-garde with its even greater range, e.g. the Ligeti cello concerto starting from pppppp. Yet even among my friends who are really obsessed with such music, seeking a better listening environment seemed an extreme measure to take. So, people who a generation ago would have invested in higher-end equipment (not audiophile snake oil, just better speakers) and who would have sought silence are now giving in to listening to music on their phones or cheap computer speakers. It’s a big shift.
24/192 文件的价值不在规格本身，而在于它们通常来自更好的母带音源
想下载 24/192 有个很直接的实际理由：做这些文件的多是发烧友，抓轨质量往往极好，常来自 SACD、hdtracks 等高质量母带。下载 mp3 或用 Spotify，你拿到的很可能来自更差的音源。320kbps mp3 听感上其实一样，但找 24/192 是平均获得更高质量音乐文件的捷径。
lz400查看英文原文
There's a very straightforward practical reason why you might want to keep downloading 24/192. The people who put those together are on average "audiophile" snobs so the rips tend to be perfect, often from high quality sources like SACDs, hdtracks, etc. The source of the recording is usually the best master known for that record (special remastered editions, etc.). If you download mp3/spotify chances are the copy you download is from a worse source. Sure, you don't need 24/192, a well ripped 320kbps mp3 would be the same in practice, but looking for 24/192 in the internet is an easy way of getting better quality music on average in practice.
高采样率在录音/转换环节有真实工程价值，但这不等于消费者需要 192kHz 文件
文章观点不错但有缺陷。问题在于人们只谈采样理论，却完全忽略了在 44.1kHz 下实现完美模数/数模转换链的实际困难：需要在模拟域做 22kHz 的砖墙滤波，这既昂贵又会带来振铃、纹波和相位偏移。所以 2017 年的转换器应该以更高采样率（如 192kHz）工作，让模拟滤波容易得多——只是这和'卖给你 192kHz 文件'是两回事。
flavio81查看英文原文
Good points in the article, but it has some flaws. The problem whenever somebody writes about digital audio, is that it is very tempting to hold on to sampling theory (Nyquist limit, etc) and totally discard the problems of implementing an actual Analog-Digital and Digital-Analog chain that works perfectly at 44100Hz sample rate. I agree with the assesment that 16 bit depth is good enough; even 14 bit is good enough and was used with good results in the past (!). However, the problem is with the sampling rate. > All signals with content entirely below the Nyquist frequency (half the sampling rate) are captured perfectly and completely by sampling; Here lies the problem. This is what theory says, however, when using 44KHz sample rate, this means that to capture the audio you need to low-pass at 22KHz. And this is not your gentle (6, 12 or 24db) low-pass filter; no, this needs to be HARD filtering; nothing should pass beyond 22KHz. And this must be on the analog domain, because your signal is analog. To implement such a filter, you need a brickwall analog filter and this is not only expensive, but it also makes mess with the audio, either 'ringing' effects and/or ripple on the frequency response and/or strong phase shifts. So on Analog-to-digital in 2017, converters should be operating at a higher rate (say, 192KHz), because this makes analog filtering of the signal much easier and without side effects. Now, for Digital-to-Analog, if your sample rate is 44KHz, you have two alternatives: a) Analog brickwall filtering, with the problems noted above or b) filtering on the digital domain + using oversampling the article mentions: >So the math is ideal, but what of real world complications? The most notorious is the band-limiting requirement. Signals with content over the Nyquist frequency must be lowpassed before sampling to avoid aliasing distortion; this analog lowpass is the infamous antialiasing filter. Antialiasing can't be ideal in practice, but modern techniques bring it very close. ...and with that we come to oversampling." So they are mentioning alternative (b). The problem is that oversampling does not solve all problems. Oversampling implies that the filtering is done on the digital domain and there are several choices of filtering you could use, for example FIR (Finite Impulse Response), IIR (infinite impulse response), etc. And each one of these choices have side effects... In short, the problem is that with 44KHz sampling rate, your filter cutoff (22KHz) is too short to your desired bandwidth (20Hz-20KHz). Using a sample rate of 192KHz gives the DAC designer much more leeway for a better conversion. And CONVERSION is the key to good digital sound. >What actually works to improve the quality of the digital audio to which we're listening? It is interesting that the author mentions things such as "buying better headphones" (agree), but he never mentions "Getting a better Digital to Analog converter", which is highly important !! On the other hand, he backs up his claim that "44KHz is enough" with an interesting AES test i was already aware of in the past: >Empirical evidence from listening tests backs up the assertion that 44.1kHz/16 bit provides highest-possible fidelity playback. There are numerous controlled tests confirming this, but I'll plug a recent paper, Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback, done by local folks here at the Boston Audio Society. This is a very interesting paper, and I did have the copy, however the test equipment should be checked. There are systems and better systems. The AES paper cited above had the particularity that the ADC and DAC used were provided by exactly the same machine (a Sony PCM converter), with the same strategy: no oversampling, brickwall analog filters. I can bet (99% sure) that the brickwall filters were identical on the ADC and the DAC on that machine; Murata-brand filters in a package. The devil, as they say, …
双盲测试也说服不了相信高规格音质的人，这是心理问题而非技术问题
我曾花约两小时给一个学声音设计的朋友讲这些，他不信。我甚至把 24/192 的 Pink Floyd FLAC 转码后让他做双盲测试，证明他每次都更喜欢稍响的那首——哪怕响的那首是 192kbps。他确实选了响的，但至今仍不信，仍觉得自己能听出 FLAC 和 MP3 的区别，而他现在是录音师。有些人的想法是讲道理改变不了的。
errantspark查看英文原文
I once spent ~2 hours explaining all of this (well maybe not all of it) to a friend of mine who was studying sound design at the time. He didn't believe me. I even transcoded some 24/192 FLAC Pink Floyd I had lying around and made him do a double blind test to show him that he'd prefer the slightly louder song every time, even if the louder song was 192kbps vs the FLAC. He did. He still doesn't believe me. He still thinks he can hear the difference between FLAC and MP3 to this day. He works as a sound engineer now. I don't think any amount of reasoning will make some people change their minds. Some people buy $500 wooden knobs to make their volume pots sound better. (or was that a hoax? i can't tell anymore)
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