The notable differences...Large and Small device responses.

There’s a very large difference between composing with headphones and a quality audio system. The lower frequency in the 60Hz and below range are cut off when using common $50 headphones and will sometimes bleed into the 100Hz and higher range to compensate. The same goes for the high quality $80 audio systems where the high end of 20Khz and above will bleed into the lower frequencies, but are not as much noticeable with headphones. So when composing with my Bose audio system and then listening through headphones my music sounds completely different because of how the speaker response operates. Though Bose audio systems are not a common household item to have, my headphones are which makes it difficult on how to compose and master between high and low quality outcomes.

Just remember, how it may sound on your system doesn’t mean it’ll sound the same on others. People using 2.0 speaker systems will not hear the bass, the common $40 2.1 speaker systems have a habit of bleeding ranges, the $80 ones have a habit ignoring high pitches, and Bose and other quality systems just like being themselves and process everything through as if it were in the theaters.

And to just solve all your problems, though not recommended, is to cut off the 60Hz and below and 20Hz and above ranges via a master EQ. There will still be bass, and the high pitches can still be heard well enough without cracking household audio systems.

Have fun.

Wait a second – Draqo – are you related to Maestro Rage? You’ve got his symbol

nah, we switched symbols. We work together on many things, and since my name is Maestro[rage], we figured he having the blue, and I having the red would be better.

Regarding the topic itself, I honestly have no clue if my sennheiser HD 280 Pro’s do any kind of bleeding, though I don’t doubt it, i’m not sure how I can gauge it.

I’m not sure what you mean by “bleeding” into different octaves…can you clarify this?

Another long type…

When you have high-polyphony instruments being layered all at once, the speakers must cope with the output. This can sometimes lead to music sounding like it’s full volume and in ranges where it shouldn’t.

When it gets really bad, on an EQ display it will show ranges being amplified as a result for compensation. (This is even so at 44.1KHz-48KHz sampling rate.)

So at a certain point, your audio system starts telling you that there’s too many samples/variations and ends up bleeding over (response time/vibration factor.) Beyond that, the computer itself is indicating too many samples/variations at once.

Generic 2.1 speakers systems have the most problems being designed to send frequencies to 2 individual parts (the satellites, and the sub) without intelligently splitting it up. So you technically have 2 audio systems, both playing the same tune, but different specifics. For example, why would you need to hear a sub-bass tune being played through the satellites when it was specifically designed for the subwoofer? Now add a instrument like a violin and play almost the same tune the bass is, it gets worse. And this is what I call a bleed.

Now, when using headphones and 2.0 audio systems, deep bass is gone yet its ghost still remains on the higher ranges, and sometimes drowning the song.

With higher-quality audio systems like a Logitec X240 or even BOSE C3 2 .1 sounds are better self-EQ’ed and in the proper audible ranges. This is thanks to lower response times and better vibration management (magnets are magnets and obey to Newton’s law of motion.)

I hope that explains what I’m trying to point out.

(And there’s a typo in my first post, instead of “20Hz and above,” it should be “20KHz” and above.)

Gotcha, so I assume you just mean the low-level compression and non-linearity that the amplifier and speakers driver start to move into.

THD is an interesting thing…I think it might be interesting to mention a funny thing I thought of after talking to an electronics guru a while back.

This particular guy working at a electronics parts store (the type that actually sells things like hex inverters, 5 watt stereo audio amp component kits, and parallax sets) seemed very knowledgeable about most everything electronic so I posed to him a question that always seems to mystify even the most critical of audiophiles “can normal electronic circuits be microphonic?” Well, somehow it moved into a discussion of THD which he mentioned that humans (and remember, I have only heard this from him before…) cannot tell a difference between anything below 10% total harmonic distortion in terms of reliability of the ear and the brain’s ability to make sense of the information. Instead, he quoted to me that the biggest audible problem of amplifiers was the voltage drop that occurs when hitting transients and peaks in high-wattage output (or really any output level for that matter). Which got me thinking: how would you qualify the difference in waveforms between the output of an amplifier with massive capacitors versus one that suffers from a severe voltage drop and resultant compression on the peaks? Well, that would really be described as nothing other than a form of distortion (which would then fit into the general THD category…hmmmm)

So after a deal of thought, I came to figure that just in the same way our eyes can detect more levels of color than any monitor can even display (the RGB spectrum is of course completely encapsulated by what our eyes perceive outside of it, neon colors for instance…) that our ears (and more importantly, minds) can also be made to be aware of a ridiculously high level of detail in audible changes in sound pressure or waveforms. Technically, any difference in the original waveform and it’s representation after being transmitted through a circuit or transducer can simply be spoken of as a “distortion”. And really, it is these ultra-tiny “distortions” that tell us we are listening to a 10 thousand dollar reference system and not quite still hearing it in real life.

Just something to think about…

I love more knowledge. ^_

There is also the digital factor that no matter how sensitive or bit-depth, it will never reach absolute precision. This is the same reason why vinyl is the closest thing to near perfect. So in attempt to blow my mind, let me blow yours…

When we had 16 color screens, we would use different combinations to make a new color, and zoom out so that it blends as a solid color in our perception. (GIF Encoding does the same thing with dithering.) Now, when we had 8-bit MIDI synths, we would blend two different combinations to make a new tone, and then speed it up (higher octave) to blend as a solid tone. Now if you continue this process all the way up to today’s technology, we’re doing the exact same concept on a never-ending struggle to increase quality. Effects and samples are processed at lower octaves first, DAW ’s and studios take over with algorithms to clean every curve while blending with the next sample. This is the same concept of zooming-in and adding a new pixel color, then on the final product it is zoomed out. Though in real time this takes less than a second.

Still, no matter how high the sample rate, it can only come within 99.99% absolute precision. Same reasoning behind that we can only get to 99.99% of absolute zero temperature, 99.99% color depth, and why 1/INF is theoretical 0, and 1 – 1/INF is 99.99%. Our bounds are between 0 and 1, never equaling.

you are both now, my gods.

famous producer and mix engineer Rob Fabroni would agree with you 100% concerning the vinyl issue…

Sad that some people refuse to consider that it is the whole pulse-code-modulation effect of sudden and instant changes in dynamic level (or bit depth) which likely wreaks havoc with the terrible slew rates of most cheaper amplifiers…one of the reasons that cds (this is my theory of it) only tend to sound close to vinyl when played back on 5 thousand dollar cd players and with amazingly linear op-amps and high-end amplifiers. I would think that theoretically the reason cds sound ‘worse’ than vinyl is the voltage overshoot that occurs when moving from one portion of the dynamic scale to another instantaneously. Because vinyl doesn’t need to produce the instantaneous ramp-ups and ramp-downs that cds contain (naturally, any digital representation will never be anything more than a series of dynamic “steps”), vinyl also likely puts less strain on the necessity of operational amplifier and circuits to create a linear output with no voltage overshoot or drop.

Most people believe it is the whole nyquist theorem that makes them right when arguing that cds sound “exactly the same”. Have they ever considered that it may just be the resultant distortion? I don’t think I’ve seen any articles detailing such a thing which miffs me a little considering how many “engineers” there are in this world.

Interesting info’s, bookmarked. Somehow smells like Newgrounds suddenly… mmmm…