- Automation
- Over compressing toms (thanks to me...)
- L/R OH levels more balanced
- 2bus compression probably gels things better (this isn't a good example because we didn't use it for the record but I think it would have help a few songs on here.)
- Take advantage of room mics and automate them with song sections
- Compressors grabbing too much and making things dull. Use sidechains!
- Pay attention to the bass and how much impact it is giving to the song.
- Utilize Sidechains and effects, but don't overdo it.
- Take some time to dial in verb and then make send levels according to where you can see them on a visual stage. Create a better sense of depth.
- Understand that the MTA board isn't perfect and be ready for a ton of minor adjustments to make sure it isn't changing sounds on you. Though it might to something surprisingly welcoming like the feedback in brainwashed...embrace the spontaneity of the board!
- Pay attention to the final limiting application so that you aren't compromising the sound and not just going to the volume.
Wednesday, July 14, 2010
The Brushfire War Album Reflection
Friday, May 7, 2010
Brian Eno Presentation
Eno started out as a minimalist painter growing up and going to art school Britain. The first band that he was involved him that gain attention was Roxy Music. After a couple of big releases Eno wasn’t getting along with the front man and he left the band. Though Eno has produced a lot of famous, great sounding records, his contribution to music is mostly through his philosophical ideas toward the art. When growing up, he was really interested in the echo in Elvis's song ‘Heartbreak Hotel,’ and was always curious as to how the effect was achieved. While attending school, he was interested in the production of art and the process, rather than participating directly in it. One Eno’s thoughts is that you don’t have to play the music if you understand how its done and can appreciate it. One of Eno’s unique contributions is his ‘Music for Airports’ project. For the project, Eno took many aspects of an Airport into consideration. He wrote the music on the record based on public spaces, sounds that fit in the frequency spectrum that were above and below the human range as to not interrupt human speech, the songs should last a long time and they don’t interfere with public spaces and can still be considered art. He worked with David Bowie on three albums between 77 and 79: Low, Heroes, Lodger. He is quoted as saying the song Heroes saying ‘it was like a game,’ in that he was trying to find a really unique sound and then bettering what he would find. He also worked with Robert Fripp and Talking Heads to develop creative sounds. Another influential idea that is created to Eno, as well as partner in the idea John Zorn, is ‘Oblique Strategy Cards.’ These cards ‘take the chance out of chance,’ and allow an idea to be made by choosing a card and deciding fate. When working with the popular band, Coldplay, Eno made the band create music by listening only to the lead singer’s demo tapes. From here they decided the best approach to record the songs. Another famous songwriter, Joni Mitchelle, uses a similar approach in that a guitar part will be recorded, then it will be muted and then recorded again. When listening back, different parts can be put together to create a unique sound where the instruments are more spontaneous in nature. ‘Eno wants to be inspired by something and then go for it... He hopes to provide counterpoint to todays faster/cheaper thinking mind set and promote slower/better thinking.
Friday, April 30, 2010
408 Production Synopsis 9
Wednesday, April 28, 2010
EQ for Cyndi HeadLauper
Tuesday, April 27, 2010
The song ‘All Along the Watchtower’ was written by Bob Dylan and is featured on his album entitled John Wesley Harding which was released in December of 1967. The track only took 5 takes to record and in the end take 3 and take 5 ended up getting spliced together for the master. When listening to the recording, it is obvious that very minimal instrumentation was used. This is due to the fact that Dylan was trying to get back to his folksy roots and move away from the sound of his 3 previous electric albums. The song has an ABABAB structure where the Verse section (A) lasts for 16 bars, or 2 phrases of 8 bars. The Channel section (B) last for 8 bars and features Dylan on harmonica. The actual instrumentation of the song consists of Dylan’s vocals, acoustic guitar and harmonica with bass guitar and drum accompaniment. The feel of the song creates a forward movement with a fairly simple steady folk drum beat (that only changes once throughout the whole song, bass drum pattern changes at 1:30) and the bass playing the root and function notes of the chord. The song starts with a 4 bar intro of guitar, then drums, electric bass and harmonica come in to support the main focus of the song, which happens to be Dylan’s lyrics. The song is unique in that the lyrics never repeat themselves and there is no chorus. Also, the main chord progression, C#m-B- remains the same throughout the entire song, which also reflects how important the lyrics are intended to be understood.
In contrast, Jimi Hendrix’s version of ‘All Along the Watchtower’ was released on the album ‘Electric Ladyland’ in September of 1968. Hendrix actually heard Dylan’s version of the song on January 21, 1968 when he was starting the recording process of the album at Olympic studios in London. Jimi’s engineer for the album, Eddie Kramer, noted that Jimi started working on recording the song soon after he heard it. Session player Dave Mason was in the studio and played the acoustic rhythm parts for the song. During the session, Jimi’s regular bass player at the time, Noel Redding, had some issues with how things were being handled and ended up leaving. Jimi took up the electric bass and played the part for the recording we hear today. Kramer’s final mix of the song happened 5 days later, but when Jimi listened back to it, he felt that it needed something more. Jimi went on to overdub guitar parts during June, July and August, at a studio on New York called the Record Plant. The master tapes of the song moved from the original 4 track to a twelve track, and finally to a sixteen track recorder. This allowed Hendrix the time and opportunity to swap out solo and go back and forth between takes.
Jimi’s version contains an extended solo section, a revision of the guitar rhythm from the one Dylan had created, dense instrumentation and many more differences. The rhythm motif of the piece happens a little differently in Hendrix’s version, with a slower half time feel with the second chord landing on the second beat, while the Dylan rhythm emphasizes the the change on the ‘and’ of 3. This difference creates a much heavier feel. Along with the big drum fills, Jimi practically playing solo bass lines, newly invented guitar tones, auxiliary percussion, and many overdubs, the whole sound is plain huge and heavy sounding.
Many of the sounds Jimi had in his mind when producing and composing songs such as ‘All Along the Watchtower’ can be traced back to the principals of African music. Many of these same principals can be used to explain theoretically what Jimi was hearing when he chose the rhythms and voicing he used in his songs.
Much like what is heard in African music, Jimi uses interlocking pitches and rhythms as individual parts; when these parts are joined together, we get the powerful sound that we hear in songs such as ‘All Along the Watchtower. An example of this can be heard in the into of the song when the vibra-slap is used to fill the quarter note rests. The song keeps the energy moving forward in a much more consistent manner, rather than letting the momentum drop with space in between the rhythmic hits. Joined with
Buddy Miles playing tom fills between the rests, the prominence of the two percussion instruments create a very powerful effect. The giant sound is then continued with tambourin panned to the opposite side of the vibra-slap, consisting of constant eighth notes to make an even denser mix.
The next quality of sound that can be traced to African music is the songs ‘buzzy’ timbre. Hendrix was one of the first to use very high amp levels when recording electric guitar which caused harmonics to be produced which are very pleasing to the human ear. The performance that we are hearing is very clean, though most of the audio is overdriven to the point where it sounds sweet and inviting. I think that the reverb and delay that was used in the song helps to accent the overdriven sound because it feels like the dirtyness continues throughout the whole song as every instrument is super wet as though it is being sustained. Jimi’s solos never seem to end with this type of mixing treatment.
During the extended improvisational period of the song, each phrase of Jimi’s solos seem to be independent of each other. There is a call and response type feel to what the guitar is playing. The guitar is being panned from left to right and then back again to embellish the technique and create an open rhythmic sound similar to that of a hocket, or a back and fourth melody line between instruments. In the case of Jimi and his virtuosic guitar playing, he can do it by himself.
Throughout the entire Hendrix version, the same 4 bar chord progression is heard for the whole song. Though it might seem redundant on paper, the song is kept on its ‘toes’ by including tasteful variations of the ostinato rhythm throughout. The way that the chord progression starts out in the song is very different as the song progresses. Right away at 10 seconds in, the acoustic guitar strumming pattern expands from the rigid hits that we hear in the beginning. Jimi also does strong electric guitar strums in between vocal phrases to add emphasis and continuation to the lyrical line. I believe that the guitar solos that separate the verses could actually be considered choruses when related to pop music. They serve as the hook to the song as the same tone is introduced every time, other than during the extended guitar break (which would be where the conventional guitar solo section would be anyways).
That leads to the discussion of the solo section and how each 8 bars introduces a unique guitar tone and feel. During the first 8 bars, Jimi plays a solo that is in the form of a minor pentatonic and in context, is a fairly straight forward sound for the time. This set up Jimi to go in numerous directions; he decides to bring in a slide (“METal on METAL”) to create an open and wandering feel. This leads to a wah explosion where the back and forth panning occurs. The last phrase is when the instrumentation comes together as Jimi chugs out chords in an acceding manner to reach the climax of delivering the last verse.
The last point that is worth a mention is how the tempo fluctuates and changes throughout. The song has a very ‘alive’ feel in that the song seem to ‘breath’ faster and slower, wavering around a stead beat. The beginning of the song starts out somewhat slow, and when the guitar part comes in, it is sped up slightly until it winds back down during the verse sections. Every section where the guitar is a focus point, the drums and rhythm section speeds up as if to keep up with Jimi and also try and hold him back. I believe that in relating this to the theory of music and that of African Culture this can be tied to a type of community participation. The repetition of the chord progression further exploits the example of community participation because as the song moves along, the more momentum it gains. Every band member is feeding off someone else and you can hear the energy though out the recording.
Works Cited
Bjorner, Olof (May 7, 2000). "Still on the Road: Bob Dylan Recording Sessions". Olof Bjorner.
McDermot. Kramer. “Setting the Record Straight.” 1992. Grand Central Publishing.
McDermot, Kramer & Cox. “Ultimate Hendrix.” 2008. Hal Leonard Corporation
Friday, April 23, 2010
408 Production Synopsis 8
Thursday, April 22, 2010
techniques of 408 6
- Create 4 Stereo Aux tracks.
- Create independent bus mixes from each instrument track and send to any of the stereo aux outputs.
- Send the 4 HP aux tracks to B1, B2, B3, B4. These are now mono summed HP outputs.
- Go to the I/O menu to name each buss HP mix is going to be for each player (drum hp, bass hp, vox hp, guitar hp).
- Send TT out of B1, into HP 1 amp, out into A of room 100 HP patchbay.
- For multiple outputs of the same mix, create a send under the send section from an already outputted HP mix.
- Stereo In, Aux mono out; otherwise only 1 side of HP will work.
- Apple, 5 on numeric keypad. Memory location menu comes up of all memory locations.
- Press Enter to pull up memory location dialog box. Select 'no time properties' (none). Under general properties, track show/hide, zoom settings, track heights, group enables.
- Erase everything you don't want to see (all except drums).
- Name 101 and select what you want (drums).
- Vocals 102...ect.
- To recall memory location, on keypad press . 101 . or 'period 101 period'.
- Line 1 inputs
- Unity gain all faders
- Pan positions set
- Activate line 1 buttons
- Line 1 pot fully counter clockwise
- Engage mix button for each track
- Take off 2 track and mix button on master
- engage group 3-4 and take out mix button (after initial settings, sums to stereo)
- Put up group 3-4 monitor level almost full and pan hard left and right.
- One dry, one compressed...mix to taste.
- Engage group 5-6 for compressed sound.
- Group insert send, to compressor.
- Group insert return, from compressor.
- Bring up dry drums, then mix up compressed drums underneath to taste.
- Take group 3-4 and make insert send to Millennia.
- Instead of going back to insert return on group, send to 2 new channels on board.
- Use the 2 new channels on the board and group to 5-6.
- Now Millennia compression is going into Distressor and then out. Adjust to taste.
- Doing by hand is the most efficient approach.
- 1st create new audio track.
- Select drum section that you would like to use and copy.
- Make sure the tab to transient button is engaged.
- Press tab to go to transient.
- Press semi colon to go to blank audio track below.
- Press V to paste audio.
- Press P to go back up to original track.
- You should you the version that is the latest copied version so that it is the closest to the unadjustable threshold of the tab to transient.
- Apple + or Apple - is right hand side of region adjustment (on keypad).
- Option + or option - is left hand side of region adjustment (on keypad).
- Adjust the grid resolution to fine tune region selection.
- + or - to nudge region back and forth.
- Used to tweak performances.
- 2 modes of analysis.
- 5 different plugins.
- Polyphonic - for guitar, voice, orchestras.
- Rhythmic - drums and percussion.
- Monophonic - for bass, violin.
- Varispeed - affects pitch and acts like a tape deck.
- X - form - renders audio before playback so that CPU isn't utilized.
- To affect Vocals: Choose Polyphonic, go to analysis, protools selects markers for regions (event markers).
- Assign markers in analysis view.
- Option click to subtract markers.
- Double click to add markers where needed.
- Or remove by select selection, delete.
- You can also right click audio for elastic audio menu (when is analysis mode).
- Adjust sensitivity of event markers.
- For long sustained sound, don't use a lot of markers.
- Usually at start of transient.
- Add a bunch for glitchiness sound.
- After analysis, create warp markers .
- In order to change things, use 3 types of warp.
- Telescope - default setting - good for long phrases.
- Accordion - Shift click for 3 warp markers.
- Range
- Use the trimmer tool (time compression expansion tool or TCE) to adjust and warp region.
- Monophonic is just used for a different sound.
- Varispeed changes pitch and time.
- Most elastic audio plugs have window selection - window acts as a gate, while 'follow' is how the plug handles the regions dynamics.
- In rhythmic, 'delay rate' in window adjusts the gate time reaction.
- To fit bass and kick together, use monophonic. Assign analysis points. Put the bass transient slightly behind the kick (up to 15 milliseconds) so that tone sounds like it is coming from kick and EQ/Compression of the both instruments don't conflict.
Friday, April 16, 2010
408 Production Synopsis 7
Techniques of 408 5
- Bring up automation window by apple, 4 (on keypad) or under automation, in the window tab.
- You can 'suspend' all automation if you want for all tracks.
- Click plug-in to automate plug in.
- When automation tab is red, it is enabled.
- Go to write mode, put up volume level so you can hear what you are doing.
- Record and move fader to apply automation.
- Write mode overrides any previous automation points.
- Make sure selection is enabled in automation window.
- 'Off' can be enabled to turn off any automation.
- 'Touch' mode only adds automation when you touch fader and then returns to what is already has been automated.
- Can be used to combine old automation with new adjustments.
- Under preferences, mixing, automated time (for setting up how slow/fast you want the touch mode to return to previous automation.)
- 'Latch' mode plays back automation that has been made, and once you move the fader, it stays in place and moves only when you adjust.
- 'Trim' function tweaks automation only slightly, creating a composite of both old and new automation.
- Trim can be enabled with the other functions as well.
- Use the grabber tool to create break points.
- Option, Click to erase break points.
- Apple, Click to add break points.
- The little arrow at the bottom left of each track can be used to show automation of specific automation. Click the '+' for other independent automation views to be opened.
- Double click to select entire track.
- Highlight section and Press Delete to get rid of all viewable automation.
- Use pencil tool to draw in automation.
- Select Wave type under pencil tool and select the resolution (bars and beats, and 1/8 or other appropriate beat type).
- Edit, Copy Special to send automation of volume, to pan as well, or whatever other automation you choose.
- Make sure to un-group to do automation for 1 track of group.
- Use selection tool to highlight and move automation.
- Hold Command key to fine tune automation.
- Control, Option, Apple, Click pot button and select automation option.
- Go to latch mode with the plug in automation selected and edit in real time with the plug in.
- Write mode is destructive.
- Latch mode only applies to one automation.
- Use Escape key to scroll through edit menu tools.
Friday, April 9, 2010
Digital Audio
by James Meder
History - Audio’s Transformation from Analog to Digital
1976 - “The first 16-bit digital recording in the US was made at the Santa Fe Opera on a handmade Soundstream digital tape recorder developed by Dr. Thomas G. Stockham.” (Pohlmann)
1982 - “The first digital audio 5-inch CD discs marketed, merging the consumer music industry with the computer revolution.” (Pohlmann)
1988 - “For the first time, CD sales surpassed LP sales, leaving CD and cassettes as the two dominant consumer formats...” (Pohlmann)
1998 - “Jonell Polansky produced the first 24-bit 48-track digital recording session at Ocean Way on Nashville's Music Row.” (Pohlmann)
2001 - “Apple Computer introduced on Oct. 23 the iPod portable music player. (solid-state iPod released Jan. 11 2005).” (Pohlmann)
Digital Audio Basics
“At its most elementary level, it is simply a process by which numeric representations or analog signals(in the form of voltage levels) are encoded, processed, stored, and reproduced over time through the use of binary number system.” (Huber)
Digital Music Systems use a Binary (2 base) System which is encoded in media as: 1 or 0, On or Off, Voltage or No Voltage, Magnetic Flux or No Flux, and Optical Reflection Off of a Surface or No Reflection.
Sample Rates, Bit Depth and Resolution
In Analog, continuous signals are passed, recorded, stored, and reproduced as changes in voltage levels (change over time).
Depending on the sample rate, periodic periods of time are processed by creating a binary word that represents the signals level and waveform. These binary words are stored and can later be recreated during the D/A process.
A sample rate of 48 kHz represents a sample every 1/48,000th of a second.
The higher the sample rate, the higher bandwidth of signal is available. This results in the ability to clearly represent higher frequencies of a digitally recorded sound (higher resolution).
“During the sampling process, an incoming analog signal is sampled at discrete and precisely timed intervals (as determined by the sample rate). At each interval, this analog signal is momentarily ‘held,’ while the converter goes about the process of determining what the voltage level actually is, with a degree of accuracy that’s defined by the converter’s circuitry.” (Huber)
A binary encoded number is then given to the computer (hard drive or storage device) that represents the analog level.
From here, the audio can be stored and referred back to in the future when the timed interval is reassembled with the binary word during the D/A process.
The converter then continues the process by quickly moving on the the next sampling period.
“The Nyquist Theorem states that in order for the desired frequency bandwidth to be faithfully encoded in the digital domain, the selected sample rate must be at least twice as high as the highest frequency to be recorded. If you wish to record and capture a frequency of 20 kHz (upper level of human hearing), the sample rate must be at least 40 kHz.” (Huber)
‘Quantization’ represents the amplitude level in relation to the sampling process. Voltage levels of the signal are processed and stored as binary digits so that they can be stored and later recreated.
Currently, the most common binary word length for consumer audio is 16-bit (cd quality).
Added internal headroom at the bit level, helps reduce errors in level and performance at low-level resolutions.
Also, the higher the bit level, the more headroom is allowed, creating less of a chance for clipping and digital distortion to occur. In the near future, system of 32- and 64-bit resolution will become the norm.
Analog-to-Digital and Digital-to-Analog Conversion
“In its most basic form, the digital recording chain includes a low-pass filter (or anti-alias filter), a sample-and-hold circuit, an analog-to-digital converter, the circuitry for signal coding (or multiplexor), and error correction.” (Huber)
Low pass occurs in order to block frequencies that are greater than half the sample rate frequency from having to be converted.
A Sample-and-Hold (S/H) circuit holds and measures the analog voltage level for the duration of a single sample period, its length is dependent on the sample rate.
For the next step, A/D conversion happens by encoding the signal into a binary a word. This is the most important step because the converter has to efficiently create the binary word from a DC voltage ? level that is correctly quantized to the nearest step level, very quickly in order to move on to the next sample.
The digital data then needs to be sent to storage, but before that happens, more processing and conditioning takes place, which includes: data coding, data modulation and error correction “(synchronization and address information)”. Rather than just keeping the data raw, it is coded into a form that can be easily and accurately stored and found.
An important process of digital audio data coding is pulse-code modulation (PCM).
“The density of stored information within a PCM recording and playback system is extremely high.” (Huber) In order to compensate for any problems in the audio due to the large amounts of audio data, many forms of error correction is used.
A mathematical pattern is used with PCM and error correction, where the binary word is sent in random order to be stored in a binary bitstream. This process also allows for the corrupted audio to become available later when a ‘puzzle piece’ of the bitstream is regenerated during the D/A conversion process.
Without this part of the coding process, digital audio would be close to useless because it ultimately saves the quality of the audio.
The digital reproduction chain (D/A) for recreating the binary word in order to be heard, is much like the A/D conversion happening in reverse.
The recorded data is restored to its modulated binary state and into pulse code form.
Next up is the digital-to-audio conversion where the analog voltage levels are reinstated from the binary word.
A sample-and-hold happens to distinguish the most-significant to least-significant bit.
The final step is a low pass filter so that the signal doesn’t distort due to high frequency harmonics.
Within every step of the digital ‘reproduction chain,’ summing each 1 or 0 together determines the voltage output.
Digital Audio Transmission
“When looking at the differences between the distribution of digital and analog audio, it should be kept in mid that, unlike its counterpart, the transmitted band with of digital audio data occurs in the megahertz range...” (Huber)
Digital audio is very susceptible to errors and signal irregularities due to the large bandwidth that is required.
Useable transmission of digital audio signals include: AES/EBU, S/PDIF, MADI, ADAT Lightpipe, TDIF, and mLAN.
AES/EBU (Audio Engineering Society and the European Broadcast Union) audio transmission is used between professional audio devices and is capable transferring two channels of interleaved audio through a single XLR cable. This is possible by having pin 1 of the XLR acting as the ground and having pin 2 and 3 carry the two signals.
S/PDIF (Sony/Phillips Digital Interface) is used to connect consumer grade 2 channel audio devices to professional interfaces and the like. Can be used with a single conductor unbalanced phono (RCA) cable, as well as an optical ‘Lightpipe’ connection. Can work as a link between multichannel data devices like that of a surround sound system.
ADAT Lightpipe uses the same optical Lightpipe cable as S/PDIF, but is capable of handing up to 8 channels in one direction. The 8 channels can link multiple audio devices and with two cables, is also able to handle 8 In/Outs simultaneously.
TDIF (Tascam Digital Interface) uses a 25 pin D-Sub cable that is able to send and receive 8 digital audio signals bidirectionally. This is the Cable system that is used to run audio from the 192’s to the HD PCIe audio cards in the Mac.
A digital distribution device can be used to route audio devices together to prevent jitter and wordclock errors.
Word clock and Jitter
“Jitter is a time base error...It is caused by varying time delays in a circuit paths from component to component in the signal path. The two most common causes of jitter are poorly designed Phase Locked Loops (PPLs) and waveform distortion due to mismatched impedances and/or reflections in the signal path.” (Katz)
Jitter occurs when long cables are used with incorrect impedances or the source impedance is not correctly matched at the load. It can lead to sound waves that were initially square to become round, fast amplitude times can become slow, and the zero crossing point of the waveform can be less accurate.
However, when looking at the binary word of a square wave that is encountering jitter, it would be the same as the original ‘unjittered’ binary number. The only way you would be able to hear the difference is through obvious distortion; which usually is clicks or tics in the audio.
During D/A conversion, the process where jitter occurs is during the sample and hold period. If this process isn’t stable, then the digital audio won’t be able to return to an analog signal quickly enough and result in “loss of low-level resolution caused by added noise, spurious (phantom) tones, or distortion added to the signal.” (Huber) With a jitter problem, a digital converter can shrink the dynamic range of a recording by a lot. When listening back to such a recording, it can sound grainy, have loss of definition, have loss of stereo width, and obvious signal loss.
The purpose of Wordclock or a Master Clock is to reduce jitter and to link multiple converter devices to the same sample rate times, as well as the same sample-and-hold periods.
Only one Master Clock can be used at once so that each devices’ internal clock can run “within a connected digital distribution network.” (Huber)
A wordclock is only necessary when using systems with various conversion devices.
Sound Quality
Tech Side...
There are ups and downs to both analog and digital audio. In analog recording, the tape is able to capture a continuous signal. If this same signal were to be captured in digitally, a bit depth would have to be chosen and the signal would need to be quantized.
The introduction of bit depth and quantization brings about the signal-to-noise issue and how accurately a signal can be reproduced. In a similar manner, when recording in analog, a signal-to-noise (S/N) ratio is introduced and can cause the dynamic range of a recording to suffer.
Dither is another component that separates analog from digital. The digital process of sampling and quantizing can create a ‘squared off’ understanding of a waveform and when recreated, low level harmonic distortion can be heard due to the ‘square wave’ sound of a sampled and quantized piece of digital audio. This happens during the A/D process when the Least Significant Bit (LSB) is encoded into the binary word. It can be avoided by adding dither, which is a small amount of noise. When noise is added, a converter has an easier time encoding, or deciding whether to use a 1 or 0 as the LSB.
Ultimately, the low level harmonic distortion is a lot more obvious when recording at a lower bit depth.
When converting from a higher bit depth (24) to a lower bit depth (16), it is a good idea to use dither. The converter will have an easier time deciding the LSB for the 16 bit signal, thanks to the dither.
If you are transferring an analog recording to digital, you don’t need to worry about dither because the tape noise will naturally decide the LSB.
Opinion Side...
Some engineers like recording to tape because they believe that it will yield a warmer, more natural and vintage sound compared to the sterile sound of digital. Other engineers like recording digitally because the S/N ratio is important to them and they feel that it will give a more exact rendition of the performance. Digital can be a lot more dependable and can be a lot more portable. If you like the sound of analog, there are many tape emulators on the market; hardware and plugin.
Works Cited
Huber, David Miles. 2005. Modern Recording Techniques. Massachusetts: Focal Press.
Katz, Bob. 2002. Everything You Always Wanted To Know About Jitter But Were Afraid To Ask. Mastering Audio, The Art and The Science. Massachusetts: Focal Press.
Pohlmann, Ken C. 1995. The Digital Revolution. New York: McGraw-Hill. Retrieved from http://history.sandiego.edu/gen/recording/digital.html
Recording Presentation Week 2 Notes
Friday, April 2, 2010
Recording Presentation Week 1
408 Production Synopsis 6
Friday, March 19, 2010
Techniques of 408 4
- Under Preferences, go to the operations tab of the setup menu. Select: automatically create a new playlist when loop recording, latch record enable buttons, link record and play faders.
- After creating and naming a new track for loop record pass, enable loop record via operations, loop record or use the shortcut option, 'option L.'
- Highlight a section to define the loop recording area.
- Add a bit of pre-roll if need be.
- Record takes!
- You can then select the 'playlists' tab and bring up all the takes.
- 'Promote' the best takes up to the top enabled track by selecting the area wanted, and pressing the arrow button on the left hand side of each track.
- Used as a technique to achieve a 'fatter' drum sound, while preserving the natural attack of the drums.
- You could approach this technique by duplicating tracks and adding compression, but it's a lot easier by using busses.
- Start by creating a new stereo bus send, to a stereo aux track (summed drums)
- Create a second stereo aux track that has the same input as the previous aux track. Add a compressor to this track and adjust settings to taste.
- This second compressed track should be used to act as 'glue.' Creating a stronger image of the drums. Bring fader up to an appropriate level. It can also be useful to put this aux mix panned center to create a punchier and more full presence.
- Serial compression is essentially two or more compressors in a row.
- It is typically applied to run lower ratios to achieve more compression with less of the negative sounding effects that one compressor using high setting might contain.
- To start, set the first compressor to a 6:1 ratio, fast release (aggressive), a high threshold and little gain. Set the second compressor to a 3:1 ratio, normal release (smooth), a lower threshold and have this compressor handle all the make up gain.
- Start by taking the original signal and splitting it into two sends.
- The first signal bus goes to compressor 1. The second signal bus heads to compressor 2.
- Finally, send the compressor 1 channel to the compressor 2 channel.
- Adjust both section to taste. Remember that Compressor 2 channel now has the master output of the signal.
- Something to remember: send the source using the audio output path for compressor 1 and the sends section to send the signal to compressor 2.
- In the I/O section of protools, you can name the busses (verb, compression, headphones). This is useful for being able to visually follow a signal path.
- Plug-ins should be placed in the following order when working with kick/snare/toms: EQ, Compression, Gate.
- Plug-ins should be placed in the following order when working with gated verb: verb, EQ, compression, gate.
- Don't mix through a master fader!
- Making individual Kick and snare aux sums can be done even before going to the drum sum. Use serial compression: have a compressor on the actual kick channel, then put a second compressor on the kick sum channel.
- On the drum sum, you can place an EQ plug-in and boost the lows and highs to bring about more 'simmer' and more 'umph.'
- Change between active playlist track - control, P or Control, ;
- Promoting Playlist Section - control, option, V
- New Track Menu Options - control, command, up, down, left, right
- Toggle 'shuffle, spot, slip, grid' - tilda key
- Solo a track - shift, S
- Enable/Disable track - control, command, click
- Consolidate - shift, option, 3 (name the track afterwards so you know what it is)