Welcome to No Limit Sound Productions
Tuesday, January 27, 2015
Q. How can I warm up my recording without using EQ?
Sound Advice : Recording
I've put a lot of effort into creating and editing a recording of solo mandolin — played quite slowly — and although I like the final result a lot, on consideration the tone is too trebly and cold, almost like a photograph with too sharp a resolution. A friend mentioned he thought I could perhaps 'warm it up' using compression, perhaps of a type designed for vocals. Can you give me some guidance on how best I might do this? Of course, I realise I can use EQ, but would specifically be interested in any thoughts on how compression/limiting could be used on an existing take to get a warmer result. I've used Logic and the recording is clear, undistorted, and free from ambient sound.
Simon Evans via email
SOS contributor Mike Senior replies: There are ways to warm up a mandolin sound subjectively using compression, although none of them are likely to make as big an impact as EQ. Fast compression may be able to take some of the edge off a mandolin's apparent tone, for instance, assuming the processing can duck the picking transients independently of the note-sustain elements. There are two main challenges in setting that up. Firstly you need to have a compressor that will react sufficiently quickly to the front edges of the pick transients, so something with a fast attack time makes sense. Not all of Logic's built-in compressor models are well-suited to this application, so be sure to compare them when configuring this effect; instinctively I'd head for the Class A or FET models, but it's always going to be a bit 'suck it and see'. The second difficulty will be getting the compressor not to interfere with the rest of the sound. The release-time setting will be crucial here: it needs to be fast enough to avoid pumping artifacts, but not so fast that it starts distorting anything in conjunction with the attack setting. Automating this compressor's threshold level may be necessary if there are lots of dynamic changes in the track, for similar reasons. Applying some high-pass filtering to the compressor's side-chain (open the Logic Compressor plug-in's advanced settings to access side-chain EQ, and select the 'HP' mode) may help too, because the picking transients will be richer in HF energy than the mandolin's basic tone.The Advanced Settings panel in Logic's built-in Compressor plug-in contains side-chain equalisation facilities that can be very useful if you're trying to sensitise (or desensitise!) the compressor to a mandolin's picking transients.The Advanced Settings panel in Logic's built-in Compressor plug-in contains side-chain equalisation facilities that can be very useful if you're trying to sensitise (or desensitise!) the compressor to a mandolin's picking transients.
Another way to apparently warm up a mandolin is to take the opposite approach: emphasise its sustain character directly while leaving the pick spikes alone. In a normal insert-processing scheme, I'd use a fast-release, low-threshold, low-ratio (1.2:1 to 1.5:1) setting to squish the overall dynamic range. Beyond deciding on the amount of gain reduction, my biggest concern here would be choosing an attack time that avoided any unwanted loss of picking definition. In this case, shelving a bit of the high end out of the compression side-chain might make a certain amount of sense if you can't get the extra sustain you want without an unacceptable impact on the picking transients.
Alternatively, you might consider switching over to a parallel processing setup, whereby you feed a compressor as a send effect, and then set it to more aggressively smooth out all the transients. The resulting 'sustain-only' signal can then be added to the unprocessed signal to taste, as long as you've got your plug-in delay compensation active to prevent processing delays from causing destructive phase-cancellation. Using an analogue-modelled compressor in this role might also play further into your hands here, as analogue compressors do sometimes dull the high end of the signal significantly if they're driven reasonably hard, giving you, in effect, a kind of free EQ.
I've put a lot of effort into creating and editing a recording of solo mandolin — played quite slowly — and although I like the final result a lot, on consideration the tone is too trebly and cold, almost like a photograph with too sharp a resolution. A friend mentioned he thought I could perhaps 'warm it up' using compression, perhaps of a type designed for vocals. Can you give me some guidance on how best I might do this? Of course, I realise I can use EQ, but would specifically be interested in any thoughts on how compression/limiting could be used on an existing take to get a warmer result. I've used Logic and the recording is clear, undistorted, and free from ambient sound.
Simon Evans via email
SOS contributor Mike Senior replies: There are ways to warm up a mandolin sound subjectively using compression, although none of them are likely to make as big an impact as EQ. Fast compression may be able to take some of the edge off a mandolin's apparent tone, for instance, assuming the processing can duck the picking transients independently of the note-sustain elements. There are two main challenges in setting that up. Firstly you need to have a compressor that will react sufficiently quickly to the front edges of the pick transients, so something with a fast attack time makes sense. Not all of Logic's built-in compressor models are well-suited to this application, so be sure to compare them when configuring this effect; instinctively I'd head for the Class A or FET models, but it's always going to be a bit 'suck it and see'. The second difficulty will be getting the compressor not to interfere with the rest of the sound. The release-time setting will be crucial here: it needs to be fast enough to avoid pumping artifacts, but not so fast that it starts distorting anything in conjunction with the attack setting. Automating this compressor's threshold level may be necessary if there are lots of dynamic changes in the track, for similar reasons. Applying some high-pass filtering to the compressor's side-chain (open the Logic Compressor plug-in's advanced settings to access side-chain EQ, and select the 'HP' mode) may help too, because the picking transients will be richer in HF energy than the mandolin's basic tone.The Advanced Settings panel in Logic's built-in Compressor plug-in contains side-chain equalisation facilities that can be very useful if you're trying to sensitise (or desensitise!) the compressor to a mandolin's picking transients.The Advanced Settings panel in Logic's built-in Compressor plug-in contains side-chain equalisation facilities that can be very useful if you're trying to sensitise (or desensitise!) the compressor to a mandolin's picking transients.
Another way to apparently warm up a mandolin is to take the opposite approach: emphasise its sustain character directly while leaving the pick spikes alone. In a normal insert-processing scheme, I'd use a fast-release, low-threshold, low-ratio (1.2:1 to 1.5:1) setting to squish the overall dynamic range. Beyond deciding on the amount of gain reduction, my biggest concern here would be choosing an attack time that avoided any unwanted loss of picking definition. In this case, shelving a bit of the high end out of the compression side-chain might make a certain amount of sense if you can't get the extra sustain you want without an unacceptable impact on the picking transients.
Alternatively, you might consider switching over to a parallel processing setup, whereby you feed a compressor as a send effect, and then set it to more aggressively smooth out all the transients. The resulting 'sustain-only' signal can then be added to the unprocessed signal to taste, as long as you've got your plug-in delay compensation active to prevent processing delays from causing destructive phase-cancellation. Using an analogue-modelled compressor in this role might also play further into your hands here, as analogue compressors do sometimes dull the high end of the signal significantly if they're driven reasonably hard, giving you, in effect, a kind of free EQ.
Monday, January 26, 2015
Q. Are some analogue signal graphs misleading?
Sound Advice : Mixing
I read your feature about 'Digital Problems, Practical Solutions' (/sos/feb08/articles/digitalaudio.htm), which said that digital audio can capture and recreate analogue signals accurately, and that the 'steps' on most teaching diagrams are misleading. Does that mean that the graph should really show lines, or plot 'x's, instead of looking like a standard bar-graph?
Remi Johnson via email
SOS Technical Editor Hugh Robjohns replies: Good question! The graphs in that article are accurate as far as they go, but offer a very simplified view of only one part of the whole, much more complex, process.
When an analogue signal (the red line on Graph 1: Sample & Hold) is sampled, an electronic circuit detects the signal voltage at a specific moment in time (the sampling instant) and then holds that voltage as constant as it can until the next sampling instant. During that holding period the quantising circuitry works out which binary number represents the measured sample voltage. This, not surprisingly, is called a 'sample and hold' process, and that's what that diagram is trying to illustrate. Graph 1: Sample & HoldGraph 1: Sample & Hold
So the sampling moment is, theoretically, an instant in time, best represented on the graph as a thin vertical line at the sample intervals (the blue lines in the picture Graph 1: Sample & Hold), but the actual output of the sample and hold process is the grey bar extending to the right of the blue line.
However, the key to understanding sampling is understanding the maths behind that theoretical sampling 'instant', and that means delving into the maths of 'sinc' (sin(x)/x) functions, which is the time-domain response of a band-limited signal sample. At this point most musicians' eyes glaze over…
As we know, the measured amplitude of each sample from an analogue waveform is represented by a binary number in the digital audio system. When reconstructing the analogue waveform that number determines the height of the sinc function.
The important point is that we are not just creating a simple 'pulse' of audio at the sample point, because the sinc signal actually comprises a main sinusoidal peak at the sampling instant (and of the required amplitude), plus decaying sine wave 'ripples' that extend (theoretically for ever) both before and after that central pulse. The reconstructed analogue waveform is the sum of all the sinc functions for all the samples.
The clever bit is that the points where those decaying sinc ripples cross the zero line always occur at the adjacent sampling instants. This is shown in the next diagram (Graph 2: Two Sinc Functions) where, for simplicity, just two sample sinc functions are shown for samples 23 (red) and 27 (blue). You can see that at the intermediate sample points (26, 25, 24 and so on) the sinc functions are always zero.Graph 2: Two Sinc FunctionsGraph 2: Two Sinc Functions
That means that the ripples don't contribute to the amplitude of any other sample, but they do contribute to the amplitude of the reconstructed signal in between the samples, with the adjacent sample sinc functions having the greatest influence, and lesser contributions from the more distant samples. This is shown in the next diagram (Graph 3: 3kHz Sinc Addition), in which the sinc functions of a number of adjacent samples are shown, and when summed together produce the dotted line that is a sampled 3kHz sine waveform. Graph 3: 3kHz Sinc AdditionGraph 3: 3kHz Sinc Addition
These last two diagrams have been borrowed from a superb paper by Dan Lavry (of Lavry Engineering), which explains sampling theory extremely well, and can be found here:
www.lavryengineering.com/documents/Sampling_Theory.pdf.
I read your feature about 'Digital Problems, Practical Solutions' (/sos/feb08/articles/digitalaudio.htm), which said that digital audio can capture and recreate analogue signals accurately, and that the 'steps' on most teaching diagrams are misleading. Does that mean that the graph should really show lines, or plot 'x's, instead of looking like a standard bar-graph?
Remi Johnson via email
SOS Technical Editor Hugh Robjohns replies: Good question! The graphs in that article are accurate as far as they go, but offer a very simplified view of only one part of the whole, much more complex, process.
When an analogue signal (the red line on Graph 1: Sample & Hold) is sampled, an electronic circuit detects the signal voltage at a specific moment in time (the sampling instant) and then holds that voltage as constant as it can until the next sampling instant. During that holding period the quantising circuitry works out which binary number represents the measured sample voltage. This, not surprisingly, is called a 'sample and hold' process, and that's what that diagram is trying to illustrate. Graph 1: Sample & HoldGraph 1: Sample & Hold
So the sampling moment is, theoretically, an instant in time, best represented on the graph as a thin vertical line at the sample intervals (the blue lines in the picture Graph 1: Sample & Hold), but the actual output of the sample and hold process is the grey bar extending to the right of the blue line.
However, the key to understanding sampling is understanding the maths behind that theoretical sampling 'instant', and that means delving into the maths of 'sinc' (sin(x)/x) functions, which is the time-domain response of a band-limited signal sample. At this point most musicians' eyes glaze over…
As we know, the measured amplitude of each sample from an analogue waveform is represented by a binary number in the digital audio system. When reconstructing the analogue waveform that number determines the height of the sinc function.
The important point is that we are not just creating a simple 'pulse' of audio at the sample point, because the sinc signal actually comprises a main sinusoidal peak at the sampling instant (and of the required amplitude), plus decaying sine wave 'ripples' that extend (theoretically for ever) both before and after that central pulse. The reconstructed analogue waveform is the sum of all the sinc functions for all the samples.
The clever bit is that the points where those decaying sinc ripples cross the zero line always occur at the adjacent sampling instants. This is shown in the next diagram (Graph 2: Two Sinc Functions) where, for simplicity, just two sample sinc functions are shown for samples 23 (red) and 27 (blue). You can see that at the intermediate sample points (26, 25, 24 and so on) the sinc functions are always zero.Graph 2: Two Sinc FunctionsGraph 2: Two Sinc Functions
That means that the ripples don't contribute to the amplitude of any other sample, but they do contribute to the amplitude of the reconstructed signal in between the samples, with the adjacent sample sinc functions having the greatest influence, and lesser contributions from the more distant samples. This is shown in the next diagram (Graph 3: 3kHz Sinc Addition), in which the sinc functions of a number of adjacent samples are shown, and when summed together produce the dotted line that is a sampled 3kHz sine waveform. Graph 3: 3kHz Sinc AdditionGraph 3: 3kHz Sinc Addition
These last two diagrams have been borrowed from a superb paper by Dan Lavry (of Lavry Engineering), which explains sampling theory extremely well, and can be found here:
www.lavryengineering.com/documents/Sampling_Theory.pdf.
Q. How do I record a hammered dulcimer?
Sound Advice : Recording
I've offered to have a go at recording a couple of friends of mine who play as a duo. For some of the songs, one of them will be playing a hammered dulcimer, which is something I've never tried to record before — do you have any advice?
Carl Turner
SOS Editor In Chief Paul White replies: Ideally, you need a stereo pair to give the sound some width and to help keep the volumes even across the strings. I usually place a pair of small-diaphragm, cardioid capacitor mics above the instrument, spaced by about two-thirds the instrument's width and around 500mm above the strings. Large-diaphragm models also work fine. You can't mic a hammered dulcimer too close because of the way the instrument is constructed and where the sound comes from — you'll lose the balance between the different strings. This means that, with this particular instrument, you need to be especially careful of spill.You can't mic a hammered dulcimer too close because of the way the instrument is constructed and where the sound comes from — you'll lose the balance between the different strings. This means that, with this particular instrument, you need to be especially careful of spill.
You need to keep other sounds away, though, otherwise you'll pick up a lot of spill; you can't mic much closer without upsetting the string balance and changing the tone. Spaced omnis would also work if the recording is an overdub being done in a sympathetic room where spill is not an issue. I know some players use contact mics for live work, so if you are recording at the same time as other loud instruments, it could help to take a feed from a pickup as well, just in case the overhead mics pick up too much spill. You may even be able to blend the contact mic and the overheads, as long as you compensate for any phase problems that might arise by time-aligning the waveforms and checking their polarity in your DAW.
I've offered to have a go at recording a couple of friends of mine who play as a duo. For some of the songs, one of them will be playing a hammered dulcimer, which is something I've never tried to record before — do you have any advice?
Carl Turner
SOS Editor In Chief Paul White replies: Ideally, you need a stereo pair to give the sound some width and to help keep the volumes even across the strings. I usually place a pair of small-diaphragm, cardioid capacitor mics above the instrument, spaced by about two-thirds the instrument's width and around 500mm above the strings. Large-diaphragm models also work fine. You can't mic a hammered dulcimer too close because of the way the instrument is constructed and where the sound comes from — you'll lose the balance between the different strings. This means that, with this particular instrument, you need to be especially careful of spill.You can't mic a hammered dulcimer too close because of the way the instrument is constructed and where the sound comes from — you'll lose the balance between the different strings. This means that, with this particular instrument, you need to be especially careful of spill.
Saturday, January 24, 2015
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