Entourage Paul Tucci
EntourageForum Replies Created
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Jesse,
I have a question for you first. Did this clap emanate from the vibe players playing position? Or maybe where you were playing from.PT
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Drew,
I might be the first to say it but I’d bet anyone who reads your question will wonder what “pulling down the vocal in the mix” sounds like. A before and after would help me hear exactly what your symptom is and what problem might be. How might you be using the delay? As a plugin on a channel? How much time are you using? Enough to create a slap back or little enough to create comb filtering which literally could cause level to drop? Mix percentage of the delay if it is on the channell? Equal level (50%) mix of a dry and delayed signal is the perfect recipe for comb filtering.
Gain staging?? Is the output lower with the delay engaged in line but at 0 time. Does the vocal level actually get louder with the delay plug in out?
So many questions….
Is it your terminology accurate? … Are you saying the actual vocal level goes down or might you mean that, with the delay effect, the vocal recedes back into the music. Instead of in-your-face-leading-the-charge vocals, they’ve moved back into the mix?
I ask if it might be heretical to USE an “Elvis” slap back delay sound in your working genres.
As Waylon Jennings once wrote “I don’t think Hank done this a way.”
Excellent subject line by the way. Reminded me of an old entry in my to-do list of rap lyrics.
“You quote Shakespeare, I quote Dylan.
I like Beastie Boys when they’re illin’ “
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Jesse,
Hell yea it’s a good idea, however, I don’t think the most important point of the “Only mid-range” story is the most important one to be considered.. Certainly listening to a bandwidth limited mix will show you if the killer kick drum you’ve created that is shaking the NBA’s balls even makes an appearance in a small speaker listening environment. There’s harmonics of the low stuff and attack of the kick that can be used to imply the presence of the kick without the low end present. Does the musical story get told if the speaker system is less than ideal, and probably more typical of many listeners’ situation?
My takeaway is that the actual topic is about changing perspective whilst listening. The small mono speaker on the shelf listened to off axis or even from outside the room is like downing a couple slices of ginger to cleanse the palate when you’re out for sushi. New perspective to be sensitized to the new taste. …From the audio perspective, if I let go the attachment and pride I feel about inflating up those basketballs to above league standard and listen with fresh ears…am I still in the game? For the football fans amongst us, and Patriot fans in particular, it Deflate-Gate, but reverse polarity.
I used a version of this thinking in my years being responsible for making sure that what was in the preacher lady’s heart made it to the audience members’ hearts. Highly intelligible was the goal. When walking around the outer concourse and poking my head into every seating entrance area I could check my success. Also I could see which vendor had the freshest pretzels. A man has got to have hobbies, you know.
PT
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Nate,
This sounds like a job for a detective. The YLM screenshot of the other song is a mere 10 seconds long. Not sure if that’s an issue here or not. I’ll go listen in on Spotify to both and see what you’re chasing.
PT
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Jesse, yes. I’ve had this one in the lineup for a good long while. It’s free, and if you use a fractional octave setting you’re comfortable with…useful. The Slope editor function is the Devil.
PT
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Drew,
There’s nothing like a big ol’ bass bomb from Phil or an acoustic bass to grab peoples’ attention. There’s a certain joy to be had with a big PA and subwoofers 🙂 . That acoustic bass can own the low end, especially in your genre.
So the lowest fundamental of an open E on the 4 string bass is 41Hz. Capturing it is a different story. You have to have those freqs recorded on tape/in the computer before you can can process them to fit in the music. EQ alone may not work if the relative level of 40 is way lower than the 100Hz area as your graphs showed. Multi-band compression won’t help much if there’s precious little content down there.
One trick I’ve used in similar circumstances is when the kick drum doesn’t really have deep bass but I want deep bass (without utilizing a sample) is to low pass a duplicate channel of the problematic instrument so that only low freqs are present. Then drop it down an octave via whatever tuning effect you have available. Where you low pass (high cut) the dupe channel is critical. If you want more below 50 Hz, I’d aim that low pass filter at 100ish Hz with a steep filter. Feather the newly created deep bass into the mix against the original. High passing (low cut) the original bass channel may clear some mud from the combination of sounds. It’s a cheat but effective if you can’t capture the low end off the pickup or microphone of the original recording.
PT
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Jesse,
Yes, I’m trying to convey the use of two different instances of the phone audio. One for the ambience of being live in the woods listening to the music so it doesn’t come across bone dry. (Who have I become?) If that stereo phone track needs a little EQ to help it talk with the music better, so be it. Anything that helps the illusion is fine.
The second use of the phone tracks would be as the driving source of the highly reflective/small room plate sound that I’m suggesting be the “unnatural” effect-y ambience sound. Perhaps wider to exaggerate the space. Tonal seasoning to taste but I’m guessing a low passed version here will do two things. One, let the low end of the close miced instruments retain their low end very distinct and intact, and two, allow you to subtly and magically change the listeners perception of the environment when and if the music hits its stride, becomes greater than the sum of its parts, and transports the listener.
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Jesse,
P-$ here.
“I’m a white man, a white man in black socks.
I wear grey shorts; tank tops and dreadlocks.”
No, I’m sorry. I think you wanted the other one…
Regarding your question. Yes, I was suggesting you use the phone mic as an ingredient in your mix of close miced parts. Something to give ambience and support the viewers’ point of view of being in the woods. Slight breeze, rustling leaves, woodland critters crunching the leaves as they scoot through. Speaking of critters, I would avoid the geese motif you used in the rowboat series, that’s so last summer.
I’ve not tried it but I do believe using a separate copy of the phone mic recording, slightly (100 Hz) high passed as the send to a plate reverb (H3000 Tight and Bright) might could add a sense of hearing reflections off the trees. Some sort of plate verb program meant for percussion that includes a bunch or early reflections and small room ( <1 sec) parameter options.
Creating an inviting and believable audio setting first can help you hook your audience before the musical story unfolds, much like the cinematic effect of looking out over desert in springtime bloom but hearing what appears to be a rattling sound. Adios cowboy.
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Jesse,
You’ve definitely opened a can of worms thinking about managing different arrival times, huh?
Your initial though of pushing earlier arrival times back in time (guitar amp/ vibes mics) to coincide with the latest arriving one (phone mics) is logical. It’s how we manage delay tower speakers at festival. Hold (delay) the signal feeding the delay tower until the main speakers have made that trip through the slow medium of air traveling at the speed of sound. As Dana pointed out, roughly a foot per millisecond. The two sources of sound can combine constructively if time and polarity-aligned. That translates to greater intelligibility, and an improved S/N ratio (signal to noise.) More dry signal level than ambient signal level.
I’m really disappointed that the Stonehenge joke in the first reply didn’t land.
Because dropping and dragging start times in the DAW is so easy you can opt to “pull” the late arrivals back to your self declared start time. That’s what we were suggesting. “Pulling’ the phone mic arrivals back to the vibe overheads. Your close mic’ed guitar would be the first arrival time to show up on the DAW session if you’re both playing the ONE! Just pull all the simultaneously struck “ONES” back to your amp arrival. Voila! Signal alignment may help clean up the sound OR not. If there’s not too much other-than-intended-signal in the microphone, the time misalignment may add spatial character which could be a better choice.
I believe it was Aristotle that said, and I’m paraphrasing here, “The more you know, the more you realize you don’t.” I just added the worms part.
-@PT
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Jesse,
Although I couldn’t spot it, I’m guessing the phone was somewhere on the “desk” on the stand. I was just checking it was visually in front of the band.
As you compare wave forms of the stereo vibe mics and your stereo phone recording we clearly see that the phone recording, which is further away from the noise source, the vibes, displays the start of the sound recording further to the right on the DAW tracks’ display. We’re basically looking at an X-Y graph. The X axis (horizontal) displays Time, the Y axis (vertical.) We’re looking at your handclap. For a white man in flannel, that is a pretty damn funky lookin ‘clap. It arrives at different times because the physical distance from the source to the microphones are different. The phone mics are further away physically so the sound of the clap (vibrations through the air) arrives later. Later shows up as a smidge to the right. View the screenshot… check? CHECK !
I’m going all science-y on the explanation here so as to provide backstory for those that may not yet understand tech. I’ve had 4 kids from the local college sound recording program shadowing me at the historic State Theatre this month and found a groove when I went really basic and broke things down to easily digestible nuggets.
So back to the original question …
The relative polarity of the phone recording to the vibes mics matters, especially if you slide the (later arriving) phone recording to align with the start time of the first arriving (vibes) signals. Find the first upward peak on the vibes, use that as your target. Find the first peak on the phone recording, if is downward facing..reverse polarity and the slide it over to the left to align with the first arriving positive peak of the vibes mic. Realize this is fucking time traveling magic. With the click and drag of a mouse, we rearrange time. Think about the poor bastards working at Stonehenge this weekend. As we fall behind with a quick digital reset, they’ve got to move all those big boulders one o’clock back.
I would first attempt signal aligning the phone recording with the mics, add a healthy level of it to see if that’s good au natural ambience, ie gentle wind, leaf crunching, geese, etc.
Another experiment would be to use the phone recording as the high passed send to a reverb.
re: milliseconds of delay
When combing two identical signals together but one is later, Inside of 20ish milliseconds they sound as one but with some tonal consequences. 50 milliseconds apart the two sounds will start to separate into two distinct arrival times. 100 milliseconds apart and you have arrived at Graceland, ie the “Elvis slap back)
Choose wisely
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Bar,
This is moody and pretty already. I was hoping @JBear would suddenly appear vocally and take it further.
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Nate,
Haven’t forgotten you. My curiosity has led me to some new-to-me info that I’m digesting before I nail the answer to your dilemnaa. A couple things seem obvious so far. Those with deep experience in record making don’t pay much attention to the LUFS. It’s like the plumber who can diagnose the problem and fix it in 15 but charges a couple hundred for the knowledge. Better yet, the analogy of Picasso pulling out a blank canvas and making a minimalist masterpiece in seconds because of all his previous work completed, having technique down cold, and then… intent. Those of us stumbling around in LUFS land currently will one day get it. So sayeth the alread -knowers.
I’m also thinking it’s possible to cheat the loudness numbers to make the Integrated number lower so as not to have the song turned down by the streamers. Because that Integrated number is averaged over the entire song, any EXTENDED sections that are above the threshold level of the noise gate will be included in the overall measurement and thereby “dilute” the time weighting of the loudest sections. Your reference track (Greater Heights) Integrated LUFS number was diluted by 2dB according to my experiment….I deleted the quiet intro and outro to that song and low and behold, the Integrated LUFS level jumped up by 2. This MAY be why you heard your track lower in comparison. If we’re only listening to which is the louder thing we hear and dismissing the quiet bits, we might could get fooled.
I’m not committing to that quite yet but am enjoying the exploration.
PT
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Drew,
Happy to help and look forward to putting an ear to your creation. Do me a favor though, if Dana and my name are in the same sentence, please do put his name first. Know what I’m sayin? We are, after all, in his house. 🙂
-PT
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I think it’s really a gain staging question.
You could send all the signal you want as long as you don’t overload the input to the verb. If you clip the verb input, that nasty digital schmutz will be baked into the verb return no matter what level you choose to use at the fader. You will have boxed yourself into a corner of somewhere between quiet distortion or “Oh Fuck!” fader dependent. In the old days, analog gear would be quieter when operating at healthy, but below clipping levels. The newfangled digital stuff seems to have a much better S/N ratio allowing a bit of slop in gain staging. There’s an argument to be made that the resolution of the reverb would be finer if all the digital bucket is filled with signal. No idea how audible that would be.
-@PT
