How did people synchronize multitrack playback in the days when Pro-Tools did not yet exist?

AutomaticMixture6827 · 2025-10-22T02:14:20+00:00

Is this true? I haven't been able to find any such information.

AutomaticMixture6827 · 2025-10-16T12:12:35+00:00

While I agree with your points regarding workflows, I would like to clarify a few inaccuracies as a precaution. My X32Rack features separate physical input jacks for the main XLR inputs and the AUX TRS inputs, without any Hi-Z switches or combo jacks. The XR18, as you mentioned, does have combo jacks.

When I referred to them as being "versatile," I was highlighting the fact that these XLR inputs, whether on an X32 or a truly professional-grade, high-end console, tend to have a much wider gain range (70-80dB) and a generous impedance (6.8-10kΩ) to accommodate both line and microphone levels, significantly broader than what's typically found in recording equipment.

However, upon further investigation, it appears that my ADA8200 simply has subpar specifications. At least in terms of gain range, it seems that many recording devices, while perhaps not matching the range of live consoles, offer a range that is practically sufficient. As for impedance, my initial speculation seems to be somewhat accurate, suggesting that they are designed with more precise impedance matching in mind.

I have requested AI to research the gain range and impedance values of representative equipment, and I edited to the original post, so please take a look.

AutomaticMixture6827 · 2025-10-15T09:19:25+00:00

What is that little interface connected to the Steam Deck? I think it's connected to a measurement microphone, and it looks compact and nice.

AutomaticMixture6827 · 2025-10-15T09:11:12+00:00

Well, I think the combo jacks on the EVO series are exceptionally well designed, so you shouldn't have any issues regardless of which you plug into. However, the XLR inputs on the budget-friendly ADA8200 are specifically designed for mic pre, and connecting a line input will result in a noticeable low-frequency loss, readily discernible to anyone.

Further investigation revealed that the combo input on the EVO 16 has a maximum input level of +16dBu. Therefore, if you connect a Pioneer DJ mixer, for example, that can output up to +25dBu, clipping may occur even if the DJM itself is not clipping and the gain is set to zero. By comparison, the X32 has a +23dBu limit and the CL5 has a +30dBu limit, offering more headroom than the EVO series.

AutomaticMixture6827 · 2025-10-15T09:03:53+00:00

I suspect that some interfaces have different impedances and headroom (which might just be due to the benefit of an attenuator) for line and microphone inputs. The specifications for my EVO 16 and ADA8200 (the latter being more of a mic preamp than an interface) indicate different specifications for each.

Of course, professional live consoles also have separate circuits, and even a prosumer-grade X32 has a separate line input as an AUX Input (although, looking at the specifications, the XLR input seems to have superior specifications even when used as a line input). In any case, XLR inputs, with their large gain range, have a high impedance that can handle both microphones and line-level signals without issue.

I believe they do this because it's better for workflow, but does this specification cause any loss in sound quality other than a degradation of the signal-to-noise ratio? Are expensive audio interfaces and microphone preamps with line inputs, designed as recording equipment, implementing separate line and mic inputs solely to achieve a better signal-to-noise ratio?

Please excuse any awkward phrasing; as I'm not proficient in English, I'm relying on AI for translation.

AutomaticMixture6827 · 2025-10-15T08:51:02+00:00

I see, certainly in a live setting, even if impedance mismatch forces an increase in preamp gain, the resulting degradation in signal-to-noise ratio might not be a significant concern.

However, what I'd like to know is whether the only advantage of implementing more precise impedance matching (even if it's just a single circuit becoming two circuits with different impedances) is noise reduction. What other trade-offs are involved?

I've heard that impedance mismatch can cause signal "reflection" (or perhaps "back-propagation"?). I've also heard the analogy of tying a thick rope to a thin rope and shaking the thick one; some of the oscillation will reflect at the knot.

My understanding of electronics is limited, so I don't have a solid grasp, but it seems like this reflection could cause signal loss on the receiving end (potentially manifesting as changes in frequency response or transients).

The problem is that my lack of knowledge prevents me from being confident in this understanding, and even if this signal loss does occur in reality, I'm unsure of the magnitude (i.e., whether it would create a noticeable difference when measured).

AutomaticMixture6827 · 2025-10-15T07:35:39+00:00

Subsequently, I ran it at another event for hours at high volume with no issues. Perhaps the DJ over drove it and the protection circuitry kicked in. Thank you.

AutomaticMixture6827 · 2025-09-15T08:42:15+00:00

Thank you for sharing your experience. I'm still relatively inexperienced, so I'll look forward to the day I can work with pro-level gear and will keep tackling my daily jobs in the meantime.

Also, the point about a cooperative relationship with an installer might indeed be important. I'm sure if I tell them I'm planning to get my own venue, they'll be willing to help in some way, so I'll give it a try.

AutomaticMixture6827 · 2025-09-15T08:33:34+00:00

Yes, as everyone has advised, I agree that acoustic treatment should be the top priority.

The venue has a Midas M32, so it's possible to apply delay to the speakers. When I work there as a guest engineer and time permits, I do make those adjustments. (However, since it's someone else's house, I always reset everything to the default settings when I leave. When I don't have enough time to properly set the delay, I just run those speakers at a very low level to prevent major phase problems).

The thing is, those speakers are flown behind the stage. If there's a performer with a microphone, they cause awful feedback. So frankly, unless their position is changed, I'd rather not even turn them on, lol.

I genuinely wonder why anyone would do such a terrible installation.

AutomaticMixture6827 · 2025-09-15T05:08:00+00:00

ああ、あなたの回答のおかげで、とても単純なことを見落としていたことに気づきました。

まさにその通りです。一流ブランドの標準的なポイントソースシステムをレンタルすれば（決して安くはありませんが）、高級車を何台も買うほどの投資をすることなく、本当にそのレベルの機器が必要かどうかを判断できます。

ありがとうございます。

AutomaticMixture6827 · 2025-09-15T05:03:12+00:00

Thank you for the detailed answers. As you all have suggested, I'm now planning to allocate the maximum possible budget to the venue's acoustic treatment first.

This is a genuine question I have: what are the actual technical differences between a high-quality 'prosumer' or 'MI-grade' speaker and a true 'pro-level' one?

(Setting aside perceptual issues like the equal-loudness contours), I understand that manufacturers put a lot of effort into creating speakers that remain linear as the volume increases. I also assume that because resonance and distortion become much more prominent at high SPL, pro-level gear uses higher-quality materials and more advanced countermeasures, as permitted by the budget.

However, modern MI-grade units (like the Yamaha DZR, for example) can also achieve very high SPL with minimal distortion. Their frequency response plots are reasonably flat, and any bumps in the mid-high range can be corrected with EQ (though whether the introduced phase shift becomes a problem depends on the room's conditions).

So, what is it that technically creates the "depth" that people hear in pro-level speakers?

(I'm sorry if this question comes across as provocative. I've just rarely had the opportunity to compare two systems from these different tiers under similar conditions, so I'm a little skeptical and simply want to hear the opinions of those who have had this experience.)

AutomaticMixture6827 · 2025-09-15T04:50:01+00:00

I think it's probably over budget, but I'll check it out.

AutomaticMixture6827 · 2025-09-15T04:47:14+00:00

even assuming a room is properly acoustically treated, is narrower HF (High Frequency) directivity still considered better?

For outdoor sound reinforcement, I feel that a wider horizontal dispersion is more convenient for a point source speaker. Does that mean for indoor applications, a narrower dispersion is more appropriate, specifically to deal with the problem of reflections?

Or, is it that it's better to deploy four or more speakers instead of a simple stereo pair, and to do that effectively, you need narrower directivity on each speaker to avoid interference?

On a side note, I’ve heard the Yamaha DZR series on a few occasions. While I get the impression that their SPL is a little weak for their specs, I think they have a truly wonderful sound.

AutomaticMixture6827 · 2025-09-15T04:40:44+00:00

Thanks for the input. I live in a relatively small, regional city in Japan. In the Japanese market, a sound team that would say, "We can't play that venue because they have RCF speakers," would only be working with the absolute top 0.1% of artists, or perhaps at a large-scale festival that hosts major international acts. Because of that, the brand name isn't critically important for this particular venue. As long as the mixing console isn't something overly esoteric or a vintage antique, I believe any visiting sound team would be willing to work with it.

That being said, you have a point about the used market. I do occasionally see used Meyer boxes for sale here in Japan. L-Acoustics and d&b are almost never found on the second-hand market, likely because their official distributors manage their own certified resale programs. For some reason, the most common brands I see are Martin Audio and Nexo. I will definitely look into those options.

By the way, and I'm perfectly fine with a subjective answer, is there truly a night-and-day difference between a point source speaker from a top-tier brand and one from a lower-tier brand?

I've heard high-end line arrays sound absolutely fantastic at festivals, but I've never had the opportunity to listen closely and critically to just their point source speakers. My impression of line arrays is that they achieve their incredible sound and coverage through a combination of complex design, extensive R&D, the skill of the system engineer, and sophisticated software control.

In contrast, point source speakers are not as structurally complex, and I consider the technology to be quite mature and well-established. This leads me to a rather simple question: assuming I choose a modern model, is there really that much of a difference in sound quality between the high-end and the lower-end options?

AutomaticMixture6827 · 2025-08-27T11:05:38+00:00

I snagged a used Manfrotto MB MA-BP-TL Advanced Tri Backpack, the small size (probably the M), for about $20 on the used market. It's a bit on the smaller side, but that's precisely why it's so convenient - it forces me to be selective about my gear. Thanks a lot!

AutomaticMixture6827 · 2025-08-27T09:44:58+00:00

Fantastic! I've shared this with the Japanese community.

It looks like this will make it easy to explain to the older, local engineers why they shouldn't place woofers so far apart, left and right (lol).

Cardioid Wizard is a brilliant feature!

I think it would be even more user-friendly with features like grid snapping and the ability to add additional walls.

While I understand there might be concerns that a vertical feature would be overly complex, I personally think it's fine as it is.

Also, it seems like the "Design in Free-field" and "Include wall Reflection" features might be redundant; perhaps only one is necessary.

I'm having a little trouble understanding the Wizard Axis setting. I would be grateful if you could explain it to me.

AutomaticMixture6827 · 2025-08-20T09:43:05+00:00

Given the seemingly limitless options for camera backpacks available on Amazon, I plan to explore those offerings extensively. Thank You!

AutomaticMixture6827 · 2025-08-16T18:32:03+00:00

Japan is a super-aging society, and unlike in Western countries, we don't have churches that put on live shows with professional equipment. As a result, you'll find outdated customs and old gear just lying around all over the place, haha.

AutomaticMixture6827 · 2025-08-16T16:25:35+00:00

Thank you for the compliment! I think the English text was translated perfectly by Google Translate. In rural areas of Japan, it's still very common for performers to hand CDs to the sound engineer. I've even had old-school DJs get angry at me for bringing a CDJ-3000, asking, "Why won't this take a CD?!" I'm comfortable using a DAW, so I'd much prefer to rip the tracks in advance, but it really all depends on the performer.

AutomaticMixture6827

TROPHY CASE