Jordan_J

^ Мерси !

http://soundcloud.com/visiondann/low-life-...gh-places-cover Надяваме се да ви хареса! Поздрави!

От сватбата ни вчера! Наздраве!

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^ Всички снимки са мноооого яки! @Пандемониум - честито!! Познай кой ще направи същото 10 дни след вас! )))))))) @Фен - малката мачка с този ДТ медальон!! Голям е сладур!

^ Благодаря ти за коментара! Наистина има доста за работа по това, но тогава много бързах - буквално го записах за 15 мин - и още 5 мин за микс Трябваше ми да го добавя за една компилация и нямах никакво време ... Всичко е записано "от раз" - няма даже компресии - нито на вокалите, нито на китарата. Като мога ще го презапиша. И мен иначе много ме кефи - още първия път като го чух и адски ми хареса идеята и текста! Поздрави!!

Много гот, ман ! Бравос!

Имах час-два преди работа и реших да запиша нещо през това време. Дано ви хареса ... LANDSLIDE - Acoustic cover ...

Скоро ще има още интересни (надявам се) неща! Благодарим за отзивите!

Жестоко!! Мачка! ))

Странно, но не намерих нищо по темата във форума ... Затова имам намерения да разбуня духовете! Давайте инфо - всеки със своя опит, забележки и впечатления! What is dither? To dither means to add noise to our audio signal. Yes, we add noise on purpose, and it is a good thing. How can adding noise be a good thing??!!! We add noise to make a trade. We trade a little low-level hiss for a big reduction in distortion. It's a good trade, and one that our ears like. The problem The problem results from something Nyquist didn't mention about a real-world implementation—the shortcoming of using a fixed number of bits (16, for instance) to accurately represent our sample points. The technical term for this is "finite wordlength effects". At first blush, 16 bits sounds pretty good—96 dB dynamic range, we're told. And it is pretty good—if you use all of it all of the time. We can't. We don't listen to full-amplitude ("full code") sine waves, for instance. If you adjust the recording to allow for peaks that hit the full sixteen bits, that means much of the music is recorded at a much lower volume—using fewer bits. In fact, if you think about the quietest sine wave you can play back this way, you'll realize it's one bit in amplitude—and therefore plays back as a square wave. Yikes! Talk about distortion. It's easy to see that the lower the signal levels, the higher the relative distortion. Equally disturbing, components smaller than the level of one bit simply won't be recorded at all. This is where dither comes in. If we add a little noise to the recording process... well, first, an analogy... An analogy Try this experiment yourself, right now. Spread your fingers and hold them up a few inches in front of one eye, and close the other. Try to read this text. Your fingers will certainly block portions of the text (the smaller the text, the more you'll be missing), making reading difficult. Wag your hand back and forth (to and fro!) quickly. You'll be able to read all of the text easily. You'll see the blur of your hand in front of the text, but definitely an improvement over what we had before. The blur is analogous to the noise we add in dithering. We trade off a little added noise for a much better picture of what's underneath. Back to audio For audio, dithering is done by adding noise of a level less than the least-significant bit before rounding to 16 bits. The added noise has the effect of spreading the many short-term errors across the audio spectrum as broadband noise. We can make small improvements to this dithering algorithm (such as shaping the noise to areas where it's less objectionable), but the process remains simply one of adding the minimal amount of noise necessary to do the job. An added bonus Besides reducing the distortion of the low-level components, dither let's us hear components below the level of our least-significant bit! How? By jiggling a signal that's not large enough to cause a bit transition on its own, the added noise pushes it over the transition point for an amount statistically proportional to its actual amplitude level. Our ears and brain, skilled at separating such a signal from the background noise, does the rest. Just as we can follow a conversation in a much louder room, we can pull the weak signal out of the noise. Going back to our hand-waving analogy, you can demonstrate this principle for yourself. View a large text character (or an object around you), and view it by looking through a gap between your fingers. Close the gap so that you can see only a portion of the character in any one position. Now jiggle your hand back and forth. Even though you can't see the entire character at any one instant, your brain will average and assemble the different views to put the characters together. It may look fuzzy, but you can easily discern it. When do we need to dither? At its most basic level, dither is required only when reducing the number of bits used to represent a signal. So, an obvious need for dither is when you reduce a 16-bit sound file to eight bits. Instead of truncating or rounding to fit the samples into the reduced word size—creating harmonic and intermodulation distortion—the added dither spreads the error out over time, as broadband noise. But there are less obvious reductions in wordlength happening all the time as you work with digital audio. First, when you record, you are reducing from an essentially unlimited wordlength (an analog signal) to 16 bits. You must dither at this point, but don't bother to check the specs on your equipment—noise in your recording chain typically is more than adequate to perform the dithering! At this point, if you simply played back what you recorded, you wouldn't need to dither again. However, almost any kind of signal processing causes a reduction of bits, and prompts the need to dither. The culprit is multiplication. When you multiply two 16-bit values, you get a 32-bit value. You can't simply discard or round with the extra bits—you must dither. Any for of gain change uses multiplication, you need to dither. This means not only when the volume level of a digital audio track is something other than 100%, but also when you mix multiple tracks together (which generally has an implied level scaling built in). And any form of filtering uses multiplication and requires dithering afterwards. The process of normalizing—adjust a sound file's level so that its peaks are at full level—is also a gain change and requires dithering. In fact, some people normalize a signal after every digital edit they make, mistakenly thinking they are maximizing the signal-to-noise ratio. In fact, they are doing nothing except increasing noise and distortion, since the noise level is "normalized" along with the signal and the signal has to be redithered or suffer more distortion. Don't normalize until you're done processing and wish to adjust the level to full code. Your digital audio editing software should know this and dither automatically when appropriate. One caveat is that dithering does require some computational power itself, so the software is more likely to take shortcuts when doing "real-time" processing as compared to processing a file in a non-real-time manner. So, an applications that presents you with a live on-screen mixer with live effects for real-time control of digital track mixdown is likely to skimp in this area, whereas an application that must complete its process before you can hear the result doesn't need to. Is that the best we can do? If we use high enough resolution, dither becomes unnecessary. For audio, this means 24 bits (or 32-bit floating point). At that point, the dynamic range is such that the least-significant bit is equivalent to the amplitude of noise at the atomic level—no sense going further. Audio digital signal processors usually work at this resolution, so they can do their intermediate calculations without fear of significant errors, and dither only when its time to deliver the result as 16-bit values. (That's OK, since there aren't any 24-bit accurate A/D convertors to record with. We could compute a 24-bit accurate waveform, but there are no 24-bit D/A convertors to play it back on either! Still, a 24-bit system would be great because we could do all the processing and editing we want, then dither only when we want to hear it.)

^ Мерси, ман!

Понеже няколко колеги от форума ме питаха едно и също нещо (общо взето) за бекинг вокалите и това как са записани. По този повод искам да споделя, че при тях няма никакви трикове - всички партии са записани на живо (даже не са компресирани, а само нормализирани). Вокалите са записвани с CAD VSM tube микрофон. Китарата е писана с Меса Триаксис - директно в миксера. Пианото е ВСТ. Поздрави на всички и благодаря за интереса!

Мерси много! Радвам се, че ви е допаднало! Поздрави от нас!

http://soundcloud.com/visiondann/separate-ways-cover Надявам се да ви хареса ...

Итрих ги ман ! Затова не се отварят ... Сметнах темата за приключила и затова ... Сори!

^ В крайна сметка аз се оказвам с 2 трака, записани от два различни микрофона. През същото време един колега казва, че мисли, че чува дефазиране в суровия запис ... Други колеги пък точно суровия запис им харесва най-много ... Въпроса се свежда до това кое е по-добре и как да го оставя (примерно).

Как третия е най-тих? На мен третия ми се струва най-силен - слушам ги алтернативно едно по едно. Сънчо, нямам друга китара в момента при мен в студиото. Ето ги накуп трите варианта: 1. Суров запис 2. С инверсна фаза 3. Суров запис напаснат ръчно

Ето тук без фазова инверсия - само съм напаснал "на ръка" стрймовете и мисля, че стана най-добре и най-дефинирано. Ще ми е интересно вашето мнение! http://soundcloud.com/visiondann/xy-record...se-synchronized

Между другото, точно това ми казаха и други колеги преди минути - че първия вариант им харесва повече. Изобщо какъв е смисъла от инверсия на фазата, ако на практика двата микрофона нямат никакъв бленд, а са си по 100% ляво и дясно?

При мен обаче микрофоните сочат съотвентно бриджа и 12 позиция.

Това определено не е и моя начин на запис на акустични китари, но го направих заради теста и понеже ме попитаха колеги от форума. Това за басите е много относително - аз даже намирам, че с малките диафрагми микса става по-фокусиран, и ако изпълнителя е в правилния спот - то резултата е доста "по-пробивен". В бързината не сметнах за необходимо да обръщам фазата, понеже двата микрофона са с по 100% панорама в ляви и дясно. Но сега обърнах фазата на левия канал и сякаш е по-добре Вие ще кажете БЕЗ инвертирана фаза на левия канал: http://soundcloud.com/visiondann/xy-recording-test СЪС инвертирана фаза на левия канал: http://soundcloud.com/visiondann/xy-record...t-phase-inverse

Понеже няколко колеги от форума се интересуваха - направих един непретенциозен тест за XY запис - няма компресии и еквилайзии. Китарата е евтиното ми Walden-че Разтоянието от микрофоните е около 40-50 см. Единия сочи към бриджа, а другия към 12 позиция някъде Няма никакви други поспроцеси, освен върба за цвят Линка е: http://soundcloud.com/visiondann/xy-recording-test

Митак, днес получих албума! Супер е! Мерси много и успех!!