The blog has not been written in for some time: this is not due to a lack of musical activity, rather a flurry of intense musical activity which has left me with no time to update.
Today I was comparing a jazz recording I did to some old Stan Getz/Sonny Sitt recordings.
The difference was in the noise and fuzz - those old recordings were not clean by any means and mine was pristine clean.
To test it out I threw the logic 'clip distortion' over the master buss and I loved the effect! If I turned the drive down the result was better - more character, compression, glue.
So tomorrow in mastering I aim to 'dirty up' my mixes, either by printing to tape or by running through a preamp with a bit of overdrive.
Friday, March 19, 2010
Monday, February 8, 2010
City at War Mastering
My 4-track broke - channel 2 died. So I couldn't finish transferring the old City at War recordings onto computer. But today I got to borrow my friend Alex's, and my very good luck allowed it to be the exact same model as I have, allowing the dbx system to work and I know the playback speed will be right.
I mastered to an RMS of 20-22dB. It will be quiet, but commercial appeal has had nothing to do with this project.
I panned bass, guitar, and drums left and vocals and keys right, allowing me to get 2 channels into logic. I then compressed the vocal channel separately to the instruments to even out the vocals and summed into into mono to bounce.
The CD:
I mastered to an RMS of 20-22dB. It will be quiet, but commercial appeal has had nothing to do with this project.
I panned bass, guitar, and drums left and vocals and keys right, allowing me to get 2 channels into logic. I then compressed the vocal channel separately to the instruments to even out the vocals and summed into into mono to bounce.
The CD:
Wednesday, January 13, 2010
A chordal idea that makes me happy
What links these three songs? (you only have to listen to the first 20 seconds or so of each track to see what I am talking about)
Toto - Africa
MGMT - Electric Feel
John Frusciante - Unreachable
The chord changes in these songs all give me a similar feeling that makes me very happy .... the essence is all three have a strong chord change on beat 4 of the bar that is held into the next bar, preceding the more traditional place to put the strongest chord - beat one.
In Africa and Unreachable, it hits chord I on beat 4, the strongest chord of all!
In Electric Feel, it hits chord VI, also very strong (please note electric feel is on 6/4, and the the others are 4/4)
Any other tracks that do this? Please comment!
Toto - Africa
MGMT - Electric Feel
John Frusciante - Unreachable
The chord changes in these songs all give me a similar feeling that makes me very happy .... the essence is all three have a strong chord change on beat 4 of the bar that is held into the next bar, preceding the more traditional place to put the strongest chord - beat one.
In Africa and Unreachable, it hits chord I on beat 4, the strongest chord of all!
In Electric Feel, it hits chord VI, also very strong (please note electric feel is on 6/4, and the the others are 4/4)
Any other tracks that do this? Please comment!
Saturday, January 2, 2010
Recording speech at Woodford Folk Festival
I spent the last week volunteering at Woodford, a 6 day music festival, recording speeches and forums at the Greenhouse stage. Topics covered the environment, politics, and social science. They were interactive - 4 radio mikes were passed around in the crowd for audience participation. My job was very simple - take a mono feed from the desk, edit it, export as an MP3, and upload it onto the Greenhouse website (where you can now have a listen):
http://www.thegreenhouse.org.au
Prof Ian Lowe gave the highlight of the festival: Has Human Society Reached it's Limits? (To which he answered "Yes, any questions?").
And my view from the audience:
Codecs
My teammate, Ben, and I were going for minimum file size to make uploading easy so we encoded at 56kbps MP3, mono (half the filesize of a stereo file). The loss in quality is noticeable at this bitrate, but it's not a significant loss for speech. 50 minute files were spat out at around 20MB.
Dynamic Range
Most of the forums had many speakers, talking at different volumes through different mikes, so there was typically large volume differences. There were also regular 'pops' on the P's which would put a spike in the recording. I set my levels when recording so the softest sections peaked at -30dBFS and the loudest ones peaked at -10dBFS.
Knowing quality was not an issue and audio of this nature may be listened to streamed from the web on mobile phones with crappy speakers, I wanted a maximum volume RMS, without clipping. The solution was a hard limiter, limiting at -0.3dBFS (with lookahead). I pushed the limiter so the loud sections hit it hard, with up to 10dB of gain reduction, and the soft sections hit it occasionally - with this, the volumes were evened out, and erroneous pops and handclaps were soaked up. An example:
http://www.thegreenhouse.org.au
Prof Ian Lowe gave the highlight of the festival: Has Human Society Reached it's Limits? (To which he answered "Yes, any questions?").
And my view from the audience:
Codecs
My teammate, Ben, and I were going for minimum file size to make uploading easy so we encoded at 56kbps MP3, mono (half the filesize of a stereo file). The loss in quality is noticeable at this bitrate, but it's not a significant loss for speech. 50 minute files were spat out at around 20MB.
Dynamic Range
Most of the forums had many speakers, talking at different volumes through different mikes, so there was typically large volume differences. There were also regular 'pops' on the P's which would put a spike in the recording. I set my levels when recording so the softest sections peaked at -30dBFS and the loudest ones peaked at -10dBFS.
Knowing quality was not an issue and audio of this nature may be listened to streamed from the web on mobile phones with crappy speakers, I wanted a maximum volume RMS, without clipping. The solution was a hard limiter, limiting at -0.3dBFS (with lookahead). I pushed the limiter so the loud sections hit it hard, with up to 10dB of gain reduction, and the soft sections hit it occasionally - with this, the volumes were evened out, and erroneous pops and handclaps were soaked up. An example:
Wednesday, December 16, 2009
A look at logic's meters
Digital metering is apparently not easy to get right, because every piece of software does it differently. Most digital meters are peak meters: they respond quickly to transients (very useful to see exactly what's going on), but they do not show the 'average' level of the audio, which is what our ears perceive as volume.
The golden rule for setting levels is: 0dBv =-18dBFS . 0dBv is the analog standard of RMS (average, loudness) level for audio - the right level of output of a preamp, the output of a compressor, the optimal input for a tape recorder etc... However, RMS is a slow response measure, and peaks will always go above this. Analog equipment often has +24dB of headroom before clipping to cater for the peaks. In digital land, if you aim to record levels at around -18dBFS, you have 18dB of headroom before clipping at 0dBFS.
Therefore, -18dB is the centrepoint for recording audio, it is the loudness that things can move either side of. So it makes sense that is in the centre of the meter, no?
Apple's Logic Pro comes with two types of meters in the mixer window:
a) Exponential meter:
b) Db sectional linear:
Both these meters are displaying the same signal, a keyboard peaking at around -12dB on the digital scale. On the exponential meter, it seems like a very weak signal, even though it is above our centre mark of -18dBFS and is a very healthy level indeed!
The Db sectional linear meter is far superior. The -18dbFS mark is about halfway up (the centrepoint!), and there is great resolution between -40 and -18 - the major chunk of the working dynamic range of a lot of instruments and if you were recording classical music, much of the quiet passages would be in this zone. The exponential display shows the -40 to -18 range as only the bottom %15, a seemingly useless twittering at the bottom of the scale.
For some silly reason logic's default is the exponential meter, but this can be changed in Preferences -> Display.
For interests sake, check out the REAPER meter:
Very well designed, and is even more generous in it's placement of -18dBFS high up the scale. The designers of this program obviously understand the huge dynamic range digital recording has, and it's meters show that.
Protools meter:
Better than the default (exponential) logic meter certainly, but still places the centrepoint too far down for my tastes, and devotes the whole top half of the meter to only the top 10dB of dynamic range, making one believe they are tracking too quietly when in reality it may be a fine level.
The golden rule for setting levels is: 0dBv =-18dBFS . 0dBv is the analog standard of RMS (average, loudness) level for audio - the right level of output of a preamp, the output of a compressor, the optimal input for a tape recorder etc... However, RMS is a slow response measure, and peaks will always go above this. Analog equipment often has +24dB of headroom before clipping to cater for the peaks. In digital land, if you aim to record levels at around -18dBFS, you have 18dB of headroom before clipping at 0dBFS.
Therefore, -18dB is the centrepoint for recording audio, it is the loudness that things can move either side of. So it makes sense that is in the centre of the meter, no?
Apple's Logic Pro comes with two types of meters in the mixer window:
a) Exponential meter:
b) Db sectional linear:
Both these meters are displaying the same signal, a keyboard peaking at around -12dB on the digital scale. On the exponential meter, it seems like a very weak signal, even though it is above our centre mark of -18dBFS and is a very healthy level indeed!
The Db sectional linear meter is far superior. The -18dbFS mark is about halfway up (the centrepoint!), and there is great resolution between -40 and -18 - the major chunk of the working dynamic range of a lot of instruments and if you were recording classical music, much of the quiet passages would be in this zone. The exponential display shows the -40 to -18 range as only the bottom %15, a seemingly useless twittering at the bottom of the scale.
For some silly reason logic's default is the exponential meter, but this can be changed in Preferences -> Display.
For interests sake, check out the REAPER meter:
Very well designed, and is even more generous in it's placement of -18dBFS high up the scale. The designers of this program obviously understand the huge dynamic range digital recording has, and it's meters show that.
Protools meter:
Better than the default (exponential) logic meter certainly, but still places the centrepoint too far down for my tastes, and devotes the whole top half of the meter to only the top 10dB of dynamic range, making one believe they are tracking too quietly when in reality it may be a fine level.
Monday, December 7, 2009
Projects, Projects, Projects
Project 1: Live Recording of the Marsden Lee's
My good friends, Aaron, Jared, and Paul are in a quite fantastic pop/indie band. We are planning to record a studio track in the next few weeks. I recorded their live show to have a reference point, as as they say, practice makes perfect!
The band:
My vantage point from the wings (which happened to be the pool table room, and I was kinda in the way):
You can see my trusty Heil PR-40 as a mono drum overhead (which did a good job at capturing highs, even though it's a dynamic). 57 on the kick, 58 on the bass (which didn't get the sub-80 lows, but still sounds essentially like a bass), sennheiser e835 on the guitar.
A LESSON: Live guys don't always know good gain staging, or they are running into amplifiers with fixed gain so they operate their desks way under 0dBu. The solution: run your desk send (for me it was the vocals) into a preamp rather than a line in on your recording equipment , you can control the level and record it at exactly the level you like. Unfortunately, you just have to put up with the raised noise floor.
This is the result. I used limiting on the drum kit to tame these extremely annoying peaks in the waveform the heil somehow picked up. The drum overhead was sitting nicely at around -18dBFS, but for some reason, the odd transient (just on some snare hits) jumped up into the red and clipped .. when you look on the waveform, it's a single 1/2 wavelegnth long peak ... most strange .. light compression and reverb on the vocals, moved the guitar a few milliseconds out of phase in the right channel for 'fake' stereo guitar .. done
Sarah - The Marsden Lee's by Adam Friend Audio
Project 2: Contemporary works for flute and woodwind at the Artisian gallery, Brisbane
This was an extremely interesting setting, in an art galley with works hanging from the roof:
I borrowed some Behringer C2's to use as an XY stereo pair, and threw my Heil PR-40 up right in from of the players, so I could blend in a dryer signal if I wanted (the gallery place was quite wet).
And finally, I put my Zoom H4 in the corner exclusively for ambience:
Of course, sound will be forthcoming. One thing was, whether it was the air conditioning in the gallery or the self noise of those C2s I got a lot of background noise ... will make a conclusion on the viability of Behringer C2s for classical recording in due course.
Last, I present to you a track from the City at War recording fresh off tape, Ramble and Tamble. Enjoy:
Ramble and Tamble - City at War by Adam Friend Audio
My good friends, Aaron, Jared, and Paul are in a quite fantastic pop/indie band. We are planning to record a studio track in the next few weeks. I recorded their live show to have a reference point, as as they say, practice makes perfect!
The band:
My vantage point from the wings (which happened to be the pool table room, and I was kinda in the way):
You can see my trusty Heil PR-40 as a mono drum overhead (which did a good job at capturing highs, even though it's a dynamic). 57 on the kick, 58 on the bass (which didn't get the sub-80 lows, but still sounds essentially like a bass), sennheiser e835 on the guitar.
A LESSON: Live guys don't always know good gain staging, or they are running into amplifiers with fixed gain so they operate their desks way under 0dBu. The solution: run your desk send (for me it was the vocals) into a preamp rather than a line in on your recording equipment , you can control the level and record it at exactly the level you like. Unfortunately, you just have to put up with the raised noise floor.
This is the result. I used limiting on the drum kit to tame these extremely annoying peaks in the waveform the heil somehow picked up. The drum overhead was sitting nicely at around -18dBFS, but for some reason, the odd transient (just on some snare hits) jumped up into the red and clipped .. when you look on the waveform, it's a single 1/2 wavelegnth long peak ... most strange .. light compression and reverb on the vocals, moved the guitar a few milliseconds out of phase in the right channel for 'fake' stereo guitar .. done
Sarah - The Marsden Lee's by Adam Friend Audio
Project 2: Contemporary works for flute and woodwind at the Artisian gallery, Brisbane
This was an extremely interesting setting, in an art galley with works hanging from the roof:
I borrowed some Behringer C2's to use as an XY stereo pair, and threw my Heil PR-40 up right in from of the players, so I could blend in a dryer signal if I wanted (the gallery place was quite wet).
And finally, I put my Zoom H4 in the corner exclusively for ambience:
Of course, sound will be forthcoming. One thing was, whether it was the air conditioning in the gallery or the self noise of those C2s I got a lot of background noise ... will make a conclusion on the viability of Behringer C2s for classical recording in due course.
Last, I present to you a track from the City at War recording fresh off tape, Ramble and Tamble. Enjoy:
Ramble and Tamble - City at War by Adam Friend Audio
Tuesday, December 1, 2009
Recording for simplicity vs Recording for a mix
I have been wondering recently what is the right way to record _______ (instrument X). I came to the conclusion that one needs to take a different approach to recording instruments depending on if that instrument is designed to stand by itself on the final product (for example, a solo piano piece) or if the instrument is fitting into a mix of different types of sounds (for example, piano on a pop track).
1) Levels
This thread:
http://www.gearslutz.com/board/so-much-gear-so-little-time/420334-reason-most-itb-mixes-don-t-sound-good-analog-mixes.html
Dismantles the theory that one should track as hot as possible ... you only need to track to around RMS = -18dBFS and all the parts will sum to a good level on the master buss without having to attenuate it.
This makes perfect sense to me - for a busy mix.
For a simple recording, say a stereo pair on a piano that is to stand by itself, you have the liberty to track a little louder, to the level you wish the recording to appear on the CD.
Normalisation is not good for your audio. From http://www.playgroundstudio.com/blog/?p=85:
2. Never ever “Normalize” your tracks, especially if it’s the entire mix. Not only are you introducing further processing into your mix, but you are inviting quantization errors and digital artifacts. Again, even if we receive mixes that only peak at -10db, 9 times out of 10, it will end up sounding better than mixes that reached 0db or hit the “red” once or twice during mixdown." - KEITH CLEVERSLEY
I would argue changing the gain on a single source after recording it digitally is effectively normalising. So record loud to start with!
2. Transient and frequency response in microphone choice
Rock engineers love dynamic microphones (and colored condensers) .. classical engineers love accurate condenser microphones. Why is this?
Dynamic microphones normally have a colored response which involves a slower response to transient sounds and a non-flat frequency curve. Engineers can use this to their advantage: they choose a mic especially tailored to the frequency area that instrument is to occupy in the mix. For example, the standard for guitar amps is an SM57, which rolls off below 200Hz and above 12kHz - highlighting the exact space you wish the guitar to sit in the mix (midrange) - leaving room for the bass below and the cymbals and vocal sibilance above. There is also a 'presense peak' around 7k that gives the guitars 'edge' and makes them 'stand out'. The slower transient response is known to smooth sound sources that are not designed to serve a percussive element in the mix. Another industry standard is the AKG 541 for drum overheads - which has a boost in the 8kHz - 20kHz, the region that drum cymbals live in.
For classical recording, the mics are often set as a stereo pair (whether it be MS, bluemline, or XY) - and there is very little post mixing involved. These engineers want the most transparent and accurate registration of what is in the concert hall - and they go for accurate, flat-as-a-board mics such as Earthworks, DPA, and Schoeps. The same goes for simple recordings - say, a acoustic guitar and vocal. You do not want the mics to themselves be colored, because you do not need to fit them into a mix - the source is free to occupy the whole frequency range rather than have to fit into a conflicting mix of frequencies.
So, to sum up using the piano example: if you were doing a solo piano piece, you would use accurate mics to capture the detail of the piano as precisely as possible, with all the highs and all the lows. For piano in a pop mix, you would typically EQ out a lot of the body below 250Hz to prevent clashes with the bass, scoop out some upper midrange where the clarity of the vocal sits, enhance some highs around 8-10k for highlight the attack of the hammers, and perhaps heavily compress it for aesthetics (see: Lady Madonna by the Beatles), and to let it sit in snugly behind the drums and vocal.
Recording and instrument to stand alone: Go for accuracy and transparancy
Recording for a mix: Record according to the contribution that instrument is to make to the final mix, so the final mix is as balanced as good-sounding as possible.
1) Levels
This thread:
http://www.gearslutz.com/board/so-much-gear-so-little-time/420334-reason-most-itb-mixes-don-t-sound-good-analog-mixes.html
Dismantles the theory that one should track as hot as possible ... you only need to track to around RMS = -18dBFS and all the parts will sum to a good level on the master buss without having to attenuate it.
This makes perfect sense to me - for a busy mix.
For a simple recording, say a stereo pair on a piano that is to stand by itself, you have the liberty to track a little louder, to the level you wish the recording to appear on the CD.
Normalisation is not good for your audio. From http://www.playgroundstudio.com/blog/?p=85:
2. Never ever “Normalize” your tracks, especially if it’s the entire mix. Not only are you introducing further processing into your mix, but you are inviting quantization errors and digital artifacts. Again, even if we receive mixes that only peak at -10db, 9 times out of 10, it will end up sounding better than mixes that reached 0db or hit the “red” once or twice during mixdown." - KEITH CLEVERSLEY
I would argue changing the gain on a single source after recording it digitally is effectively normalising. So record loud to start with!
2. Transient and frequency response in microphone choice
Rock engineers love dynamic microphones (and colored condensers) .. classical engineers love accurate condenser microphones. Why is this?
Dynamic microphones normally have a colored response which involves a slower response to transient sounds and a non-flat frequency curve. Engineers can use this to their advantage: they choose a mic especially tailored to the frequency area that instrument is to occupy in the mix. For example, the standard for guitar amps is an SM57, which rolls off below 200Hz and above 12kHz - highlighting the exact space you wish the guitar to sit in the mix (midrange) - leaving room for the bass below and the cymbals and vocal sibilance above. There is also a 'presense peak' around 7k that gives the guitars 'edge' and makes them 'stand out'. The slower transient response is known to smooth sound sources that are not designed to serve a percussive element in the mix. Another industry standard is the AKG 541 for drum overheads - which has a boost in the 8kHz - 20kHz, the region that drum cymbals live in.
For classical recording, the mics are often set as a stereo pair (whether it be MS, bluemline, or XY) - and there is very little post mixing involved. These engineers want the most transparent and accurate registration of what is in the concert hall - and they go for accurate, flat-as-a-board mics such as Earthworks, DPA, and Schoeps. The same goes for simple recordings - say, a acoustic guitar and vocal. You do not want the mics to themselves be colored, because you do not need to fit them into a mix - the source is free to occupy the whole frequency range rather than have to fit into a conflicting mix of frequencies.
So, to sum up using the piano example: if you were doing a solo piano piece, you would use accurate mics to capture the detail of the piano as precisely as possible, with all the highs and all the lows. For piano in a pop mix, you would typically EQ out a lot of the body below 250Hz to prevent clashes with the bass, scoop out some upper midrange where the clarity of the vocal sits, enhance some highs around 8-10k for highlight the attack of the hammers, and perhaps heavily compress it for aesthetics (see: Lady Madonna by the Beatles), and to let it sit in snugly behind the drums and vocal.
Recording and instrument to stand alone: Go for accuracy and transparancy
Recording for a mix: Record according to the contribution that instrument is to make to the final mix, so the final mix is as balanced as good-sounding as possible.
The vinyl version offers some hope for Californication
Californication (1999) by the Red Hot Chili Peppers, mastered by Vlado Meller (a mastering engineer so infamous he is known simply as Vlado) stands as one of the worst and well-known victims of the loudness war. The album has no dynamic range - it has been squashed flat by compression and limiting - and is full of digital clipping, the level amplified to beyond the maximum volume the CD format can support. I found the vinyl version is a slightly better master.
Currently, the best listenable version of this CD is the 'unmastered' version (which is really a different mix altogether in some instances) available at: http://www.megaupload.com/?d=FBKM7IOB
I decided to run a test to see if the vinyl version of californication was in fact the same digital master as the CD. This was to counter the arguments post by these forum goers:
"Of got Californication on vinyl, and it sounds the same as the (terrible) CD. You get to hear an analog reproduction of digital clipping, why I record company allowed it to be released on vinyl in that state I have no idea."
from http://www.hometheaterforum.com/forum/thread/249591/rhcp-stadium-arcadium-cd-lp-mastering-comparison
"The vinyl sounds the same as the CD, with all the clipping reproduced accurately. :(
from: http://www.stevehoffman.tv/forums/archive/index.php/t-106260-p-2.html
I recorded the vinyl from my Sony turntable into my M-audio profire interface, into logic software, and compared the waveform with the CD version. Here are my results. Track: Parallel Universe. Two different waveform zooms.
Currently, the best listenable version of this CD is the 'unmastered' version (which is really a different mix altogether in some instances) available at: http://www.megaupload.com/?d=FBKM7IOB
I decided to run a test to see if the vinyl version of californication was in fact the same digital master as the CD. This was to counter the arguments post by these forum goers:
"Of got Californication on vinyl, and it sounds the same as the (terrible) CD. You get to hear an analog reproduction of digital clipping, why I record company allowed it to be released on vinyl in that state I have no idea."
from http://www.hometheaterforum.com/forum/thread/249591/rhcp-stadium-arcadium-cd-lp-mastering-comparison
"The vinyl sounds the same as the CD, with all the clipping reproduced accurately. :(
from: http://www.stevehoffman.tv/forums/archive/index.php/t-106260-p-2.html
I recorded the vinyl from my Sony turntable into my M-audio profire interface, into logic software, and compared the waveform with the CD version. Here are my results. Track: Parallel Universe. Two different waveform zooms.
The top waveform in the images is the vinyl, the bottom the CD.
You can see he waveforms are NOT the same, which means they are different masters. Some peaks get through on the vinyl which are clips on the CD. Sonically, the vinyl sounds more open and has space, and I did not hear the clipping. The CD is cold and flat. It is also a VERY loud vinyl though - you can listen to it without speakers, the resonance of the turntable itself is so loud. It appears they have been passed through the same limiter/compressor to remove the bulk of the dynamic range ... however the transients that got through the attack time of the limiter were preserved on the vinyl, and on the CD they simply clipped.
hmm... i should do some sound samples
A 414 discovery
My tascam 414 has dbx noise reduction built in. Tape has a significant noise floor - hiss. The louder you record to tape, the better the signal:noise ratio.
Dbx reduces the hiss by compressing the audio on the way in (pre-tape) by a 2:1 ratio to ensure the soft parts of the program are recorded at a healthy level above the noise floor. Then when the tape is played back the audio is expanded (the opposite of compression) by 2:1 to recover the initial dynamic range. The noise floor in the quiet sections drops down in the expansion - however, for a loud signal, the noise is boosted in the expansion, so the noise 'breathes' or 'pumps' with the volume of the audio (one of the main disadvantages dbx has compared to dolby NR).
The cool part is I can turn the NR (noise reduction) on and off whenever I want. I can record to tape with NR on and play back with NR off, which means the audio has been compressed, but isn't expanded on the way out, so I can score some free compression! Alternatively, I can record to tape with the NR off and playback with NR on - I then score expansion - which is a eerie envelope -almost like a gate - where the sounds seem to rush up at you out of silence and then quickly fall off back into the noise floor.
The compressed versions of the sounds are very bright - lots of high frequencies - I suspect the dbx process knocks off some of the tops, so they purposely put a high-frequency EQ boost in the compression stage.
Now to my discovery. There is a third 'sync' mode for the NR. As I was flicking the switch back and forth listening to my City at War recordings sync mode sounded the best. But what is it?
From the tascam manual:
When it is set to the SYNC position, Track 4 is disconnected from the dbx system, so the process does not affect the sync signals going to and from track 4, but tracks 1-3 still go through the dbx encode/decode process. Use the SYNC position for recording and playback of FSKsync or SMPTE time code.This can be used to save slightly out-of-tuneparts, or to create sound effects such as flanging.
I put the vocals on track 4 - by doing this I unwittingly gave myself a choice if I wanted to use vocal compression or not! If I use sync mode, the vocals which were recorded to tape with NR ON (ie compressed) can be played back without the expansion stage, whilst leaving all the other tracks expanded normally by the dbx. Sweet!
I am not the first one to do this, Richard Adams (U2) used a similar technique:
VOCALS: Bono's vocals were largely recorded with an SM58 and compressed with a Summit compressor. Adams: "Instead of using the Summit, what we did on several of the tracks on Achtung Baby was to record his vocals on tape with Dolby SR and play it back without Dolby SR. It tightens up the vocal sound and gives it more brightness and presence. It makes his voice sit really nicely in the mix and easier to balance."
(source: http://www.soundonsound.com/sos/1994_articles/mar94/u2robbieadams.html)
It's a kind of compression you don't find elsewhere - no attack or release times, no threshold, it's just a permanent tightening of dynamic range across the whole signal.
Sunday, November 29, 2009
Recording City at War
I received a request: to record City at War, a 5-piece band onto a 4-track cassette deck. After recording digitally in a Brisbane studio and being unsatisfied with the result (added reverb on everything did not match the aesthetic they were going for), this band was keen to go analog. I have a tascam 414 - a neat little unit out of production by the early 00s, and had its prime in the 80s and 90s as a popular choice for home recordings. They have an illustrious history - Bruce Springsteen's Nebraska, John Frusciante's Niandra Lades and Usually Just a T-shirt, and Elliot Smith's Roman Candle were all recorded on such machines..
The limitation, of course, is 4 tracks, enough to make a modern recording engineer claustrophobic just thinking about it. But, practically every album from the 60s, including Jimi Hendrix's first two records, Revolver, and Sgt. Peppers were recorded onto 4 tracks (albeit large 1" ampex tape machines, not cassette). I had to do what they did and submix and make mix decisions on the spot *gasp*.
In the photo you see 2 small mixers which each combined 2 mics to one output
Track 1: Drum kit. One overhead (sm57), and one kick mic (a beat up phillips karaoke mic that was lying around).
Track 2: Bass. DI.
track 3 2 guitar amp mics (sm58) and a DI'd acoustic on some occasions.
Track 4: Vocals and keys. This track was a feed from the PA ... balancing the voice and the keyboard proved to be difficult. Vocal mic 1: Heil PR40, Vocal mic 2: Sennheiser e835.
Now, this was my first experience submixing, and it was not made any easier by being in the same room as the band with a pair of not-very-isolated headphones. A lot of it was essentially guesswork and watching the red meters on the tascam jump. I had to take a close-enough-is-good-enough approach to setting levels, and pray everything made it onto tape. Only the bass, which wallowed in luxury on it's own channel was easy.
I haven't mixed it yet, but from the first listen, it worked!
Stay tuned for sound
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