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An evaluative production report of Linear productions, and the process and understanding of Linear recording.

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Final Stereo Mixe

Session Files

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Following on from last year’s assignment where we established and rehearsed techniques for recording various instruments using a blend of analog and digital resources, this term we are further developing these techniques. I am performing on the guitar and shall be utilizing both analog and digital workflows, recording vocals, and programming drums through EZ Drums which originated from analog recordings but are performed digitally. I  shall then produce a synchronous multitrack recording of the ensemble, planning and outlining the project according to a defined production process, including pre-production, rehearsal, demo recordings, brainstorming, discussion, planning, scheduling, and then mixing and mastering. For this process, I am using a Cranborne Audio 500R8 and the 500ADAT expansion, allowing for twenty-eight inputs and thirty-two outputs which enables me to take full advantage of 8 high grade compressors, 8 exotic EQs, and 2 industry-standard Lexicon reverb units. alongside this the SSL xLogic rack which I am using to sum into the analog domain for mixing and mastering. The guitar is routed through the Camden 500 preamp, while vocals are recorded through the SSL's VHD Pre. Audio will be summed into the SSL via the 8-input summing module and the 4-input summing module, taking full advantage of SSL renowned analog warmth and clarity. The final mix will then be completed in Pro Tools/Reaper employing dynamic, spectral, temporal, and spatial effects as appropriate. In the video I have included a detailed layout of my patch bay configuration, showcasing how the signal flow and patching are set up for this recording project.

Linear Production

I began by connecting my guitar into the Camden 500 preamp, adding some saturation using the Mojo knob to introduce a bit of low-end warmth. I adjusted the gain and set the levels accordingly. I then routed the signal into FL Studio's mixer channel and checked the incoming audio levels. I then would load a plugin  called Guitar Rig 7. Guitar Rig is a software that emulates a variety of amplifiers, cabinets, and effects via impulse responses of those original equipment used offering a wide range of customizable tones. I used it to dial in the specific sound I was aiming for. Whilst browsing for the sound I was aiming for, I played a riff from a chord I was strumming. As I experimented with different sounds and settings, this riff became the foundation for my melody structure. I eventually found a sound that I loved and was aiming for a tone that felt very dreamy with a swelling quality achieved with the use of heavy reverb. As I strummed each note would blend seamlessly into the next creating an ethereal and immersive sound I was looking for. I began recording my melody, which consisted of a chord strummed individually in a fingerstyle technique, alternating between two chords. To capture this riff  I armed the mixer channel with my guitar and started recording. i would use the metronome to help keep me in time. I then recorded another melodic pattern of the same chord but this time introducing a variation to make it feel more interesting with the 1st recording.

After recording the second melody I duplicated it and made it a unique audio track meaning any effect done on this audio track would not affect any other of the tracks similar to it . On this duplicated audio track, I applied a half-time effect to create a counter-melody for a new layer. To ensure the counter-melody blended well with the original melodies I filtered out some of the low end and adjusted the volume, allowing it to sit nicely alongside the other two melodies. I then raised the formant of the half timed audio to enhance the high-end frequency information giving the counter-melody a richer tonal quality. To further refine the sound, I added compression using the LA-2A to reduce the dynamics making the counter-melody feel more cohesive and full in the mix of the other two melodies. To finish off the arrangement I opened a VST plugin from Kontakt and used a piano library. I played a few sparse notes that complemented my melody. These notes were deliberately open and minimal adding a mysterious tone that enhanced the overall feel of the overall piece.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Moving on to the drums I used Easy Drummer alongside the Native Instruments "Maschine" MPC to create a dynamic and natural drum performance. The MPC's touchpads allow for expressive control where the intensity of each drum hit varied depending on how lightly or forcefully the pads were tapped. Easy Drummer uses authentic drum sounds that were captured using the various common microphones and configurations reflecting real-world variations in how drums sound when played at different velocities. This made the drum performance feel organic and true to life with each hit matching the dynamics of the performance as shown in the beginning of this video. I started off by playing an intense kick and tom pattern which reminded me of the rhythmic style often found in bands like Coldplay. The combination of the kick and toms created a powerful driving rhythm bringing energy to my composition. I then added hi-hats and open hi-hats and the odd few clashes to my groove complementing the tom and kick pattern which gave the performance a fuller more dynamic feel.  I played the groove alongside my guitar performance which tied everything together and created a cohesive full sound.. I also experimented with other drum kits within EZDrummer such as the Nashville kit, which provided a thicker fuller sound due to its different microphone configuration. I would revisit my performance which had been recorded in MIDI and started adding more variation to enhance the groove i created. I focused on humanizing the performance, adjusting the velocity of the notes to make them feel more dynamic either by softening or intensifying how hard the notes were hit. Additionally, I added some randomness and subtly shifted the timing of certain notes to create a more natural and organic feel further tightening the groove making the overall performance more expressive. At this point I also took the opportunity to structure the drums into an arrangement breaking it down into sections like the intro and verse. I made sure to add variation to the drum patterns for instance by incorporating extra hi-hats or subtle changes in the rhythm as the loop repeated. These small variations helped build interest and kept the groove more engaging, ultimately capturing the listener's attention more effectively. I also added fills at certain points to build transitions particularly leading into the chorus. Thise would include increasing the intensity of the velocity in specific sections to enhance the dynamics. Since I'm not a professional drummer I focused on paying close attention to how expressive the performance would be if played by a drummer. It's the small nuances like variations in intensity and timing, that I carefully considered to make the performance feel more natural and dynamic.

 

I then edited my play style in Easy Drummer, adjusting the velocity of the performance and the microphone used for capturing the audio. This allowed me to tweak the overall intensity and tone of the drums to match the track's progression. I also added extra variations to the hi-hats and filled additional areas that required more movement and energy."

For the drums, the amazing feature of EasyDrummer 3 is its ability to route individual drum sounds into separate channels in my DAW. The microphone configurations for each sound, such as the kick in, kick out, snare top, snare bottom, overheads, and room mics, can be routed just as they would be in a real-life recording environment. I would set a configuration similar to what I had at university for my previous linear productions, using channels for the kick in, kick out, snare top, snare bottom, hi-hat, hi-tom, mid-tom, low-tom, overhead, and room. Though there were additional channels for extra toms or room mics, I routed each to individual tracks, enabling me to mix them within my DAW like a real-life drum setup. I then routed each individual drum sound to my FL Studio mixer and tested each one to ensure they were sent to their respective mixer channels. I set the drums to their original mix, meaning they hadn’t been enhanced or panned yet, so I could mix them from scratch without any pre-mixing. As in any real recording environment, some microphone bleed occurred, where sounds from other instruments would be captured in each mic. EasyDrummer also features a bleed function, replicating this natural bleed you’d get with multiple microphones in the room. Additionally, room microphones capture ambient sounds or other noise that other mics might miss, adding realism to the drums. With all the tracks routed into my DAW’s mixer, I was able to fully utilize my mixing workflow. This setup mirrors the layout of a console, with two UF8 control surfaces that give me hands-on control over my mix, similar to an SSL large format mixing console. This allows me to have more precision and flexibility as I mix my drums. I would also name my tracks respectively to the drums, microphones, and color-code them for better organization. Once this was all done, I would begin my linear recording of my drums, arming my tracks and preparing to record my drum performance.

 

 

I took advantage of this moment to create a new drum pattern for a different genre, as one of my other assignments required me to work on a hybrid genre production. To do this, I found a drum kit that was more percussion-led, which allowed me to blend the original rock pattern I had created into a new hybrid genre. I was working with my colleague, a singer, who would provide direction on where she wanted the production to go. With this in mind, and the track I had already started, I focused on fully establishing the song while keeping the hybrid genre in mind. I fused elements from both genres, integrating my colleague’s ideas with my own. I then routed these tracks to my FL Studio mixer, making sure each sound source and microphone were placed on individual tracks. I also named and color-coded the FL Studio tracks before recording my performances

 

 

With a more developed composition in mind, I was now ready to begin my linear recording for my guitar parts. Initially, the guitar sections were just brainstorming ideas for the linear production, but now, with a fuller composition, I was set to record my guitar parts individually, giving them their place in the overall arrangement. To achieve the right tone and presence, I took advantage of the top-tier gear available to me. The guitar signal was fed into a preamp, which added some mojo saturation to the low end. From there, I routed it into an LA-3A compressor. The LA-3A combines characteristics of both the 1176 and the LA-2A: it uses a solid-state transformer like the 1176, meaning the attack is fast, but it also features a T4 optocell, similar to the LA-2A, so the release characteristics reflect the tube-based, vacuum compressor style. I also experimented with the LA-3A in LA-4A mode, which has a faster attack and release, giving the sound a punchier and more immediate response. Once I dialled in the compression to my liking, I routed the signal into the Marg EQ-4. This musical EQ allowed me to shape the guitar’s tonality to fit the mix. Finally, I ran the signal through the Kush Clarophonic, a parallel EQ known for its specialty in the high-frequency range. This EQ has bands for sheen, lift, presence, and silk, which I used to bring out the high-end clarity and brightness of my guitar, adding that final touch to the tone. With my signal processing set up and ready for my guitar performance, I would begin with a warm-up performance to get comfortable with the parts. Afterward, I would arm my track and prepare to begin my linear recording for all of my guitar performances. . I also recorded alternative performances, such as strumming harder during transitions to emphasize changes in the song. Additionally, I recorded lead guitar parts, using this time to improvise new riffs and enhance my lead sections.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

My next task was to begin the linear recording of my vocals. I started by setting up my microphone, which is connected to the XLR patch bay. The microphone I used has two inputs—one for the front capsule and one for the back capsule. This is because the microphone is designed to emulate different microphones' frequency responses by capturing both the front and back audio sources. Using this dual-capsule setup, I can change the polar pattern, frequency response, and axis of the recording, giving me flexibility in sound. Although the track might appear stereo due to both capsules being active during recording, the software merges the two captured sources into a mono track after processing. This allows for the emulation of various microphones' frequency responses and contributes to the overall sound quality and character of the vocals.  my XLR cables from the microphone were routed to the inputs of the SSL VHD preamps. I used two of them for the signal flow. From there, I would add some VHD harmonic drive, which adds 2nd or 3rd harmonics enhancing the warmth and presence of the vocals. I enabled the 48V phantom power and applied a slight low-cut filter on both channels. Then, I added the legendary Listen Mic Compressor from SSL. This is something I learned by tracking vocals in a compressed state, where they are fat and less dynamic, it helps achieve an in-your-face sound, which is great for rock, pop, and hip-hop vocals. I introduced just a subtle amount of compression, making my vocals less dynamic but not too squashed.

From the SSL VHD preamps, the signal would be routed to the Xlogic Master Module. The Xlogic Master Module is derived from the SSL 9000K console’s master section, providing the renowned SSL clarity and analog warmth, characteristics that make it highly sough. Once the signal passed through the Master Module, it would be routed to my patch bay outputs 9 and 10, as this is where the master module’s summing output is configured in my setup. From there, the signal would be routed into inputs 19 and 20 of the patch bay, which I use as inserts in my DAW to send the signal into the digital domain. Additionally, in my DAW, I would send the vocals to an empty track, where I would route the audio out into the analog domain and into my Lexicon PCM96 Reverb Processing Unit. After processing the reverb, the audio would be sent back into the digital domain, allowing me to blend the reverb with the dry vocal signal and finalize the reverb effects. In regards to my vocal chain, the advantage of having the Cranbourne Audio unit is that it allows me to not only send audio out into the analog domain but the ability to chain each module together, helping to create a powerful vocal chain. I have a great vocal chain setup with high-end EQs and compressors, configured to complement my vocals. Thanks to the ability to link each 500 series module, I’m able to build the ultimate vocal chain for my recordings, making full use of the analog equipment.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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Once my signal flow and setup were sorted out, I began by determining the key of the song. Once the key of the composition was identified, I selected an emulation of a legendary microphone, the AKG C12, using my microphone's software. The C12 is known for its smooth and warm sound, ideal for the recording I was aiming for. I then did a test recording to ensure everything was set up properly. Since I handle everything myself—recording, mixing, and everything in between—I did a demo take where I looped the section. This allowed me to fine-tune my vocal setup with the analog gear, ensuring everything was ready for the final recording. Once I had my vocal recorded, I started adjusting each section within my analog vocal chain, beginning with the Bluey 500, which is an 1176 Blue Stripe hardware unit. With a fast attack, fast release, and a 12:1 ratio, this helped to aggressively compress my vocals, giving them a thick, in-your-face quality, which is commonly used for genres like hip-hop, rock, and soul. The fast attack and release keep the vocal consistent, while the 12:1 ratio ensures the vocals stay controlled without losing energy. Next in the chain, I used a harmonic EQ, which allowed me to add or attenuate harmonic saturation. This tool is great for adjusting the clarity or sheen of my vocals without introducing harshness. It also lets me control the low-end frequencies by adding harmonic content to them, which helps keep the low end tight while still adding warmth. Following that, I used the LA-3A compressor. The LA-3A has a similar character to the 1176, but with a more gentle attack and release, similar to the LA-2A. This compressor helped me control the tail end of my dynamics, smoothing out the vocals while the first compressor took care of the initial fattening and level control. After that, I routed the signal into the MAG EQ. This piece of gear allowed me to shape the mids, high-mids, and low-end, and it's particularly good for managing subsonic frequencies. While I’m still learning to control this unit, I've found that when I introduce gain in one area, I have to compensate by attenuating a frequency elsewhere, which is a key lesson. To finish off the vocal chain, I used the Cush Clariphonic, which focuses on enhancing the high-end frequencies, adding more presence and sparkle to the overall sound. This final step gave my vocals the clarity and air they needed to sit perfectly in the mix

With my vocal chain set, I was ready to record. I deleted my test take and started looping the section of the song. The reason I looped this section was so that I could record all my vocals as separate takes. This is a technique I’ve learned from working with vocal professionals, including songwriters and artists.

The benefit of this workflow is that it allowed me to focus on the creative side of things, specifically vocal top lining. By looping the section, I was able to record multiple takes—three, four, or five—without overwhelming myself. This provided me with enough variation, which I could later use to create the best possible take by slicing together the strongest parts from each.  After recording, I moved on to my vocal composition. This involved slicing all the takes to create the best performance from each. Once I had this done, I would then glue all the slices together. Reaper has a feature that allows me to merge all my takes into one continuous audio track, which is perfect for this stage of the process. When I was happy with the result, I would open a plugin called Snapshot. This tool allows me to save a photo of my entire analog configuration, something that I’ve learned is essential when working in the analog domain. In the digital world, you can easily recall previous settings, but in the pre-digital era, this wasn't possible. So, by using Snapshot, I can take full advantage of the digital environment while working with analog gear. This allows me to save my vocal chain configuration, making it easy to return to the exact setup later if needed.

 

 

i would finish off by doing an initial mix which would be revisited later when i have a fresh pair of ears and mind. To start off my mix I would start gain stage my drums followed by grouping them for later when summing into the analogue domain. i would balance the levels and move on to the next task. i would pan some sounds and also adjust the stereo field more to give a more spatial feel to some sounds. (Bass sounds I usually place as full mono to avoid problems in mixing the low end) .

i would adjust the levels of all other instruments to fit well with the drums and would ensure i am allowing headroom for the mastering stage 

I would save my 1st mix revision and would be ready to master.    

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Summary

In this project, my main challenge was working with the analog domain, as I wasn't fully experienced with it. The main difficulty was the difference in how hardware reacts compared to digital plugins. For example, I was used to using the Mark EQ4 in the digital domain, but when I moved to the physical hardware, the way it handled audio was different. Additionally, recalling settings in the analog domain was another problem. Unlike in the digital domain where undo and recall functions are available, once you’ve set something in analog, you can’t easily go back or replicate it unless you’ve documented everything or experienced. whilst some hardware has stepped knobs/potentiometers you would have to remember your last sessions settings. This lack of recall ability made it harder to retrace my steps. Despite these challenges, I’ve managed to overcome them through the use of tools, a solid workflow, persistence and gaining experience over time as this kind of knowledge only comes with practice and exposure.

 

 

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