Audio-Quality - Page 1
Forum section: electronics / general

author: Write_to_Smokegenerator
author_rank: Regular Contributor
timestamp: May 18, 2012, 11:42:42 am
content: Hello EveryoneToday I saw a comment on a forum, someone claimed that 16bit 44.1 KHz is to "low" and wanted 24bit 196bit (for Cd's, DVDs etc.).First i thought wtf? But I remembered that a friend of mine also claims that he hears the better audio quality.Well I never heard any difference so I can't say anything about that.But if my understanding is correct it seems it is complete unnecessary to raise the sample frequency, because of Shannon's theorem and the fact that you can only hear up to 20kHz.Next issue is the resolution: I played with audio-samples years ago. Once I did replace the 2 last bits (of 16bit )with "0" to see what happens, well I couldn't hear any difference at all. So in my opinion it is completely unnecessary to go up with the ressolution (we're talking about home entertainment)What do you think about that?Well if my understanding is false correct me pleasecheers.

author: Psi
author_rank: Super Contributor
timestamp: May 18, 2012, 12:25:46 pm
content: Quote from: Write_to_Smokegenerator on May 18, 2012, 11:42:42 amYeah, 24bit at 192khz is totally insane for listing. (however for editing/mixing it is useful)Under some circumstances you can tell the difference between say 44khz and 96khz.eg the right audio track and expensive headphones/speakers like Sennheiser, Klipsch etc.. but it's pretty hard unless you have the full 20khz hearing range.Personally, i find shannon's theorem a bit hard to accept.I know it probably is true but still.. 20khz signal being sampled at 44khz only gives you around 2 samples per waveform.

author: DrGeoff
author_rank: Frequent Contributor
timestamp: May 18, 2012, 12:31:11 pm
content: Quote from: Write_to_Smokegenerator on May 18, 2012, 11:42:42 amTry listening to a 10kHz sine wave, and then switch it to a square wave. Perceive any difference? Neve reckons that frequencies up to 50kHz are important and designs his equipment as such.However think about spatial resolution. Human hearing can distinguish between signals arriving at each ear separated by a very small amount of time. When recording acoustic performances in natural settings (orchestras, choirs etc) the perception of space has a lot to do with the accuracy at which the signals arriving at each microphone can be separated. If they get quantized in to a single sample then you lose spatial accuracy. So a higher sampling rate is useful in this regard. Of course this does not really have any effect on close mic'ed instruments or pop/rock productions since the mix engineer is responsible for placing each player in the sound stage.Quote from: Write_to_Smokegenerator on May 18, 2012, 11:42:42 amIt makes a huge difference, particularly in natural reverb tails, where the quantising noise is very apparent (this is also where you will hear a lot of codec artefacts). You will need to use headphones to really pick this up.Also, when mixing digitally, many complex algorithms are used to manipulate bits. To avoid rounding errors compounding as more and more algorithms are applied (gain, eq, dynamics, reverb etc) DAW's usually start with a 24-bit audio recording and turn this into a 32 or 64 bit floating point number, so the internal calculations can be accommodated with a high degree of accuracy. It's only when the material is finally mastered for CD that the bit depth is reduced, and usually dithering is added to the LSB to eliminate quantising noise.

author: EEVblog
author_rank: Unknown
timestamp: May 18, 2012, 12:32:30 pm
content: For playback, 16bit 44.1K is the CD standard, so most people wouldn't have a system (or golden ears) capable of truly hearing major differences in double blind experiments, especially with complex content like music.44.1K is fairly "low" when it comes to recording high frequency reference quality content though, so much high sampling rates are used in recording studios.24bit audio recording is commonly used because the greater dynamic range can be very useful, and I think is mostly the professional industry recording standard?In effect, when recording and mixing master quality stuff, you'd "guild the lilly" and go much higher than actually required, so 24bit/196K for example.But to actually play back say 24bit/196K audio vs 16bit/44.1K, yeah, you be hard pressed to tell the difference I suspect.And as always, what people claim they hear is usually rubbish. Only true controlled double blind experiments can give an indication if there is a real difference.Dave.

author: ejeffrey
author_rank: Super Contributor
timestamp: May 18, 2012, 12:39:51 pm
content: Yes, 16 bit, 44.1/48 kHz is fine for storing audio data. 16 bits is plenty of dynamic range and 22 kHz is plenty of bandwidth.Oversampling in ADCs and DACs is not a bad thing. This means that you digitally upsample your 16-bit / 44.1 kHz audio to a higher sample rate (and possibly bit depth) before sending it to a DAC or that you convert with a high sample rate ADC and use digital filtering to downsample to CD quality. This won't create signal that isn't in the original, but it can ease the difficulty of the analog anti-aliasing filter, shifting the burden onto a digital filter which is cheaper and better. This is in fact the principle of sigma-delta converters. A 16 bit 44.1 kHz sigma-delta DAC is perfectly fine for audio use, even if a 16 bit 44.1 kHz resistor ladder DAC might not perform very well or require a very complex anti-aliasing filter.Likewise, 24 bit audio has some value in audio signal processing applications. The extra dynamic range means that your don't need to be as careful about rescaling and it avoids the buildup of quantization error through multiple cascaded digital filters. Generally signal processing is done better with floating point numbers, but if you are stuck with fixed point DSP hardware you might want to go with 24 bit internal representation for some extra headroom.But for waveform storage, CD quality is good enough. It has plenty of dynamic range and sample rate to hit the limits of most humans hearing. A few people (mostly children) apparently do have demonstrable hearing capacity out as far as 30 kHz. Even so, it is rarely desirable to actually produce any sound in that frequency band. More often you just want to make sure that your hi-fi isn't putting out unwanted noise at 30 kHz that doesn't exist in the source material.A completely unrelated issue is that high resolution sources like SACD and DVD-audio usually have different mixes than the consumer CD/MP3 crap. So a DVD-audio may have considerably better sound than a CD of the same music, but the difference is in the mixing not the bit depth.

author: amspire
author_rank: Super Contributor
timestamp: May 18, 2012, 12:47:14 pm
content: Quote from: Write_to_Smokegenerator on May 18, 2012, 11:42:42 amIf you had a "perfect" 22KHz filter, 44KKhz sampling would be fine. Unfortunately there is nothing close to a perfect filter. If you make a filter that is flat to 20KHz and -90dB down at 22KHz (still pretty impossible), it would have massive phase shifts at the higher frequencies. 196KHz means you can extend the filter passband well above the audible range,and the filter can have much better phase response, and a much lower order (less stages) which is all good.If you amplified a 10KHz squarewave, the 44KHz sampling and filtered waveform would not look a lot like a square wave. The 196KHz would look close to a squarewave. So 44KHz is great for pure single frequency sinewaves. 196KHz is better for complex waveforms.24 bit gives you much better dynamic range. 16 bit means noise somewhere at at around -90dB, but say you wanted to give yourself 20dB of headroom above the average sound level, your noise is now at -70dB which may be audible, particularly in the quiet parts of the music. Another way to look at this is if you turned the volume up to 120Db, the noise would be at 30dB which would be very obvious in a quiet period. About the same as a quiet PC supply fan. With 24 bit, you can give yourself an extra 48dB of headroom above the digitizing noise level. If you turned the volume up to 120Db, the digitizing noise is still inaudible.16 bit 44.1kHz means that the digitizing will degrade the audio a little. 24 bit 196KHz means that the degradation is essentially insignificant and outside any audible range.If the music you were listening too had some compression to limit the volume variations and it was normalized to maximize the signal amplitude, going to 14 bit digitizing would probably not be audible. If you were listening to an uncompressed rendition of the 1812 Overture where the true headroom for the cannons has been added, you would notice the effects of the missing bits during the quiet parts.Richard.

author: Psi
author_rank: Super Contributor
timestamp: May 18, 2012, 12:49:16 pm
content: You're far better off to spend your money on a good set of headphones/speakers than on high bit/sample rate equipment above 44khz/16bit.The difference between a normal pair of headphones in the NZ$50-100 range verses a top quality brand in the NZ$300-600 range isverynoticeable.Personally i really like the Sennheiser(headphone) and Klipsch(speaker) brands but there are other good ones too.If you ever get a chance to hear a NZ$300+ pair of Sennheisers or NZ$2000+ Klipsch Reference speakers you should definitely try them out.I still have a pair of 15 year old Sennheisers (and they still sound quite good). But recently i bought a pair of Sennhesier PC360's which are a gaming headphones+mic. And DAMN, music on them is like sex in your ears.I was expecting them to be very good and was happily surprised that they were better than i expected

author: DrGeoff
author_rank: Frequent Contributor
timestamp: May 18, 2012, 12:49:53 pm
content: Quote from: EEVblog on May 18, 2012, 12:32:30 pmAnd most releases are highly processed anyway, so you'd never know if there was any difference. Pumping drums, synth riffs and screaming lead breaks still sound the same at 16/44 as they do at 24/96.Quote from: EEVblog on May 18, 2012, 12:32:30 pmMost people still record at 44/48/96 in studios. Not many use 192. When you are mixing with 100 tracks or so, that ends up wasting a lot of disc space and putting unnecessary load on the DAW. If you are recording and mixing for a CD release, 44.1k is still fine for recording, since you don't have to sample-rate convert to the final master. For video, 48k is usually used.Quote from: EEVblog on May 18, 2012, 12:32:30 pmParticularly when a large proportion of the listening public are using mp3, which has another set of artefacts thown in.196K is over the top. 44/16 is adequate for most. The bit depth is only important when you have natural acoustic recordings and are fussy about the detail in the reverb tails (now that has to be a minority).Sampling rate of 96k can be useful for spatial accuracy, but really it's not that important that anyone would notice Classical acoustic recordings at 24/44 are perfectly adequate for most.

author: olsenn
author_rank: Frequent Contributor
timestamp: May 18, 2012, 12:50:23 pm
content: My PC soundcard (Asus Xonar D2X) can be switched between 16/24-bit and 44.1KHz - 192KHz sampling, and I honestly can hear A difference, but I'm not sure I'd say it's better, just different. Furthermore the qualitative differences in settings changes between songs, so I wouldn't be surprised if it's not at all the sampling rate and bit-depth that I am hearing as much as different paths along the circuit having different noise floor's etc.One thing that makes a HUGE improvement in audio quality is to use ASIO instead of the default Windows audio kernel.

author: jahonen
author_rank: Super Contributor
timestamp: May 18, 2012, 01:55:24 pm
content: One thing what is often missed is the time domain implications of the sampling theorem. As it is defined in the frequency domain, it does not say anything about the shortest burst of the tone which can be recorded. Closer we get the Nyquist limit, more samples we need to reconstruct the signal. That means that transient response of the tones which are close to Nyquist limit is going to suffer (short transients require more bandwidth of course). For fs/2, infinite number of samples is required.While there are obvious restrictions on human hearing, the computing power which is used to turn it into hearing experience is quite impressive, so one should not immediately jump into conclusions. All aspects of human hearing are not still known, perception of pitch for example. To find out if something is really audible or not, requires careful testing.But on topic, data storage is quite cheap nowadays, what is wrong with some headroom?Some other thoughts on the subject:http://www.hypex.nl/docs/papers/AES123BP.pdfRegards,Janne

author: IanB
author_rank: Super Contributor
timestamp: May 18, 2012, 02:21:47 pm
content: Quote from: Psi on May 18, 2012, 12:25:46 pmTake a look at the oscilloscope displays in Mike's recent pill cam tear down video:http://youtu.be/bH6i3bfie_EJump to 15:30 or thereabouts to see how the sine wave is reconstructed from few points.

author: Psi
author_rank: Super Contributor
timestamp: May 18, 2012, 02:56:35 pm
content: Quote from: IanB on May 18, 2012, 02:21:47 pmI guess the question is, do some(all) sound cards interpolate the data and reconstruct the waveform or do they feed it out to the DAC direct?

author: jahonen
author_rank: Super Contributor
timestamp: May 18, 2012, 03:01:05 pm
content: Quote from: Psi on May 18, 2012, 02:56:35 pmI think most do interpolation (aka oversampling), that is a built-in feature of delta-sigma DACs which is common audio DAC topology.Regards,Janne

author: SeanB
author_rank: Super Contributor
timestamp: May 18, 2012, 03:19:37 pm
content: From my own experience a higher sample rate is better, as well as a higher bitrate in the recording> I digitised a few of my records, and recorded at the best the sound card could do ( 48kHz and 16 bits) so that I could have the best possible original material. Then I tried various bitrates for saving into MP3, and settled on 320k MP3, as I could not hear the difference between this and 240kHz, so went with the higher.The biggest problem is most material is such poor bitrate to begin with, the DVD audio ( rare of course) having the best quality, aside from having the original master tapes.

author: ErikTheNorwegian
author_rank: Frequent Contributor
timestamp: May 18, 2012, 03:22:10 pm
content: And don't forget age... we loose "frequency response" as we age.. ehh.. You said what.. speak up!!My hearing is now lost, and i think is in the earea of MP3 quality all over. So good hearing is somthing we sould take care of from we are young, and sadly this generation is loosing out bigtime.Read more..https://mustelid.physiol.ox.ac.uk/drupal/?q=acoustics/clinical_audiogramsAnd don't forget psychoacoustics..Read more..http://en.wikipedia.org/wiki/Psychoacoustics(And funny about this tread.. i go a MLSSA card (ISA Card, Windows 98!!) incoming next week.. fabulous measuring tool for sound measurement.. Doug Rife`s little masterpeace.)

author: SeanB
author_rank: Super Contributor
timestamp: May 18, 2012, 04:31:45 pm
content: I still have a PC running Win98 at work, and it has ISA slots..........I also have a Novell server with ISA slots as well.

author: Unknown
author_rank: Unknown
timestamp: May 18, 2012, 10:09:10 pm
content: Quote from: jahonen on May 18, 2012, 01:55:24 pmGood point. I wonder how common this is in music, however, since I don't think the transient response of our ears is very good either, especially at high frequencies.Quote from: jahonen on May 18, 2012, 01:55:24 pmNo problem in the frequency domain, since an infinitely repetitive signal is assumed. In real life we don't use true sinc(x) interpolation either, so it's better to stay away from this Nyquist limit.

author: beaker353
author_rank: Unknown
timestamp: May 18, 2012, 10:23:04 pm
content: Where you would see an advantage with higher bit depth and sample rate are highly dependent on what the material is. Lets first look at bid depth.Most popular music people buy commercially like on iTunes or a good old CD-DA has the dynamic range of next to nothing. Most of the producers compress (as in audio dynamic compression) the music to within an half inch of its life because people don’t want to constantly adjust the volume on their playback device and it makes for better conversion when lossly data compression like iTunes is used. For this material, I doubt even the best industry ears could blindly pick out a 16-bit verses 24-bit version of the same material. However, or material that is more minimally processed and has a high dynamic content, like classical music, a higher bit depth means that you still have high resolution on a quieter sections. Each bit you add or subtract you gain or loose 6dB of dynamic range. A 16-bit audio signal has the dynamic range of 96dB. A 24-bit audio signal has a 144dB dynamic range. Master source material is often recorded in 24-bit these days because it offers a healthy safety margin for the operator while still making sure your digital noise floor will be eventually lobed off the bottom once mastered.Let’s take the average drum kit for how the source material determines sample rate needs. The higher the sample rate, the more accurate the stored information represents the original signal in particular the higher frequency ranges. Take the kick drum. Other than some odd harmonics that are so far buried in the noise floor you have no chance on recording them, there would be no difference in a recording at 44.1kHz and 192kHz of a kick drum. None. However, record the overheads cymbals and even a untrained ear could pick out at least identify a difference between the two sample rates due to the high frequency content with harmonics that reach way up beyond human hearing. The general rule of thumb is your sample rate needs to be at least twice your highest frequency you want to sample. Most engineers, me included, feel that that ratio really should be 4:1. This is why 96kHz seems to be the sample rate of the top end recorders.Other huge part of the discussion that I will only briefly touch on resides around the high end digital effects processing you will find in ProTools and like systems. In these cases bit-depth and sample rate become much more important as the very complicated math inside these possessors works much better simple with more data points.-EM(Full Time Audio Engineer for Cirque du Soleil, Part Time EET Student and Electronics Hobbyist Inspired by Dave…)

author: Psi
author_rank: Super Contributor
timestamp: May 19, 2012, 12:02:06 am
content: Just did a hearing check with Cool Edit Pro.I generated a 20second ramp from 1Hz -> 20,000Hz in 32bit/96khzWith some good headphones (that i know can do 20khz) i stop hearing the tone at 16 seconds which correlates to 16khzSo my range is up to 16khz (I'm 31).I'd be interesting to know how far others can hear + age.Here's the 96k/32bit file as well as a 48k/16bit version.(On both versions i stopped hearing it at 16 seconds, so for a hearing test it doesn't seem to make much difference)(14MB)http://psi.abcom.co.nz/1-20000Hz_96k_32bit.wavIf the tone isn't pure and you hear beating then your hardware/software isn't playing the file at its full sample rate. Try the 48/16 version below.(3.6MB)http://psi.abcom.co.nz/1-20000Hz_48k_16bit.wav

author: DrGeoff
author_rank: Frequent Contributor
timestamp: May 19, 2012, 02:33:18 am
content: Took a break from mixing and put these together for your personal amusement.A small except (30 seconds) from a classical recording made at 24/44k a few months ago.All files are here:http://www.axtsystems.com/dl2012/mp3tests/1. hov_arek_24_44.wav : 24 bit 44.1k file2. hov_arek_16_44.wav : 16 bit 44.1k file, dithered (as it would be for CD release)3. hov_arek_160k.mp3 : a 160k mp3 render4. hov_arek_256k.mp3 : a 256k mp3 render5. hov_arek_24_16_diff.wav : 16 bit undithered file of the difference between the (1) and (2).6. hov_arek_16_256mp3_diff.wav : 16 bit undithered file of the difference between the 256 mp3 and the 16 bit CD quality render. It's essentially a file of mp3 artefacts.7. hov_arek_160mp3_256mp3_diff.wav : The difference between the 160k mp3 render and the 256k mp3 render.Files were all rendered using Steinberg Cubase 6.5.

author: EEVblog
author_rank: Unknown
timestamp: May 19, 2012, 04:19:19 am
content: Quote from: Psi on May 19, 2012, 12:02:06 amI can just hear 17kHz with my Samson SR850 headphones.Dave.

author: Lightages
author_rank: Unknown
timestamp: May 19, 2012, 05:23:02 am
content: I have not tested myself for a long time. When I was much younger I could hear past 23k. "Ultrasonic" alarm systems drove me absolutely crazy in some buildings. I would be happy if I could hear to 16k now.

author: SeanB
author_rank: Super Contributor
timestamp: May 19, 2012, 05:48:49 am
content: You get Yoof Repellors that play loud ultrasonic noise, and they are said to work.I do have a dip at 1kHz, years of jet noise. The high end stops at around 15kHz, or that could be the speakers.

author: JuKu
author_rank: Frequent Contributor
timestamp: May 19, 2012, 07:17:57 am
content: Quote from: jahonen on May 18, 2012, 01:55:24 pmUmm... Yes, it does include the time domain, too. The key point is that the signal needs to be band-limited. If so, the theoremproves, that the reconstruction, if done right*, will beperfect, in time and in frequency domains. On band-limited transients, when the frequency goes up, the lenght goes up, too. There is no such thing as a band-limited burst at f/2.*: But no chip on the market can do this. It makes very good sense to use higher sampling rates or even upsample the signal before conversion, so that the DA artifacts close to f/2 move to unaudible range.QuoteOn this, we 100% agree! I'm an engineer, I believe that the quality of signal reproduction can be fully defined by about 30 measurable parameters - of these, maybe ten are known. I also believe that many traditional wisdoms about audibility are wrong or at least, not universally applicable. When the quality of the reproduction goes very high and starts to approach reality, we are in very different domain. When the question isn't anymore about how good the reproduction is but does this sound a reproduction at all or is this real, incredibly small differences seem to matter. The difference in 0.05% and 0.001% THD+N should not be audible - and yet, in a right setup, it is obvious. And so the endless discussions about the merits of different op- amps and DA chips... Maybe, if we would know what really makes the difference between the magical, hair raising "this is real" experience and just extremely good reproduction, we could also measure it.-Juha

author: ejeffrey
author_rank: Super Contributor
timestamp: May 19, 2012, 11:08:37 am
content: Quote from: JuKu on May 19, 2012, 07:17:57 amYes, this is the most misunderstood part of the nyquist-shannon theroem. The digitized representation provably contains all the information, but you have to know what to do with it. In particular, everyone focuses on the DAC, but the reconstruction filter is actually more important. The closer you push your signal frequency to the nyquist frequency, the better reconstruction filter you need.QuoteEvery chip on the market upsamples before conversion, that is fundamental to how sigma-delta converters work. Even so, there is little reason to store and transmit audio at sample rates above 48 kHz. Oversampling to prevent aliasing belongs at or immediately before the DAC.In the interest of fairness, there is one caveat to the whole thing: there is no such thing as a band limited signal. A truly band limited channel cannot be used to transmit information. Thus if you want to get technical, what you have to show is that the spectral density falls off outside the band fast enough that the total signal outside the band is an insignificant fraction of the power. Then you can show that the aliases of the leaked spectral content does not degrade the SNR by an unacceptable amount.

author: StubbornGreek
author_rank: Frequent Contributor
timestamp: May 23, 2012, 08:17:12 pm
content: Its really a joke what marketing has done to the audio world (and I love my audio gear, including many headphones). Most has been covered already in this thread but the statement below needs addressing.Quote from: olsenn on May 18, 2012, 12:50:23 pmThe reason you can hear a difference when changing the sampling rate inside your OS is that you most likely are changing volume from within Windows. Software volume control works by 'stealing bits' from the digital signal before converting it to analogue. If you were to set you windows volume panel to 100% and use an external device (i.e. a headphone amp or anything of quality with a volume pot) to change the listening level, you would be hard pressed to hear a difference.As far as ASIO, absolutely good stuff. I use WASAPI as I've had more success on various systems when ASIO was just being a plain old PITA. I assume you're using foobar?