32 bit DAC??? Is that even possible? - Page 1
Forum section: electronics / projects

author: Circlotron
author_rank: Super Contributor
timestamp: April 20, 2021, 01:20:18 am
content: I thought at after 21 or 22 bits anything smaller was just noise. 32 bits sounds like fairy tales. Or has analog technology advanced to that point while I wasn't looking?https://fscdn.rohm.com/en/products/databook/datasheet/ic/audio_video/audio_processor/bd34301ekv-e.pdf

author: oPossum
author_rank: Super Contributor
timestamp: April 20, 2021, 01:35:15 am
content: 130 dB is about 22 ENOB.32 bits probably refers to the DSP that is in front of the actual DAC.

author: JohnnyMalaria
author_rank: Super Contributor
timestamp: April 20, 2021, 01:38:44 am
content: I suspect it just means it can do some of the on-board DSP on 32-bit data.The SNR is 130dB - equivalent to the LSB of approx. 21 or 22 bits.

author: oPossum
author_rank: Super Contributor
timestamp: April 20, 2021, 01:41:03 am
content: "Supports 16-bit to 32-bit data formats"So you can feed it 16, 24, or 32 bit audio. The output will be essentially 22 bits due to SNR.

author: ledtester
author_rank: Super Contributor
timestamp: April 20, 2021, 01:45:48 am
content: According to this article:https://www.androidauthority.com/why-you-dont-want-that-32-bit-dac-667621/QuoteIt also mentions that the LG V10, a phone released in 2015, has a 32-bit DAC.For more perspective also check out the comments made by users at the end of the article.

author: Ed.Kloonk
author_rank: Super Contributor
timestamp: April 20, 2021, 01:54:27 am
content: Quote from: Circlotron on April 20, 2021, 01:20:18 amCoupling a DAC >24-bits to your ears is pointless. And borders on retarded if the user intends to use it at high volume with ear buds.

author: Algoma
author_rank: Frequent Contributor
timestamp: April 20, 2021, 03:18:35 am
content: Give the marking team time and they'll push the engineers to create even higher numbers.If the goal wasn't audio output.. I'm curious if there could be other possible applications.

author: NiHaoMike
author_rank: Super Contributor
timestamp: April 20, 2021, 05:09:40 am
content: Quote from: blueskull on April 20, 2021, 03:33:11 amIsn't it that at least some modern (lossy) audio codecs are floating point and therefore do not really have a fixed bit depth?

author: Circlotron
author_rank: Super Contributor
timestamp: April 20, 2021, 05:34:57 am
content: Say you're playing a 16-bit audio signal on a 24-bit DAC. From one 16-bit sample to the next, say there is a 1 LSB (relative to 16-bit) step change. Could you use your 24-bit DAC to output samples at 256x the 16-bit clock rate and linearly (or otherwise) interpolate between 16-bit samples to ease the post DAC filtering requirements? Is that what oversampling is?

author: Ed.Kloonk
author_rank: Super Contributor
timestamp: April 20, 2021, 06:11:14 am
content: Quote from: Circlotron on April 20, 2021, 05:34:57 amI have a sound card that does just that. The Motorola DSP chip does the voodoo by taking what ever the computer sends it and over-samples the life out of it. 20 year old sound card. The company has gone out of business so I can't quickly find more tech info. Found one for sale tho.https://www.ebay.com.au/itm/Echo-Gina-PCI-Audio-Interface-Sound-Card-24-bit-96khz-Echo-Audio-/254387533631?mkevt=1&mkcid=1&mkrid=705-53470-19255-0&campid=5337590777&customid=&toolid=10001A  newer card, but I believe it's similar.https://www.extremetech.com/computing/55104-review-echo-audio-indigo-io/2

author: Kleinstein
author_rank: Super Contributor
timestamp: April 20, 2021, 06:50:28 am
content: Oversampling is doing the interpolation, but no need to use additional resolution for this. It is enough to have the same, in the extremes even slightly reduced resolution.AFAIK oversampling was used from the very beginning of the CD in the first phillips player and likely also in the Sony version.Modern Audio DACs are essentially all a kind of sigma delta DAC and this kind of does faster ouput and filtering anyway. So the modulation and oversampling for the DAC may be kind of combined. Even if the DAC externally looks like 24 Bits, it may internally do 8 Bits with massive oversampling and some filtering.

author: magic
author_rank: Super Contributor
timestamp: April 20, 2021, 07:11:04 am
content: Echo advertised oversampling rates like 64x or 128x which suggests they meant the internal oversampling of the delta-sigma DAC, which as Kleinstein remarked, is common for all audio DAC ICs produced in this century.The DSP enabled arbitrary hardware mixing of 8 stereo / 16 mono streams into the two streams fed to the stereo DAC. Perhaps also upsampling them to a common format, but I'm not sure of that. SoundBlasters of the era had similar capability using Creative's proprietary chip, and these were known to upsample the streams to 48kHz causing no end of (perhaps justified) butthurt. SBs also could run a host of effects on their DSP, but CPUs are fast enough that such offload became a nonissue.Decade earlier, there were standard R2R audio DACs which oversampled to 88 or 176kHz to ease up the demands on hardware reconstruction filters and move part of the filtering to DSP. Recently one vendor revived the concept using Analog Devices R2R industrial DACs and I'm pretty sure you can still buy megabuck gear with discrete R2R.32 bit is of course marketing wank and parting audiophools from their money.

author: floobydust
author_rank: Super Contributor
timestamp: April 20, 2021, 08:04:37 am
content: It's just marketing wank, more bits = better. BD34301EKV $94 and the EVAL board is $818 at Digi-Key. But finding 32-bit recordings?I found it extremely hard to design mixed signal 24-bit DAC that actually meets it's performance specs.I tried ESS parts and they were terrible, they have a cult following and hype, and high price etc. but a simple listening test is hilarious. A cymbal sounding like syran wrap lol interpolate that.DAC IC's have auto-mute which mutes the output with a zero stream, it's blatant cheating to get "130dB S/N" etc. and I laugh at the hiss and noise jump when that switches off.

author: Berni
author_rank: Super Contributor
timestamp: April 20, 2021, 08:30:36 am
content: Yeah you can just crank up the number of bits on a sigma delta converter as far as you want, but weather those bits are actually any use is another question.In practice 16bit is all you need when you have analog volume control after it, but when you do digital volume control then the 24bit is a good idea. But for DSP it makes sense to use 32bit since that tends to be the native number of bits for a lot of processors. The extra bits come in useful when audio is mixed or filtered, preventing clipping and overflows from being problematic.As far as 32bit float audio, it is never used on a hardware level because floating point numbers are a pain to work with. It is also rarely used in DSP because most DSPs are significantly faster with fixed point math. All the common audio formats also use fixed point. Pretty much the only common use for floating point audio is in audio editing and music creation software. They enjoy not having to worry about clipping at all along the pipeline while x86 PCs are practically just as fast with floating point math as they are with fixed point math, so there is no performance penalty to doing it.

author: Gyro
author_rank: Super Contributor
timestamp: April 20, 2021, 08:53:06 am
content: Quote from: Kleinstein on April 20, 2021, 06:50:28 amActually, that was the first divergence. Philips felt that they could only achieve 14 bits with decent linearity at launch, they introduced 4x oversampling as a form of compensation for this. Sony launched with 16 bits and no oversampling. The Philips players were generally regarded to sound much better as the oversampling pushed the required post-DAC analogue filtering well above the audio frequency range. Sony had to use a brick-wall analogue filter which caused lots of audible phase and amplitude errors in the audio passband.By the next generation, Philips had perfected decent 16 bit DACs but retained the 4x oversampling for the more relaxed filtering requirement.

author: Ed.Kloonk
author_rank: Super Contributor
timestamp: April 20, 2021, 08:59:01 am
content: In the early days of sound blaster for the PC, there was DOS project that aimed to convert 8-bit to 16-bit.In the period before the 16-bit soundblaster, apparently people had recorded stuff in 8-bit.Whilst it couldn't do much about the sample source being recorded in 22050Hz, the 8-bit data was 'dithered' to 16-bit. What that means is they rounded the sharp, jagged edges using alot of floating point. You had to have a 486DX if memory serves.I remember giving it a go once. I recorded a song in 8-bit mode and ran it through the software. I think it took about 25 minutes for a 3 minute song. It sounded not as harsh as the 8-bit, but it wasn't great either.When expanding out the bit depth, I have to wonder what transform is being used. I hate to use this analogy but it's like when they tried to colorize b/w movies. Sometimes, you just shouldn't.

author: Ed.Kloonk
author_rank: Super Contributor
timestamp: April 20, 2021, 09:05:09 am
content: Quote from: Gyro on April 20, 2021, 08:53:06 amA bit later, before the 1-bit DACs, a few cheap designs came out with a DAC running at 88.2kHz. Some switching logic and a 4066 on the output side. Because two DACs was too expensive. I've got an old schematic here somewhere. I'll have to upload it sometime. You look at the output section and go

author: SeanB
author_rank: Super Contributor
timestamp: April 20, 2021, 11:36:43 am
content: Quote from: Ed.Kloonk on April 20, 2021, 09:05:09 amMany of the cheaper ones still do this for the DAC, using an analogue switch and a sample and hold to generate the stereo output, as in general the DAC values are close enough at high frequency to not cause much slew, and the crosstalk is not too bad as a result. You can still hear it for audio coming out of one channel only as a faint high frequency signal in the other quiet channel.

author: SiliconWizard
author_rank: Super Contributor
timestamp: April 20, 2021, 03:08:14 pm
content: Couple points:* If you're dealing with 32-bit samples, a true 32-bit DAC is likely going to yield a better SNR than a 24-bit fed with samples down-quantized from 32-bit to 24-bit;* Sigma-delta DACs are really all digital. And to achieve a given SNR, it can be cheaper to design a 32-bit SD DAC than a 24-bit one, for instance.Of course, if the question is: can you get close to the ideal SNR of ~192 dB, the answer is no, but that would be missing the two above points - they are not made for this.Now sure there can be some marketing - which allows to sell those 32-bit DACs at a higher price point than 24-bit DACs, whereas as I said above, they can be actually easier to design for a given target SNR, and with possibly simpler digital filters needed, even possibly take actually a bit less silicon area...

author: David Hess
author_rank: Super Contributor
timestamp: April 20, 2021, 04:54:11 pm
content: Quote from: Kleinstein on April 20, 2021, 06:50:28 amAnalog Devices published an excellent document describing the three basic architectures, single-bit, multi-bit, and MASH (multistage noise shaping):https://www.analog.com/media/en/training-seminars/tutorials/MT-023.pdfBriefly, single-bit delta-sigma converters have higher linearity but require extra modulator stages to suppress idle tones and this makes them susceptible to instability from overload. Multi-bit delta-sigma converters can have fewer modulator stages, but DAC linearity limits linearity.Quote from: SiliconWizard on April 20, 2021, 03:08:14 pmDo you have any examples where that is the case and it is not just marketing? TI's (Burr-Brown) highest SNR audio ADCs and DACs are 24 bits and have a higher claimed SNR than the "32-bit" converters from AKM.Some instrumentation converters claim slightly more than 24 bits of noise free resolution.

author: SiliconWizard
author_rank: Super Contributor
timestamp: April 20, 2021, 06:29:35 pm
content: Quote from: David Hess on April 20, 2021, 04:54:11 pmTo make sure my point is fully understood:1. I am assuming a 32-bit DAC and a 24-bit DAC with the same SNR.2. I am explicitly talking about dealing with 32-bit samples, which, if it was not clear enough, means samples that are natively generated as 32-bit numbers.3. If you have 32-bit samples, and feeds a 24-bit DAC, you have to down-quantize. Merely truncating creates truncation distortion which is rather nasty. The usual way of mitigating this is to *dither*, which basically is adding some random noise to the signal. It's a trade-off. Usually less annoying than pure distortion, but added noise nonetheless. Then end-result will be noisier than the 32-bit samples directly fed to a 32-bit DAC with the same SNR.4. Of course, if you're dealing with 24-bit samples, using a 32-bit DAC wouldn't make sense. Actually, you'd get worse results too since you'd have to up-quantize here, which would inevitably make the signal noisier, even if just a little.5. All in all, it's best to use a DAC with the same resolution as the samples you're feeding it with. Any change of quantization WILL add noise.6. Some may question the benefits of 32-bit samples to begin with. And if you're considering recorded music played on an audio player, you'd be right: there is no native 32-bit material available. But there still are use cases for this. For instance, digital mixing consoles, or any device that does DSP with a native width larger than 24 bits.

author: ejeffrey
author_rank: Super Contributor
timestamp: April 20, 2021, 07:51:54 pm
content: Quote from: SiliconWizard on April 20, 2021, 06:29:35 pmThat is only true for a very specific meaning of "natively generated as 32 bit numbers" Most audio you will find in any format has been subjected to some combination of signal processing and then truncated to a target output resolution. Taking 32 bit audio produced this way, then truncating them to 24 bits is likely going to produce equivalent results to if the original producer had simply truncated the original to 24 bits.QuoteThat depends on your DAC. If the 32 bit DAC doesn't have any better linearity (especially DNL) than the 24 bit DAC that is going to set your distortion floor and dithering is not really going to fix it. And if the SNR of the 24 bit DAC is < 24 bits (say 20 bits), then you can easily add enough dither to decorrelate the quantization noise without materially affecting the overall SNR. Technically it still adds noise, but the added noise can be inconsequential, and in any cases it is exactly the same noise you would have to add to generate native 24 bit audio in the first place.QuoteAn ideal 32 bit DAC with the lowest 8 bits zeroed is an ideal 24 bit DAC. If you have a 32-bit DAC and 24 bit audio dithering and up-quantizing might improve the distortion products for very particular waveforms at the expense of noise, but not re-quantizing is just as good as the 24 bit DAC would be anyway.QuoteBut completely impractical and not important at all in practice. Sample depth conversions are easy, common, and have insignificant impact on SNR when done properly. At the same time it is very useful to have intermediate formats that are 24-bit, 32-bit, or floating point, but since not all sources are the same so it is mandatory for mixers to combine them. For a final product 16 bits is about all the dynamic range you can use anyway, so carrying around an extra 8 or 16 bits is just a waste, but again it can be useful to re-quantize to 24 bit or higher for signal processing at the playback end.

author: NiHaoMike
author_rank: Super Contributor
timestamp: April 21, 2021, 12:29:43 am
content: Quote from: blueskull on April 20, 2021, 07:47:05 amWhat I'm getting at is that in practice, some codecs already "transport" the audio using more than 16 bits (after uncompressing) per sample. How many "integer bits" of precision those floating bit samples have can be a complex question, not to mention how much degradation the lossy compression really introduces.At least in theory, if the floating point samples do not exactly map to 16 bit integers, mapping them to 24 bit integers means less error.

author: bson
author_rank: Unknown
timestamp: April 21, 2021, 10:31:39 pm
content: Quote from: blueskull on April 20, 2021, 03:33:11 amIt's more that a good stereo pot like an ALPS costs $40-$50 (maybe less in 1ku volume), while attenuation in firmware costs 2kB of code space.

author: NiHaoMike
author_rank: Super Contributor
timestamp: April 22, 2021, 03:01:46 am
content: Many DACs also have PGAs built in, so just a little bit of code to write to the register(s) that controls the PGA.

author: Berni
author_rank: Super Contributor
timestamp: April 22, 2021, 05:27:40 am
content: Most audio DACs have no extra control bus to even have registers. Usually its just a few extra pins that can be set high or low to enable or disable some features. On the lower pin count ones you don't even get that. Its just I2S in and analog out the other side.At the same time you can buy a 24bit stereo audio DAC from name brands like AKM, Texas, Cirrus for $1.20 to $2.00 on digikey. So there is not really much room for saving money by going with less than 24bit. They have good specs and will not be the limiting factor once paired up with cheep opamps in the analog stages. Because lets face it if you bought a $1.20 DAC chip, then you are not going to pair it up with a pair of $2 opamps per channel. Whatever is producing the audio signal likely also has a hardware multiplier so volume control is literally a few CPU cycles to do. On the other hand a external digital pot for volume control is going to cost at least as much as that DAC while doing a poor job designing it might result in I2C bleeps and bloops leaking into the audio output.Its the same argument as 32bit ARM MCUs doing stupid simple tasks that your oldschool classical 8 banger could also easily do. Yes you don't need a 32bit ARM core to do the job, but at this point the cheep tiny ARMs now cost similar as name brand 8bit MCUs cost. So you might as well just go for 32bit and enjoy the luxuries such as a CPU core that was actually designed for running C code rather than hand coded assembler, actual interrupt vectors, proper debug functionality... etc.

author: magic
author_rank: Super Contributor
timestamp: April 22, 2021, 05:34:05 am
content: There surely are audio DACs with volume control over SPI or similar interface, though the examples I could name aren't cheapAnd I think they used digital filter for it, not PGA.

author: Berni
author_rank: Super Contributor
timestamp: April 22, 2021, 06:10:05 am
content: Quote from: blueskull on April 22, 2021, 05:34:58 amYeah the 32bit also provide a lot of efficiency in computation since we very often work with numbers that don't fit into 8 bit nicely.But my point is more for applications where performance doesn't matter at all. Like some thermostat with a 7 segment display and a few buttons, a electronic combination lock, an remote control receiver for a garage door... etc. Things that almost any crappy slow MCU could do even when running at 1/10 its rated clock speed. Yet these days it makes sense to use cheep little 32bit MCUs for these dumb tasks since they cost about as much as any other name brand MCU.Yes i know about dirt cheep Chinese MCUs that literally cost a couple of cents each and tend to be a 8bit MCU (pretty basic even at that). But unless you are doing something that is really high volume and cost sensitive then people from the west are going to stick to western name brand MCUs since they know how to use them, where to buy them and can read the documentation without a translator. These western 8bit MCUs are nowhere near as cheep.

author: Ed.Kloonk
author_rank: Super Contributor
timestamp: April 22, 2021, 07:15:44 am
content: If you ever wanna see some cool audio software engineering, check out the MAD mp3 decoder library. 24-bit output, integer-only code. No floats.Nearly 20 years old now.

author: Jan Audio
author_rank: Frequent Contributor
timestamp: April 26, 2021, 03:44:04 pm
content: Quote from: Berni on April 20, 2021, 08:30:36 amThings change.You now have 64 bit floating point in STM32 ( sold out )It should be faster doing floating point math since you have no shift.*i dont have STM32 due to package restrictions, i can see the advantage of 64 bit floating point in 1 instruction.

author: Jan Audio
author_rank: Frequent Contributor
timestamp: April 26, 2021, 03:48:37 pm
content: Suppose they make a 32 bit DAC on -15 to +15 volt, the s/n ratio is improved right ?Or maybe high voltage for the best quality.

author: Ed.Kloonk
author_rank: Super Contributor
timestamp: April 26, 2021, 04:03:52 pm
content: Quote from: Jan Audio on April 26, 2021, 03:48:37 pmThere was a use case theory ages ago, among us HQ audio snobs, where that concept could be used in active speaker modules. Sounded good, in theory. What we got instead were active speaker modules with sloppy old 16-bit DACs, class-d amps and really terrible power supplies.

author: bson
author_rank: Unknown
timestamp: April 26, 2021, 05:10:18 pm
content: Quote from: blueskull on April 22, 2021, 02:19:51 amTrue, but you also need some way of controlling the volume.

author: Bassman59
author_rank: Super Contributor
timestamp: April 26, 2021, 05:17:36 pm
content: Quote from: bson on April 26, 2021, 05:10:18 pmMeh, that's what knobs are for!

author: Humanoid
author_rank: Regular Contributor
timestamp: April 27, 2021, 09:30:26 pm
content: I work with audio a lot, so here's my 2¢ (or maybe 10).I use a pro audio interface (MOTU 1248) and don't need a dedicated ADC or DAC for my work. Most modern ADCs and DACs on interfaces handle 32-bit conversion, if they cost more than $20. 32-bit 192k is pretty standard specs even for mid range desktop interfaces now with 2 ins and outs. I personally see no benefit to using those higher sample rates as they can add in harmonics and other garbage above 20k during a mix which brings the entire level of the mix up. Some engineers swear the highs are clearer at 192 and 384, but I have no need for it. Tokyo Dawn Labs has an old plugin specifically made for filtering those ultrasonic frequencies out of a mix. 32-bit could be useful though if you need it.In the late 90s when I went from 16-bit to 24-bit, there was a big difference because the signal to noise ratio is so much better with 24. No longer did we need to record hot signals. These days I try and record levels around -18dB to -12dB so I have plenty of headroom when mixing. 24-bit to 32-bit the noise floor is usually the same. It's the massive headroom over digital 0 that differs. If you hear noise in your 24-bit recording at rational listening levels, it's probably interference, or you're recording too low and raising the gain too much, or your monitoring signal chain is noisy, or maybe you have grounding problems (typically hum though).I used to use 96k 24-bit back when many plug-ins didn't upsample because I could hear a difference in the output, but these days pretty much everything upsamples to some ridiculous degree to avoid aliasing and other artifacts and then downsamples on the fly, unless you are using some old plug-ins that aren't being updated. I can hear no difference in output anymore so I stick to 48k sample rate as a standard unless a client demands a higher one. It keeps files smaller, avoids the garbage above 20k, is the standard for video and cinema, and drastically frees up your CPU to use way more plugins. I can do mixes on my modern 10-core CPU using dozens and dozens of tracks with tons of plugins that would need accelerator hardware in the past. My heaviest mixes usually don't pass 50% CPU usage. God bless multi-core.As for bit rate (or depth if you prefer), 32-bit float I find to be beneficial when recording because of the extra headroom. You can record values past digital 0 to avoid clipping and then bring down the gain when mixing (or surgically with bite volume or in a 3rd party app like RX or Audacity). I think the headroom surpasses the decibel levels of sounds we can even make on earth. I find it useful when recording voices and sound effects, because you can get spikes if someone goes closer to the mic and ruin a take. If you like to record hot, 32-bit is also your friend for the same reason, you can always mix down to 24 once things are tamed and most DAWs these days let you mix different bit rates in the same sequence if you really want to.In pro studios, 24-bit is the standard because they won't record hot and most engineers won't mix to the ceiling. Some swear by 88k because it's double 44k used for CDs, and supposedly there will be no rounding errors when downsampling. You may find 32-bit being used in the final mixing phase at some studios so if something goes over the mastering engineer can tame it and not be dealing with a clipped square wave. It all depends on the workflow. These days people send stems to mastering engineers sometimes so nothing will be clipping (in theory). If you have the hard drive space and want to avoid having clipping problems, use 32-bit float. If not, 24 is fine. There's even free plugins you can drop on your tracks and buses to check if there are inter-sample peaks the meters don't register.I personally do all my recording and audio mixes at 48k 32-bit float for A/V projects to avoid clipping and because Adobe After Effects does not natively export 24-bit. There is no audible difference between 24 and 32, it's merely so I can avoid clipping and don't have to change formats when doing intermediate renders and when working with different clips, or different audio sources and music. I keep everything at 32-bit 48k throughout the entire production phase so regardless of what software I have to use, no problems arise. If I have to convert sample rate or bit depth, I do it at the start using iZotope RX7 because it has one of the best software converters. Some people prefer Weiss Saracon or Saracon and RX (or a few others), though RX is price tiered and has more tools.If you work on A/V for the web, YouTube even accepts 48k 32-bit float linear PCM embedded in video (even if the video is compressed as H264), so you can let them do the final audio compression after upload for the best audio quality and avoid double compression.So basically, if you have a good "pro-sumer" or professional level interface, the bit depth/rate you choose will really depend on what you're doing and most likely it will matter most in the software when recording or doing a final mix. All the well-known manufacturers use great preamps and their ADC and DAC is top notch. Even modern soundcards on motherboards handle 32-bit audio playback these days. Use 32-bit for practical reasons, not because of a sales pitch.

author: JohnnyMalaria
author_rank: Super Contributor
timestamp: April 27, 2021, 10:01:23 pm
content: Quote from: Bassman59 on April 26, 2021, 05:17:36 pmAs long as they go to 11 and you can make everything louder than everything else.