Arch prepared the test files in accordance with the OP's question: so anyone who might care about Arch's test is anyone who cares to be on-topic with the intent of this thread. (recognizing that OP obviously meant 44.1 kHz.)
To be clear about the above, the source is 24/88.2, downsampled to 24/44.1 and to 16/44.1? Now I just gotta figure out how to play this-maybe USB stick into Oppo? I don't recall if our AVM 70 has such an input.
Correct @head_unit . 24/88.1 --> 24/44.1. Bit-depth reduced to 16/44.1 using the 24/44.1. @rexp - will send you a link soon later today...
Yup. The question was always about 24 vs. 16-bits and it's good that @Spaceboy specified 44.1; the choice of "Giorgio" was raised by @rexp himself. A bit more regarding 44.1kHz - not only is this smaller for listeners to download and transfer to USB/phone/etc., but compatibility better than 88.2. Over the years I've come across DACs that do not handle 88.2 well but can perform excellently in 96kHz. Also, certain products, especially Apple players are limited to 44.1/48 but can perform better objectively with 24-bit data. One more thing, for the survey released to the broad public this weekend, I want to include a "wireless headphone" choice for listeners just to see if I can capture that growing segment. Even though wireless, some codecs these days can still encode in 24-bits and other than LDAC being most common capable of 96kHz at higher bitrates, I think they're almost all 44.1/48... Curious about that group.
Just to confirm, I have PM'ed @rexp with the test samples for the 88.2kHz version. I've done my end of the deal and look forward to the response. For full transparency, if you want rexp, feel free to leave a note in the final comments that it was the 88.2kHz version you listened to. I've also added the "Wireless headphones" options for listening system choice. I've never tried a blind test with specifically looking at this as a popular option these days so will be interesting to just be able to isolate this population of listeners/music lovers. Again, a thanks to those already submitted! As I noted earlier, if we don't want to just be old guys talking about opinions which could just be dramatics with no factual basis, I would highly encourage participating in a test like this. This IMO, is what the "next level" looks like if we want to call ourselves experienced, knowledgeable, audiophiles.
Thanks for setting this up! I hope those who are confident in their ability to discern between 24 & 16 take this challenge! I won’t bother because I know that I can’t differentiate between high res and CD quality. In fact I haven’t been able to successfully differentiate between CD quality and 256 AAC in blind testing so I suppose I need to turn in my audiophile card
Well, ya never know man! Might still be good to have a quick listen or even try quick A/B switching with something like foobar ABX Comparator. There was one time I didn't expect to hear anything and went ahead anyway to listen to a 24/48 vs. 24/96 test and was able to ABX with good confidence. Turned out the person running the test didn't use a transparent setting for the 96 --> 48kHz downsampler and the high frequency roll-off was audible. I remain open to the idea that the dithering settings could still affect stuff like subtle frequency shifts. Ultimately, while I have my beliefs ("hypotheses" about the outcome), I'll keep an open mind until the end of the trials and see what the data shows. Yes, I would love to have those who believe 16-bit vs. 24-bit can be discerned with confidence give this a try. Worse that could happen is that you learn that you don't like Daft Punk! But then again, who doesn't like Giorgio Moroder?! "The Father of Disco"!
Not trying to derail this thread, but need help and thought it would fit in here without starting a new thread I've been recording music from my computer to my reel to reel going through my Dac. I never look at my Dac, but took a look and the reading was stuck on 48kHz for everything. I thought the Dac might be broken, tried lots of stuff, then realized that my windows setting changed somehow and it was set at 32 bit 48khz for max output instead of 24 bit 192 kHz that it once was set at. Is it worth it to go back and rerecord all the tracks to get that 24 bit 192 kHz?
I AM an audiophile. But I am also a scientist, and I refuse to believe in things like "audiophile" power cables, USB cables, etc
While the perception of audio quality is subjective and can vary widely among individuals, there is scientific evidence to suggest that claims about the impact of certain audio equipment, such as power cables and USB cables, may be unfounded. For example, a study conducted by Ethan Winer, a prominent audio engineer and researcher, found that "expensive speaker wires and interconnect cables are typically no better than inexpensive ones," and that "most exotic cables provide no improvement in sound quality over plain, low-cost cables." Similar studies have been conducted on the impact of power cables and USB cables on sound quality, with many researchers concluding that the differences between different cables are not audible or are very small. Here are a few more studies that support the idea that expensive audio cables and other equipment may not necessarily provide noticeable improvements in sound quality: A study conducted by The Wirecutter found that "cheap HDMI cables work just as well as expensive ones." The study involved blind listening tests using a variety of HDMI cables, and found that "no one could hear any difference between any of the cables tested." A study conducted by the Audio Engineering Society (AES) found that "high-end" audio equipment, including cables, did not necessarily provide improvements in sound quality. The study involved blind listening tests using a variety of audio equipment, and found that "there was no consistent, audible difference between the high-end and low-end systems." A study conducted by the Consumer Electronics Association (CEA) found that "a majority of listeners could not hear any difference in sound quality between a high-resolution audio file and a CD-quality file." The study involved blind listening tests using a variety of audio files, and found that "most listeners could not distinguish between high-resolution audio and CD-quality audio." These studies suggest that claims about the impact of expensive audio cables and other equipment on sound quality should be approached with caution, and that objective measurements and controlled listening tests should be used to make informed decisions about audio equipment.
FYI - the blind test has been posted on my blog and rolls out broadly now: Internet Blind Test: 24-Bit vs. Dithered 16-Bit Part Deux - Daft Punk Edition! And the honest desire to seek truth as audiophiles. Thanks for the discussions so far and I'll definitely update this thread when the results are collated at the end of the listening test...
I can see Arch accidently is stirring up the hornet's nest again. And, we're off! For the record--I can't tell the difference between any 16/44 and 24/192 album mastered exactly alike.
It was a bit of an eye (ear) opener when I took a 16/44 audio file and reduced the bit depth to 15, then 14, etc. The music sounded the same but white noise became increasingly louder as the bit depth was decreased.
Wrong! Oversampling - Wikipedia See especially the section on resolution! DSD is a specific form of noise-shaped sigma-delta modulation. As an oversampled signal, it already satisfies the Nyquist criterion. I would not quote that link on the subject of digital audio. It would be more correct to say that a continuous waveform can be broken into an infinite number of data points. However, what Shannon and Nyquist showed was that you do not need an infinite number of data points to represent a band-limited signal, in fact, ideally, you need only two! Moreover, it doesn't matter where, in time, you start sampling the band-limited signal, the digital data will still restore exactly the same original band-limited signal. The role of the reconstruction filter in a digital-to-analog converter is to restore the time continuous signal. Keep in mind, in PCM digital audio we are ONLY concerned with the band-limited signal produced by the anti-alias filter, not the original signal itself. Clearly, the goal is to determine what sampling rate and bit-depth are sufficient that audible differences between the original and the band-limited version are minimized. That depends on the implementation. If you drink the Kool-Aid espoused by the likes of Audio Note, that is exactly what you will get: Audio Note CDT One/II CD transport & DAC 2.1x Signature D/A processor Measurements Non-oversampling, terrible jitter performance and no reconstruction filter, the worst of all worlds. Fools and their money....
I forgot to add. Blind-testing/double blind-testing comes up a lot in these discussions. There is no reason to double blind-test anyone if they cannot differentiate between the reference and the alternative in a sighted test. So, both files should be made available (the reference and the alternative) such that listeners can become acquainted with the characteristics of both signal representations. Only when they feel they can reliably distinguish between them should they proceed to ABX testing. In formal tests of this nature, the musical passages are also kept short, approx. 10 seconds, due to the limitations of short-term audio memory. For more details: https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1116-3-201502-I!!PDF-E.pdf
While I appreciate your efforts, I really think you should scrap this test as the 24/44.1 doesn't sound as good as the original 24/88.2. There are plenty of original 24/44.1 recordings you could use instead.
Apples and oranges. That's talking about using oversampling to increase the resolution of the DAC in the playback device. I was talking about the sampling rate and resolution (bit depth) of the original recording. Oversampling cannot magically invent greater resolution than what existed in the original recording -- a CD is only ever going to give you 16-bit audio, regardless of what kind of DAC you use. Oversampling just helps the DAC to (hopefully) reproduce those 16 bits with better accuracy and less noise.
Go back and read again what you wrote ("resolution is actually determined by bit depth, not sampling rate."). You were emphatic that resolution is only determined by the wordlength in PCM. Oversampling can be, and is, used in both ADCs and DACs.
My takeaway from this thread is that bit depth determines noise floor level and potential dynamic range, and sample rate determines maximum audio frequency range. The CD format is capable of rendering just over 22Khz audio frequency range and about 96db of dynamic range. Those specifications more than amply meet and far exceed what my ears are capable of hearing. That said, what's left to consider except mastering quality? Streaming and download sites of digital music love to boast about the numbers, but do not discuss mastering sources and/or processes. This reminds me of what marketers of portable radios commonly did in the 1960s by boldly stating the number of transistors incorporated into their circuitry. The more transistors, the better; Right? No! The number of transistors were only part of the equation in determining how well radios performed in the day. I won't argue against some folks claiming they hear better resolution of soundstage, bass, or whatever, but I find those that assert that I need to "train your ears" to be somewhat condescending. I know good SQ when I hear it and it rarely, if ever, comes from HI-REZ audio files, but rather from carefully mastered sources of them.
Again, oversampling does not magically invent greater resolution. It is just allows you to use the existing resolution more efficiently.
