Violin sound

how to I play just a single violin with garageband? same with trumpet...I can only find sections, not the single instrument.

Roger is correct. That is the best way to get your solo violin (something sadly missing from JP4)
If you want real sounding solo instruments, consider GPO
http://garritan.com/GPO-features.html
And if you want to go flawless solo violin, consider...
http://garritan.com/stradivari.html
Demo
http://garritan.com/Strad/fabio-CapriccioNo1.mp3

Similar Messages

Mainstage 2.1 - EXS sound stop when playback loop start

Hi,
i use with MacBookPro 4(Go) and audio Kontrol (Native instrument)
i have some concert with exs24 sound violin , when i play a note and start a loop in Playback after, the violin sound stop!
do you have an issu?
Thanks

Sure, a quick reply: create a multi audio track song in Logic using markers. Bounce this to AIFF files - one for each instrument say.
Using the MS multi track mixer template, drag the bounced files into playback instances and set the synch option to "wait for marker"
Start MS playback then use the jump back/forth between markers buttons to do that. MS should continue playing untill it reaches the end of the section, then jump. It does but the results are unpredictable. Mostly I get white noise. This used to work under 2.1
This is very well documented in another post and YT vid which I will try to find and append later today.
Here it is: http://discussions.apple.com/thread.jspa?threadID=2372092&tstart=75
Message was edited by: randallp

Jam pack strings poor sound

Hello,
2 questions.
1) Why is it that when I assign the JamPack violin sound to a track another track with the same name appears in the list? I seem to remember reading some where that some of these sounds are stereo and require two tracks though it seems to make no difference to the sound if I delete one of the two.
2) The sound itself is lacking depth and is scratchy. I find it hard to believe that this is the sound that the engineers intended. It seems like I'm only getting part of the sound. It's the same with all the string sounds. Can this be so?
Thanks, Julian

Ah, I see you are on 7. It may well be different. Anyway, I've just installed "Jam Pack Symphony Orchestra" myself, & obviously I'm not getting this behaviour, although that isn't surprising given the differences.
I need to know exactly +how & where+ you installed your Jam Pack constituent parts, since this is a very well-known bone of contention. If everything just went where the installer suggested it should go, then that's not the issue (although it means your boot drive now has loads of 'stuff' on it).
However, if you made different choices, or moved stuff yourself, then I suggest you read this discussion (especially towards the end) & then get back: [Make Room!|http://discussions.apple.com/thread.jspa?threadID=2026399&tstart=0]
You should also know that installing Jam Packs can result in previously-installed material being deleted. There's a warning about that in the installer Read-Me.

File Export menu in CS6 does not work properly?

Two main issues:
1. If I downsample a 96kHz audio file to 44k1 for example I have the option to impose an anti-alias low-pass brickwall filter in the export path or not. This is a good idea. However as implemented in the Audition CS6 system we have, Adobe has the tick-box configured incorrectly. With the box ticked the filter is switched out, and with the box unticked the filter is switched in. This should be debugged and fixed. It makes me worry that other configurations might be wrong.
2. The defaults for the implementation of dither in the export of a file are inconsistent. The default should be that whenever audio data are changed by any processing, dither is applied in the exported file. This is inconsistent in Audition CS6 on our system and we have to go through all the setup for each export.

Thanks Steve.
I agree about the quality of Audition's sample-rate conversion. It is very good indeed and one reason we bought Audition. I also agree about the perverse logic of Audition having a tick in the box to turn the low-pass filter off, rather than on. This is is counter-intuitive, even if the excuse is that it is how it has always been historically in Audition or maybe even CoolEdit (which I have never used). Maybe Adobe could consider retitling this function so the logic is more obvious to newcomers. The riddle of how to combine requirements for 96k/24 high-res masters for one market with the needs for 44k1/16 delivery for CD, downloads and streamers remains a riddle if we are not to make our 44k1/16 masters (the main market) sound inferior to having made them 88k2 or 44k1 originally.
As for dither. My experience has been that one stage of unintended truncation does typically approximately five to ten times as much damage (irreparably) to a clean recording as does one stage of unintended dithering. I am only talking about default settings which can or should be easily overridden either in 'settings' or 'preferences' somewhere within Audition, or overridden on any single or multiple instance of exporting. The mathematics are against you on this one. If you have a 96k/24 original file and want to export a straight 96k16 file or 44k1/24 or 44k1/16 version, for me dither is obligatory. The same if you are exporting a 96k/24 file after processing. Whatever happens next. If you are exporting 96k/32 then maybe, I stress maybe, no dither is ok. Otherwise you get truncation errors, with a shower of unwanted side-effects and squeaks. That is one problem with digits rather than analogue. With digits you don't get extra random relatively benign noise with repeated undithered generations of copying, processing, transfers. The artefacts are much better described as errors and they are a shower of correlated noises. They are programme related, clock related or a combination of both so the ear is much more likely to pick them out as something less benign than two bits of pk-pk flat dither.
I understand the effects may depend on the programme material, but even two bits of pk-pk flat dither of a 16bit stream is typically 16dB to 20dB below the broadband background noise of a classical chamber music recording in a typical real-world venue. With commercial music the dynamic range tends to be compressed, so the dither will be even less perceptiable because there are fewer extended quiet sections of programme. I would prefer the effective noise level to increase by even a dB or so, if the alternative is gradually making a violin sound more and more like breaking glass from the effects of truncation.
We have been doing listening tests on dither in association with the development of new high-res delivery products. It has been interesting that no listeners detected dither as a noise 'problem' during any of the tests - even when we tried adding a lot more than 2bits' worth. In general either one noticed no change or a slight improvement subjectively with more than 2bits' worth of dither. The only negative thing we detected was that adding multiple layers of so-called random dither sometimes brought attention to the fact that it was sometimes not quite as 'random' as it should be.

How to best use Multi-Channel Plug-in: Miroslav Philharmonik

Hello everyone,
I recently purchased the Miroslav Philharmonik sound library and am learning to use it. You can assign different instruments and sounds in up to 16 different channels within the plug-in. I've quickly learned that this is a very important feature to use in order to have, for example, all the different violin articulations (e.g. legato, staccato, pizzicato, detache, etc.) at your disposal at once and within one instance of the plug-in. The only problem is, I'm not really sure how to manage changing MIDI channels within the arrange window.
Let's say I have the 4 previously mentioned violin sounds loaded into Channels 1-4 in the Miroslav plug-in, which I have inserted into the Channel Strip for Inst 1. Let's say I play four bars of legato, then want to switch to staccato for four bars. How do I do this? I know I can switch the MIDI Channel from 1 to 2 in the channel strip info window, but is this automatable for play-back? I have just a basic M-Audio MIDI keyboard with pitch bend, mod wheel, octave +/-, advanced functions button, and volume. I've never worked with a multi-channel plugin before and would love any and all suggestions on how best to use this feature! Thanks!
David

Doesn't the Miroslav Plug In support keyswitches? Using these there would be no need to setup a Midi channel for each articulation as all articulations would reside in the same patch on the same Midi channel and you would just toggle between them by using pitcches which are not within the range of the corresponding Instrument.

Midi loop not sounding same at arrange area

Hi,
when I drag a midi loops (green loops) into the arrange area it will not play the same sound........ mean I'm not getting the same effect of the midi loop at the arrange area & I don't want use these loop on audio track. I know if I drag the midi loops into audio track it will create an audio loop sounds same effect what I'm listening at the Loop browser but I want to use these midi loops in there own midi track.
Please help me guys

Thanks Jorge for getting back to me.
I tried what you said and when I do get the first Capital Violin 2 midi loop and turn it to green. There is a slight staccato sound but not much. When I move the modulation wheel and select the first sound in the piano roll, yes I can hear a dramatic staccato sound. But the minuet I get the play head to the front of the loop and play it from start it is gone again. Quite frustrating. I would love to get that energetic Capita F Violin 2 midi sound but it doesn't have the power or anything like what it sounds in the loop browser.
Any extra tips on how to KEEP the sound after playback would be a great boost to my musical loop understanding.
But thanks for those suggestions!
I am using a 49 E keyboard and it has a Pitch Bend Wheel and a Modulation Wheel to the left of the keys. The volume control on the 48 e is all the way up. That is another problem I have is even when I push the volume up high in the arrangewindow more often than naught the loops don't seem strong like they do in the loop browser.
But again, my main problem is a lack of strong staccato sound.
Bill
Message was edited by: William Furner1

How do i change the play style (e.g on a violin from arco to pizz) in Kontakt while still using the same instrument track, mid way though the song

So if the title is not clear, im really stuck. i got Kontakt 2 and the EWQL sound sets the other day from my uncle and I've been trying them out in logic. My problem is, that in my full orchestral piece, I have a violin 1 which plays pizz at the start. Mid way trhough the song i want it playing arco or legato. I cant do it without physically creating another track to play it on. Is it even possible to change it half way?
Thanks guys
Jonathan

Most keyswitched instruments have that already built into the Kontakt patch. It's probably possible to add that functionality either by editing in Kontakt or doing fancy setup in Logic's environment window, but either of those would be pretty complicated.
If a patch already has those, it should show up in the keyboard view in kontakt (colored keys away from the rest of the instrument range, sometimes a different color).

Recording LPs with smart recorder and sound quality off.....

Hi,
I have X-Fi Extreme music card on my PC and trying to record some LPs with SmartRecorder. These are classical LPs and I find that when played back, the orchestra (violins, choir, etc) sound like this was recorded in a tin can. Any ideas why this would be? I connected all my cables accordingly and?am still?getting this problem.

Which type of connection are you using there?
Do you have preamp stage w/ RIAA there before soundcard (you have connected the PC into your stereo system just the way
like you connect a casette recorder or you have connected your turntable w/ RIAA preamp directly into soundcard) or
have you connected your turntable (w/o RIAA + preamp) directly into soundcard?
If you're using the latter method, you need to have EQ w/ RIAA compensation added into signal (you need some ASIO/VST
capable recording software which allows you add FXs into signal (Hermann Seib's VSTHost) and an EQ capable for
compensation curve +20dB (20Hz) - -20 dB (20 kHz) (ddmf:IIEQ). It's possible that you
can't get levels set high enough w/o adding some gain plug-in into signal path. You can use Sonalksis FreeG VST (stereo)
for this (in VSTHost, place (route) the FreeG either after or before the EQ (try which works better for you)).
Also, check if the recording source is set correctly (w/ "What U Hear" option you get all added effects added
into signal (EAX, CMSS) so it's best to set input as Line-In) and turn all effects OFF.
If you record into lossy format as mp3 or wma, check that the bitrate is set high enough (I suggest you use bitrates
256 - 320). Also, do not use addons like Chrystalizer, etc..
If needed some software, http://www.head-fi.org/forums/showpost.php?p=282263&postcount=7
jutapaMessage Edited by jutapa on 05-08-200704:02 PM

Classical Orchestral Sound: Panning, EQ, and Convolution Filters

Hi,
I'm now studying the sections of the rather large, but quite informative
book, The Guide to MIDI Orchestration, 3rd Edition, by Paul Gilreath
to obtain details about how to use my non-ambient, dry samples of
the orchestra to create a true, classical orchestra sound (not a
hollywood sound), though my music is not 19th-century classical.
I'm also studying the Apple Logic Pro plug-in book concerning the
convolution filter, Space Designer.
Being a beginner, my learning curve is rather slow. But, I am feeling
more and more confident. And, this confidence is boosted by the
fact that I have a 3.0 gHz Mac Pro that can handle, most likely,
the processing required.
I post this thread to see if there are others who might have similar
goals and might have hints or ideas on this topic.
For instance, does anyone out there go through the details of
panning to place the instruments of the orchestra in their
correct locations?
Does anyone lower the EQ of the percussion section because they
are placed further back in the orchestra?
Does anyone deal with the details of singling out a particular section
of the orchestra and consider "delaying" the sound, such as the
percussion section, or perhaps the French Horns? If so, how do you
achieve this? Do you use a plug-in for each section, or single out
some sections for a plug-in, and plug every other section into another
plug-in?
So, if you all have any experience in these matters, or some "best settings"
for some of these panning, EQ, and convolution filters (or even simple
reverb plug-ins), I'd love to read about your experiences while I
continue to study my books.
Thanks,
Sys6
P.S.
My orchestra is almost all non-ambient samples. Garritan Orchestral
Strings. Westgate Studios Advanced Woodwind Collection. And,
a new dry, brass collection which has not yet been delivered. My
percussion is Apple Garage Band percussion (which I'm assuming is
relatively dry).

Hi,
Here are some details of my set-up. I continued to read the MIDI
Orchestration book, and the Logic Pro manuals where applicable.
I set everything up, and then I visited the above links to read them
in more detail, to see how they solved the problem. First, I don't pretend
to have "solved the problem", but this is my first start.
Following the MIDI orchestration book, I discovered that my Garritan
Orchestral Strings were in Stereo, so I used them in stereo, specifically
using the insert, Direction Mixer.
Violins I and II, Violas, have a Direction Mixer spread of 1.3. Following
the book, the Celli and Bassi were given a Direction Mixer spread that
was wider, and I chose 1.5.
I made all the brass monophonic, and used the pan pot on the bus.
My Westgate Studios Advanced Woodwind Collection is mono, so
again I used the pan pot on the bus.
The percussion samples (piano, harp, timpani, and everyone else)
are stereo, so I used dirmix also.
For the most part, each section of the orchestra, such as either
Violins I or oboes (and English Horn), each feed into their own
bus. So, every section can be panned using the pan pod if
they are mono, or using dirmix otherwise.
Every section of the strings feeds into another bus dedicated to strings.
Every brass section feeds into a similar bus dedicated to brass.
The same for woodwinds, but there are two buses: one for the
front woodwinds (flute family and oboe family) and one for the
woodwinds sitting behind them (clarinet family and bassoon family).
For some of the percussion: timpani and other miscellaneous
percussion, for now, these go to their own bus. Also, piano
and harp go to a pianoAndHarp bus. Tubular Bells,
Marimba, and Xylophone share a bus. All these sub buses use
dirmix for stereophonic spread and panning, all all these sub buses
feed into a master percussion bus.
Everyone audio instrument has its individual volume level set to
-15 db. I may go over every instrument again and make it even
lower, such as -20 db.
There are six master buses (nearly at the top of the food chain):
1. String Bus.
2. Woodwind Bus Front
3. Woodwind Bus Back
4. Piano Harp Bus
5. Brass Bus
6. Percussion Bus
Each of these higher-level buses received one insert: Space Designer
(a convolution reverb unit).
And, for each, Space Designer (at least at this time), is practically
identical: that is, I chose
33 Misc. Reverbs > Misc. Reverbs - Short Dark > 1.5s_Dark & Small
which I like (for now, for chords anyway; later I may need a smaller
room for music with notes moving at a faster clip and for music
that is more chromatic).
There is one parameter that differs for each of these six Space
Designer inserts:
1. String Bus: Pre-Delay 0 ms
2. Woodwind Bus Front: Pre-Delay 1 ms
3. Woodwind Bus Back: Pre-Delay 2 ms
4. Piano Harp Bus: Pre-Delay 3 ms
5. Brass Bus: Pre-Delay 4 ms
6. Percussion Bus: Pre-Delay 5 ms
Other parameters may differ over time. For instance, space designer
seems pretty clever, it can add a pre-delay, it looks like it can
act as a low pass filter, and so forth. So, maybe, over time, I might
use it as a low pass filter for the percussion section, not sure yet
(hard to tell, since my percussion section is not very good, it is
Apple Jam Pack Orchestra).
All these six busses feed into a final bus, which at this time
does nothing, but pass everything on to Output 1-2.
I focused my musical chords to be very soft or very loud in
the score editor.
That is, the goal is not to have the final mix exceed 0 db.
Unfortunately, I failed in this regard, for in a rather loud
section (at the end of the above sound file), the chimes played
loudly brought me over the 0 db level. So, for now, I turned
down the bus that these chimes feed into. There are other
more advanced solutions, but for now, I am focusing on
simple solutions (though I am always happy to hear about
advanced solutions which I may implement at a later date).
Over time, of course, I may adjust pannings, stereo pan width,
and the like. But, being a beginning "sound recording engineer,"
I am impressed with the results! Wow!
As an aside, each of my 3.0 gHz processors is being used;
two are used for effects processing (such as for the 6 Space
Designer plug-ins), and two are being used for audio processing.
No disk activity occurs (I use the latest Lacie external drive, the
biggest I could find, and perhaps it is buffering enough files
so that the drives are never accessed?).
Thanks again, if there are other details I have left out, I'll
update.
Sys6

Tips for making strings sound more real

Basically i'm having trouble getting my strings from the exsp24 to sound real. mainly the violins. does anyone have any good tips?

It very much depends on what sort of music you are writing and what you want the strings to achieve.
As well as having decent sounding samples, you must make sure that what you are writing are typical string parts. You can have the most wonderful string samples in the world, but if you play "Chopsticks" with two fingers they'l still sound like some chimp playing "Chopsticks" with two fingers.
Broadly speaking, chords should be structured with wider spacing between the bottom notes than the top. Although complex classical music can have chords made up of 8 or more notes, pop music tends to have much simpler writing with perhaps only 4 parts or fewer.
Dynamics are also very important. Listen to any piece of classical string music and you'll notice there is a very wide dynamic range within phrases, and even within the length of a single note.
In general, the sort of sound that comes with Logic are good for long legato lines, or sustained block chords, or perhaps short stabbing sounds (think "Psycho). Because the violin is a fretless, fluid instrument, parts that require more sophisticated articulations will require a more up-scale library that features these articulations, and even then the results are not always great.
Mac OS X (10.4.8)

5.1 Digital Surround Sound Output

So what I need is to learn if AA3 has the ability for me to play only those First Violins (Rear Left), Second Violins, Piccolo, Oboe, Clarinet (Front Left), Brass Section & Cymbals (Front Right), Woodwind section (Center), Pipe Organ 16' bass, Timpani & Percussion etc (Sub-woofer), Double Bass (right rear), Violas, Keyboards, Organs, etc (Front Right).
Does anyone know how I can do this?
I need to experience what an Orchestra Conductor hears by digitizing through Multi-Track the ability to hear what he hears, but extended almost 360 degrees. My source material is mono-aural John Phillip Souza's marches recorded in the 1960s through Reader's Digest, who have given their permission for this type of work to be done.
I want to have the resulting tracks using MixDown back to 2 channel stereo in .WAV format but in 5.1 digital output in MP3 or MP4 format for all our cellphones that have MP3 players. If I can come up with my own arrangements of my own compositions, or my own arrangements of others' compositions that I have helped to "orchestrate", then this is the sound I need. Anyone with a digital home theatre system will appreciate the forethought gone into creating something of this magnitude.

Hi Ian,
this system has the design feature of discarding standard usenet quotes and
anything thereafter, but not writing usenet style would be confusing to any
other user of the miserable, bodged, offline feature of this here
re-invented messaging system. Consequently I'll try to beat the design issue
with a different quoting prefix ... just in case I'll re-iterate the main
point made up here: you can not uncook a stew.

How to make this synth sound?

I have a sample audio (a crappy song, but its the only one which had the tone I needed) - https://www.youtube.com/watch?v=4_pioLA47AU#t=20 I want to know how to synthesize the sweeping oscillating sort of tone that goes on from 0:20s to 0:24s. Could anybody help me?

Well, I might as well answer my own question.
I just bought NI Massive yesterday as SmokinSound suggested because I found a complete list online of what softsynths the artist uses to create his sounds and NI Massive was on it, but Nexus wasn't. The other softsynths didn't seem like they could make pad sounds like this one because of their names.
I mucked around with some of the presets and after an hour or two of doing that I got the synth so I'm pretty sure it was made with NI Massive in Logic pro. I used the "Dusty Violin" synth pad preset and turned the reverb and chorus knobs down a bit, raised the resonance knob, changed the attack and release, turned the Delay knob down, and all that.
The only thing throwing me off quite a bit is the autofilter. I still can't completely figure that one out so I posted a new question on how to set up the autofilter to go like that and posted the link to the video again.
Thanks you guys for the help. I really appreciate it. :-)

Recording multi-track violin

Hi, I've been laying down a production using Logic Pro 7 in which I have recorded singly, three viola tracks, and six violin tracks through a Rode NT5 condenser mic, via an M-Audio 1814 Firewire into a PowerMac G5 2.3 with 4.5 Gb of Ram.
Each track is recorded at as high an input as possible without more than a very little clipping of the signal. Each track, singularly, sounds fine with no signal deterioration where the instruments are played loud. However, when I put the 'nine' tracks together, I tend to get a lot of signal overload and break-up.
What's causing this? Do I need more Ram? Is it the hardware I'm using? The problem is worsened when I use any EQ or Limiter.
Anyone had this problem before? If so, I'd appreciate some help.
Thanks
Terry, Ohio USA

Terry,
What you're really talking about is considered, in part, "mastering". There is a ton of information here, as well as other forums, as to how to best go about that, as far as attempting at doing it yourself.
But yes, setting the overall level of each track in a CD, is usually handled in mastering software, such as Waveburner.
FYI, that is the very short explanation... there's actually a lot more to this, but that should get you started.

How do you get instruments into garageband i.e. violin, sax etc.

how do you get instruments into garageband i.e. violin, sax etc.

You can install additional sound fonts, see GarageBand '11: Play Soundfonts in GarageBand
or install Jam Packs: If you buy (for a moderate fee) Mainstage 2 at the AppStore, this will install all five JamPacks, so that the loops and instruments can be used with GarageBand. This way you will get all Symphonie orchestra instruments, World Music, Rhythm Section and many more.

Mp3 sounds bad on iphone

I am new to iTunes and iPhone 4.
I have an existing collection of cd's, that I did copy into MP3, while using 320kbits sampling rate (I only remember the number).
If I listen to them, I can hear that the quality is a bit lessor then the original. But, I can live with it, as I will never listen to the MP3 on my hifi set.
I did drag the MP3 out of explorer, into a iTunes folder, which seems to be coupled to my iPhone 4. That resulted in creating, more then one album per cd, but I found out how to repair that (lot of work).
Then I started to listen on the iPhone with the apple headphone set.
My first thought was, pretty clear, but... I miss something.
I could not put a finger on it. So I plugged in my old Sony headset.
It sounded better, but I was still missing things.
Then I heard it.
It sounded like the high tones and basses where cutoff.
And there where the music should be very rich in detail, it sounded cluttered up.
There where a violin stops playing, you normally here it fade out to 0, but now it stopped half way the fading process.
Then I tried to listen through the PC speakers.
I loaded iTunes and what seems to be, loading music from the iPhone and played via the speakers. Meanwhile I compared it, by listening to the original MP3, while using Windows Media Player 11.
I could hear a huge difference. The music via Media Player was more open, No high tones or basses cutoff, no cluttered, but detail, and fading out, did not cutoff either.
Via iTunes, I heard exactly, what I heard on my iPhone.
Cutoff high and basses, no detail, all cluttered up, and cutoff fade outs.
When I look under Info, it says that the MP3 was created at 320kbs.
My question is, why has an MP3 get such a low quality on my iPhone?
BTW, I did not change any settings in iTunes.

I have just tried something out:
I used iTunes. Made several MP3 with 320kbps setting, and the same numbers and settings for AAC.
What I heard was, that the MP3 sounds a bit muffled when played in windows media player and iTunes.
The AAC sounded a bit better then the MP3 with Windows Media player, but it sounded high pitched in itunes.
I could here more detail with the cd.
I listened to the Zauberflöte.
Very odd, when put on the iPhone, I only see the numbers once. but which one?

Violin sound

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