HiFi Music: Listening, Sources, Tracks?

Howdy folks. I want to up my music listening game. Where are the best places, and what are the best methods for me to do so? A few more questions I have:
  • Does streaming high quality on Spotify count?
  • Do I need to use a service like Tidal?
  • What details should I look for besides the music being "lossless"?
  • Can I truly get the most out of HiFi music with standard equipment?
There has to be more to this than I realize and I'll put it to you all to let me know. And if you have links/examples to tracks as well that would be awesome so please share. 🗣THANK-YOU

Feb 1, 2021
For me, I can't really tell the difference between Tidal and Spotify. I'm more of a music discovery guy. Finding music is more important than the quality, and sometimes I'll download Spotify to MP3 and save them to my computer for my music collection.
Aug 31, 2019
First of all, Tidal has more songs than Spotify. Second of all, Tidal offers a free 60 day trial if you sign up for Hi-Fi. While on Hi-Fi, you're able to switch your playback between all 4 different quality tiers. You can switch on the fly for A/B testing. Going from Standard (MP3-128) to Premium (AAC-256 / MP3-320 equivalent) to Hi-Fi (FLAC 44.1/16) to Master (MQA 96/24) I can hear a major improvement with each step up in tier. To my ears, it's no small margin, either. Is it worth the extra $10/month, though? Well, you could always get a couple of friends to join the Family Plan with you, which cost $30/mo. Split 3 ways, that's only $10/mo each, which is the same price as all the other services. If you can mange it, you can have up to 6 members in a Family Plan, and that's only $5/mo ea. Also, you can use PayPal, and PayPal has a feature that lets you instantly send money to another PayPal account. So, you don't have to worry about collecting cash from your friends. You can even set it up to automatically send the money each month. You can also bill them. It really takes the sweat out of it. Getting your friends involved means you'll be able to share playlists, talk about stuff, and get to enjoy a more fun and social listening experience. So, Tidal does not have to be expensive, if you're willing to share. I recommend you start a 60 day trial, and decide for yourself. Get your friends involved, and ask their opinions. You never know, they might already have a subscription, and be willing to let you join their family. You'll be helping each other out, in the end.
Jul 12, 2019
Quick comment... I think it depends on the kind of music you listen to. I listen to a lot of acoustic jazz and symphonic, chamber music. When I listen to rock music (I like Radiohead) I hear a lot of dynamics compression, so there's a distinct lack of dynamic range (difference between quietest and loudest sounds). Contemporary pop music is just loud, all the time. That means having this music in high-res audio formats is a waste. There will be no difference between CD quality or anything higher resolution. BUT... If you listen to high quality masterings of good acoustic recordings, with 12dB or more dynamic range, then you *might* hear a minor difference between a CD quality FLAC and a 24-96k FLAC. (Maybe.) I find that I hear only the slightest difference between 16-44.1k and 24-96k, but it's worth it for me. I can't for the life of me hear any difference between 24-96k and 24-192k. So, long story short, if all you listen to is pop recordings, even 'acoustic' pop music, save your money and go with something of standard 'CD quality' res. Just my humble opinion. FWIW.
Jul 16, 2019
+1. You can put me down for this too... With just one (already discussed elsewhere in this thread) caveat: *Sometimes* there are differences between the 24-bit "Hi-Rez" release and the CD/MP3/etc other than just the sample rate and bit depth. Not always. But sometimes HiRez gets a "better mastering" built to sound better for"audiophiles" who aren't listening as background music but REALLY focusing on a quiet space using good gear. The CD release is (sometimes) made to sound "better" in people's cars, where there's lots of ambient background noise and people don't want to be fiddling with the dial while the background noise changes as a function of how they drive... When it's purely apples to apples and the CD is the same mastering as the HiRez, my experiences and opinions are nearly identical to yours...
Jul 11, 2019
Hi Duncan, The best advice i can give is this: Can you hear a difference? With G6 > MCTH > HD800 the difference between lossless and lossy audio formats is painfully obvious. Its still very clear running G6 to AKG k553. MP3 is by far the worst. Harsh, whistling and pathetic. AAC and MP4 just lack any kind of engaging texture, soundstage, timbre or realism. Moving up to high res yields improvements, but only really on well recorded, mixed and mastered material. Thats what i hear, with my equipment. Maybe to you it wont matter? Please tell us more about how you listen. It could be that there are other areas that you could make more significant improvements such as your DAC and amp? In brief, dont ask. Try it yourself! If there is a difference, you'll hear it!
Jul 5, 2019
I attended a lecture on high resolution audio by a pioneer in the field, and he basically said that high res audio is a scam. He explained that many ultra-high resolution recordings actually sound worse than CD quality, because they mix in the noise floor which is filtered out of CD quality recordings. His words, not mine. I've had both Tidal and Spotify. I did a lot of research and couldn't find any evidence that anyone can hear the difference between the Spotify's sound quality and Tidal. In fact, the scientific evidence points to the contrary. I have golden ears and I really care about this question, but when I went back to listen critically, I confirmed it for myself. The main advantage of Tidal is you can pair it with Roon. (On Android, USB Audio Player Pro plays a similar role.) Depending on your platform, Roon can sometimes squeeze out a bit more fidelity by optimizing system resources for audio playback. I've tried Roon and it does seem to produce a just noticeable difference on my Mac. But it has nothing to do with the bitrate. Long story short, I gave up Tidal and kept Spotify Premium because of its larger selection and lower price. I haven't regretted my decision.
Jul 22, 2019
Very interesting! Thank you for your impressions. Please feel free to update if your ideas develop. Personally I have never touched DSD. I do hear discernible improvement with higher resolutions with my rig. As for MQA, I only heard it once on a vastly superior rig (DCS Rossini > SPL Phonitor). Needless to say it was superlative. I believe much of it has to do with the ability of the DAC to make use of a higher bandwidth description of the analogue signal. With USB for example, a more volatile usb mode is necessary to communicate more bits, and as a result errors are much more common. I didn't set out to write all that, but since you are taking this so seriously I thought you might be interested.
Jul 30, 2019
Since the Liquid Spark amp seemed like a weak link in my system, I upgraded it to a Graham Slee Solo Ultra Linear, which is near-reference quality. My system is now Tidal app on Galaxy S10+ --> Audioquest Jitterbug USB decrapifier --> Audioquest Carbon USB cable --> Chord 2Qute DAC --> Cardas Quadlink 5C interconnects --> Graham Slee Solo Ultra Linear headphone amp --> Sennheiser HD6XX. The upgrade to the amp fixed the problem with MQA where it was sounding worse than CD quality. MQA at 192 kHz decoded from the Tidal app now sounds excellent on good recordings. I can no longer distinguish MQA from CD quality, but that goes in both directions. Meaning I don't hear any clear improvement of MQA over CD quality. On DSD high-res test tracks I can't reliably hear any difference from the identical track in CD quality. There might be some microscopic differences but they're not better or worse overall. If anything the soundstage collapses at DSD256, but that might be a limitation of my DAC.
Jul 3, 2019
Ohhh Niko..... so limited... and yet,..... so vocal ...
May 5, 2019
Here is my take: Spotify Premium is 320kbps MP3 which is very compressed. CD's and Tidal HiFi ($20/month) are 44.1khz (which translates to about 44,100kbps) so more than 100 times the sampling rate of Spotify Premium. Tidal HiFi are FLAC files. That means they employ data compression similar to ZIP. The difference between this type of compression and the compression employed by the MP3 is that the decompressed version of a FLAC file is an EXACT copy of the CD source. This is FAR from the case with even the best MP3's. There are other lossless formats: ALAC (used by Apple) and WAV (no compression). To get the benefit lossless formats offer you will need a decent Digital Audio Converter. Forget the one in your PC. Unless you have a high end mixing board, you can be certain it is garbage. If you use your PC to stream, you will need to make sure you are sending the pure digital stream to your external DAC. Some phones have decent on board DACs: Top of the line LG and Samsung units for example. Even in these units, you will get a benefit by incorporating an external headphone amp. TIDAL HiFi (which I subscribe to) produces FLAC files of CD quality with some recordings that are 192Mhz at 24/bit (as opposed to 44.1/16 bit for CD) using a technology called MQA which allows them to be played by devices that do not support 192/24 input. Since I use an external streaming device that does not support MQA (Cambridge CXN) I cannot comment on the difference in quality. Clarifying a little to get 192/24 performance from tidal, your DAC must specifically support MQA as well as the 192/24 rate. This is because MQA in effect sends both 44.1/16 and 192/24. Units that do not support MQA will ignore the 192/24 signal. It is also significant to note that at the time of this writing TIdal Masters recordeings are only supported on the TIDAL desktop app and a few select streaming devices. The list includes ROON, and there are a large number of Network Audio players that support ROON. ROON is a subscription service: $199/year or $499/lifetime. Even higher sampling rates are available in other formats (DSD for example). I do not know of any streaming service that includes such files. Generally you purchase downloads of these ($30-$40 for an 'album') and store them on your own media - usually a hard drive or 6. These types of files require high end equipment to decode properly and the files tend to be quite large, so if you want to go that route, prepare your wallet. Another downside of these files is they are generally not playable on portable devices. A word about TIDAL in your car: Android Auto does support TIDAL, but the current state of the Android Auto interface is it sucks, so I would recommend interfacing directly to your phone. Depending on the capabilities of your head unit (radio) you can (in order of preference) (1) send the digital stream to your head unit and use its DAC; (2) send the audio stream to your head unit (via the 3mm jack) In this case you are using your phone's DAC which could be good or bad depending on your phone; (3) Send audio via Bluetooth to your head unit.
Aug 19, 2020
"MP3 is to be avoided as it sucks." Well, it will greatly depend on the sampling rate. A 128 kbps mp3 sucks, a 192 kbps mp3 usually does not, 320 kbps even better but probably overkill. Of course may vary with certain types of music, and some people have "golden ears". Whatever, but making blanket statements like mp3 sucks is grossly misleading. If you want to archive your music in a lossless format, don't use mp3. But if you just want to enjoy listening to it, it's a tried and true and still very good format.
I just don' t get the reason behind continuing to use lossy codecs for local files in 2020. I have over 500gb of storage between my onboard and sd card for my phone. I get it for the iPhone crowd as they have fixed storage but for most android users and people with DAPs there is just no point. I don't need to carry around 112,000 songs with me at all times, I have hard drives for that. Streaming services allow you to save content offline so mobile data limits shouldn't be a problem for most folks. Lossy codecs were created due to storage and bandwidth concerns over 20 years ago... we don't have those same limitations today.
May 4, 2019
Tidal supposedly has Master tracks...ive just signed up for the trial and so far Tidal tracks seem cleaner to me. Im a noob in this HiFi stuff anyway, so take that with a grain of salt. What sounds better to me may not sound the same for you.
Apr 13, 2019
They all coint, but for kost of the music, the source file matters the least.
Apr 27, 2019
In fact, the hires standard itself proves nothing whether human hear them(they use a different sensor for higher range) and I personally cannot blind test myself between standard loseless and 96 24. Plus, one of the audiophile hearing test from Philips actually implies that mp3 identification was much harder over frequencies, soundstage, noise... What hires pcm could do is that higher sampling most of the time means a lower noise in the file, dsd has same advantage when they simply pushes all noise to higher inaudible frequency and cut off using low pass filter.(yes, one of the reason why you feel dsd is much more clear, even if you pass a standard loseless to upscaler like dcs Vivaldi for conversion) Seriously, the file itself matters the least unless you are listening some stupid 128k stuff.
Aug 19, 2020
One of the better comments. 192 kbps mp3 sounds great to me. As long as the butrate is not too low, you should be fine but let your ears be the judge.
Mar 24, 2019
A good recording makes a huge difference, especially the mastering phase. Most sound engineers tend to compress the sound (don't confuse this with data compression) so that it sounds louder (google for loudness wars). This usually means that parts of the audio spectrum are misrepresented and basically forces you to a specific type of listening experience that dates back to radio broadcasting (on radio broadcast you had to be loud to survive the RF noise floor). This "tradition" is present on a lot of studio releases, even now that we have better ways of storing, reproducing, and transmitting audio. You 'll have to look for a release (or a label) that focuses on sound quality, and doesn't mess up with the mix, but presents it in the most neutral/natural way possible so that -if you want- you can "tune" it later on to your own preferences, or just listen to the thing as it really was. No matter how much money you are going to pay for your HiFi system or the X subscription service, nothing can fix a bad recording ! The best experience to me is a live performance, nothing beats the real thing, especially for jazz or classical music, you also better support the artists this way, it's a win win situation. So to answer questions about what details you should look, check out the various releases of an album you are after and get the one that has the best reviews in terms of sound quality, communities such as discogs or musicbrainz can help you with that (whatcd used to have a lot of information on the sound quality of each release, too bad it's dead). Once you have a good recording, you are looking for a good media that can hold it. Studios use ADCs and processing tools, usually with 24bits of dynamic range and 96KHz of bandwidth. This way they can sample audio signals (google for Shannon-Nyquist sampling theorem) up to 96KHz / 2 = 48KHz, with a dynamic range of 144dB (~6bits per dB so 24*6 = 144, in reality it's mostly 118 - 120dB at best). Our ears have a dynamic range of almost 120dB under ideal conditions, so 24bits are just what we need since it goes in multiples of 8bits (an octet), 16bits (what the CDs use) give us ~96dBs which is mostly OK and 32bits are just too much (but it's more "natural" for e.g. a PC so it helps when processing). As for the frequencies of the audio signal, the audible spectrum is (theoretically) 20Hz - 20Khz, we can feel sounds outside that spectrum (especially the lower end of it, with proper speakers) but in most cases, most people are not able to listen to anything above 20KHz, especially as we get older, where we loose perception of higher frequencies. So why do we go for 48KHz instead ? In digital recording the audio signal is split in pieces/samples (or if you prefer quanta), it's not continuous as in an analog recording, this affects both the recording and the reproduction of the analog signal. To put it differently there are volume changes that are too small or too short to be represented. A rough example for a small change is to have a value of 1.4 and only be able to write 1, 2, 3 etc, so 1.4 will become 1 instead and some accuracy will be lost. This is known as rounding error (or quantization error) and is a thing both ADCs and DACs has to deal with (google for dithering). A rough example of a short change is to think of a repeating pulse, if for example there is a beep every 1 second and we sample every 2 seconds we can't distinguish this beep from another beep that goes on every 2 seconds or from a continuous beep. In a worse scenario if we miss the bit for less than a second we'll never hear it ! So from a beep that goes every 1 second we get three possible signals (including the silent one) or "aliases" along with the original. A similar thing happens on the DAC side when in the process of re-creating the analog signal, if we get the 2 beeps/sec signal, and the DAC reads 2 samples/sec, it will also produce a constant beep along any other analog signal that could result these 2 beeps/sec, e.g. a 1 beep/sec, 1beep/0.5 sec etc, all with higher frequencies than the original, another version of "aliasing" also known as "imaging" in this case. In order to avoid such noise, we need to have a low-pass filter on the ADC input so that signals with frequencies higher than the maximum frequency the ADC can sample do not get recorded, and on the output of the DAC so that any generated "aliases"/"images" don't reach our ears. This filter is called an anti-aliasing filter when used on ADCs, and a reconstruction filter when used on DACs. Its cutoff frequency is the ADC's/DAC's Nyquist frequency, which is half its bandwidth, so for an ADC/DAC with 96KHz bandwidth, the cutoff frequency would be 48KHz and for one with 44KHz bandwidth it would be 22KHz which is just above the audio spectrum. So now you know why the CDs have a bandwidth of 44KHz ;-) Since the aliases produced by the DAC will be above the audible spectrum, people have argued that there is no need for a reconstruction filter or any further processing needed. This makes sense especially for DACs that operate on a single bandwidth such as the DACs on CD players that only work on 44KHz, and have an adequate dynamic range of at least 16bits. This approach is known as filter-less, non-oversampling (NOS) DAC design and is only possible for DACs, on ADCs we always need an anti-aliasing filter, as for the oversampling, we'll come back to that later on. In practice a filter is not ideal, you can't have a filter that goes from 100 to 0dBs instantly, the same way a car can't stop instantly, it needs some "space" to work (aka transient response), which means that you have to start filtering before 22KHz in order to reach the desired level at 22KHz. So an ADC/DAC that operates on 44KHz of bandwidth will need to start filtering from ~18KHz in order to reach e.g. -60dBs when it reaches 22KHz, which means you will loose some of the audio on high frequencies (to some people this seems like a "warmer" sound and they are OK with it). To deal with this, we may use a higher bandwidth, so for example instead of 44KHz, we can use 48KHz and filter anything above its Nyquist frequency (24KHz), only now the DAC will have 2 more KHz to work with so it will start filtering from ~20KHz instead, and get closer to the "20 - 20" range. We could be fine with 48KHz (and many people are) but there is still another issue we have to deal with. The filter's response is very important for the sound quality, it doesn't only affect the level of signals above the cut-off frequency (in this case the Nyquist frequency), but also their phase. In other words the signal gets a delay while it gets filtered, relative to its frequency, which in turn affects among other things the stereo imaging of a stereo recording. Stereophonic audio depends on the L and R signals being in-sync, if you create a time difference between them -aka change the signal's phase- you break this and you get less separation between the 2 channels, hence the "sound stage" becomes narrower. This distortion may happen on frequencies even lower than the cutoff frequency which means we need to move the cutoff frequency (hence the Nyquist frequency / bandwidth) even more, since the "steeper" the filter the greater the distortion. Creating good filters is a challenge, especially in the analog domain, on the digital domain on the other hand we have more options available. Instead of filtering the signal outside the ADC/DAC, we can do the filtering on the data itself, at a much higher bandwidth than the one used for A/D conversion, a process called oversampling. So for example in the case of the ADC we'll get an input signal of 24KHz bandwidth, do the A/D convertion using 48KHz bandwidth, oversample it to 96KHz (*2), 192KHz(*4), 384KHz (*8), 768KHz (*16) or more, do the filtering there at a much better resolution and either output the oversampled signal for processing, e.g. a 24/96KHz signal, or down-sample it to the desired bandwidth. This way we'll not only be able to filter out the aliases, we'll also be able to further filter out the noise created due to quantization. On the DAC side, an input signal of 48KHz will arrive, it'll get again oversampled to 96KHz (*2) or more, do the filtering there and output either the oversampled signal, or a down-sampled version of it. That oversampling now can be a huge mess because the DAC needs to "invent" new samples to place in-between the existing ones, a process called interpolation. If this is done poorly, instead of making things better, it can add distortion to the audio and "dry" it up, it may also create phase distortion. Even worse the bandwidth of the oversampled output signal may not be a multiple of that of the input signal, for example 48KHz is half of 96KHz, so if you get an input signal of 48KHz and oversample it to 96KHz you need exactly twice the samples, which makes it easier since you'll just need to place one sample between each two successive samples. If however you get an input signal of 44.1KHz and the output bandwidth is 96KHz you are in trouble, you'll first need to resample the input signal to a rate that is divisible by both 96 and 44 (aka Least Common Multiple), do the processing there and then downsample it to the output bandwidth (44 -> *24 -> 1056 -> / 11 -> 96), or even use multiple resampling steps (e.g. 44 ->* 12 -> 528 -> / 11 -> 48 -> *2 -> 96, so that it re uses 48 to 96 conversion). To avoid this some DACs also support native handling of 44.1KHz so they can also output 88.2KHz (*2), 176.4KHz (*4) and so on. In general resampling is tricky and can add distortion, especially the "uneven" kind, you can get an idea here -> http://src.infinitewave.ca/. The reason I mention all this technical stuff is to give you an idea of how complex a DAC can be, a hint of why e.g. a 32/768 DAC may not be better than a 24/96 DAC, and also to point out that this whole marketing with higher sampling rates and bit rates doesn't make much sense if the rest of the DAC is not implemented properly. Other things that are important for ADCs/DACs is to have a solid clock, one that doesn't loose ticks (google for clock drifting), when that happens the ADC/DAC will skip samples and generate a very ugly noise known as jitter or "digititis". A poorly designed filter even with good resampling in place, can be too steep for example and generate "ringing" noise. In general don't let the bit rate and sampling rate numbers fool you, as with any audio equipment, you should listen to a DAC before buying it, or at least look for trusted reviews. Just don't settle for anything less than 24/96 and look for DACs that can also do multiples of 44.1KHz as well such as 88.2KHz (*2), 176,4KHz (*4), 352.8 (*8) etc, also it's a good idea to check the specs of the DAC chip, since some times the manufacturers of the device will fail to mention that capability since the marketing is mostly on multiples of 48KHz. In terms of digital audio encoding, FLAC is probably your best option, then comes ALAC that's also lossless, and then you go for lossy encodings such as OGG/Vorbis with a high quality setting and then MP3 above 192Kbit/s with multiple passes. What you want is to have the whole spectrum available so that you don't loose any spectral content and the more lossy the encoding, the more spectral content you loose. Obviously if you get a lossy file and re-encode it to lossless, that won't fix your problems and there are marketing people out there that do just that so watch for it. You may use a tool such as spek to figure out if your file is truly lossless and retains the full spectrum. Because of the lost spectral content, if you try to post-process a lossy MP3 or OGG, you'll get various artifacts because the filter or the audio compressor / limiter will see an incomplete spectrum and add noise for the missing parts, you can get away with it with various re-construction tools that try to re-create the lost spectrum by generating harmonics (look for "bass enchancer" or "crystalizer" or "exciter" etc) but again these need a lot of tweaking to get it right and are not suited for all types of recordings. Analog media such as Vinyl or Tapes have a dynamic range of less than 110dBs (110 is a perfect vinyl, it's usualy 60 - 70dBs) and in case of Vinyl you can go up to 50KHz in terms of frequencies (quadraphonic records used such frequencies to encode the rear channels). You don't need anti-aliasing filters used on ADCs/DACs but you need pre-epmhasis/de-emphasis filters such as the RIAA curve on Vinyl (that's what a phono stage does) and the various Dolby filters on Tape. In general the analog path is closer to the way our ears operate and it sounds more naturally, for example in case of overdrive (think of electric guitar distortion), a digital recording behaves worse than an analog one (google for clipping), it distorts the sound pretty badly, where the analog recording will behave more like the real thing (obviously there are high quality tools for dealing with this during mastering so hopefully you won't get such issues on the final mix). However in order to get that experience you'll have to pay a lot of money for your equipment and properly maintain/store your media since they tend to degrade with time. It's also useless to go for analog media when the recording itself is digital, unless your only alternative is the CD version which can indeed be worse since it has to be downsampled to 16/44.1. Another thing to keep in mind is that some engineers do a different mix for the digital media than for the Vinyl version so the Vinyl version may sound better because of that since it's more focused for audiophiles, but in general Vinyl or Tape is not better or worse than a digital recording, again there are lots of details to take into account. As for the rest of the system:
  • For the amplifier check out its linearity and total harmonic distortion. The best amplifiers are those that behave the same across all frequencies (aka linear amplifiers), they are the most expensive and eat the most power, this property is found mostly in A-class amplifiers. If you think of a speaker moving back and forth, A-class amplifiers amplify both the forth and the back movement, B-class amplify only the forth movement and let the speaker come back on its own due to its elasticity, saving power in the process, AB are a hybrid between them, C is garbage, D use PWM for more efficiency and are more common and cheap, you can get high-quality D-class amplifiers but it takes a lot of work to design one.
  • For the Headphones/Speakers the larger the better IMHO, in case of speakers make sure you get something that fits your room specs, getting speakers to loud or too weak for your room will do you no good. Listen before you buy, even reviews here won't help you much, get something that matches your ears. It's easier to get the Speakers wrong than headphones so I suggest you start with a good pair of headphones such as the HD6xx, it'll cost you less and you also don't depend on the room. A good reference is if you have frequency response diagrams from a trusted source.
  • For the cables, prefer balanced systems over unbalanced if your cable lengths are longer than a few meters to avoid interference. The lengths of the cables matter, especially for multi-channel setups, but not that much, most people don't notice anything related to the stereo imaging and different cable lengths. The materials of the cables are mostly a marketing thing, just make sure it's not crap and the connectors are properly attached.
Since you asked about tracks, this is what I use for determining the quality of audio gear:
  • Nardis from Bill Evans (the Riverside recordings) to check how well the bass is represented (it's hard to get jazz bass right, it's usually "muddy")
  • Because from Beatles (the original stereo mix) to check the stereo imaging (huge stereo separation on this song)
  • Various tracks from Ozric Tentacles (I usually go with Kick Muck) for the dynamic range
  • Various tracks from Alan Parsons Project (Alan Parsons is a legendary sound engineer, among other things), especially for verifying a neutral post-processing (these are engineered so well that any post-processing messes them up, just don't touch them !)
  • The Dark Side Of The Moon by Pink Floyd (look for the Japanese releases) (Alan Parsons was the sound engineer on this one btw) because I've heard it a thousand times
  • Rudy Van Gelder (RVG) recordings from Blue Note (another legendary sound engineer)
  • Symphonic music in general (I love Tchaikovsky, look for Swan Lake, check out Deutsche Grammophon and other relevant labels, also check out for any good recordings of Moriconne)
  • Various disco / synth-wave etc tracks for checking dynamic range (80s productions are in general awesome, for more recent stuff check out Daft Punk and more recently I found Chrome Brulee that I enjoyed)
In general pick some tracks that you are familiar with, you'll instantly notice the difference, and most importantly don't let anyone tell you what "sounds good" or not, just because it's not FLAC or it's 16/44.1 instead of 24/96 or 32/768. Hope this helps + it wasn't too much :-) P.S. For more information https://www.audiosciencereview.com is a great resource ! It also contains trusted reviews on DACs and other equipment. P.S.2 Here is an interesting site that monitors the dynamic range of various recordings, it should help you verify if a recording was overly compressed or not -> http://dr.loudness-war.info/
Jul 3, 2019
Here is a chart with the frequency range produced by most musical instruments, it shows a range of approximately 25 hz to 14Khz , CD quality 16/44.1 is more than adequate to represent any kind of music , anything beyond that is totally redundant and mostly marketing hype (like audio cables above a $100) if one considers that most people can't hear any frequencies above 18Khz and musical instruments don't produce any frequencies above say 15khz

Oct 15, 2020
Oh no! Audiosciencereview?? A "great resource"? You must be kidding me. Read "a new methodology for audio frequency power amplifier testing" by Daniel Cheever. Essentially, THD may be useful for engineers, but it is COMPLETELY INADEQUATE as a figure of merit! ASR assumes that THD is a figure of merit, and ranks DACs as such. However, there is in fact no correlation between the measured THD of a DAC and how good it sounds. Stop spreading ignorance and superstitious scientism.
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May 17, 2021