Monday, January 19, 2015
Saturday, January 17, 2015
Q. Why is 88.2kHz the best sample rate for recording?
I have read that the optimum sample rate to record at is 88.2kHz. The reasons include simple integer-ratio sample-rate conversion, avoiding the phase shifts and ringing of anti-alias filtering at 20kHz, and less data to move about compared to 176.4kHz. Is there any truth in these assertions?
The Benchmark DAC1 is one of many converters that achieves excellent audio quality without using synchronous (integer-ratio) sample-rate conversion.The Benchmark DAC1 is one of many converters that achieves excellent audio quality without using synchronous (integer-ratio) sample-rate conversion.
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: These claims are partially true! Let's start with the simple integer-ratio sample-rate conversion issue. Simple ratios were important in the days of 'synchronous' sample-rate conversion, but that technology went the way of the dodo a long time ago. It was a relatively simplistic approach that did work best with simple ratios between the source and destination sample rates. However, it had limited resolution in terms of the practical word-lengths achievable, with the noise floor rarely being better than the equivalent of 18 or 19 bits. Moreover, the approach is hugely wasteful of computational effort, calculating millions of intermediate sample values no one has any interest in.
Modern 'asynchronous' sample-rate conversion is far more sophisticated and works by analysing the source and destination sample rates and working out only the required sample values with huge precision. This achieves a technical performance that is significantly in excess of any real-world converter and very close to the 24-bit theoretical level — and that's achieved with any ratio of input-to-output sample rate. There is no measurable difference in performance between using simple integer ratios or complex ones.
In fact, it's interesting to note that some of the best performing D-A and A-D converters from the likes of Benchmark, Crookwood, Cranesong, Drawmer, and others, all use non-integer sample-rate conversion as an inherent part of their jitter-isolation process. For example, D-A converters using this approach typically up-sample the incoming digital audio to something like 210kHz, or the rate at which the physical D-A converter chip achieves its best performance figures: no simple-ratio conversions going on there, yet class-leading performance specifications!
Moving on to the second point, there is a (weak) argument for sampling original material at a rate higher than 44.1kHz in some cases. The reason is that a lot of A-D systems are designed with relatively imprecise anti-alias filtering, which typically only manages 6dB attenuation at half the sampling rate, instead of the 100dB or more that is theoretically required. It's done in that way because it makes the A-D converter's digital filtering a lot easier to design, and in most cases it makes little difference. However, aliasing could result if the input audio contains a lot of strong extreme HF harmonics. Cymbals, orchestral strings and brass can all generate enough HF energy, if close-miked, to cause this problem with some converters. In such cases, switching to a higher sample rate might sound noticeably sweeter.
The issue then, of course, is how to down-convert to 44.1kHz for release without suffering the same problem in the sample-rate converter (SRC) anti-alias filtering. Clearly, a properly designed digital filter is required in the SRC, and while some software SRCs do this properly, some don't. The Infinite Wave SRC comparison web site reveals the scary truth! (See http://src.infinitewave.ca).
The 176kHz (or 192kHz) quad-sample-rate idea is really just about being able to say 'mine's bigger than yours'. There's a very good white paper about sampling theory on Lavry's web site (www.lavryengineering.com) where Dan Lavry points out that the higher the sample rate, the greater the proportion of error in sampling time and the lower the actual audio resolution. Lavry argues (very sensibly, in my opinion) that the optimal sample rate would actually be 60kHz. In the real world, 96kHz can be useful, for the reasons mentioned above, but the quad rates are a folly and Lavry refuses to support them!
The Benchmark DAC1 is one of many converters that achieves excellent audio quality without using synchronous (integer-ratio) sample-rate conversion.The Benchmark DAC1 is one of many converters that achieves excellent audio quality without using synchronous (integer-ratio) sample-rate conversion.
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: These claims are partially true! Let's start with the simple integer-ratio sample-rate conversion issue. Simple ratios were important in the days of 'synchronous' sample-rate conversion, but that technology went the way of the dodo a long time ago. It was a relatively simplistic approach that did work best with simple ratios between the source and destination sample rates. However, it had limited resolution in terms of the practical word-lengths achievable, with the noise floor rarely being better than the equivalent of 18 or 19 bits. Moreover, the approach is hugely wasteful of computational effort, calculating millions of intermediate sample values no one has any interest in.
Modern 'asynchronous' sample-rate conversion is far more sophisticated and works by analysing the source and destination sample rates and working out only the required sample values with huge precision. This achieves a technical performance that is significantly in excess of any real-world converter and very close to the 24-bit theoretical level — and that's achieved with any ratio of input-to-output sample rate. There is no measurable difference in performance between using simple integer ratios or complex ones.
In fact, it's interesting to note that some of the best performing D-A and A-D converters from the likes of Benchmark, Crookwood, Cranesong, Drawmer, and others, all use non-integer sample-rate conversion as an inherent part of their jitter-isolation process. For example, D-A converters using this approach typically up-sample the incoming digital audio to something like 210kHz, or the rate at which the physical D-A converter chip achieves its best performance figures: no simple-ratio conversions going on there, yet class-leading performance specifications!
Moving on to the second point, there is a (weak) argument for sampling original material at a rate higher than 44.1kHz in some cases. The reason is that a lot of A-D systems are designed with relatively imprecise anti-alias filtering, which typically only manages 6dB attenuation at half the sampling rate, instead of the 100dB or more that is theoretically required. It's done in that way because it makes the A-D converter's digital filtering a lot easier to design, and in most cases it makes little difference. However, aliasing could result if the input audio contains a lot of strong extreme HF harmonics. Cymbals, orchestral strings and brass can all generate enough HF energy, if close-miked, to cause this problem with some converters. In such cases, switching to a higher sample rate might sound noticeably sweeter.
The issue then, of course, is how to down-convert to 44.1kHz for release without suffering the same problem in the sample-rate converter (SRC) anti-alias filtering. Clearly, a properly designed digital filter is required in the SRC, and while some software SRCs do this properly, some don't. The Infinite Wave SRC comparison web site reveals the scary truth! (See http://src.infinitewave.ca).
The 176kHz (or 192kHz) quad-sample-rate idea is really just about being able to say 'mine's bigger than yours'. There's a very good white paper about sampling theory on Lavry's web site (www.lavryengineering.com) where Dan Lavry points out that the higher the sample rate, the greater the proportion of error in sampling time and the lower the actual audio resolution. Lavry argues (very sensibly, in my opinion) that the optimal sample rate would actually be 60kHz. In the real world, 96kHz can be useful, for the reasons mentioned above, but the quad rates are a folly and Lavry refuses to support them!
Friday, January 16, 2015
Q. Can I pay someone to run my WAV files through a Studer?
I keep reading about the desirable tape effect on a mixdown, and keep hearing people talk about these Studer tape machines. Is it possible to send your WAV song to someone who owns one of these, so they can run it through and back to WAV?
While the effects of running a track through an analogue tape machine at mixdown might be desirable, you may be better off using a plug-in if you have a very specific effect in mind. The possible parameters involved in using an old tape machine, like this Studer model, for example, can create large variations in the sound achieved.While the effects of running a track through an analogue tape machine at mixdown might be desirable, you may be better off using a plug-in if you have a very specific effect in mind. The possible parameters involved in using an old tape machine, like this Studer model, for example, can create large variations in the sound achieved.
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: You could, but this isn't a one-size-fits-all process. The crushing and bending effect of tape varies dramatically with the type of tape, the bias level, the record EQ, the overall signal level, the tape speed, the tape width, and the electronics in the tape machine itself. The art and skill is in choosing and using just the right parameters to achieve the effect you're looking for. Plug-ins are beginning to get there now, however; I'm really quite impressed with the UAD offerings, for example.
SOS Reviews Editor Matt Houghton adds: Analogue tape can do nice things on a bus, but it's no magic bullet: you need to tweak settings to achieve the desired effect. For example, running tape at 15ips will add a low-frequency 'head bump' that can be really flattering on a rock drum bus, but could screw things up elsewhere. Similarly, it's not a case of running a source louder to get more effect: the tape effect changes in nature with level. While people will happily do what you ask, I prefer using plug-ins or outboard tape simulators (eg. Anamod ATS1, RND Portico 5042), as these offer far more control and I can use them as I might a bus compressor, for example.
While the effects of running a track through an analogue tape machine at mixdown might be desirable, you may be better off using a plug-in if you have a very specific effect in mind. The possible parameters involved in using an old tape machine, like this Studer model, for example, can create large variations in the sound achieved.While the effects of running a track through an analogue tape machine at mixdown might be desirable, you may be better off using a plug-in if you have a very specific effect in mind. The possible parameters involved in using an old tape machine, like this Studer model, for example, can create large variations in the sound achieved.
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: You could, but this isn't a one-size-fits-all process. The crushing and bending effect of tape varies dramatically with the type of tape, the bias level, the record EQ, the overall signal level, the tape speed, the tape width, and the electronics in the tape machine itself. The art and skill is in choosing and using just the right parameters to achieve the effect you're looking for. Plug-ins are beginning to get there now, however; I'm really quite impressed with the UAD offerings, for example.
SOS Reviews Editor Matt Houghton adds: Analogue tape can do nice things on a bus, but it's no magic bullet: you need to tweak settings to achieve the desired effect. For example, running tape at 15ips will add a low-frequency 'head bump' that can be really flattering on a rock drum bus, but could screw things up elsewhere. Similarly, it's not a case of running a source louder to get more effect: the tape effect changes in nature with level. While people will happily do what you ask, I prefer using plug-ins or outboard tape simulators (eg. Anamod ATS1, RND Portico 5042), as these offer far more control and I can use them as I might a bus compressor, for example.
Q. Can I comp together tracks from different takes?
've not used the group editing feature of Cubase 6.5, having only recently come over to this version from Cubase SX3, but Matt Houghton mentions in the June 2012 issue of SOS (/sos/jun12/articles/cubase-6-5.htm) that the combination of group editing and comping can help a lot with editing multi-take, multi-part background vocals.
However, in a YouTube tutorial video I came across (Cubase 6 Tutorial Level 1 — Group Editing, found at www.youtube.com/watch?v=7UDdKgxfMeA) it is mentioned that group editing is "not meant to combine different tracks from different sessions”. But isn't comping of background vocals by definition working with tracks from different sessions? How do I reconcile that statement from the video with what Mr Houghton writes?
I'm clearly having trouble visualising how to use group editing with the comp tool for background vocals. Could you give me an idea about how I might approach this?
Via SOS web site
SOS Reviews Editor Matt Houghton replies: I hope you got something from the review. As I said towards the end, it really feels, to me, as though the latest incarnation of Cubase is reaching maturity. There are only a handful of gripes and wish-list features for me now, but nothing worse than I'm aware of in any other DAW.
When I talk about 'multi-part' vocals, I mean, for example, four-part harmony backing vocals with each singer singing a different 'part' into their own mic. You record each part to a separate track simultaneously from one performance, and record several passes to obtain the best 'take'. You can then comp the best overall backing vocal take using Group Editing and the Comp tool. It all worked fine for me during my tests. Unless you want to chop up your tracks and 'fake' a decent take, the Comp tool in Cubase 6.5's group editing feature should be all you need to create a great overall take from different parts. We're talking here about multi-part backing vocals, for example, where multiple performers are being recorded at the same time onto different tracks.Unless you want to chop up your tracks and 'fake' a decent take, the Comp tool in Cubase 6.5's group editing feature should be all you need to create a great overall take from different parts. We're talking here about multi-part backing vocals, for example, where multiple performers are being recorded at the same time onto different tracks.
The point made on the video is different. It's suggesting that each of the equivalent takes on each track must be from the same performance. It's easy to find yourself getting this wrong if you inadvertently click and drag a part from one lane to another (for example, imagine that take one of an inside kick mic is on lane one of one track, but the outside kick mic from the same performance is on lane two of another track).
Actually, you can work with group editing from entirely different performances if you really want to fake things up by dragging and dropping parts on to the relevant lanes. But you have to put a bit of work in to make sure everything's on the right lane in each track, and that the parts are of the same length and start and end at the same points (easy enough with the Split Loop command, or with Scissors and Snap To Grid).
Thursday, January 15, 2015
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