The users that do want their music mixed or mastered by Aplus Audio Mastering Services should consider to use AAMS first. When you are in need of Aplus Audio Mastering Services by our Professionals, you can read about our excellent audio mixing and mastering services further here.
Aplus Audio Mastering Services
Welcome to APlus Audio Mastering Services!
High quality audio mastering services with low prices.
The users that do want their music mixed or mastered with more needs then the AAMS program supplies directly, or users that want their mastering done by a human ear or just want more equipment or mastering tricks to be used, can visit our mastering services. But also be sure to try out AAMS V4 first and only when your needs are not met, maybe consider your masterings to be done with the human factor in place, and you can supply us with the information about you want your sound to be like. You can consider mastering your audio by Aplus! When you are in need of Aplus Audio Mastering Services by our Professionals, you can read about our excellent audio mixing and mastering services further here.
AAMS Auto Audio Mastering System For windows users we provide AAMS complete audio mastering software package for Audio Mastering, free of use. Process your Mix to a commercial great sounding Master.
Process your Mix to a commercial great sounding Master. With AAMS you get a clean transfer of your mix to a good sounding master. AAMS will do this lifting up the mix, but trying to preserve the same sound. Therefore when your mix has a particular sound, AAMS will not hurt it but will only try to compare source and reference and apply that with mastering techniques available. That is great for most users who do their own mixing and creating the sound is mostly done at the mixing stage. AAMS improves the mix towards quality commercial levels for al kinds of musical styles.
Audio Mastering
Mastering is audio post production, is the process of preparing and transferring recorded audio containing the final mix towards a data storage device the master. From which all copies will be produced via methods such as pressing, duplication or replication. Mastering is a crucial gateway between production and consumption and. It involves technical knowledge as well as specific aesthetics.Results still depend upon the accuracy of speaker monitors and the listening environment. Mastering engineers may also need to apply corrective equalization and dynamic compression in order to optimise sound translation on all playback systems. It is standard practice to make a copy of a master recording, known as a safety copy, in case the master is lost, damaged or stolen.
Welcome to APlus Audio Mastering Services!
High quality audio mastering services with low prices.
We maximize your audio material to clear crisp audio mastered versions. Giving detail, clarity, definition, warm low end punch, stereo depth and prisitine sound using Analog and Digital equipment. Audio Mastering is most important part of the audio chain and we give your sound high definition and clarity. Every single track and every song of your album collection wil sound complete next to each other, with taking care from start to end. Ready for distribution and radio. That is what Aplus Audio Mastering Services stands for, excellence! Your audio material or mixes will become an adequate to commercial radio, CD or MP4/MP3 streaming services, just to fit in correctly. We do not attend the Loudness War, you need appropiate levels and professional quality!
Aplus Mastering will make single tracks or songs stand out in excellence.
Aplus Mastering will make your whole Album Sound perfect and give it togetherness.
Aplus Mastering will create a professional sound for all of you.
We make your music shine!
Supply us with your audio material, mixed down to stereo in one of the formats below.
We prefer to accept the following formats:
Sending or submit your mixes to Aplus Mastering over the Internet or post, from yours to us and get it back in no time.
Upload your final mix with our easy to use upload system.
Goto the 'Shop' and decide what option you need :
- For single audio track mastering, choose Single Track.
- For multiple tracks that need mastered as single tracks, choose Multiple Tracks.
- Finally when you have multiple tracks that need to be sounding as an album, choose Full Album.
Instructions will follow afther you have chosen.
On your own User Account page, you can sign up for your account. Or go directly to our Shop and chose. Then Upload your songs / tracks with our easy webbased upload system. We will send you back a mastered sample of your song, if you do not pay upfront. When you have paid and accept the our high quality masterings when we are finished with your content, we will send you an email and place all mastered song / tracks on your personal w web account. And you can download the finished product from your own user directory!
There are a number of audio mixing and editting tips that will help you prepare your mixes before submitting to Aplus Audio Mastering studio.
It is important to know how to prepare your mix, so you can get the best sound for your songs!
When quality is at stake, be sure to read this page and spend some time to get your mixes right.
Audio mastering is a process that stands far from mixing, it is the next stage afther mixing and it is the final stage for sound quality. Actually while mixing we do not attend the loudness much, we mix. What everybody is thinking of 'How to get our mix sound loud'! That is what Aplus Audio Mastering stands for, most likely preferred that your mix will become an adequate to commercial radio, CD or MP3 streaming levels, just to fit in correctly. We do not attend the Loudness War, but we need appropiate levels and professional quality. Also when Mastering a Full Album, Aplus Mastering will make the whole Album Sound as an Album. We name it 'the album sound'. So Aplus Mastering can do single tracks as well as full albums, and create a good quality professional sound for you. But however, mixing is an important stage before Aplus Mastering can be done. So we ask you to attend some time and thought before sending your mixes to Aplus Mastering Engineers.
Check, Check, Double Check!
0. You should do these mix check steps before you plan to hand your project to our Aplus Mastering Engineer.
1. Eliminate any noise or pops that may be in each single track. Apply fades or cuts or mutes to spots containing recorded noise, pops or clicks.
2. Keep Your Mix Clean And Dynamic. Unless there is a specific sound you need, do not put compressing or processing on the master out of the mixing bus. It is best to keep the master buss free of outboard processing or plugins. Dont add any processing to the overall mix, just to individual channels.There should never be a limiter or loudness maximiser set on the master out mix bus!
3. The loudest part in a mix should peak at no more that -3db on the master bus, leaving headroom. It does not matter How Loud your mix sounds at this time, mixing means mixing.
4. Does your mix Work In Mono? As a final reality check, switch the master buss output to mono and make sure that there is no weakening or thinning out of the sound. In any event, do not forget to switch the bussing back to stereo afther this check.
5. Only when a mix is completed and finished off, and your are happy with the overall mixing sound and quality, then the next fase is Aplus Mastering to do their work.
6. Normalising a track is not necessarily a good idea.
7. Dont add any fades or crossfades, anywhere. Dont fade beginning or end.
8. Do not dither individual mixes.
9. You can output the mix on a stereo track before sending to Aplus Mastering, save your mix in Stereo. Use a lossless format! Using digital equipment Wav 32Bit Float Stereo is a good output format.
10. Do not try to output your mix to a mp3 file, this can mean loss of information! If you do want to send in MP3 files, be sure they are of quality, prefer a bitrate higher than > 192kbps, 320kbps is quite good.
11. Export your mix out of your sequencer or audio setup in a correct and quality unharming format;
12. Finally, always back up your original mixed files! If the song is later remastered for any reason for a re-release, a compilation or for use in any other context you will want a mix thats as easy to remaster as possible.
13. Submitting reference tracks or example songs alongside your mix submission that have a similar sound desired is for a good point of view how your music must sound. Giving Aplus Audio Mastering Engineers an idea of your musical vision. This could be a reference to bands who inspire you or have a similar sound that you like.
14. Put all your files of a single mix (the stereo file, reference songs, text documents or pictures or any file that you need to send) in one single directory.
15. Use a packing program like ZIP, RAR, 7z and pack all files in that directory to one single packed file. Name this file correctly, preferably the track number and name of the track.
16. Now you can send your mix files to Aplus Audio Mastering Services!
Mastering Stems
Mastering from stems is becoming little by little more common practice. This is where the mix is consolidated into a number of stereo stems subgroups to be submitted individually. Instead of submitting a Stereo output of your mix, you can send the mixing tracks seperately. For example you might have different tracks for Drums, Bass, Keys, Guitars, Vocal, and Background Vocals. This will give Aplus Mastering more control over the mix and master. If a master from stems is desired, following the same steps listed above is best for each stem. When submitting stems each file track must start at the beginning and must durate though the end, most mixing sequencers will output this way exactly to the sample. Each stem file should be exactly the same length.
We master all Musical Styles :
- Accoustic
- Amine
- Blues
- Commercials
- Classical Music
- Chrildrens Music
- Holiday, Christmass
- Conferences
- Country Music
- Disco
- Dubstep
- Easy Listening
- Electronic Music
- Fusion
- Folk Music
- Funk
- Gospel, Inspirational
- Hardrock
- Heavy Metal
- House, EDM, Electro, Trance, etc.
- Industrial
- Instrumental
- Karaoke
- Live Performances
- New Wave, New Age
- Rap
- Opera
- Pop Music
- Reggae
- R&B
- Rock
- Singer Songwriter
- Soundtrack
- Soul
- Latin, Spanish
- Trance Music
- TV Themes
- Vocal
- World Music
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The users that do want their music mixed or mastered by Aplus Audio Mastering Services should consider to use AAMS first. When you are in need of Aplus Audio Mastering Services by our Professionals, you can read about our excellent audio mixing and mastering services further here. Aplus Audio Mastering Services Welcome to APlus Audio Mastering Services! High quality audio mastering services with low prices. The users that do want their music mixed or mastered with more needs then the AAMS program supplies directly, or users that want their mastering done by a human ear or just want more equipment or mastering tricks to…
{tab Audio Mastering} Welcome to AAMS! The introduction towards Automatic Audio Mastering!Get your music to sound good without a hassle with AAMS. As a musician or engineer working on music, you need the best sound possible when releasing material to the public and to create a sound that is working for all audio speaker systems is a difficult and time consuming task. AAMS is windows software for Fully Automatic Audio Mastering. To process your Mix to a commercial great sounding Master! Featuring 100 Band Equalizer, 8 Multiband Compression, Balancing and Loudness settings for internal DSP Processing with all audio corrections automaticly done. A Reference Database of200 Musical styles. Creating your own personal…
{tab Audio Formats} Audio Formats The list below showcases audio formats that are able to encode audio and compress it in a lossless way ensuring your music is perfectly preserved in digital form. WAV (WAVeform Audio Format) The main format for AAMS is WAV format audio files, basically internally AAMS is build with it´s 16 bit 32 Bit audio drivers and 64 Bit internal processing, so WAV format is very compatible.And musically 16 bit wav is great for normal audio and will achive good sound, the 32 bit wav floating point is really exact and is the main format for…
Welcome to the infomation page about Mixing Music The fine art of mixing single audio tracks together as a whole is difficult, specially when you do not have some guidelines. First rule for explaining tthe name 'mixing' is that it stands for mixing it al up together, to make a whole overall sound. This means adjusting overal sound levels and making use of Fader Levels, Panning, EQ, Compression, Reverb, Delay or any kind of effect towards a good balanced track. Several issues come up while mixing, technique and equipment. Also offcourse like in composing, improvisation and goofing around might help…
{tab Composing} Welcome to the information page about Composing Music. Composing Music is a first on the list of things to do, or learn. Although modern music allows crossing the borders between Composing, Recording, Mixing and Mastering. Composing is a good starting point when you newly play an instrument or you are just starting to get interested in music making. On the upside there is a lot information around and known about chords, notes, scales and structure. The downside is that this information is almost all reading and understanding, it can be experienced as uplifting or some find it very boring.…
Website Links www.curioza.com AAMS Auto Audio Mastering System. As a musician or engineer working on music, you need the best sound possible when releasing material to the public. To audio master a mix towards a professional commercial quality recording and to create a sound for all audio speaker systems is a difficult and time consuming task. Providing a windows free software package that supplies you with Full Automatic Audio Mastering. Featuring automatic Equalizer, Multiband Compression and Loudness settings, internal DSP Processing with all audio corrections done purely inside the AAMS Program. Making the Mastering Process easy and less time consuming, having…
Audio Matering Services and AAMS Auto Audio Mastering System compared! Web Audio Mastering Services – Is It Any Good? Are there other free mastering options? It’s been around for some time and is extending it’s services. I put it to the test. You’re on a budget. A tight budget perhaps or even no budget at all. You’ve got a recording you want the world to know about and you need to have it mastered (yes, you do need mastering!). And you want it done good. Web Audio Mastering Services is here to the rescue. Aren´t they? I’ve read many comments…
The Origins of House Music Like it or not, house is first and foremost a direct descendant of disco. Even Farley couldn’t deny that the foundation for Chicago house was New York underground disco. After all, Frankie Knuckles, the acknowledged “godfather” of Chicago house, got his start in Manhattan, where he was spinning records in the early ‘70s with another legendary deejay, the late Larry Levan. AAMS Auto Audio Mastering System Disco had already been going for ten years when the first electronic drum tracks began to appear out of Chicago. Dilution and racial and sexual prejudice which culminated in…
What is Dub, what is Reggae ? The remixed versions of a recording, often with vocals partly or entirely removed, and usually with it's main emphasis on the bass and drum tracks (rhythm). Most reggae singles since the early 1970's have included dub versions on their flip sides. AAMS Auto Audio Mastering System .Apart from reggae, dub is often used as a "catch-all" word to mean a recording where the mixing engineers' skills play an important role in the final product, and therefor, making the word "dub" virtually meaningless. Haile Selassie (Hayle Selasse), "common" name Tafari Makonnen (or Tafari Makwannen)…
AAMS Auto Audio Mastering System Contact us for Information!Questions, Reviews, AAMS Professional License Registration Reguests (Username, Installcode), ask for a Keycode and invoice for registration or Questions about Music, Mixing and Mastering.
AAMS How to Register for the Professional Version
AAMS Auto Audio Mastering System V4
AAMS V4.x is freeware to Download, with high encouragement to Register AAMS V4 Professional Version.
Buy AAMS V4 Professional Version!
AAMS V4 Professional Version direct pay and download!
AAMS V4 Professional Version direct pay and download!
Registration ensures users to have all functions and options opened, having full control!
The price of AAMS V4 Registered (Pro) is 65 Euro or about 75 Dollars.
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AAMS Auto Audio Mastering System
The license and keycode are for all versions of AAMS V4 and upcoming V4.x versions.
User Registration is needed for administration purposes only and offcourse to open all professional features of AAMS Software.
We do not use your user information for other purposes but to keep track of the license system, read our license agreement.
A single registration license grants you acces to all professional functions with a single AAMS V4.x version installed on one single computer you retrieved the installcode from.
So be sure you have AAMS software installed on the computer you need the License for, wise the given Keycode will only work for that computer.
Just understand when you buy for the first time a registration license and pay 65 Euro's for a AAMS V4 single computer licence, you are a registred and licensed user.
And when you send in the installcode, you will get an email with the corresponding keycode.
With this AAMS V4 registration as a user, you can register each extra copy on another computer of AAMS V4 software later on at a half price discount.
For AAMS V1 or AAMS V2 users there is a special Upgrade half price discount available towards all AAMS V4.x versions.
Please allow a maximum of 48 hours for us do our adminstration and send you the correct Keycode back.
To get send a invoice or have any questions, you can send an email or use the AAMS Contact Form below this website.
If you want to install AAMS V4.x version to another computer, you will get a different installcode.
Therefore the combination of installcode and keycodes given, are unique!
Each computer you install AAMS needs a seperate Full Registration License applied.
Therefore you can register a license for AAMS V4.x version for each single computer and it's installcode / keycode.
Every other computer (you have 2 or more computers) as a registered user there is a half price discount.
Because as a registred user can have one or more licenses at cheaper rates, but not the first license.
For AAMS V1 or AAMS V2 users there is a special Upgrade half price discount available towards all AAMS V4.x versions.
Use our contact form for any keycode or license questions.
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Audio Formats
The list below showcases audio formats that are able to encode audio and compress it in a lossless way ensuring your music is perfectly preserved in digital form.
WAV (WAVeform Audio Format)
The main format for AAMS is WAV format audio files, basically internally AAMS is build with it´s 16 bit 32 Bit audio drivers and 64 Bit internal processing, so WAV format is very compatible.
And musically 16 bit wav is great for normal audio and will achive good sound, the 32 bit wav floating point is really exact and is the main format for more demanding users.
For audio files to be written in a lossless fashion, wav is a good choise, wav can handle also different types of samplerates, like 44.1 Khz or higher.
The WAV format isn't thought of as the ideal choice when choosing a digital audio system for preserving your audio CDs, but still remains a lossless option. However, the files produced will be larger than the other formats in this article because there isn't any compression involved. That said, if storage space isn't an issue then the WAV format has some clear advantages. It has widespread support with both hardware and software. Much lower CPU processing time is required when converting to other formats because WAV files are already uncompressed -- they don't need to be uncompressed before conversion. You can also directly manipulating WAV files (using audio editing software for instance) without having to wait for a de-compression/re-compression cycle in order to update your changes. Short for WAVeform Audio Format, it is normally used in an uncompressed format on the Microsoft Windows platform. This raw audio format, which was developed jointly by IBM and Microsoft, stores audio data in blocks. On the digital music scene, its usefulness has diminished over time with the development of better lossless audio formats, such as FLAC and Apple lossless. It is a standard that will probably be used for some time yet due to its widespread use in professional music recording and is still a very popular format for audio/video applications. The file extension associated with WAV is: .WAV
FLAC (Free Lossless Audio Codec)
The FLAC format (short for Free Lossless Audio Codec) is probably the most popular lossless encoding system which is becoming more widely supported on hardware devices such as MP3 players, smartphones, tablets, and home entertainment systems. It is developed by the non-profit Xiph.Org Foundation and is also open source. Music stored in this format is typically reduced between 30 to 50% of its original size. Common routes to rip audio CDs to FLAC include software media players (like Winamp for Windows) or dedicated utilities Max for example is a good one for Mac OS X.
ALAC (Apple Lossless Audio Codec)
Apple initially developed their ALAC format as a proprietary project, but since 2011 has made it open source. Audio is encoded using a lossless algorithm which is stored in an MP4 container. Incidentally, ALAC files have the same .m4a file extension as AAC, so this naming convention can lead to confusion. ALAC isn't as popular as FLAC, but could be the ideal choice if your preferred software media player is iTunes and you use Apple hardware such as the iPhone, iPod, iPad, etc.
Monkey's Audio
The Monkey's Audio format isn't as well supported as other competing lossless systems such as FLAC and ALAC, but on average has better compression resulting in smaller file sizes. It isn't an open source project, but is still free to use. Files that are encoded in the Monkey's Audio format have the humorous .ape extension! Methods used to rip CDs to Ape files include: downloading the Windows program from the official Monkey's Audio website, or using standalone CD ripping software that outputs to this format. Even though most software media players don't have out-of-the-box support for playing files in the Monkey's Audio format, there is a good selection of plug-ins now available for: Windows Media Player, Foobar2000, Winamp, Media Player Classic, and others. More »
WMA Lossless (Windows Media Audio Lossless) WMA Lossless which is developed by Microsoft is a propriety format that can be used to rip your original music CDs without any loss of audio definition. Depending on various factors, a typical audio CD will be compressed between 206 - 411 MB using a spread of bit rates in the range of 470 - 940 kbps. The resultant file that is produced confusingly has the .WMA extension which is identical to files that are also in the standard (lossy) WMA format. WMA Lossless is probably the least well supported of the formats in this toplist, but could still be the one you choose especially if you use Windows Media Player and have a hardware device that supports it such as a Windows phone for example.
Uncompressed Formats
WAV and AIFF: Both WAV and AIFF are uncompressed formats, which means they are exact copies of the original source audio. The two formats are essentially the same quality; they just store the data a bit differently. AIFF is made by Apple, so you may see it a bit more often in Apple products, but WAV is pretty much universal. However, since they're uncompressed, they take up a lot of unnecessary space. Unless you're editing the audio, you don't need to store the audio in these formats.
Lossless Formats
FLAC: The Free Lossless Audio Codec (FLAC) is the most popular lossless format, making it a good choice if you want to store your music in lossless. Unlike WAV and AIFF, it's been compressed, so it takes up a lot less space. However, it's still a lossless format, which means the audio quality is still the same as the original source, so it's much better for listening than WAV and AIFF. It's also free and open source, which is handy if you're into that sort of thing.
Apple Lossless: Also known as ALAC, Apple Lossless is similar to FLAC. It's a compressed lossless file, although it's made by Apple. Its compression isn't quite as efficient as FLAC, so your files may be a bit bigger, but it's fully supported by iTunes and iOS (while FLAC is not). Thus, you'd want to use this if you use iTunes and iOS as your primary music listening software.
Lossy Formats
APE: APE is a very highly compressed lossless file, meaning you'll get the most space savings. Its audio quality is the same as FLAC, ALAC, and other lossless files, but it isn't compatible with nearly as many players. They also work your processor harder to decode, since they're so highly compressed. Generally, I wouldn't recommend using this unless you're very starved for space and have a player that supports it.
For regular listening, it's more likely that you'll be using a lossy format. They save a ton of space, leaving you with more room for songs on your portable player, and—if they're high enough bitrate—they'll be indistinguishable from the original source. Here are the formats you'll probably run into:
MP3: MPEG Audio Layer III, or MP3 for short, is the most common lossy format around. So much so that it's become synonymous with downloaded music. MP3 isn't the most efficient format of them all, but its definitely the most well-supported, making it our #1 choice for lossy audio. You really can't go wrong with MP3.
AAC: Advanced Audio Coding, also known as AAC, is similar to MP3, although it's a bit more efficient. That means that you can have files that take up less space, but with the same sound quality as MP3. And, with Apple's iTunes making AAC so popular, it's almost as widely compatible with MP3. I've only ever had one device that couldn't play AACs properly, and that was a few years ago, so it's pretty hard to go wrong with AAC either.
Ogg Vorbis: The Vorbis format, often known as Ogg Vorbis due to its use of the Ogg container, is a free and open source alternative to MP3 and AAC. Its main draw is that it isn't restricted by patents, but that doesn't affect you as a user—in fact, despite its open nature and similar quality, it's much less popular than MP3 and AAC, meaning fewer players are going to support it. As such, we don't really recommend it unless you feel very strongly about open source.
WMA: Windows Media Audio is Microsoft's own proprietary format, similar to MP3 or AAC. It doesn't really offer any advantages over the other formats, and it's also not as well supported. There's very little reason to rip your CDs into this format.
Alfabetical Order
3gp = multimedia container format can contain proprietary formats as AMR, AMR-WB or AMR-WB+, but also some open formats.
act = ACT is a lossy ADPCM 8 kbit/s compressed audio format recorded by most Chinese MP3 and MP4 players with a recording function, and voice recorders.
aiff = Apple standard audio file format used by Apple. It could be considered the Apple equivalent of wav.
aac the Advanced Audio Coding format is based on the MPEG-2 and MPEG-4 standards. Are usually ADTS or ADIF containers amr AMR-NB audio, used primarily for speech.
au = Sun Microsystems the standard audio file format used by Sun, Unix and Java. The audio in au files can be PCM or compressed with the µ-law, a-law or G729 codecs.
awb = AMR-WB audio, used primarily for speech, same as the ITU-T's G.722.2 specification.
dct = NCH Software. A variable codec format designed for dictation. It has dictation header information and can be encrypted (as may be required by medical confidentiality laws). A proprietary format of NCH Software.
dss = Olympus. Files are an Olympus proprietary format. It is a fairly old and poor codec. Gsm or mp3 are generally preferred where the recorder allows. It allows additional data to be held in the file header.
dvf = A Sony proprietary format for compressed voice files; commonly used by Sony dictation recorders.
flac = File format for the Free Lossless Audio Codec, a lossless compression codec.
gsm = Designed for telephony use in Europe. Is a very practical format for telephone quality voice. It makes a good compromise between file size and quality. Note that wav files can also be encoded with the gsm codec.
iklax = An iKlax Media proprietary format, the iKlax format is a multi-track digital audio format allowing various actions on musical data, for instance on mixing and volumes arrangements.
ivs = 3D Solar UK Ltd. A proprietary version with Digital Rights Management developed by 3D Solar UK Ltd for use in music downloaded from their Tronme Music Store and interactive music and video player.
m4a = An audio only MPEG4 file. Used by Apple for unprotected music downloaded from their iTunes Music Store. Audio within the m4a file is typically encoded with AAC, although lossless ALAC may also be used.
m4p = Apple. A version of AAC with proprietary Digital Rights Management developed by Apple for use in music downloaded from their iTunes Music Store.
mmf = Samsung. a Samsung audio format that is used in ringtones.
mp3 = MPEG Layer III Audio. Is the most common sound file format used today.
mpc = Musepack or MPC. Formerly known as MPEGplus, MPEG+ or MP+. Is an open source lossy audio codec, specifically optimized for transparent compression of stereo audio at bitrates of 160–180 kbits.
msv = Sony. A Sony proprietary format for Memory Stick compressed voice files.
ogg / oga = Xiph.Org Foundation. A free, open source container format supporting a variety of formats, the most popular of which is the audio format Vorbis. Vorbis offers compression similar to MP3 but is less popular.
opus = Internet Engineering Task Force. A lossy audio compression format developed by the Internet Engineering Task Force (IETF) and made especially suitable for interactive real-time applications over the Internet. As an open format standardised through RFC 6716, a reference implementation is provided under the 3-clause BSD license.
ra/rm = RealNetworks. A RealAudio format designed for streaming audio over the Internet. The ra format allows files to be stored in a self contained fashion on a computer, with all of the audio data contained inside the file itself.
raw = a raw file can contain audio in any format but is usually used with PCM audio data. It is rarely used except for technical tests.
sln = S Linear format used by Asterisk.
tta = The True Audio, real-time lossless audio codec.
vox = The vox format most commonly uses the Dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. Similar to other ADPCM formats, it compresses to 4-bits. Vox format files are similar to wave files except that the vox files contain no information about the file itself so the codec sample rate and number of channels must first be specified in order to play a vox file.
wav = Standard audio file container format used mainly in Windows PCs. Commonly used for storing uncompressed (PCM), CD-quality sound files, which means that they can be large in size—around 10 MB per minute. Wave files can also contain data encoded with a variety of (lossy) codecs to reduce the file size (for example the GSM or MP3 formats). Wav files use a RIFF structure.
wma = Microsoft. Windows Media Audio format, created by Microsoft. Designed with Digital Rights Management (DRM) abilities for copy protection.
wv = Format for wavpack file.
Denis van der Velde
AAMS Auto Audio Mastering System
www.curioza.com
Website Links
AAMS Auto Audio Mastering System.
As a musician or engineer working on music, you need the best sound possible when releasing material to the public. To audio master a mix towards a professional commercial quality recording and to create a sound for all audio speaker systems is a difficult and time consuming task. Providing a windows free software package that supplies you with Full Automatic Audio Mastering. Featuring automatic Equalizer, Multiband Compression and Loudness settings, internal DSP Processing with all audio corrections done purely inside the AAMS Program. Making the Mastering Process easy and less time consuming, having a good overall sound quality, to process your Mix to a commercial great sounding Master. Now you can listen to what you expect!
Home of Pleasure Fabric.
Supplies top quality Clothing and Wearables. Our shop is filled with ready to order designs.
PleasureFabric is the place for you! Custom Design Needs ? Add your own text, art, images, pictures or styles, to just about anything. Pleasure for your personality!
Home of Sined Supplies Inc.
Sined Supplies Inc, is the homepage of Denis van der Velde.
Dutch based and features Sined Supplies Inc. and Sined Studio.
This main site has a lot of info about us, music, mixing and mastering, as wel contains music to listen to freely.
And features resources on art, pictures, video and audio, games, etc.
7ZIP - Compressed Zip and Rar Archives
Adobe Photoshop - Painting and Photo Correction
Bluftitler - Video Text Titler and Editor
https://www.outerspace-software.com/blufftitler
Burnaware - CD Burning
Classic Shell - Windows Start Menu
Daemon Tools - Windows ISO CD on Harddisk
https://www.daemon-tools.cc/downloads
DOPDF - PDF Printer Driver for Windows
EMclient - Email Client for Windows
FileZilla - FTP Client
https://filezilla-project.org/
Google Chrome - Internet Browser
https://www.google.com/intl/nl_nl/chrome/
Format Factory Convertor - Audio, Video, Picture Convertor
GreenShot - Screen Capture Tool
HXD - Hexadecimal Editor
Internet Explorer - Microsoft Internet Browser
https://support.microsoft.com/nl-nl/help/17621/internet-explorer-downloads
Malwarebytes - Scan you PC for Malware
MP3tag - Batch Editor for MP3 Tags
XML Notepad - Microsoft XML Editor
https://www.microsoft.com/en-us/download/details.aspx?id=7973
One Drive - Microsoft Cloud Storage
Microsoft Visual Studio - Programming for Operating Systems
https://visualstudio.microsoft.com/
MusicBee - Media Player and Radio Stations
Norton Antivirus - Virus Protection and Scanner
Norton Utilities - Windows Cleanup and Repair
https://us.norton.com/norton-utilities?
Notepad plus - Text Editor
https://notepad-plus-plus.org/
Apache Open Office - Free Office
Open VPN - Free VPN Client
Paint.Net - Free Paint like Photoshop
Panda Antivirus Protection
https://www.pandasecurity.com/
Putty
Recourse Hacker - Software Editor
http://www.angusj.com/resourcehacker/
Robosoft Submitter - Software Submitter to Sites
Rusfus USB - Boot Start from USB
Samsung Sydesync - You Samsung on your Windows Desktop
https://www.samsung.com/levant/support/side-sync/
Sheep Desktop Pet
https://www.microsoft.com/en-us/p/esheep-64bit/9mx2v0tqt6rm?activetab=pivot:overviewtab
Skype - Messenger Microsoft
SSD Tweaker - Optimize your SSD Disks
Teamviewer - Remote Desktop
Total Commander - Command your Harddisk and Files
Ultimate Defragmenter - Defragment and Optimize your Harddrives
VLC Media Player - Play all your Video and Media
https://www.videolan.org/vlc/index.nl.html
VMWARE Workstation Pro - Run Windows in your Windows
WinAmp - Media Player
Tweaking Windows Repair - Repair All Windows Versions
https://www.tweaking.com/content/page/windows_repair_all_in_one.html
WinRar - File Compression
https://windowsreport.com/music-sequencer-software/
https://www.image-line.com/flstudio/
https://www.ableton.com/en/live/
https://www.reasonstudios.com/en/reason
https://new.steinberg.net/wavelab/
https://new.steinberg.net/groove-agent/
https://www.magix.com/us/sem/sound-forge-pro/
https://www.nch.com.au/wavepad/
https://www.nch.com.au/wavepad/index.html
https://www.nch.com.au/recordpad/index.html
https://www.presonus.com/products/studio-one/
https://acoustica.com/mixcraft
https://www.avid.com/pro-tools
https://www.native-instruments.com/en/
https://blog.landr.com/best-free-vst-plugins/
https://bedroomproducersblog.com/
https://www.izotope.com/en/products/
https://www.meldaproduction.com/
https://www.blackghostaudio.com/blog/the-ultimate-list-of-plugin-companies
https://www.spitfireaudio.com/
https://www.yamaha.com/2/proaudio/
http://www.virsyn.de/de/Home/home.html
https://www.audiopluginguy.com/deals/
https://en.wikipedia.org/wiki/Audio_plug-in
https://www.pluginboutique.com/
https://www.homestudioinfo.nl/sequencer-software/
https://www.soundonsound.com/sound-advice/pc-freeware-sequencers-editors
https://www.soundonsound.com/sequencers-grooveboxes
https://en.audiofanzine.com/plugin-sequencer/editorial/articles/the-best-daws.html
https://www.linuxlinks.com/muse/
https://www.avs4you.com/index.aspx
https://www.uaudio.com/blog/audio-mastering-basics/
https://www.landr.com/online-audio-mastering
https://www.headroom-mastering.nl
https://nl.wikipedia.org/wiki/Mastering
https://www.bandlab.com/mastering
https://www.cefeflynnmastering.com/
https://www.musicianonamission.com
https://productionadvice.co.uk/mastering-basics/
https://tomsrayaudiomastering.com/
https://www.audioanimals.co.uk/
https://www.pointblankmusicschool.com/
https://www.superaudiomastering.com/
https://www.cefeflynnmastering.com/
https://www.plugin-alliance.com/
https://www.celemony.com/en/start
https://new.steinberg.net/spectralayers/
https://www.audiosciencereview.com/
https://www.musicguymastering.com/
https://pedals.thedelimagazine.com/everything-about-audio-mastering/
https://audioservices.studio/pheeks-mixdown-and-mastering-blog
https://download.cnet.com/AAMS-Auto-Audio-Mastering-System/3000-2170_4-10352089.html
https://aams-auto-audio-mastering-system.soft112.com/
http://forum.cakewalk.com/Auto-Audio-mastering-System-AAMS-m3380705.aspx
https://www.audioanimals.co.uk/
https://www.leapwingaudio.com/
https://www.piercejohnstonmastering.com/
https://www.youtube.com/playlist?list=PL2ImSFNFVe5WcCA5WauwOvLAPUNA8jlGL
https://www.download3k.com/Install-AAMS-Auto-Audio-Mastering-System.html
https://www.kvraudio.com/forum/viewtopic.php?t=379224
https://www.reddit.com/r/edmproduction/comments/1c9wuj/aams_auto_audio_mastering_system_is_now_free/
https://forum.renoise.com/t/aams-auto-audio-mastering-system/23557
https://www.middlemastering.com/
https://forum.cockos.com/archive/index.php/t-182493.html
https://www.pro-tools-expert.com/
https://tehnoblog.org/aams-auto-mastering-basic-advanced/
https://soundcloud.com/denis-van-der-velde
http://www.sined.nl/index.php/aams/aams-vs-landr
https://nl-nl.facebook.com/AamsAutoAudioMasterSystem
https://nl.linkedin.com/in/denis-van-der-velde-1b820a24
https://aams-auto-audio-mastering-system.apponic.com/
http://forum.cakewalk.com/Auto-Audio-mastering-System-AAMS-m3380705.aspx
https://www.bytesin.com/software/AAMS-Auto-Audio-Mastering-System/
https://nl.pinterest.com/denisvdvelde/aams-auto-audio-mastering-system/
https://gitlab.com/toyginoformca/sylwh/-/wikis/AAMS-Auto-Audio-Mastering-System-v3.8-Rev-001-WiN
https://filecr.com/windows/aams-auto-audio-mastering-system/
https://www.sgonlinemixing.com/
https://aams-auto-audio-mastering-system.software.informer.com/
https://www.studioployens.com/
https://download.cnet.com/AAMS-Auto-Audio-Mastering-System/3000-2170_4-10352089.html
https://qpdownload.com/aams-auto-audio-mastering-system
https://aams-auto-audio-mastering-system.soft32.com/
https://zifyrodohasa.ml/1392.php
http://aljhajmitam.over-blog.com/2020/01/AAMS-Auto-Audio-Mastering-System-v3-keygen.html
https://www.windows8downloads.com/win8-aams-auto-audio-mastering-system-tkcsbrnr/
https://appagg.com/windows/music/aams-auto-audio-mastering-system-10966172.html?hl=en
https://www.careersinmusic.com/best-mastering-plugins/
https://www.msv.nl/computer-audio/audio-software/mastering-software
https://www.sweetwater.com/c702--Mastering_Software
https://store.steampowered.com/tags/en/Audio+Production
https://new.steinberg.net/nuendo/
https://www.soundonsound.com/techniques/audio-mastering-in-your-computer
https://www.practical-music-production.com/audio-mastering-software/
https://www.musictech.net/guides/buyers-guide/freeware-mastering-tools/
https://www.cdmusicmastering.com/what-software-for-audio-mastering.html
https://www.amazon.com/Sound-Editing-Digital-Audio-Software/
https://www.apple.com/logic-pro/
https://www.sonorissoftware.com/
https://www.vmusicpro.com/software/
https://www.youtube.com/watch?v=i-V1mUoz8JI
https://homerecording.com/bbs/threads/automatic-audio-mastering-systems.391400/
https://freeappsforme.com/auto-audio-mastering-system-software-review/
https://anymusicdownloads.com/a-guide-to-setting-up-a-home-recording-studio/
Listen to Reggae Dub Radio Stations
King Dub Radio
https://www.radio.net/s/kingdub
Roots Reggae Dub Radio France
https://www.rootslegacy.fr/
Reggae Dub Stations
https://www.internet-radio.com/stations/dub/
Reggae Dub RadioNet
https://www.radio.net/genre/dub
1fm Reggaetrade Radio
https://www.radio.net/s/1fmreggaetrade
Reggae Dub TuneIn Radio
https://tunein.com/search/?query=reggae dub
Radioforest Reggae Dub
https://www.radioforest.net/genre/reggae/dub/
Radionomy Reggae Dub
https://www.radionomy.com/en/style/reggae/dub
Rcast Reggae Radio
https://www.rcast.net/dir/reggae/page3
Shoutcast Reggae
https://directory.shoutcast.com/?q=242_Reggae_0
Shoutcast Reggae Dub
https://directory.shoutcast.com/?q=245_Dub_242
.
AAMS Auto Audio Mastering System
Welcome to the information page about Composing Music.
Composing Music is a first on the list of things to do, or learn. Although modern music allows crossing the borders between Composing, Recording, Mixing and Mastering. Composing is a good starting point when you newly play an instrument or you are just starting to get interested in music making. On the upside there is a lot information around and known about chords, notes, scales and structure. The downside is that this information is almost all reading and understanding, it can be experienced as uplifting or some find it very boring. Sometimes it will be difficult to understand when you read about technical things about music. Just because music is worldwide played and enjoyed, you can find information on the internet about ways to learn music, lessons and opinions. Maybe you will be overblown by all this info. On this page we try to give you an overview what is available and what is important to know when composing a song or track, or learning to play an instrument. On the other hand this SINED site is off course only one of the many sites about music.
Improvisation
Improvisation is the practice of acting and reacting, of making and creating, in the moment and in response to the stimulus of one's immediate environment. This can result in the invention of new thought patterns, new practices, new structures or symbols, and/or new ways to act. This invention cycle occurs most effectively when the practitioner has a thorough intuitive and/or technical understanding of the necessary skills and concerns within the improvised domain. To me, it is important to stay with an improvising mind. Whenever you’re playing an instrument of any kind, you can always relax and enjoy your instrument. So a lot of people starting will be advised to read notes and scores, learn to play them and learning a lot of technical stuff about music. That will be helping making you understand music more, it is not the only way to do so. Keep in mind that just playing your instrument out of your head exploring the possibilities, relaxing, improvising and goofing around is just as supported and commonly accepted as a part of music making. Some people do not want to learn all this technical stuff, some do. It is all up to you if you do everything by the book or not. And now days there are a lot of artists and recordings around that were made by people who never even could read a single note out of a score. They are just bashing tunes out of their instrument or equipment and are having a blast doing it. They are successful with their way of doing things, their style of making music. Experienced players who are theoretical and well educated, who know to read and play all sorts of scores, scales, chords, etc, maybe good at reproducing music that has already been made. Maybe they will not be as good in composing music or playing their instrument in an improvised session. Maybe someone will tell you that you are playing wrong chords or notes, because technically it is done in a structured way. Maybe someone will tell you that reading scores and knowing chords, scales is the most important thing to learn playing an instrument, well then it is good to know you are never doing anything wrong or right. Music is an open format; you can do anything you like. If you will be successful or not, it depends what you are looking for in Music Making. This will be you deciding what is important for you to learn or not. Investing in time and relaxing with the whole subject is a good thing to do. Watch and learn from other people, never be afraid that they might be better and more educated. If music becomes a fight to be the best and making you win the game, this will not help when you are not relaxed playing music. Just understand that everything you learn takes time to be processed by your brains, playing and repeating learn, investing time and patience is best. Playing an instrument means practicing a lot, knowing your instrument. Also teaching yourself that improvisation is just as important. Just remember not to get hasty and relax, take your time to explore what you’re doing. To me there are two kinds of players, some can improvise and some just play what has been given to them. Ok you can learn notes, chords, scales and every score you can find, you will get better playing your instrument and you will get better at learning overall music. But this will only go for music that has already been made and written, it’s good to learn from others. But it does not really help you improvising and inventing new. Mainly for composing or making music, improvisation is a key element. In a beginning rock band you might find 4 or 5 players. Most likely not all players are in the process of composing the music for the band. Most members of the band will be only playing what has been given to them by the other band members who compose or write the music. It is likely because of communication and band structure that only 1 or 2 players actually are getting involved with the composition of songs, they write almost all music for the other band members. The other players never get so much involved in composing the material they play, because this might be lack of improvisation on their part or just band structure agreement, who is running the show. Some players are just happy to just play that has already been written. Some players have learned so much about what has already been written, they know their chords and progressions, because they have learned them by the book. It’s ok, but it might keep them from inspiration, improvising and invention.
Improvising, Inspiration and invention are needed to create!
The creative side of music is the most comprehensive and rewarding thing once you get the hang of it. Learn yourself that although doing things by the book, does not mean that you will be best off. It means you are educated, it is proven ground. You can learn from what other people do and have done in the past. Remember that people are creative by themselves. You can create new melodies and chord progressions, you can learn by improvising and you will re-invent yourself while doing it, exploring the unproven ground. Composing and Making Music is getting inspiration, improvising and invention from others or you reach into yourself to get it out. So if you are playing or making music, just understand that time is the main factor. Dividing time into learning and goofing around (improvising) is important. Once you have invested time playing and improvising your way, you will be more educated by yourself and how you like to play your instrument or equipment. Learning is time consuming, take your time and do not get frustrated by overdoing the practice. Your mind most be settled down and you most understand what you’re doing, this might take more time then you think it will be. Rushing things is never good. It is good to know there are always people better (or worse) then you make up your own mind in these situations. Learn from it, instead of competing. Players who compete with other players is a good thing, but over competing is not. Music is a set of instruments that each play their part of the whole. This means that competition is not a factor in written music, getting the best out of it is the way to go. If this means sacrificing yourself to have more room for inspiration and improvising, that will be a good thing. Making and creating music is by itself easy. Sit down and play. But for newcomers the vast available information might get them off-track, they are egar to grab hold of any information or learning process. There is a good chance you will be advised to learn all kinds of stuff and you will be over educated. There is a chance that by learning by the book, you will only play by the book. The more time you spend on playing by the book, the more chance there is you will get stuck by being thought structured and you might be hanging onto what other people have done before. Remember music is more free then that and there is more success to be made with the basics, making things complicated is most likely not the way to do music. Once your improvisation has resulted in new music or compositions, you will get inspired even more with the results.
Step 1: Writing a Song
Without a song, how could one possibly begin to record. Typically, writing a song starts with an idea or an inspiration. It may begin with lyric and a melody, a chord progression, a unique sound/loop or an improvisation that takes on a life of its own. Once this idea has developed enough to stand on its own merits, the music production process can then begin. A music production must support, in every way possible, the message or prevailing emotion of the song. The most common mistake I see today with young producers and songwriters is that they focus on the sounds or production elements before the song is finished being written. For certain styles of music this can work if the production invokes a feeling or emotion that inspires the lyric and melody. In many cases though, the production sounds disjointed because the lyric and melody end up being limited to the production style or arrangement. What happens next is that the arrangement must be adapted to the lyric and melody and the production can easily lose the coherency necessary for the song to carry its message.
Traditional Songwriting
Traditionally, writing a song is done with a singular instrument, a lyric and a melody. That's why so many songs start out as a piano and vocal or acoustic guitar and vocal. If you were writing a song with a group of musicians, they would likely become bored or disinterested if you spend too much time experimenting with melodies or new lyrics. When writing a song, it is typically best to work through these issues alone or with a writing partner that will help you quickly dismiss ideas that just don't work. Once you have flushed out all of these issues the music production process can really begin in earnest. When carefully crafted, a song will hold the interest of the listener. A song tells a story that conveys ideas and emotions. If the story is something the listener can relate to then they will listen as long as it is told in a compelling way. Great storytellers are very dynamic and interesting people as are great recording artists. They convey the emotions and events in a song with vivid imagery that takes you on a journey. Although the recording artist and the songwriter are not always the same person, the pairing of artist and songwriter is critical to the success of a song. Sometimes they work together in the process of writing a song so that the artist can add their input and perspective of what the song is about. If the artist cannot relate to the song from their own personal experience, then it will typically sound hollow. The passion must be there for the song to be taken in by the listener.
Modern Songwriting
The blessing of the process for writing a song today is that there are so many resources available, you don't need a band to make a music production. You can create a template production that allows you to work on your ideas without wearing other people out by making them play the same parts over and over again. The use of music loops and samples is an exceptional way of getting the creative juices flowing and setting the stage for writing a song that's inspired. This process can also have pitfalls. One of the most common is that the songwriter may fall into the trap of focusing on the production elements instead of just writing a song. Without a good sense of judgment, the songwriter may ignore the real problems which may be that the lyric or the melody just isn't very good. By focusing on the production elements they may waste hours, days weeks or months trying to salvage a song that is not really ready for the music production process. It is for this reason that I believe most of these tools are best used in the demo stage of the music production process. I've seen too many songwriters lose their flow while writing a song because they spend hours trying to work out technical issues instead of just writing. Keep the songwriting process simple. Always have a recording device with you to capture an inspired idea. If you have a smart phone, your one app away from having a portable recording device with you at all times. For those that struggle with writing a song, good lyrics and melodies or finding good subject matter to write about, there are many websites and forums on songwriting to hone those skills. Writing a song is an art form in itself. However, to start the music production process, the quality of the song cannot be ignored. If you want to become a music producer, you cannot ignore good songwriting skills as a necessary part of your repertoire. The ability to assess issues and make necessary corrections will go a long way to helping you be successful. It is the song, after all, that the listener will relate to most, not the production. To be very clear, the process I have been talking about here is all about songs that are meant to be the center of one’s attention. Although many of the ideas presented here will also work for other forms of music, the focus here is on lyric driven music. Since all music carries some story or emotionally driven feeling, the concepts here can be adapted to the production style to achieve similar results. A jazz or classical record, for example, also convey emotions that tell a story. Even though the story may not be as explicit as a lyric driven song, the same process can be used to aid the listener into the interpretation of that story. The second is the way RNB or Rap music is made, that can be done with 12 bit samples or with a sequencer set on quantisize 1/16. But this might be for composing. Anyway the less is more. Sometimes people dont't need the full content of the song tekst you have written, keep in mind that suggestions in the text or song, leaving things out, might be better and more enjoyable for the listener.
Basic Principles of Writing a Song
To help lend a broader understanding of writing a song, let's go over some of the key elements of good songwriting and how they affect the music production decisions you make. These four basic points of focus must be addressed before the song enters the recording phase of the music production process. What the hell is your song about? What feeling are you attempting to convey? Love, jealousy, hate, anger, fun, etc… These decisions lay the groundwork for EVERY other decision that is made including what sounds and instruments are selected in the production process. Writing a song about heroin addiction, for example, is not going to have bright tinkling bells as part of the music production. In this example, the musical elements of the song will need to be dark and oppressive sounding so that they support the prevailing message of the song which is most likely about depression and helplessness. Conversely, writing a song that's meant to make people party and dance is not going to be filled with dark heavy depressing sounds. The elements used here will be brighter, punchy and focussed. They will need to pump and breathe at the pace a person would dance to. While this may seem obvious on the surface, the real artistry of writing this type of music is doing something unique while remaining within these parameters.
Telling the Story
How do you plan to convey this message? These decisions all start with the prevailing message or feeling from the song. This can be as simple and using a minor key for a sad song versus a major key when the message is more positive. The blend of melody and lyric must support each other in every way. If the prevailing message is one of irony or sarcasm, writing depressing lyrics in a major key could convey a sense of humor or show a person trying to cover up their true feelings about the subject matter. There is no way to underestimate the importance of this relationship. The human brain is wired to receive and process information in a very particular way, if you go too far outside of these parameters, the message will be lost on most that care to listen. When presented well, you open a doorway to the listener's consciousness. From there it is up to you to keep the door open by continuing to hold the interest of the listener. Of all the topics surrounding the music production process, this is the one with the least number of technical solutions. No plugin, compressor or effect will cover up a bad song for very long. No processor will change the attitude or feeling of a song. These tools can only enhance an energy that must already be present. In the example above, heavily compressing the recording may help to convey the feeling of being trapped. This approach may work against you, however, if the song focusses on the feeling of freedom while on the high. This is the reason that the songwriting process is so critical to get right before even attempting to start to make a music production out of it. If a song can't hold the interest of a listener when presented in its most simple form, then it likely can't withstand the music production process without becoming and endless parade of band aids.
Holding the Attention of the Listener
How would you like to present the song? How will the dynamic energy of the song flow? Do you want it to start out simple and end big? Do you want it to start out big, drop down in energy and then explode in the end? Do you want it to maintain an even energy level throughout? Any one of these methods can work if the selected method supports the message of the song. The classic structure for a song starts with a verse which presents a story or situation. It tells you what happened, how you got into this situation in the first place. The chorus section then conveys the emotional result of the story that has just been told. It tells the listener what has resulted from the events told in the verse. Usually, there is a back and forth profession between verse and chorus that may lead into a breakdown or bridge section. The breakdown or bridge sections will take you to another perspective of the story. It may be the truth of what has transpired, it may represent a reprieve from the story so that the impact of the remaining story is more dynamically felt when the next chapter is told. This traditional method of songwriting is not necessary if a creative way to keep the interest of a listener is created. A song about the repetitive nature of living and working in a big city may benefit from a repetitive loop or programmed rhythm. The programed, repetitive nature of the production may help to convey the feeling of living a robotic life, repeating the same pattern of living day after day. The reason why the traditional verse, chorus, bridge method works is that there is a template that will most likely to hold the interest of the listener if presented well. Every story has a setup (verse) a problem or dilemma (chorus), a realization or solution (bridge or breakdown), and an ending. The ending can be any of the other song elements or something completely different depending on how the story ends. A song is basically a 3-5 minute movie in audio form. I like to use visual references when talking about any kind of audio because the reality is that sound is a secondary sense to sight. Up until the age of synthesis, every sound that we ever heard came from a physical object that we could visualize. This programming has been built into us for thousands of years and serves us well as a survival mechanism. Sound allows us to perceive and interpret things we may not be able to see. Sometimes they are dangerous things, like a car racing through an intersection you are about to cross. Sounds presented well in musical form also help to support or create the images or feelings that are presented in the song. A song is no different than any other form of audio. A song can create images in a person’s mind. They listener may recall past events in their life that relate to the story being told in the song. The music production helps to support that imagery. When properly done it may bring a person back to their own personal experiences that they can remember and relive through your song.
Feeling Over Thinking
When a song is well written the dynamic of the song will be clearly spelled out by the story. It will help you decide if you should use a breakdown section instead of a bridge section. It will help you decide whether to fade out on chorus sections, a vamp section, end the song with big crash or just a simple melody. Unless you are writing a song that is meditation music or attempting to put the listener into some kind of hypnotic state, people will respond most to differences in things, not sameness. Without this progression of dynamic changes, people will get bored and turn you off. The best way to judge whether a song is ready for the music production process is to FEEL it instead of listening to it. In other words, stop thinking and just let it speak to you. Pay close attention to any section of the song where you lose interest or feel your attention is taken somewhere else. Does the song hold your full attention from beginning to the end? Does it drag on too long? Do you feel cheated of shortchanged by the song because it is too short? Do you feel satisfied after listening to it ? Remember, feeling will always outweigh thinking! If you find yourself trying to convince somebody why a song is good, then you should already know that something is wrong. If you have completely lost your perspective, shelve the song for a while until you can listen with fresh ears. Listening to the same thing over and over can have the effect of burning it into your consciousness. You lose the ability to be objective. Finally, when writing a song, never ask somebody what they think of it. Unless they are a professional producer or artist and are brutally honest people, they will usually BS you because they are your friend and trying to support you. The best way to judge a song is to play it in the background and just watch for reactions without soliciting one. Do they move their head or body to the beat? Do they leave the room singing the lyric or melody? Do they ask you about whose song this is? These are clear signs that something is right because they are feeling it, not listening to it. You may want sometimes to write down some melodies or chords.
Using symbols C, C#, D, D#, E, F, F#, G, G#, A, A#, B for writing down notes can be helpful when you need to capture melodies, baselines, etc.
Using common chords symbols like C, Cm, Am, or D can be handy to write down chords. Also for chords it can be handy to store how many times the chord is played. Using C/ Am/ E/ C for wiring down your chord list might not be enough. So you can write C/ Am / E (2/ 2/ 4), this means the C chord is played 2 times, the Am chord is played 2 times, the E chord is player 4 times. You can safely say that (2/2/4) is the same as (4/4) , but at least you can write down what your intending to do with the chords, take in account of how many times a chord is played inside bars. Anyway it is important to have a system for writing down notes, chords for melodies and composition. I do not use any scores for writing down, scoring is a better way of doing things, but it can be more complicated to learn and more time consuming.
Keyboard Playing
Playing on midi keyboards is a common thing, now days music making on a single computer can stand with one single keyboard for inputting notes and chords. When you are new to playing a keyboard we will first explain how it works while composing music.
Note C.
You can see we will start basic with the note C.
Octaves 1 to 5.
On this keyboard there are 5 octaves and the C note can be played five times from C1 to C5.
Notes C, C#, D, D#, E , F , F#, G, G#, A , A# and B.
Let’s investigate one of the octaves. The white notes labels are C, D, E, F, G, A, B. The black notes labels are C#, D#, F#, G#, A#. Together they form one octave.
Major Chords.
Now here is where it is getting interesting, to remember all Major Chords, remember the sequence 1-4-3!
The Cmajor chord is played by pressing C, E and G together.
Minor Chords.
To remember all Minor Chords, remember the sequence 1-3-4!
The Cminor chord is played by pressing C, D#, G together.
Major Septime7 Chords.
To remember all MajorSeptime7 Chords, remember the sequence 1-3-4-3!
The CmajorSeptime7 chord is played by pressing C, E, G, A# together.
Major Kwint Chords.
To remember all MajorKwint Chords, remember the sequence 1-3-4-1!
The CmajorKwint chord is played by pressing C, E, G, G# together.
Just a quick overview:
Major Chords 1 - 4 - 3
Minor Chords 1 - 3 - 4
Major Septime7 1 - 4 - 3 - 3
Major Kwint 1 - 4 - 3 - 1
You may want sometimes to write down some melodies or chords.
Using symbols C, C#, D, D#, E, F, F#, G, G#, A, A#, B for writing down notes can be helpful when you need to capture melodies, baselines, etc. Using common chords symbols like C, Cm, Am, or D can be handy to write down chords. Also for chords it can be handy to store how many times the chord is played. Using C/ Am/ E/ C for wiring down your chord list might not be enough. So you can write C/ Am / E (2/ 2/ 4), this means the C chord is played 2 times, the Am chord is played 2 times, the E chord is player 4 times. You can safely say that (2/2/4) is the same as (4/4) , but at least you can write down what your intending to do with the chords, take in account of how many times a chord is played inside bars. Anyway it is important to have a system for writing down notes, chords for melodies and composition. I do not use any scores for writing down, scoring is a better way of doing things, but it can be more complicated to learn and more time consuming.
The Keyboard.
Advice for all computer and keyboard users is that a modern midi keyboard will be sufficient. But there are some traps to avoid. Get a midi keyboard that supports aftertouch and is touch sensitive. Cheaper keyboards do not support this kind of features, so really be sure you have these features on board. Touch sensitive playing. When you press a note on the keyboard, not only the note is send to the computer but also how fast and hard you are hitting it, this is most useful when playing natural instruments alike a piano. You can vary playing soft and hard, this is more expressive then playing a keyboard without Touch Sensitive keys.
Aftertouch.
Another things the Touch Sensitive keyboard can do, is when you are holding a note you can press softer or harder when to note progresses. For like Organ playing this can be a handy feature to control the effect of the rotary speaker. With Touch Sensitive Keys and Aftertouch you can at least input what your fingers are doing and record it more natural, both are highly recommended. It is likely when you do not need these features, when you do have them you can always turn them off. But you are at least sure that you’re getting the maximum out of your midi keyboard.
Pitch Control.
The place for pitch control on a keyboard is mostly on the left hand side, next to the keyboards lowest keys. Most common are wheels, joysticks and faders for this kind of operation. It is handy to have pitch control on your midi keyboard.
Modulation Control.
The place for modulation control on a keyboard is mostly on the left hand side, next to the keyboards lowest keys. Most common are wheels, joysticks and faders for this kind of operation. It is handy to have pitch control on your midi keyboard.
Other Controls.
Some midi keyboard do have touchpad’s and more wheels and faders for you to assign. For people using synths on a computer these can be handy to adjust parameters. But these can be done with the mouse or controller equipment, it is not as important as it can be fun. Most controlling on cheaper midi keyboard is quite good, but one you start to be a control freak maybe you will later look for more controlling options.
Midi or USB connections.
There has to be at least one Midi-Out to connect to the computer. This can be done using an USB-cable or midi cable. I still prefer the keyboard to be connected to a real midi cable, instead of using USB. So when there is a midi in and output on the keyboard or USB, choose midi to connect to your computer. Although USB is faster than midi, midi is more stable then USB. It is a matter of timing, when notes arrive to a computer. Until now USB is fast but can deliver notes that are just not the same as you played. Mainly a good written USB driver will help, but some manufacturers do better just then others by making USB drivers for their keyboard. Midi is proven, so connecting the midi cable to your computer is a good safe option.
Do Re Mi - Finding Chords
Minor 1 - 3 - 4
Major 1 - 4 - 3
Septime7 1 - 4 - 3 - 3
Kwint 1 - 4 - 3 - 1
White and Black Keys
The white keys are C, D, E ,F, G, A, Flats ' b' and sharps '#' are all black keys called
C#, D#,F#,G#,A#. The ' b' symbol can be used when ever needed.
One Octave are 12 notes in a row called C, C# or Db, D, D# or Eb, E, F, F# or Gb, G, G# or Ab, A, A# or Bb, B.
Major
Any time you see a letter on its own for example “F” this is called F Major or “F Maj”.
The Major Scale are seven notes in order 1 2 3 4 5 6 7 .
Root - Tone – Tone – Semitone – Tone – Tone – Tone – Semitone
Do – Re – Mi – Far – So – La – Ti – Do
You choose a note or chord like C then count 4 notes up for E and 3 notes up for G, so C, E, G make up for Cmajor or just talking chords C.
Major = 1 - 4 - 3
Key Root 2n d 3rd 4th 5th 6th 7th
Cm c d e f g a b
Gm g a b c d e F#
Dm d e f# g a b c#
Am a b c# d e f# g#
Em e f# g # a b c# d#
F# F#/Gb G#/Ab A#/Bb B/Cb C#/Db D#/Eb E#/F
Db-m Db Eb F Gb Ab Bb C
Ab-m Ab Bb C Db Eb F G
Eb-m Eb F G Ab Bb c d
Bb-m Bb C D Eb F G A
F-m F G A Bb C D E
Minor
Every major key has a corresponding relative minor key. You choose a note or chord like C then count 3 notes up for D# and 4 notes up for G. So C, D#, G make up for Cm.
Minor = Notes 1 - 3 - 4
The minor key will be in the same key signature, and will contain the same notes as the major key. The only difference between the two is that the minor key simply starts on a different note. In the key of C Major, the relevant, corresponding minor key is A minor.
You can always find the relative minor key by counting up six notes from the root of the Major key. So in the C Major example: C, D, E, F, G, ->A<-, B, C.. The minor key starts on A.
Root – Tone – Semitone – Tone – Tone –Semitone – Tone - Tone
So if as an example we use the A minor scale which is the relative minor scale of C Major, we have the following sequence of notes:
A B C D E F G A
If we were playing in F Major, the relative minor would again begin on the sixth note in the key, which would be the D, and the sequence of notes would be:
D E F G A Bb C D
Minor.
If you see "Fm" or "Fmin" this is called Fminor.
Key Root 2nd 3rd 4th 5th 6th 7th
A-min A B C D E F G
E-min E F# G A B C D
B-min B C# D E F# G A
F#-min F# G# A B C# D E
C#-min C# D# E F# G# A B
D#- Eb- D#/Eb E#/F F#/Gb G#/Ab A#/Bb B/Cb C#/Db
Bb-min Bb C Db Eb F Gb Ab
F-min F G Ab Bb C Db Eb
C-min C D Eb F G Ab Bb
G-min G A Bb C D Eb F
D-min D E F G A Bb C
A semitone (or half step) is the smallest increment on a western musical instrument. On a piano, it is represented by moving from one key to the next, and on a guitar, it is represented by moving from one fret to the next. As an example, on a piano, moving from middle C to the black key directly next to it on the right, we would get a C# would be a semitone. Moving from middle C to the next WHITE key on the right, which is the D, would be a tone from the middle C (also known as two semitones or a whole step). On a guitar, moving from the open A string to the first fret on the A string A# would be a semitone, whilst moving from the open A string to the second fret B would be a tone (two semitones).
Major Again
So if we look at the C Major scale, it looks like this:
C (root note)
Then up a TONE to D
Then up a TONE to E
Then up a SEMITONE to F
Then up a TONE to G
Then up a TONE to A
Then up a TONE to B
And finally up a SEMITONE again to finish back on C.
All major keys follow this pattern, and you can start a Major scale on any note.
A couple of things to be aware of: Some notes have the same sound, but different names depending on which KEY they are in. For example, an A# is the same note as a Bb as if you move up ONE semitone from A it becomes A# and if you move down ONE semitone from B it becomes a Bb. Again, you don’t need to worry too much about this if it’s confusing you as we’re going to stick mainly to simple chords and keys.
Intervals and Chords.
Basically each note and the following note is half a step. From the first note to the next nearest note is half a step (the same) and is called a semitone. On a piano this is each following key. On a guitar this is each following fret.
Timeline.
The purpose of a time signature is to show you what type of feel, rhythm, and speed you should play certain notes, phrases and bars. There are various time signatures in music. The two most common are Four-Four time, and Three- Four time. The first number in the time signature denotes the NUMBER of notes you will be playing, PER BAR and the second number tells you what TYPE of note you’ll be playing. So if we’re playing in Four-Four time, you would have four even beats of quarter notes, and count like this: One, Two, Three, Four, One, Two Three, Four etc. If you were playing in three four time, you’d be using the same length notes, but only count three of them per bar, for example: One, Two, Three, One, Two, Three etc. The following are the most common types of note found in Western music, and each of these notes also has a corresponding rest that has the same duration. These are also found on the examples below.
1 2 3 4 5 6 7
C D E F G A B
C# D# F F# G# A# C
D E F# G A B C#
D# F G G# A# C D
E F# G# A B C# D#
F G A A# C D E
F# G# A# B C# D# F
G A B C D E F#
G# A# C C# D# F G
A B C# D E F# G#
A# C D D# F G A
B C# D# E F# G# A#
Ionian: No Change
Dorian: 3b,7b
Locrian: 2,3,5,6,7b
Lydian: 5b
Phyrgian: 2,3,6,7b
Mixolydian: 7b
Aeolian: 3,6,7b
Major: 1,3,5
Sus2: 1,2,5
Minor7th: 1,3b,5,7b
Minor: 1,3b,5
Sus4: 1,4,5
Major7th: 1,3,5,7
Add2: 1,2,3,5
Fifth: 1,5
Diminished: 1,3b,5b
Scales
Major pentatonic c d e g a
Neapolitan minor C Db Eb F G Ab B C
Hemitonic pent3 C D Eb G B Spooky
Pent var C E G A Bb Smooth
C-majdom7 C E G B yawm, a major scale
c-mindom7 C Eb G Bb minor
Harmonic mindom7 C Eb G B phantom minor
Melodic mindom7 C Eb F A Really F-majdim7,2nd
Esoteric 6th Cb D F A Dreams ?
Augmented C E G# Tence
Diminished C Eb Gb Also Tence
Minor 3rds C Eb Gb A Tence , dreamy
Harmonic minor C D Eb F G Ab B Sad
Melodic minor C D Eb F G A B Nice sweat
Whole tone C D E F# G# A# Whoa tripping!
Augmented C D# E F# G# B# nasty tension
diminished C D Eb F Gb Ab B Sad Ab+B = Bbb/Cbb
Enigmatic C Db E F# G# A# B Indeed very strange
Bizantine (gypsy) C Db E F# G# A# B Spooky
Locrian (arabian) C D E F Gb Ab Bb Drunk
Persian C Db E F G A# B Secrets
Spanish 8 tone C C# D# E F F# G# A# Ummm
Native American C D E F# A B Bold btw which tribe
Major bebop C D E F G Ab A B Funky min/maj
Barber shop1 C D E F G B D F G B Full
Barber shop2 C G C E G B Same as 1 but sad
Rain A# D E F# G# C D F# G# Messed pissed off
Crystalline min9#7 C G B Eb G D Eb Bb D
Gb Bb F Gb Db F A Very mad as hell
Popular blues C D# F F# G A# pissed crunchy
Blues II C D# E G Ab D# G Spacy dissonant
Total disharmony C Db E F G Ab B C D Eb
F# G A Bb C# D# Ouch thunder
Sus2 C D G Cool nine
C-phuq'd C F Ab Bb
Db-grace major C F Ab Db F G Db Gb
E-blues C D E G Ab Gb Db F
Neaplotitan major C Db Eb F G A B C
Oriental C Db E F Gb A Bb C
Double harmonic C Db E F G Ab B C
Enigmatic C Db E F# G# A# B C
Hirajoshi A B C E F A
Kumoi E F A B C E
Iwato B C E F A B
Hindu C D E F G Ab Bb C
Pelog C Db Eb G Bb C
Gypsy C D Eb F# G Ab Bb C
Maj phrygian C D F E F G Ab Bb C
Maj locrian C D E F Gb Ab Bb C
Lydian min C D E F# G Ab Bb C
Overtone C D E F# G A Bb C
Arabian C D E F Gb Ab Bb C
balinese C Db Eb G Ab C
Gypsy C Db E F G Ab B C
Mohammeddan C D Eb F G Ab B C
Javanese C Db Eb F G A Bb C
Persian C Db E F Gb Ab B C
Algerian C D Eb F G Ab B C D Eb F
Aeolian C D Eb F G Ab Bb C
byzantine C Db E F G Ab B C
Hawaian C D Eb F G A B C
Jewish E F G# A B C D E
Mongolian C D E G A C
Ethiopian G A Bb (b) C D Eb (e) F (F#) G
Spanish C Db E F G Ab Bb C
Egyptian C D F G Bb C
Japanese C Db F G Ab C
Chinese F G A C D F C E F# G B C
New pentatonic C D E F# A
jap penta C Db F G Ab
Bal penta C Db F Gb A freaky = Db-maj7dim4?
Pelog penta C Db Eb G Bb dreamy
Musical Definitions, Terms relating to TEMPO.
GRAVE - Very slow and solemn
LARGO - Very slow and broad, with dignity
LENT or LENTO - Very slow
ADAGIO - Very slow and expressive
LARGHETTO - Not as slow as LARGO, but slower than ANDANTE
ANDANTE - Rather slow, but with a flowing movement ("Walking tempo")
ANDANTINO - A little quicker than ANDANTE
MODERATO - Moderate speed- not fast, not slow
ALLEGRETTO - Light and cheerful, but not as fast as ALLEGRO
ALLEGRO - Merry, quick, lively, bright
VIVO - Lively, brisk (usually with ALLEGRO, as ALLEGRO VIVO
VIVACE -Vivacious, faster than ALLEGRO
PRESTO -Very quick, faster than VIVACE
ACCELERANDO - Abbreviated: accel. To increase the speed gradually
STRINGENDO - Abbreviated: string. To increase intensity by increasing tempo
AFFRETTANDO - To increase the speed gradually
ALLARGANDO - Abbreviated: allarg. Slower and louder
RITARDANDO - Abbreviated: Ritard. or Rit. Gradually slackening the speed.
RALLENTANDO - Abbreviated: Rall. Slowing down, gradually.
RUBATO - Literally means "Robbed"- a lingering on some notes and hurrying of others; free from strict tempo, but preserving the value of the rhythmic notation.
A TEMPO - Return to original tempo after a RITARD
TEMPO I (PRIMO) - Return to original tempo after a RITARD
Words that often accompany TEMPO Markings:
MOLTO -Very much. MOLTO RITARD means to slow down exceedingly
MENO - Less. E.g., MENO MOSSO means less fast (slower)
PIU - More
NON TROPPO - Not too much, e.g., ALLEGRO NON TROPPO means fast, but not too fast
POCO A POCO - literally "little by little". Used in combination with tempo markings. e.g., ACCEL. POCO A POCO means to increase the speed gradually over a span of measures.
Terms relating to DYNAMICS (from soft to loud):
PIANISSIMO -(abbr: pp). Very soft
PIANO - (abbr: p). Soft
MEZZO - Medium or moderately
MEZZO PIANO - (abbr: mp). Medium soft
MEZZO FORTE - (abbr: mf). Moderately loud
FORTE - (abbr: f). Loud
FORTISSIMO - (abbr: ff) Very loud
DIMINUENDO - (abbr: dim.) or the sign means gradually getting softer
CRESCENDO - (abbr: cresc.) or the sign means gradually getting louder
POCO A POCO - Little by little. Indicates a gradual increase or decrease in volume of sound.
ACCENT - A stress on notes so marked
SFORZANDO - (abbr: sfz) A strongly accented note or chord
SFORZATO - (abbr: sfp) strongly accented by then immediately PIANO
SUBITO - Suddenly. Usually to indicate a dramatically sudden change in dynamic level of sound.
AGITATO - With agitation- excitedly
ALLA - In the style of (always used with other words).
CON - With (as a connecting word), e.g., ANDANTE CON AMORE- slowly, with tenderness
ANIMATO - With animation, in a spirited manner
APPASSIONATO - With intensity and depth of feeling
BRILLANTE - Bright, sparkling, brilliant
BRIO - Vigor, spirit
CANTABILE - In a singing style
DOLCE - Sweetly and softly
ENERGICO, CON - With expression
FUOCO, CON - With fire or much energy
GRANDIOSO - In a noble, elevated style
GRAZIA, CON - With a graceful, flowing style
LEGATO - Smooth and connected, in a flowing manner (Opposite of STACCATO)
MAESTOSO - With majesty and grandeur
MARCATO - In a marked and emphatic style
PESANTE - Heavily, every note with marked emphasis
QUASI - In the manner of; e.g., QUASI UNA FANTASIA- in the style of a fantasia
SCHERZANDO - In a light playful and sportive manner
SCHERZO - A jest, one of the movements of certain symphonies, a composition of light and playful character
SECCO - Dry, plain, without ornamentation
SEMPRE - Always; e.g., SEMPRE STACCATO- to continue playing in a short and detached style
SPIRITO, CON - With spirit, or animation
STACCATO - Short and detached, with distinct precision (the opposite of LEGATO)
TENUTO - Sustained for the full time-value
TRANQUILLO - With tranquility, quietly, restfully
LARGO MA NON TROPPO - Slow, but not too slow (ma = but)
ADAGIO CANTABILE E SOSTENUTO - ('e' = and) Very slow and in a sustained and singing style
ANDANTINO, CON AFFETUOSO - Faster than ANDANTE, with tender feeling
ALLEGRETTO CON GRAZIA - A moving tempo with a graceful flowing style
ALLEGRO AGITATO - Quick with agitation
POCO PIU MOSSO - A little quicker
ALLEGRO CON MOLTO SPIRITO - Fast with much spirit
ANDANTE MAESTOSO - Rather slow-moving tempo, majestic feeling
PRESTO CON LEGGIEREZZA - Very fast with lightness and delicacy
ACCIDENTALS - Flats and double flats, naturals, sharps and double sharps
ALLA BREVE - Cut time. The half-note is the unit of the meter
ARPEGGIO - A broken chord (Each note of the chord played in succession)
ATTACCA - Begin the next movement immediately
CADENCE - The close or ending of a phrase
CADENZA - An elaborate solo passage with fancy embellishments to display the proficiency of a performer.
CHROMATIC - Proceeding by semitones
CODA - Literally "A tail"- the closing measures of a piece of music
CON - With; e.g., CON SORDINO means "with mute"
DA CAPO - (abbr: D.C.) from the beginning
DAL SEGNO - (abbr: D.S.) to the sign
DIVISI - Divided, one performer plays the upper notes, the other plays the lower notes
FERMATA - A pause, marked
FINE - The end
G.P. - General Pause; a dramatic moment of silence for the entire ensemble
SEGUE - To the next piece without pause
SENZA - Without; e.g., SENZA SORDINO means without mute
SORDINO - A mute (used by brass and string players)
TACET - Be silent
TEMPO PRIMO - (Sometimes TEMPO I), means to return to the original tempo after a RITARD or ACCEL.
V.S. - Abbreviation found at the lower right corner of a music page and means to turn the page quickly.
COL LEGNO - Applies to string instruments.
GLISSANDO - To slide. Pulling or drawing the finger quickly up or down a series of adjacent notes. Also poss. on trombone and other inst..
That is all for now...
Denis van der Velde
AAMS Auto Audio Mastering System
www.curioza.com
Sequencers
Without a doubt the computer is mostly where you go for the good sequencers, but in the older day they only had hardware and tape recording. So there are two ways to do music, that is knowing what you can learn. Or not knowing to play chords and not kwoning the key etc, but still fiddle around to get something at random. Anyway for composing music a midi sequencer will do, mostly also audio will you get also. If it is hardware or software, most sequencers on the market do have rizen to having it all. Even scoring and pluging instrumnets and effects, mixing and also maybe mastering. So mostly with one sequencer that is good, is worth all for composing and making music production happen.
Recording Music
The art of recording music is filled with information that is mostly technical. There is loads of information on miking techniques, what microphones and preamps to use, and how to process them. What is less often talked about are the fundamentals that underlie those techniques and choices. The acoustics of the recording space and the quality of the musician will define the sound of the recording more than any mic technique or processing chain ever will. Behind the techniques lies the real foundation of making great recordings. It's the information you don't often get because most are not keenly aware of its existence. Many have only worked in professionally treated acoustic spaces designed for recording and can often forget that their audience is not working in the same conditions. Many work through the problems that arise with intuition rather than taking the time to really understand what lies underneath all the technical choices they make. In reality, it usually takes years to become great at recording music. During that time, allegiances to different pieces of gear will come and go and solutions will be based mostly on experience. When problems arise, it is often easier to blame the studio, available mic selection, recording console or the recording space. If you want to make great recordings, regardless of the recording space and equipment you are working with, you will have to learn something that is more fundamental. Essentially, no two recording situations are identical and each requires a discerning ear and eye. Most of the great engineers learned by experience, trial and error, and working under great engineers before them that understood how sound works. They learned how to use acoustics to their favor, and learned how to work with musicians to get the best performances out of them.
The 3 Types of Recording
In this article I will break down the art of recording music to its most basic elements. The articles that follow in the links at the bottom of the page will get into details about recording specific instruments and the best way to manage those recording situations
Essentially you can break down the types of recording into 3 basic categories:
1. Acoustic Recordings
2. Electronic Recordings
3. In the Box Recordings
Even though, electronic and in the box recordings are not dependent on the acoustic space, the principles of acoustics are still very much at play because that is the only way we know how to perceive sound. Let's take a closer look at each:
Acoustic Recordings
Recording music in the acoustic realm is all about capturing sound waves through microphones and converting them into an electronic signal so they can be captured and recorded. Today, those recordings are mostly into computers and onto hard drives. Whether you are recording analog or digital, the basic process hasn't really changed a whole lot of over the last century or so. Music, for the vast majority of its history to humankind, has always been acoustic. It is only in recent decades that music has gone to purely electronic sources.
The concept of recording, came into play in late 1800s with the inventions of Thomas Edison. Music recording soon followed, although the capabilities were very limited. Primarily, all recording was acoustic material. The technical issues of capturing music in recorded form have undergone immense development over the last 100 years or so. In years past, the mechanical limitations of recording devices limited the engineer's options. Today, those options are seemingly endless. The irony is that the greater number of options available today have taken many engineers away from the fundamentals of acoustics and focused them on new gear and plugins instead. As the quality of recording technology increased, so too did the importance of the acoustic recording space. The decisions made about how to manage the recording space became critical to the quality of a music recording. If you want to achieve a very big live drum sound, you are not going to get it by recording in a small dry space. In the end, no mic will make a recording space sound bigger than it is.
Managing Acoustics
When you place an instrument in a recording environment, that instrument will sound different, sometimes radically different, depending upon how and where you place it in the room. This is especially important for recording music in spaces that are smaller than 20 x 20 feet. There is no microphone that will solve all of the problems with a bad acoustic environment. Even with the best gear all you will get is a very accurate recording of very limited acoustic environment. This does not mean you have to spend thousands of dollars on acoustic treatments. Even professionally treated recording studio environments require careful placement and attention. The most important thing, in any recording situation, is to listen carefully as you move the instrument around room. Find a place in the room that enhances the sound of the instrument without making it sound unnatural. Acoustics is really the key to capturing great recordings and is often overlooked by most novice engineers. If you just dump an instrument anywhere in the room and throw a mic in front of it, you are basically rolling the dice and hoping that a good sound comes up. If you're a bit more conscious about how you place an instrument in a recording space, then you will get significantly better results, with much less effort, and be much happier in the end, even if you are using inexpensive recording equipment. How you choose the right acoustic environment, and how you treat the immediate space around the instrument is unique to each instrument and the sound you are trying to achieve. These guidelines and methods will be covered with more detail in the individual recording instrument links at the bottom of the page.
Electronic Recording
The second method of recording music is electronic recording. Electronic recordings go back to the invention of keyboards and synthesizers, and also with basses and guitars. The idea of using a direct electrical signal is that you are bypassing the acoustics altogether. For many instruments like bass and guitar, the amplifier is a huge part of the sound you are trying to create. Without the speakers and acoustic environment, you have to count on the electronics you are using to create the sound for you. The typical method for capturing electronic audio is through a DI box. The DI box will take any signal that comes from a high impedance unbalanced source, like from a guitar or bass and convert it into a balanced signal so it can be plugged into a mic preamp and recorded. The balanced lines help to keep the signal quiet with a minimum of degradation. Long guitar cables will pick up loads of noise and you can end up with significant signal degradation. Always keep unbalanced cables to a minimum in terms of their length. When recording music with keyboards, you are dealing with electronics that are controlling oscillators to generate synthesized sounds that are sometimes meant to emulate acoustic instruments. The older ones are typically connected to a DI although many of them now have balanced line level outputs. This allows them to be brought directly into a line level input in a recording console. They can then be recorded without having to add a significant amount of gain, thus keeping noise to a minimum. The only issues from a technical perspective are selecting the sounds and editing them until you get it to sound the way you like. If it is a bass, you will need to change pickups, adjust the tome knobs or switch between picks or fingering methods to get the sound you're looking for. Many direct boxes, designed for bass, have pre-amplification stages that include distortion equalization and tube components that allow you to add some character. The same can be done with guitar using pedals and effects to add warmth and depth to the sound before gets recorded. Otherwise, the only other issues that you are making sure the signal passes cleanly, is full frequency, and that there are no buzzes, hums or noises. Most DI boxes have ground lift switches that help to eliminate these problems.
In the Box Recording
The third method of recording music is in the box recording. In the box recordings are primarily referencing to computer recordings where all of the recording work is done inside the actual recording application. There is no audio coming in externally into the recording device. Recording music inside the box is most often, or at least to some degree, MIDI recording. Essentially, you are capturing the technical aspects of a performance through a midi keyboard or other midi instrument. Once you have captured the performance, you have the ability to grab any sound from the vast number of software synths and sample libraries available and edit them till you get the sound you want. A performance played with a flute sound, can easily be changed or adapted to be a clarinet or oboe sound. This, of course, is not possible with acoustic recordings. The art of this types of recording lays in the ability to make these artificially generated sounds seem like the real thing. When recording music in the box what you are actually recording is MIDI control signals, not the actual audio. This allows you to edit your performance, fix wrong notes or sloppy passages. You can also change the dynamics if you play a note too loud or soft by adjusting the velocity. You can change the length or sustain of notes and countless other parameters until you to get exactly the performance you desire. When dealing with loops, you may be dealing with audio loops or MIDI loops. Audio loops are essentially acoustic recordings or electronic ones that are premixed and effected. You will have limited control in affecting audio loops, which is why the libraries are typically so vast. MIDI loops, by contrast, can be edited and manipulated in exactly the same way that any MIDI performance can including quantization and sound selection.
Moving On
All these methods for recording music are still primarily about capturing performances. To make great recordings, the goal must always be to capture great performances. Great performances will transcend the recording techniques used. Sometimes a low fi recording captures the essence of a performance better than a squeaky clean full frequency one will. Playing with this concept, is truly the art of recording music. Select from the list below for the detailed recording techniques of specific instruments.
RECORDING VOCALS PART 1
The lead vocal is typically the most important part of any song. As a result, recording vocals almost always requires the most attention to the details of performance and sound. Capturing a great performance is a byproduct of preparation, a good recording setup and great communications skills. Because the vocal is the primary focus of most music productions its importance cannot be overstated. The following tips should help to make the process of recording vocals less stressful. To capture a great performance requires as much attention from the producer and engineer as it does the artist.
The 2 Aspects Of Recording Vocals
There are 2 basic aspects to recording vocals. The technical aspect and the emotional and psychological aspect. The technical aspect of vocal recording is simple once you understand the basics principles of audio that most fundamentally affect the sound quality of a recording. These basic principles will help you to make great recordings regardless of the quality of the gear you are using. They set the foundation for all the other techniques and tricks you use. The second aspect of recording vocals is the emotional and psychological aspect. You need to make the performer feel comfortable and confident in what they are doing. This process is supported by creating the space from which they can perform well. The recording studio is a very unnatural environment, and most people don't perform well without some level of inspiration. Many feed off the excitement of an audience or the energy of a live performance with a band. The recording studio, however, is a completely different experience. Careful planning of the technical and psychological aspects of recording vocals is absolutely necessary to get the best performance possible.
The Technical Side of Recording Vocals
The technical aspects of vocal recording often get the most attention in engineering circles, and for good reason. The ability to hear subtle inflections in a performance and the ability to clearly understand the lyric and melody go a long way to adding to the listener's experience. Unfortunately, this is not the whole solution to getting great performances. Discretion must be used when applying the techniques that give you a great 'sound' so as not to put the performer in an uncomfortable position. Ultimately, it is their performance that will make people want to listen, not the quality of the recording. The following sections will break down the technical aspects that make great quality recordings while being sensitive to the needs of the artist. If you ignore this simple principle, you may end up with a great 'sounding' recording that nobody wants to listen to.
Selecting The Best Space To Record
The most important decision to make when recording vocals is selecting the right space to record in. Selecting a space that best supports the sound of the vocal while giving the artist a comfortable space to perform requires some careful attention.
Most people do not listen carefully enough to the sound of the space unless it is doing something obviously wrong. Each person has a unique voice with a unique tonal quality. No one space will work perfectly for every artist and for every song. The decisions made here affect every other level of the recording chain, for better or for worse.
What To Look For
In a professional recording studio, most engineers will record vocals in the biggest space available. The reason has nothing to do with the reverb but rather with way the early reflections will affect the tonal quality of the voice. What happens in every recording space is that sound will travel in all directions from the sound source. The direct sound wave does not stop at the microphone, it continues past it and bounces off all the surfaces. Depending on the shape of the room and the acoustic treatments, the sound will return to the sound source and mic a short time later. How long it takes to get back is critical. If it comes back within 20 milliseconds (ms) it will merge with the original signal and tonally color the sound of the voice. This is important to understand because this will greatly affect the tonal quality of the voice. It can make any voice sound hollow, bright, muddy, clouded or harsh no matter what mic you use.
A Little Math
It all starts with the speed of sound and the distance it has to travel. The rest is simple math. Sound travels at 1130 ft. per second or 344 meters per second. This amounts to .88 ms per foot or 3 ms per meter.
To get past the 20 ms delay time, you must be at least 12 ft. or 3.5 meters away from any surface. The reason for this is that the sound must travel to the surface first before coming back to the mic. The total length of travel from the sound source will determine the delay time. Because of gravity, the reflections from the floor are mostly beyond your control. They will always create a delay within 20 ms. Most engineers use rugs to help limit this transmission. The rest of the surfaces will require a bit more attention. In a large recording space this is not an issue. In a small recording space, it can be a big one. The knee-jerk reaction is to completely deaden the space with foam or absorptive materials, but this is not a truly effective solution. This will create an unnatural balance of low to low mid frequencies that are the bigger source of the problem.
Get Out Of The Closet
I am not a big fan of recording in small spaces. I find they rarely, if ever work, and are largely uncomfortable claustrophobic spaces. Not exactly the best environment to perform in for most artists. From a technical point of view, very small recording spaces create enormous problems that far outweigh the convenience. Surfaces too close to the source signal will create an enormous amount of resonant constructive interference in the low mid frequency range. The result is often a boomy, muddy or flat sound that is unbalanced and unnatural sounding. Covering the surfaces entirely with foam will only serve to further cause imbalances in the frequency response. It's not the deadness of a space that creates the sense of dryness or immediacy. It is the balance of dry to reflected energy that creates that sense. Without the reflected energy in the sound the sense of space is entirely lost and the dry signal flattens out and loses its sense of aliveness. There is a reason nobody records in anechoic chambers. It's important to understand the difference between tonal coloration and frequency response. Tonal coloration comes from reflections that return to the source within 20 ms. Signals this close in time get merged together by our brain. This process is called temporal fusion. Once these tonal imbalances are recorded, they cannot be removed with EQ.
How To Record Vocals In A Small Room
Just because small rooms are not ideal recording spaces doesn't mean that you can't get great results. There are many ways to control the effect of early reflections without sucking all of the energy out of a room. While some of this will involve acoustic treatments, the process starts with something more fundamental. The first step in getting a great sound involves finding the best placement for the vocal. Start by having the vocalist sing the song as you walk around the recording space. If you have more than one space to work with, walk through them all until you find the best sound. Try to focus on the tonal quality of the voice and not the reverberant energy of the room. Notice if the tone becomes boomy, hollow or thin sounding. As you walk around the space take note of where the voice sounds most balanced and natural. This is the best place to start.
Treating The Space
The standard procedure for recording vocals in a professional recording studio is to build a semicircular booth around the vocalist in the biggest room. The booth is created by using gobos. A gobo is a freestanding acoustic baffle that can be easily moved around a room. The ones used for vocals stand at least 6 feet tall. The booth should be large enough to allow freedom of movement by the artist without creating a claustrophobic feeling. A rug is usually set on the floor inside the booth area. In addition to absorbing reflections from the floor it also serves to minimize noise from shoes and vibrational energy transmitting through the stand to the mic. The reason this works is that it minimizes the effect of tonal coloration from the early reflections (less than 20 ms) and also minimizes the reverberant energy from getting into the microphone. It's important to note that it doesn't eliminate them, just minimizes their effect. With a little resourcefulness, a very similar approach can be used in the home recording environment. Suspending heavy packing blankets from the ceiling around the recording area can achieve a similar effect. The difference between this approach and layering the walls with foam is that the sound will get absorbed from both directions. Leaving the booth and returning to it after reflecting off the wall. Layering the walls with foam does not minimize the early reflections nearly as well as the booth and it kills all of the higher frequency reflections that make a recording sound alive and present. By allowing those frequencies to propagate around the outside of the booth, a subtle sense of presence will be added to the vocal recording.
Selecting a Mic
Selecting a mic is the next step in getting a great vocal sound. No mic will undo a poor recording space, but once you have established the best recording space, the mic selection will take your sound to the next level. Microphones are like gloves. There is no one glove that fits everybody perfectly. There are some microphones that are exceptionally good at capturing most people's voices, but every person's voice is still unique. Unless you happen to own one of these very rare microphones, it must be selected uniquely for each person. I've always found the best results were gained by setting up as many microphones as I felt might work for the voice I was going to record. Aside from the time it takes to set up the microphones, it doesn't actually take a lot of time to pick one.
How to Pick the Best Mic
Start by recording a vocal line with each mic. The vocalist should sing a line or two from the song that you're going to be recording. One of the mics will to stand out beyond the others in terms of imaging and tonal quality. If there is a significant change in the vocal range later in the song, it may be worth recording that as well to make sure the mic can maintain the sound. It is typical that tube mics and condenser mics are selected most often for recording vocals because they give the most clarity. With voices that are very bright, a dynamic mic can also come in very handy. Dynamic mics can cut away some of the harshness of a voice and add some warmth and body when needed. It important not to rule out a mic because of its type or price tag. The most important part of recording vocals is getting the sound you're looking for no matter how you have to get it. If you need to, make the test blind so that you are not swayed by preconceived notions of quality.
Pop Filters
One of the many technical issues with recording vocals is plosives. A plosive is puff of air that is sometimes emanated by the vocalist when singing words that contain the letter P. This puff of air can strike the diaphragm of the microphone with enough force that it causes a low frequency pop or distortion. A pop filter, or pop screen, can be used to break up the puff of air while still allowing the sound to pass through to the microphone. If you don't have a pop filter handy, another way of getting rid of plosives is by taping a sharpie to the front of the microphone. It should go right down the center of the diaphragm. The sharpie will spread the air out around the diaphragm without really affecting the frequency response. Pop filters can serve a second valuable purpose even if there's not a big plosive problem with your vocalist. A pop filter also allows you to set a distance from the microphone to which the vocalist can easily and consistently return. It is very important to keep a vocalist singing at an even distance from the microphone. The distance is important for creating a consistency in terms of frequency response and tonal characteristic. If the vocalist is pulling away or moving around or turning their head when singing then will not be singing directly into the diaphragm. The result is variations in the tonal quality of the voice.
Mic Placement
The real issue with microphone placement for recording vocals is a matter of comfort for the vocalist. If the microphone setup feels obstructive to the vocalist it will take way from their performance. Doing everything possible to make the artist comfortable will always yield the best results. The primary focus of your setup will be to make sure that the vocalist is able to perform comfortably with good posture. The professional method of recording vocals involves the use of a big boom stand. The boom stand is set up out of the way to the left or right of the vocalist. The boom stand extends above the head of the vocalist and drops the mic from above down in front of the mouth of the vocalist. This typically works very well because it keeps the microphone out of the way and allows the vocalist to maintain good posture. As a general rule, the artist should never be seated when recording vocals. If standing is a problem for the artist, then the next best option is to use a stool. This way, they are at least mostly standing up. This can also be handy if you have a long session planned and want to keep the vocalist fresh.
Lyric Sheets
There is one issue that often arises when the microphone is coming from above the head of the vocalist. If the vocalist needs to look at a lyric sheet, the microphone is directly in the way of their sight line. What happens is that the lyrics will either be placed to the left of the right of the microphone. This naturally leads the vocalist to turn away from the mic and not directly into it when they look at the lyrics. There are 2 ways to deal with this issue when recording vocals using lyric sheets. One way is to set up a mic stand so the microphone can be flipped upside down and come up from below the head, not above. This way they can look straight over the top of the microphone to see the lyrics. As the singer looks at the lyrics they are actually leaning into the microphone not looking away from it. The pop screen will help to keep them at the exact distance you want them to be from the front of the microphone. Sometimes it is best to set up a vocal mic coming from the side if you have a quality heavy duty mic stand that can handle the weight of a good microphone. The benefit of this setup comes when using tube mics. The tubes will generate heat that naturally goes up towards the diaphragm if set upside down as in the previous example. This can negatively affect the quality of the sound because the diaphragm will be heated and expand thus changing its performance characteristics. Using a side position will allow the vocalist to see the lyrics and sing directly into the microphone while also keeping the heat from affecting the sound.
Mic Settings
There may be a series of options on your vocal microphone that will help control the sound quality when recording vocals. These options vary from mic to mic, but here is a list of the most common ones found on quality vocal mics.
• Polar Pattern
• Filters (EQ)
• Pads
Polar Pattern:
The polar pattern of a microphone determines the direction from which the microphone is most sensitive to sound sources. The selection of the polar pattern will also determine from what direction, if any, the mic will reject signals. When recording a single vocalist, the pattern is typically set to cardiod. The cardio pattern will, to greater or lesser degrees, reject signals from all directions except from directly in front of the mic. This polar pattern is most suitable for recording a single vocalist. When recording two vocalists with at the same time one mic, a figure 8 pattern will allow equal sensitivity from both the front and back of the mic while rejecting sounds from the sides. This pattern makes recording more comfortable because they do not have to crowd around the front of the mic. When recording a group of vocalists through one mic, it is typical that the omnidirectional polar pattern is selected. the selection of Omni enables equal sensitivity from all directions. This allows the vocalists to comfortably sing toward the mic from every direction.
Filters (EQ):
Filters are a very powerful equalization tool used to eliminate problematic frequencies that are not a necessary part of the sound source. There are 3 basic types of filters, High Pass, Low Pass and Notch. While all three types are helpful, only the high pass filter type is used for recording vocals. It is the only type you will typically find on a mic. Many large diaphragm microphones use a High Pass (low cut) filter because of their increased sensitivity to low frequencies. This allows the engineer to dramatically remove low frequency rumble that can be caused by air conditioning systems and poor acoustic isolation. Some have selectable frequencies but most are fixed to a single frequency below which signals are cut. The quality of these filters vary with the quality of the mic and it is common for many mic preamps and audio interfaces to include a High Pass (low cut) filter. Check to see which sounds best without affecting the quality of the vocal recording. If the low frequency rumble in a room is excessive, it may be necessary to use both if there is no way to eliminate the source of the rumble.
Pads:
A pad allows the electronics of the microphone to be buffered from sound sources that create high sound pressure levels. This does not necessarily mean that you won't get distortion as the diaphragm of the mic may not be able to handle the excessive sound pressure levels. If this is the case the mic will need to moved farther from the sound source. For most vocalists a pad is not necessary, and it is often better to back the vocalist away from the mic if they project too loudly. This is a very common practice when recording opera singers that are trained to project with enough power to fill an opera house.
Controlling Dynamics
Headphones
Although the headphones are technically on the performers side of the recording, their control is almost always the dominion of the engineer. Here are some tips that will help to make your headphone mixes best for recording vocals. The best place to start is by using the best quality headphones you can find. You want to avoid "open back" headphones because they are vented and the sound will bleed into the mic. Make sure you have a good cleanly amplified signal with no distortion. When preparing a mix for the artist, try to make the mix as dynamic and alive sounding as possible. The mix should be exciting so that the artist can feed off the energy of the song the same way they would in a live performance. Be very attentive to how the artist feels about the mix. Make sure they are able to hear themselves clearly and cleanly. I generally try to avoid using reverbs when recording vocals because it makes it harder to hear pitch accurately. If the headphone mix does not feel right to the artist without reverb, then add in just as much as is needed to add the presence they need to perform well. Try to avoid long washy reverbs if possible. If the artist is having problems with pitch using headphones, have them take one headphone off so they can tune acoustically. This is a very common solution to solve pitch problems when using headphones. Create a mono mix and cut signal to the unused headphone so that it doesn't bleed into the mic.
Controlling Dynamics Acoustically
There are many different thoughts about how to best control dynamics of a vocalist in the recording room. It is typical that the artist will sing louder in different sections of the song if the melody takes them to the power range of their voice. Most vocalists will pull away from the microphone when these parts arise. While this is very dramatic looking in concert it is not always great for recording vocals. Generally, I would rather have them at an even distance from the mic throughout the performance if possible. If the artist will give me a better performance because they don't have to think about staying in one position, I will deal with the sound later. Always take a quality performance over a quality sound if there's no way to get both. A well trained vocalist with good technique will not necessarily sing louder when going to the power range of their voice. Unfortunately, this is the exception rather than the rule and you will have to deal with these issues as part of the recording process. Never force an artist to think about technique when recording vocals. It is always better to work out the technical issues of a performance in a rehearsal session so that the focus is entirely on expression and feeling in the recording session.
Mic Preamps
Selecting the best quality mic preamp is the next stage of the vocal chain. A good mic preamp will have loads of headroom and will not distort if the vocalist belts away. Compare as many mic preamps as you have available to find the one that best suits the vocalist.
Always leave yourself a good bit of headroom, especially with very dynamic performers. You can always make up again at a later stage in the recording chain, but you will not be able to get rid of the distortion. If the gain is too high for your mic pre, you may need to use a pad. Most mic preamps have a pad, but make sure you compare the quality to the one on the microphone to see which sounds best. If necessary, you can ride the mic preamp gain during the performance to help even out the gain. Make every attempt to eliminate distortion at every phase of the recording chain. Be careful to monitor the gain as the voice starts to open up. Usually, a vocalist will not sing with full power until they are warmed up and singing the song full force. Be prepared...
Compression and EQ
If all of the previous details have been considered and brought into focus, what happens with the EQ and Compression should be a breeze. There are many thoughts regarding what the processing order should be after the mic preamp. Here is my general view:
EQ Before Compression
Subtractive EQ is best before compression, additive EQ is typically best after the compression. The reason for this is simple, subtractive EQ is meant to eliminate noise that you don't want. If you don't get rid of this noise before compressing then the compressor will make the noise louder and it will be harder to remove later. The most typical form of subtractive EQ is a high pass or low cut filter. The purpose of this filter is to roll off low frequency rumble or noise that is below the frequency range of the voice. Many vocal mics have this filter built into a switch. It is also common to find a filter stage built into the mic pre. Make sure that when the filter is engaged, it does not roll off frequencies from the low end of the voice. You may need to check the specifications of the owner’s manual to verify the frequencies if it is not labeled on the mic or preamp.
EQ After Compression
EQ after compression is typically additive. If you have followed all of the techniques in the prior 2 articles leading up to this point, you should need very little if any EQ. If the recording setup is too limited to accommodate all of the techniques outlined in the previous articles, then some EQ may be necessary to make up for what is missing. More on this later in Recording Vocals part 4.
Compression
Generally, most engineers agree that compression, when recording vocals, should be as transparent as possible. You don't really want hear the compression you just want to control the dynamic of the vocal. There are many ways to approach doing this, let's take a look at a few.
One Compressor Approach
One very simple approach to getting transparent compression and tighten up the dynamic of a performance is to set the ratio to a very low setting 1.5:1 or lower. Set the set the attack and release times to a medium setting and then set the threshold until you get a consistent 2-3 dB of gain reduction. The idea of this approach is to have the compressor consistently working. Most compression is perceived when overused or when it kicks in intermittently. This is especially true when the compression kicks in aggressively for high peek signals. Adjust the attack and release time until it feels musical. The attack and release times will typically mirror the tempo. Faster attack and release times for a faster tempo, slower attack and release times for a slower tempo. This very general guideline and must ultimately be based on the approach of the performance.
The Compressor Limiter Approach
It is a very common approach to use a limiter and compressor or two compressors in series to control the dynamics when recording vocals. The basic approach is very simple. Set the limiter, or first compressor, with a high threshold so that it only captures peak signals. This will help to control the amount of gain going into the compressor so that it does not have to respond to the high peaks. If using a compressor, you will want to use faster attack and release times with a high ratio to emulate the action of a limiter.
If you set the limiter stage up correctly, it should control the peak signals and allow you to set the compressor to yield consistent gain reduction. This will allow the compressor to better focus the performance.
Additive EQ and Interfaces
Okay, this is the home stretch! As we have learned, recording vocals involves attention to the fundamental principles of acoustics. We started by selecting the best recording space and treating it acoustically to optimize the sound of the voice. Next, we looked at the process of selecting the best microphone and how to set it up so that the artist can perform comfortably. Adjustments to the setup may be necessary to accommodate the use of music stands and pop screens for plosives. With the vocal setup complete, the next step was learning to control the dynamics of a performance both acoustically and electronically to keep consistency in the performance. We also looked at the recording chain and how best to apply compression and subtractive EQ. In this article, we will take a closer look at the use of additive EQ and how best to deal with recording interfaces when recording vocals. So without further ado, let's dive in to the tricky world of EQ for vocals.
EQ
I strongly believe that compression is a better way to add presence to a voice rather than EQ. If you have set up your compression well, this will be apparent in the vocal sound and the need for EQ should be minimal, if at all. If the sound is still not what was expected and I have exhausted all of my options with the setup, I will look at adding EQ. The best approach is always try and remove what you don't like first before using additive EQ. Unless I am completely convinced that the EQ I've added is exactly the sound I want I will leave it out of the record chain. If you are not 100% sure about the EQ, it is always best to leave it for later where it can approached with fresh ears.
Adding Presence
The most common reason for additive EQ when recording vocals is to add presence. Presence frequencies generally live in the 2-6K range. Unless this area is particularly deficient, I usually try to avoid adding frequencies here because you will most likely start to accentuate sibilance. Sibilance is a pronounced peak in frequencies that are usually heard when sing words with the letter S, T and a soft C. De-esser are used to rid these problems if they cannot be dealt with using EQ. De-esser are very fast limiter stages that are keyed by these frequency areas. As a result, the limiter stage only kicks in when these frequencies are overtly present. A safer way to add the feeling of presence without adding in the extra problem of sibilance is to add air to the voice. This is easily accomplished with a shelving EQ at around 10K. The shelving EQ will affect all frequencies above the selected frequency. Adding frequencies in this range will brighten and raise the vocal up in the speakers. Adding frequencies in the 2-6K range will draw the vocal out of the speaker toward you but may also draw out problems. The best approach is really a matter of taste, the style of music and the meaning of the song. Remember that there are many ways to add presence to a vocal with reverb or effects. If you are not convinced that your EQ is perfect, leave it for later.
Adding Warmth
The best way to add warmth to a voice is to move the vocalist closer to the mic. The proximity effect will add some natural warmth and body to the voice without adding muddiness. If this doesn't work it is worth considering a dynamic mic that is used for radio broadcasts.
Broadcast mics will add warmth to the harshest of voices and tone down sibilance problems as well. If a broadcast mic is not available, then you may want to consider adding the low end in before the compressor instead of after it. Adding the low end before the compressor allows the compressor to help make the added low end integrate better with the rest of the sound. It will also allow you to get better results with less EQ. In general, this is a problem best addressed in the mix session where more tools will be available to you with less pressure.
Using Interfaces With Mic Preamps
For many of the home recording enthusiasts, the only mic pre available is the one in the interface. The quality of the mic pre will vary with amount of money spent of the interface. If the interface is connected to an AC outlet you are more likely to have some headroom for recording dynamic vocals. If the interface is bus powered by USB or Firewire then you will have more difficulties recording vocals without distortion. The reason for this is very simple. The dynamic range of a mic pre is determined by the quality and strength of the power source. Many high end mic preamps will have a separate transformer used to clean up and convert the power source feed to the electronics to optimize their performance.
Dealing With Buss Powered Interfaces
If a USB or Firewire powered interface is the only option for recording vocals, good results can still be achieved with a little ingenuity. The best place to start is to keep the mic preamp as low as possible when recording so that the preamp does not clip.
Although, this approach in not optimal for signal to noise ratio, at least the vocal will be distortion free. The remaining gain issues can then be better dealt with in the box. If your interface has an insert point available, you can use an external compressor to help control the dynamics of the vocal performance and add the necessary gain before going into the ADC of the interface. This will help to create some consistency with the levels. An alternative approach would be to ride the mic preamp, in real time, with the vocal performance to prevent overload. This can be a bit tricky, however, if you are familiar with the song and vocalist. If you are familiar, then this can be a nice alternative way of controlling the gain before conversion to digital.
Conclusion
The best results for recording vocals are always achieved by getting the fundamentals straight. Start with the best sounding room and treat it well so that the voice sounds focused, full-bodied and dynamic as possible. Taking time to select the best mic will go a long way to bringing the sound to the next level. Make sure the artist is comfortable so that you get the best performance out of them. Adjust the mic setup if necessary to accommodate their needs. Select you processing stages carefully and only use compression and EQ if you are getting the sound you are looking for. Never force processing on a sound if it does not sound or feel right. Remember that as long as you can get a good clean distortion free recording, everything else can be dealt with at a later time with less pressure. Experiment with these ideas until you find the setup that works best for you. Every situation is unique and no one setup will work for every situation. The purpose of this article is to focus you on the fundamental aspects of recording vocals that have worked professionally for decades. I hope you found this series of articles on recording vocals helpful !!!
Becoming an Audio Engineer
The audio engineer is perhaps the most unheralded person in the recording studio. The impact he or she has on the outcome of any production is incredible. Every decision made regarding how a performance is recorded, stored, edited, processed and mixed can have a tremendous effect on the final product. It's no wonder that artists and producers select their engineers very carefully. Becoming an engineer is journey that takes years to develop. Having good technical skills is only part of the equation for success. You must build a bond of trust with the clients you work with on every level. The client must be able to count on you when everything else is falling apart around them. The confidence you exude in a session will go a long way to making the client feel comfortable so they can focus on their role in the production process. If you are interested in becoming an audio engineer, you must exhibit skills that include a meticulous attention to detail. In addition to learning the technical skills of engineering, a recording engineer must be organized, communicate well with people and be attentive to the needs of the client. Most importantly, you must be a good problem solver.
Experience is the Key
The majority of skills necessary to become a recording engineer can only be gained by the experience of doing it over and over again. The best way to learn is to get a job at a recording studio and learn from the people who make a living at it. People who make a full time living as an engineer are generally, though not always, free of most of the bad habits that plague the novice. Remember that you can gain valuable experience by watching others work. This is the primary role of the assistant engineer in a recording studio. Pay careful attention to what the members of a recording session are doing, how they act, and what the end results are. You can learn just as much from a bad engineer or producer (what not to do) as you can from a good one. Then, take what you have learned and try to do it yourself in your own projects. This will tell you how much you really understand about the engineering process. The term audio engineer is one that is generally used in the context of studio recording. Audio engineers, however, can take on a great number of roles that are not specific to music production. The following, is a list of some of the roles taken by engineers in their most common career paths.
Studio Music Production
• Tracking Engineer
• Overdub Engineer
• Editing Engineer
• Mixing Engineer
• Mastering Engineer
Live sound Music
• Front of House Engineer
• Monitor Mixer
• Location Recording Engineer
Theater
• Sound effects engineer
• Stage Sound and Front of House Support
• Recording Engineer
Broadcast TV
• Dialog Engineer
• Location Engineer
• Sound Effects Engineer
• Mixing Engineer
• Broadcast engineer
Film
• Location Recording
• Foley Engineer
• Sound Effects Engineer
• Sound Design Engineer
• Mix Engineer
Radio
• Broadcast Engineer
• Voiceover Engineer
• Location Engineer
Cable TV
• Location Engineer
• Voiceover Engineer
• Sound Effects
• Sound Design Engineer
• Mixing Engineer
Video Games
• Sound Design
Advertising
• Recording Engineer
• Voiceover Engineer
• Sound Effects
• Mixing Engineer
Many other roles for audio engineers exist in the world today. Almost everything that generates sound has run through the hands of an engineer somewhere in the process of development and production. Engineers are often hired as consultants for the development of products like cell phones, mp3 players, radios, headphones, speakers, home stereo systems, car stereo systems, microphones, pro audio gear and software. The list goes on and on…
Technology and the Audio Engineer
Audio engineering is an art form that is typically appreciated only by audiophiles and those who engineer or produce music for a living. Behind every engineer is the technology that helped create the sounds we hear. Ask the average person to name a recording engineer and they will most likely have no reply. Ask them to name any of the equipment used to make a recording and they will be equally dumbfounded. Although the names may not be known by those outside the music industry, the impact they have had on music production throughout the decades is immeasurable. The expertise and ingenuity of the audio engineer has brought many artists to the forefront of the industry. In many cases, those accomplishments were backed by advancements in recording technology. The Beatles, for example, were known for many amazing creative and technical feats in the recording studio. The vision of George Martin and the genius of the Fab Four had to be realized in physical form by engineers who found ways to make that vision a reality. They stretched the boundaries of what was possible, by embracing new technology, and made history in the process.
1900-1940's: Vinyl Discs and cutting lathes rule into the late 40's
During this era technology was fairly limited. Even though there were major technological developments, their use in the recording studio was limited by the marketplace. Vinyl disc sales had not quite grown big enough to support large recording budgets and thus support radical change in the design of recording studios. As you will see in the coming decades, this outlook would change dramatically. All records, during this period, were largely made the same way with the same recording techniques and recording technology. The audio engineer, sporting a white lab coat, was not generally considered part of creative process. The audio engineer was primarily there to just capture performances. Artists, composers and arrangers were largely responsible for the success or failure of their production. At this point in history, the limitations of cutting lathe technology did not allow the audio engineer enough latitude to enhance the artist's performance by any great measure.
1950's: Analog tape machines replace cutting lathes in the recording studio.
Analog recording technology was developed in the late 40's but its true impact was not felt until the 50's. The physical limitations of vinyl were coming to a head. Performances captured on cutting lathes for vinyl production were limited by the time available on a disc side, the amount of low frequency content, and the dynamics of the performance. Any of these basic issues, out of balance, could easily render a beautiful performance destroyed. Analog tape changed these parameters dramatically. At worst, performances that were too long, bass heavy or with excessive dynamics might require editing or suffer from some distortion or tape compression. Multiple performances or takes could be easily edited together to make one better performance. Performances that would not fit on one side of a record could be easily split between Side A and Side B or edited in length to fit on one side of a vinyl disc. The transfer engineer, now known as mastering, was born.
1960's: Multitrack recording technology and the release of stereo recordings.
The 1960's saw the full realization of stereo technology that was created in the 50's. The recording technology that emerged from the 60's would change the way recordings were made forever. While consumers were enjoying stereo on vinyl discs, recording engineers were working with multitrack recording. Multitrack recording allowed individual instruments to be recorded on separate tracks. Once separated they could be processed individually when mixed into stereo for the commercial release. The Mix engineer's position was born.
Sel-Sync multitrack recording (selective synchronization) allowed the audio engineer to rerecord individual performances synchronously with other tracks on the same tape machine. This would allow the vocalist to rerecord their part if the band captured a perfect take but the vocal performance was not up to the same standard. With careful forethought, It would also be possible for additional parts to be layered. Harmonies, doubles and additional instruments could be added to a performance to enhance or sweeten the sound of the recording. The term "overdubbing" was now part of audio engineer's vocabulary. This was a truly revolutionary change in the production process. The ability to separate and layer performances would grow exponentially in the coming years. It would expand the time artists spent in the recording studio dramatically. The early albums of the 60's might take a few days to complete. By the end of the 60's those same records would take weeks or even months to complete. The job of the audio engineer was taking on a greater role in the recording studio. As recording technology was getting more complicated, so too was the role of the engineer. The engineer, once seen as a technician only, was taking on a much more creative role in the music production process.
1970's: Expanded track counts lead to larger recording consoles and studios. What happened in the 70's was an explosion of technological development that saw track counts rise and recording consoles get larger. Parametric equalizers and compressors were stock features of professional recording consoles. New microphones, compressors and equalizers were entering the studio for external processing. Companies like Lexicon and EMT brought digital reverb and effects processing into recording studios with the EMT 250 and the Lexicon 224. The creation of digital effects processing would become a major part of the mixing process. New recording studios, like the world famous Power Station, were being built with isolation booths for better separation of instruments in the multitrack recording environment. Studios were being designed to record specific styles of music. The audio engineer, once an employee of the recording studio, would start to become a commodity for artists and producers. Seeing the benefit of having a great engineer, artists would start to hire the best engineers to work with them away from their home studio. The freelance engineer would become a force in the recording industry.
1980's: The compact disc, midi, synthesis and digital recording.
The 80's saw a largely unwelcome guest enter the recording industry. The introduction of the compact disc in 1980 changed the way people listened to music and brought digital technology into the recording studio. The CD was a huge success on the consumer level and ushered in a huge influx of money into the recording industry as record companies reaped the profits of reselling every previously released vinyl album in compact disc form. Most of the recording community thought it an abomination compared to the much warmer and pleasing vinyl disc. While digital technology took away the clicks, pops and skipping of the vinyl disc it also brought a cold clinical sound that was hard to swallow by professionals. Those that embraced digital technology were served well in the long-run though. Digital offered many advantages over analog, including increased dynamic range, no tape hiss and the ability to make exact copies of tracks without loss of quality. The 80's also saw midi sequencers enter the studio, allowing performances to be captured and edited until perfected. The influx of synthesizers, drum machines and samplers would usher in a whole new style of recording studio that would embrace synthetically generated sound over real acoustic instruments. Smaller recording spaces and larger control rooms would accommodate a new breed of client for recording studios, the programmer, the DJ and the electronic musician. Digital multitrack tape machines called DASH machines entered the recording studio as an alternative to analog tape machines. Although initially realized in 2 track format, these DASH machines would eventually accommodate up to 48 tracks of digital recording on 1/2 inch tape. The built-in self-synchronizing technology would allow for 96 tracks of recording capability by simply locking 2 machines together. It became evident that the much slower and limited 24 track analog tape machine was beginning a steady decline from which it would never recover.
1990's: The nondestructive recording and editing capabilities of computers
The 90's saw an explosion of radical change in digital recording. The recording industry would peak in the mid to late 90's and then start a radical fall heading into the 2000's. Computers would become a powerful force in the industry and eventually supplant all the major recording console and tape machine manufacturers as the driving force of the recording industry. The relatively cheap technology and radically enhanced editing and mixing capabilities of Pro Tools systems would allow many producers and artists to take their work into their home studios. Many commercial recording studios would close their doors forever as a result. Recording studios would also take a hit from the record companies that were lowering their recording budgets due a decrease in CD sales. The growth of the internet and the creation of file sharing websites like Napster would see piracy reach a new level never seen before in the music industry. A recording industry that once ruled with huge recording consoles and expensive tape machines suddenly had to change to a new model. This model would prove difficult to achieve as many of the big recording studios could not survive a dwindling client base and lower recording budgets. The ones that did survive now serve the high profile recording artists that have the budgets to accommodate.
2000's The explosion of new software and diversification of recording technology.
The first decade of the 21st century saw a technological explosion that has rocked all media industries including the recording industry. Increased processor speeds and hard drive capacities have made home recording a viable option for everyone. For very little money, anyone could compose, record, edit and mix their own music. The result of these rapid changes in computer technology is evident in the diversity and number of new recording software applications. Recording software design was targeting very specific markets like DJs and beat writers. Thus, filling a void left by programs primarily designed for engineers and musicians. Keeping up with this rapid growth has been a difficult task for the audio engineer. Hardware technology, once designed, would never change its signal-flow unless modified by the chief technical engineer of the studio. A recording console once learned, was learned forever. Software though, is very different matter. While the merits of software updates to fix design flaws is great, it also makes it more difficult for the new user to learn. Each update to a software program adds new features for the long-term user but also creates a steeper learning curve for someone new to the program. For this reason, a simple interface design is critical to a software's success.
The rapid development of recording software has challenged the traditional way of making records. The ability to create music inexpensively changed the way audio engineers have gone about their work. The audio engineer of today will find themselves in many nontraditional recording situations. As a result, audio engineers have been forced to be more creative in their approach to recording in an attempt to maintain a professional quality recording. Recording in these non-traditional situations requires a lot of professional recording studio experience. There are very, very few home recordings that merit a professional quality standard. The reason is simple, unless you have had a great deal of experience working in professional situations, you will have no clue what is required to create that sound. An experienced engineer working at home would create a significantly better product than the inexperienced engineer in a pro studio. Many today, aspire to become an audio engineer but few understand all that is involved. Having spent many years teaching students the art of engineering, I have come to realize the depth and enormity of information about recording audio. As you can see, an incredible amount of technology has been developed for just this purpose. Each development trying to improve on and solve the problems facing every audio engineer on a daily basis.
Denis van der Velde
AAMS Auto Audio Mastering System
www.curioza.com
Denis van der Velde
AAMS Auto Audio Mastering System
www.curioza.com
The Daw and files, processing, plugins.
Use 32 Bit float all the time, if possible. 24 and 16 bit files need to have a brick wall limiter and at the end process need dithering. Allow the highest possible bit depth. Do not dither 32 bit files. More bits more sound. As long as audio data remains in the 32 bit floating point resolution, your good, do not worry about potential over’s. With 16 or 24 bit levels above 0dB lead to artifacts. So maybe convert you whole sample database to 32 bit? What happens when a 16 or 24 bit file is saved as 32 bit? Free dynamic range! Always use the best possible resolution 32 or 64 bit float. Or convert. Use only 32 Bit floating point plugins (check this with a bit depth meter).
Live Recordings.
Only when it is clear that the bottom end frequencies are troublesome a low cut from 0 Hz to 30 Hz can be allpied. Sometimes even above 70 Hz for Vocals and Microphones. Real adjustment of sound is not recommended, live recordings must sound as played. Little EQ and Compression.
The bottom end of the mix.
The bottom end of a mix is fundamental. The headroom on the signal is the space between 0 dB. Strong low frequencies from 0 Hz to 100 Hz and lesser from 100 Hz to 1Khz are multiple times stronger then the signals between 1kz-22khz! Low frequencies makes a muddy mix and without power, even if all lower frequencies including the basedrum are maxed out, still there will not be a nice base bottom end. The problem is that sounds ranging from 0 Hz to 50 Hz are not heard but feld. Also up to 100 Hz most monitor speaker do not really play it like it should be heard. Headphones ofthen play from 20 Hz to 22KHz easy. Sub monitors have good quality to about 25 Hz. The human hearing accepts frequencies till 50 Hz more as a feeling experience then a hearing experience. Bottom end signals make a big factor in a mix, while high frequencies strongly loose power. A badly mixed bottom end cant be mastered, a good one with ease.
The Haas Effect.
Before we start discussing Headphones or Monitor Speakers, i would like to adress the Haas Effect. The Haas effect is a psychoacoustic effect related to a group of auditory phenomena known as the Precedence Effect or law of the first wave front. These effects, in conjunction with sensory reactions to other physical differences (such as phase differences) between perceived sounds, are responsible for the ability of listeners with two ears to accurately localize sounds coming from around them. When two identical sounds (i.e. identical sound waves of the same perceived intensity) originate from two sources at different distances from the listener, the sound created at the closest location is heard (arrives) first. To the listener, this creates the impression that the sound comes from that location alone due to a phenomenon that might be described as "involuntary sensory inhibition" in that one's perception of later arrivals is suppressed. The Haas effect occurs when arrival times of the sounds differ by up to 30–40 milliseconds. As the arrival time (in respect to the listener) of the two audio sources increasingly differ beyond 40 ms, the sounds will begin to be heard as distinct; in audio-engineering terms the increasing time difference is described as a delay, or in common terms as an echo. The Haas effect is often used in public address systems to ensure that the perceived location and/or direction of the original signal (localization) remains unchanged. In some instances, usually when serving large areas and/or large numbers of listeners, loudspeakers must be placed at some distance from a stage or other area of sound origination. The signal to these loudspeakers may be electronically or otherwise delayed for a time equal to or slightly greater than the time taken for the original sound to travel to the remote location. This serves to ensure that the sound is perceived as coming from the point of origin rather than from a loudspeaker that may be physically nearer the listener. The level of the delayed signal may be up to 10 dB louder than the original signal at the ears of the listener without disturbing the localization. The Haas effect is also responsible in large part for the perception that a complete complex audio field is reproduced by only two sound sources in stereophonic and other binaural audio systems and it is also utilized in the generation of more sophisticated audio effects by devices such as matrix decoders in surround sound technologies, such as Dolby Pro Logic. For a time in the 1970s, audio engineers used the Haas effect to simulate that a sound was coming from a single speaker in a stereo sound system, when it was actually coming from both. This was to compensate for the fact that a sound coming from a single speaker would be 3 db lower in volume than a sound coming from both. This technique has problems if the stereo sound is mixed to mono, as a comb filter effect would occur. Also, the aesthetics of sound mixing changed to exclude the use of solo instruments emanating from a single corner of the sound field in most popular recordings. Named after Helmut Haas who described the effect in his doctoral dissertation "Über den Einfluss eines Einfachechos auf die Hörsamkeit von Sprache" to the University of Göttingen, Germany. An English translation was published in December, 1949.
Headphones.
Before starting with Monitor Speakers, let's adress the headphones first in line. Composition wise headphones are just as good as using Monitor Speakers. But while mixing headphones might be a solution to many users, because it does not produce complaining neighbours or you might love the direct sound headphones produce. The direct single user headphone sound might be refreshing because the sound is directly inputted into the ears of the headphone listener. Actually while mixing on headphones, there is a lot of information lost. As an opposite to Monitor Speakers, you will miss out on the room you are listenging in. Basically this would be the least of your concern now, but by the end of reading this text you might think different. Headphones normally produce a direct sound, basically cancelling out the room (listening room as would be on loudspeakers). So the headphones can produce a sound that is only comming from the left, as this left sound goes directly into your left ear and hearing nothing on the right side. This is an impossible situation with loudspeakers, in this case there will be allways the room you listen will reflect sound as into your right ear. Also when using headphones and the sound just comes from one single side (left or right) it is allmost impossible for our brains to hear any distance (reverberation) in a correct manner, even a good headache can occur when headphones are used for a long time. Listening on headphones can be confusing while mixing frequency wise (dimension 2), it is likely you will reduce the mix more then you would do on Monitor Speakers / Loudspeakers. You will come to find this out when you listen your mix on different systems (i.e. listening on several speakersystems and headphones). Listening on as many sound systems is advisable anyway, but a special remark here when using headphones. Listing to distances, setting up a delay or reverb, creating a stage (a spacious domain), playing around with delaytimes , using the Haas Effect (Dimension 3), will be more difficult when using headphones. Basically setting up volume levels is a job that can be done with headphones. When you try to setup panning for each instrument or mixtrack (Dimension 1), you will notice that the direct sound of the headphones will produce a different sound-stage-setup, as you would with using Monitor Speakers. Try this if you have a good pair of Headphones and Monitor Speakers available. When you listen on Headphones (A) and suddenly switching to listing to Monitor Speakers (B), you will notice that the panning of some or all instruments change from slightly to more. The Monitor Speakers will produce a more open and reliable sound reproduction in Dimension 1,2 and 3, while listening to Headphones you will get the feeling you will miss out and mismatch direction. Basically to the direct sound Headphone produce, they can be generally only be used while for a vocalist or guitarist, bass, drums, as a prevention of the mix comming into the recording signal. While using Headphones for mxing the Haas Effect is cancelled out because of the direct to ear sound. The Haas effect occurs when arrival times of the sounds differ by up to 30–40 milliseconds, this is explained next under 'The Haas Effect'. And we have not discussed the bottom end yet! Having resolved spatial issues, bass levels provide the main obstacle for mixing on headphones. Deciding how much bass sounds 'correct' on headphones is a big problem because, although you hear bass through your ears, you don't get the physical full-body feelings that you do from the bass that emerges from loudspeakers. Regularly comparing your in-progress mix with commercial tracks of a similar genre always helps, but the bass end on many cheaper headphone models does not sound like the bass you will expect to hear from loudspeakers, so you can easily misjudge it. As a result, it is quite possible to end up with a mix where the bass guitar and kick drum levels seem to be correct, yet they sound 'bloated' when heard over speakers, with too much bass at 80 Hz and below and paradoxically, too little in the next octave between 80 Hz and 160 Hz, where your headphones offer much greater clarity. So my advise is to use Monitor Speakers anywhere you can and as ofthen as you can while Mixing or Mastering. Even for recording purposes a mixing engineers benefits from listening to Monitor Speakers in a control room. Use Headphones when you have little room, in a very small studio or workplace. Or as an effective way to cancel out the mix sound while recording with a microphone. For mixing or mastering purposes, prefer Monitor Speakers to finish the job, Headphones will not give you a clear mix and you will struggle in dimension 1,2 and 3. For setting up a mix as a stage, Monitor Speakers will produce a better and more understanable view to the dimensions 1,2 and 3. Ok, try to avoid using headphones, as an opposite there are plenty of mixing and mastering done with Headphones. So be aware of the differences and learn the differences between Headphones and Monitor Speakers. When understand that the common listener to commercial music is still listing on speakers (allthough walkmans or ipod's are quite commen these days) and that speakers setup correctly will produce a better listening envoiroment, it is easy to prefer Monitor Speakers and Headphones are to be avoided.
Audio Plugins that help out headphone listening.
HDPHX
VNoPhones
Crossfeed EQ
Roomsound
Monitor Speakers.
Use good reference flat monitor speakers. The smaller ones can give a good frequency range from 50 Hz to 22 Khz. The slightly bigger monitor speakers go from 35 Hz to 22 Khz. Be sure where you gain the bottom end, you might not even hear what your doing because your monitors will not play them well. Best Monitor Speakers will produce a good flat sound from 25 Hz to 22 Khz (or more). When your Monitor Speaker setup misses out on the bottom end range, get a good sub speaker. Generally a good sub speaker (placed in the middle or centre) can result in better frequency range because the bigger the sub speaker becomes the lower the frequency range. Most sub speakers will start at about 30 Hz and end at 120 Hz. Placed in centre the sub will produce a more centered placement for Drums and Bass (Basedrum and Bass), these are normally placed in centre anyway (stage plan). It is good to listen a lot of commercial available music on your monitors. Just to know how your monitors react. Also for setting up the sub speaker in comparing to the left and right monitor speakers, you must be certain of a nice flow (flat frequency range from 30 Hz to 22 Khz). The sub speaker will have and adjustable level and cutoff frequency, where at about 120 Hz or above you will need to have a crossover point, flowing most 30 Hz to 120 Hz frequencies out of the sub speaker, while crossing over at about 120 Hz to 22Khz on your monitor speakers (left and right). When you have done a lot of listening to your monitor speakers, listening on other speakers or your friends stereo system will let you hear the differences and will make your hearing adjusted to the sound of your control room or listening envoiroment. Also Surround sets are not really a good listening envoiroment for Stereo listening. When you have enough experience, thrust in your monitor speakers and trust on your hearing, do not listen any more on other speakers but stay with your monitor speaker setup. Stereo sound is made on speakers (not on headphones), for correct hearing dimension 1,2 and 3 a good monitor setup is prefered. Be in a good mood when you mix or listen. Monitor also on low levels. Every mix will sound good on loud levels, the level is better heard, different sounds are better seperated, the loudness and peaks are better heard. When you mix loud, it could be when you listen to it softly on low levels, there is nothing to hear or your mix will deflate. A good mix will listen also on low levels. A good mix produced with monitor speakers, will likely produce a good headphone sound. A mix created on headphones, will likely a less on monitor speakers. Prefer your monitor speaker setup as a first listening tool. The basics of setting up speakers ; Set up the speakers within a 60 angle (30 left of the middle and 30 right of the middle). Keep enough distance between the speakers and the walls (50cm), refer to the manual of your speakers. Listening distance at least 1 meter. The tweeters of both speakers should be the same height as your ears when sitting in you listening chair, about 120cm.
Some rules for mixing on Headphones and Monitor Speakers.
For mixes that sound good through speakers and headphones, it maybe quicker and easier to start a mix on loudspeakers and then tweak it for headphone listeners than the other way round. Don't be tempted to keep edging up headphone levels, or you'll end up with a headache, listening fatigue and eventually hearing damage. Try instead to take regular short breaks, which should keep your decision-making processes fresh. If you're using headphones, try experimenting with how you position them on your head. Wearing headphones slightly lower (by extending the headband) and slightly forward on your ears gives noticeably sharper imaging.
Sound isolation or soundproofing.
Despite what you may have seen in the movies or elsewhere, egg crates on the wall don't work! First, understand what's meant by soundproofing. Here we mean the means and methods to prevent sound from the outside getting in, or sound from the inside getting out. The acoustics within the room are another matter altogether. There are three very important requirements for soundproofing, mass, absorption, and isolation. Sound is the mechanical vibration propagating through a material. Thelevel of the sound is directly related to the size of those vibrations. The more massive an object is, the harder it is to move and the smaller the amplitude of the vibration set up in it under the influence of an external sound. That's why well-isolated rooms are very massive rooms. A solid concrete wall will transmit much less sound then a standard wood-framed, gypsum board wall. And a thicker concrete wall transmits less than a thinner one: not so much because of the distance, but mostly because it's heavier. Secondly, sound won't be transmitted between two objects unless it's mechanically coupled. Air is not the best coupling mechanism. But solid objects usually are. That's why well isolated rooms are often set on springs and rubber isolators. It's also why you may see rooms-within rooms: The inner room is isolated from the outer, and there may be a layer of absorptive material in the space between the two. That's also why you'll also see two sets of doors into a recording studio: so the sound does not couple directly through the door (and those doors are also very heavy!). If you are trying to isolate the sound in one room from an adjoining room, one way is to build a second wall, not attached to the first. This can go a long way to increasing the mechanical isolation. But remember, make it heavy, and isolate it. Absorptive materials like foam wedges or Sonex and such can only control the acoustics in the room: they will do nothing to prevent sound from getting in or out to begin with.
Tuning.
Instrumental tuning can be very important, while recording or mixing. Every instrument can be tuned, from start, while recording or mixing. It is a common thing to tune. Synths for instance you might believe their are well tuned, but however sweeps or calculations can make the signal off tune. Natural sounds and unnatural sounds can be tuned. So check your tunings. Tuning can be factor for mixing and making it sound right and correct. The miracle of well tunes instruments and a good mix can make the difference. Reconising tuning problems is having a lot of experience hearing. Sometime you find a tuning setting that really sounds well ? Keep it then, as long as it sounds ok. The one who is well trained in hearing tunings can make a difference. Mixing with effects and mixers is one part of the deal, tuning is also a part the must be handled and given some time and tought.
Normalizing.
Normalizing is raising the level of the peak to 0 dB and with that the rest is gained evenly. It is ofthen confused as compression, but no such thing. When normalizing to 0 dB this is done to get the most level out of a file or trakc or finished mix. Also normalizing is a common source for confusion and also is more looked at when beginners try to get the most out of their unmixed mix. Normalizing is also not mastering. Mastering is more complex and even might not include normalizing at all. On burning a compilation CD or Mp3 files, you could use normalizing to get the most level
ABC of recording/music
ABX Comparator: A device that randomly selects between two components being tested. The listener doesn't know which device is being listened to.
AC3: See Dolby Digital
AES/EBU: Balanced digital connection. For example, used to connect a CD transport to a DAC. The AES/EBU standard uses XLR type connectors.
Ambience: The acoustic characteristics of a space with regard to reverberation. A room with a lot of reverb is said to be "live"; one without much reverb is "dead."
Amplifier (Amp): A device which increases signal level. Many types of amplifiers are used in audio systems. Amplifiers typically increase voltage, current or both.
Amplifier classes Audio power amplifiers are classified primarily by the design of the output stage. Classification is based on the amount of time the output devices operate during each cycle of signal swing. Also defined in terms of output bias current, (the amount of current flowing in the output devices with no signal).
Class A operation is where both devices conduct continuously for the entire cycle of signal swing, or the bias current flows in the output devices at all times. The key ingredient of class A operation is that both devices are always on. There is no condition where one or the other is turned off. Because of this, class A amplifiers are single-ended designs with only one type polarity output devices. Class A is the most inefficient of all power amplifier designs, averaging only around 20%. Because of this, class A amplifiers are large, heavy and run very hot. All this is due to the amplifier constantly operating at full power.The positive effect of all this is that class A designs are inherently the most linear, with the least amount of distortion.
Class B operation is the opposite of class A. Both output devices are never allowed to be on at the same time, or the bias is set so that current flow in a specific output device is zero when not stimulated with an input signal, the current in a specific output flows for one half cycle. Thus each output device is on for exactly one half of a complete sinusoidal signal cycle. Due to this operation, class B designs show high efficiency but poor linearity around the crossover region. This is due to the time it takes to turn one device off and the other device on, which translates into extreme crossover distortion. Thus restricting class B designs to power consumption critical applications, e.g., battery operated equipment, such as 2-way radio and other communications audio.
Class AB operation allows both devices to be on at the same time (like in class A), but just barely. The output bias is set so that current flows in a specific output device appreciably more than a half cycle but less than the entire cycle. That is, only a small amount of current is allowed to flow through both devices, unlike the complete load current of class A designs, but enough to keep each device operating so they respond instantly to input voltage demands. Thus the inherent non-linearity of class B designs is eliminated, without the gross inefficiencies of the class A design. It is this combination of good efficiency (around 50%) with excellent linearity that makes class AB the most popular audio amplifier design.
Class AB plus B design involves two pairs of output devices: one pair operates class AB while the other (slave) pair operates class B.
Class D operation is switching, hence the term switching power amplifier. Here the output devices are rapidly switched on and off at least twice for each cycle. Since the output devices are either completely on or completely off they do not theoretically dissipate any power. Consequently class D operation is theoretically 100% efficient, but this requires zero on-impedance switches with infinitely fast switching times -- a product we're still waiting for; meanwhile designs do exist with true efficiencies approaching 90%.
Class G operation involves changing the power supply voltage from a lower level to a higher level when larger output swings are required. There have been several ways to do this. The simplest involves a single class AB output stage that is connected to two power supply rails by a diode, or a transistor switch. The design is such that for most musical program material, the output stage is connected to the lower supply voltage, and automatically switches to the higher rails for large signal peaks. Another approach uses two class AB output stages, each connected to a different power supply voltage, with the magnitude of the input signal determining the signal path. Using two power supplies improves efficiency enough to allow significantly more power for a given size and weight. Class G is becoming common for pro audio designs.
Class H operation takes the class G design one step further and actually modulates the higher power supply voltage by the input signal. This allows the power supply to track the audio input and provide just enough voltage for optimum operation of the output devices. The efficiency of class H is comparable to class G designs.
Attenuate: To reduce in level.
Analog: Before digital, the way all sound was reproduced.
Aperiodic:Refers to a type of bass-cabinet loading. An aperiodic enclosure type usually features a very restrictive, (damped), port. The purpose of this restrictive port is not to extend bass response, but lower the Q of the system and reduce the impedance peak at resonance. Most restrictive ports are heavily stuffed with fiberglass, dacron or foam.
Audiophile: A person interested in sound reproduction.
Balanced: Referring to wiring: Audio signals require two wires. In an unbalanced line the shield is one of those wires. In a balanced line, there are two wires plus the shield. For the system to be balanced requires balanced electronics and usually employs XLR connectors. Balanced lines are less apt to pick up external noise. This is usually not a factor in home audio, but is a factor in professional audio requiring hundreds or even thousands of feet of cabling. Many higher quality home audio cables terminated with RCA jacks are balanced designs using two conductors and a shield instead of one conductor plus shield.
Bandwidth: The total frequency range of any system. Usually specified as something like: 20-20,000Hz plus or minus 3 db.
Bass Reflex: A type of loudspeaker that uses a port or duct to augment the low-frequency response. Opinions vary widely over the "best" type of bass cabinet, but much has to do with how well a given design, such as a bass reflex is implemented.
Bessel crossover A type of crossover design characterized by having a linear or maximally flat phase response. Linear phase response results in constant time-delay (all frequencies within the passband are delayed the same amount). Consequently the value of linear phase is it reproduces a near-perfect step response with no overshoot or ringing. The downside of the Bessel is a slow roll-off rate. The same circuit complexity in a Butterworth response rolls off much faster.
Bi-amplify: The use of two amplifiers, one for the lows, one for the highs in a speaker system. Could be built into the speaker design or accomplished with the use of external amplifiers and electronic crossovers.
Bi-wiring: The use of two pairs of speaker wire from the same amplifier to separate bass and treble inputs on the speaker.
BNC: A type of connection often used in instrumentation and sometimes in digital audio. BNC connectors sometimes are used for digital connections such as from a CD Transport to the input of a DAC.
Boomy: Listening term, refers to an excessive bass response that has a peak(s) in it.
Bright: Listening term. Usually refers to too much upper frequency energy.
Butterworth crossover A type of crossover circuit design having a maximally flat magnitude response, i.e., no amplitude ripple in the passband. This circuit is based upon Butterworth functions, also know as Butterworth polynomials.
Channel Balance: In a stereo system, the level balance between left and right channels. Properly balance between the left-right speakers. In a home-theater system, refers to achieving correct balance between all the channels of the system.
Clipping: Refers to a type of distortion that occurs when an amplifier is driven into an overload condition. Usually the "clipped" waveform contains an excess of high-frequency energy. The sound becomes hard and edgy. Hard clipping is the most frequent cause of "burned out" tweeters. Even a low-powered amplifier or receiver driven into clipping can damage tweeters which would otherwise last virtually forever.
Class A, Class A-B etc.: In a sense,amplifying the audio signal means using the wall-current (usually either 120 or 240 volts) to increase the amplitude of the audio signal from mill-watts to watts. Different classes of amplifiers accomplish this in different ways. Turning a vacuum tube "on" or "off" with current demand increases the efficiency of the amplifier but may add switching distortion. A Class A amplifier is relatively inefficient, converting much energy to heat, but has no switching distortion.
Coloration: Listening term. A visual analog. A "colored" sound characteristic adds something not in the original sound. The coloration may be euphonically pleasant, but it is not as accurate as the original signal.
Coherence: Listening term. Refers to how well integrated the sound of the system is.
Compression: In audio, compression means to reduce the dynamic range of a signal. Compression may be intentional or one of the effects of a system that is driven to overload.
Co-axial: A speaker type that utilizes a tweeter mounted at the center of a woofer cone. The idea being to have the sound source through the full frequency range become "coincident".
Crossover: A frequency divider. Crossovers are used in speakers to route the various frequency ranges to the appropriate drivers. Additionally, many crossovers contain various filters to stabilize the impedance load of the speaker and or shape the frequency response. Some crossovers contain levels controls to attenuate various parts of the signal. A passive crossover uses capacitors, coils and resistors, usually at speaker level. A passive crossover is load dependent (the transition may not be very smooth or accurate if a different speaker is substituted for the one the crossover was designed for).
An active crossover is based on integrated circuits (ICs), discreet transistors or tubes. An active crossover is impedance buffered and gives a consistent and accurate transition regardless of load.
Crossover Slope: High and low pass filters used for speakers do not cut-off frequencies like brick walls. The rolloff occurs over a number of octaves. Common filter slopes for speakers are 1st through 4th order corresponding to 6db/oct to 24db/oct. For example, a 1st. order, 6db/oct high pass filter at 100hz will pass 6db less energy at 50Hz and 12db less energy at 25Hz. Within the common 1st through 4th filters there is an endless variety of types including Butterworth, Linkwitz-Riley, Bessel, Chebychev, etc. Salesmen and product literature will sometimes make claims of clear superiority for the filter used in the product they are trying to sell. Since the subject fills books, suffice it to say that there is no one best filter, it depends on application and intended outcome. Good designers use the filters required to get the optimum performance from the system.
Cross-talk: Unwanted breakthrough of one channel into another. Also refers to the distortion that occurs when some signal from a music source that you are not listening to leaks into the circuit of the source that you are listening to.
DAC: A Digital to Audio Converter. Converts a digital bitstream to an analog signal. Can be a separate "box" that connects between a CD Transport or CD Player and a pre-amplifier.
Damping (Damping factor, etc.) Refers to the ability of an audio component to "stop" after the signal ends. For example, if a drum is struck with a mallet, the sound will reach a peak level and then decay in a certain amount of time to no sound. An audio component that allows the decay to drag on too long has poor damping, and less definition than it should. An audio component that is overdamped does not allow the initial energy to reach the full peak and cuts the decay short. "Boomy" or "muddy" sound is often the result of underdamped systems. "Dry" or "lifeless" sound may be the result of an overdamped system.
D'Appolito: Joe D'Appolito is credited with popularizing the MTM (Midrange-Tweeter-Midrange) type of speaker.
Decibel (dB): Named after Alexander Graham Bell. We perceive differences in volume level in a logarithmic manner. Our ears become less sensitive to sound as its intensity increases. Decibels are a logarithmic scale of relative loudness. A difference of approx. 1 dB is the minimum perceptible change in volume, 3 dB is a moderate change in volume, and about 10 dB is an apparent doubling of volume
0 dB is the threshold of hearing, 130 dB is the threshold of pain. Whisper: 15-25 dB Quiet background: about 35 dB Normal home or office background: 40-60 dB Normal speaking voice: 65-70 dB Orchestral climax: 105 dB Live Rock music: 120 dB+ Jet aircraft: 140-180 dB
Dipole: An open-back speaker that radiates sound equally front and rear. The front and rear waves are out of phase and cancellation will occur when the wavelengths are long enough to "wrap around". The answer is a large, wide baffle or to enclose the driver creating a monopole.
Distortion: Anything that alters the musical signal. There are many forms of distortion, some of which are more audible than others. Distortion specs are often given for electronic equipment which are quite meaningless. As in all specifications, unless you have a thorough understanding of the whole situation, you will not be able to make conclusions about the sonic consequences.
DIY: Abbreviation for Do - It - Yourself. In audio, the most common DIY is building speakers but some hobbyists build everything from pre-amps to amplifiers to DACs.
Dolby Digital: Is a five-channel system consisting of left, center, right and left rear, right rear channels. All processing is done in the digital domain. Unlike Dolby Prologic in which the rear effects channels are frequency limited to approx. 100-7000Hz, Dolby Digital rear channels are specified to contain the full 20-20Khz frequency content. The AC3 standard also has a separate subwoofer channel for the lowest frequencies.
Dolby Prologic: Is a four-channel system consisting of left, center, right and rear channel, (the single rear channel is usually played through two speakers).
DTS: Digital Theater System. A multi-channel encoding/decoding system. Used in some movie theaters. Also now included in some home-theater processors. A competitor to Dolby Digital.
DSP: Digital Signal Processing. DSP can be used to create equalization, compression, etc. of a digital signal.
DVD: Digital Video Disc or Digital Versatile Disc. A relatively new standard that seeks to combine better-than-laser-disc quality video with better-than-CD quality audio in a disc the size of a CD. Requires special players. Seems to be a viable candidate to replace both Laser Discs and CDs, but the jury is still out.
Dynamic Headroom: The ability of an audio device to respond to musical peaks. For example, an amplifier may only be capable of a sustained 100 watts, but may be able to achieve peaks of 200 watts for the fraction of a second required for an intense, quick sound. In this example the dynamic headroom would equal 3 db.
Dynamic range: The range between the loudest and the softest sounds that are in a piece of music, or that can be reproduced by a piece of audio equipment without distortion (a ratio expressed in decibels). In speech, the range rarely exceeds 40 dB; in music, it is greatest in orchestral works, where the range may be as much as 75 dB.
Electrostatic Speaker: A speaker that radiates sound from a large diaphragm that is suspended between high-voltage grids.
Euphonic: Pleasing. As a descriptive audio term, usually refers to a coloration or inaccuracy that non-the-less may be sonically pleasing.
Extension: How extended a range of frequencies the device can reproduce accurately. Bass extension refers to how low a frequency tone will the system reproduce, high-frequency extension refers to how high in frequency will the system play.
Fletcher-Munson curve: Our sensitivity to sound depends on its frequency and volume. Human ears are most sensitive to sounds in the midrange. At lower volume levels humans are less sensitive to sounds away from the midrange, bass and treble sounds "seem" reduced in intensity at lower listening levels.
Frequency: The range of human hearing is commonly given as 20-20,000Hz (20Hz-20kHz). One hertz (Hz) represents one cycle per second, 20Hz represents 20 cycles per second and so on. Lower numbers are lower frequencies
Fundamental. The lowest frequency of a note in a complex wave form or chord.
Gain: To increase in level. The function of a volume control.
Grain: Listening term. A sonic analog of the grain seen in photos. A sort of "grittiness" added to the sound.
Haas effect: If sounds arrive from several sources, the ears and brain will identify only the nearest. In other words, if our ears receive similar sounds coming from various sources, the brain will latch onto the sound that arrives first. If the time difference is up to 50 milliseconds, the early arrival sound can dominate the later arrival sound, even if the later arrival is as much as 10 dB louder. The discovery of this effect is attributed to Halmut Haas in 1949.
Harmonics: Also called overtones, these are vibrations at frequencies that are multiples of the fundamental. Harmonics extend without limit beyond the audible range. They are characterized as even-order and odd-order harmonics. A second-order harmonic is two times the frequency of the fundamental; a third order is three times the fundamental; a fourth order is four times the fundamental; and so forth. Each even-order harmonic: second, fourth, sixth, etc.-is one octave or multiples of one octave higher than the fundamental; these even-order overtones are therefore musically related to the fundamental. Odd-order harmonics, on the other hand: third, fifth, seventh, and up-create a series of notes that are not related to any octave overtones and therefore may have an unpleasant sound. Audio systems that emphasize odd-order harmonics tend to have a harsh, hard quality.
HDCD: High-Definition Compact Disc. A proprietary system by Pacific Microsonics that requires special encoding during the recording process. Some observers report HDCD discs as having better sound. To gain the benefits requires having special HDCD in your CD player.
Headroom: The ability of an amp to go beyond its rated power for short durations in order to reproduce musical peaks without distortion. This capability is often dependent on the power supply used in the design.
Hearing Sensitivity: The human ear is less sensitive at low frequencies than in the midrange. Turn your volume knob down and notice how the bass seems to"disappear". To hear low bass requires an adequate SPL level. To hear 25Hz requires a much higher SPL level than to hear 250Hz. In the REAL world, low frequency sounds are reproduced by large objects; bass drums, string bass, concert grand pianos, etc. Listen to the exhaust rumble of a 454 cubic inch V8 engine vs. the whine of the little four banger. The growl of a lion vs. the meow of your favorite kitty. As frequency decreases we perceive more by feel than actual hearing and we lose our ability to hear exact pitch.
Hertz (Hz): A unit of measurement denoting frequency, originally measured as Cycles Per Second, (CPS): 20 Hz = 20 CPS. Kilohertz (kHz) are hertz measured in multiples of 1,000.
High-Pass Filter: A circuit that allows high frequencies to pass but rolls off the low frequencies. When adding a subwoofer it is often desirable to roll-off the low frequencies to the main amplifiers and speakers. This will allow the main speakers to play louder with less distortion. High-pass filters used at speaker level are usually not very effective unless properly designed for a specific main speaker (see impedance below).
Imaging: Listening term. A good stereo system can provide a stereo image that has width, depth and height. The best imaging systems will define a nearly holographic re-creation of the original sound
Impedance: Impedance is a measure of electrical resistance specified in ohms. Speakers are commonly listed as 4 or 8 ohms but speakers are reactive devices and a nominal 8 ohm speaker might measure from below 4 ohms to 60 or more ohms over its frequency range. This varying impedance curve is different for each speaker model and makes it impossible to design a really effective "generic" speaker level high-pass filter. Active devices like amplifiers typically have an input impedance between about 10,000-100,000 ohms and the impedance is the same regardless of frequency.
Interconnects: Cables that are used to connect components at a low signal level. Examples include CD player to receiver, pre-amplifier to amplifier, etc. Most interconnects use a shielded construction to prevent interference. Most audio interconnects use RCA connections although balanced interconnects use XLR connections.
Jitter: A tendency towards lack of synchronization caused by electrical changes. Technically the unexpected (and unwanted) phase shift of digital pulses over a transmission medium. A discrepancy between when a digital edge transition is supposed to occur and when it actually does occur - think of it as nervous digital, or maybe a digital analogy to wow and flutter.
Kevlar: Material developed by Dupont that is has an exceptional strength to weight ratio. Used extensively in bullet-proof vests, skis, sailboat hulls, etc. In audio, used in many variations for speaker cones.
Line Level: CD players, VCRs, Laserdisc Players etc., are connected in a system at line level, usually with shielded RCA type interconnects. Line level is before power amplification. In a system with separate pre-amp and power-amp the pre-amp output is line level. Many surround sound decoders and receivers have line level outputs as well.
Line-Source: A speaker device that is long and tall. Imagine a narrow dowel dropped flat onto the water's surface. The line-source has very limited vertical dispersion, but excellent horizontal dispersion.
Lobing: Any time more than one speaker device covers the same part of the frequency range there will be some unevenness in the output. (Picture the waves from one pebble dropped into a calm pool vs. two pebbles dropped several inches apart.) Lobing means that the primary radiation pattern(s) is at some angle above or below the centerline between the two drivers. Good crossover design takes this into account.
Low Frequency Extension: Manufacturers, writers and salespeople toss around all kinds of numbers and terminology that can be very confusing and misleading. "This $300 shoebox sized sub is flat to 20Hz". Right, in your dreams . . . How is that cheap, tiny box and driver going to reproduce a 56 foot wavelength with enough power to be heard? It will not to it. Good bass reproduction requires moving a lot of air and playback at realistic volumes. Remember the rule of needing to move four times the air to go down one octave. Example: You have a pair of good quality tower speakers with 10" woofers that produce good bass down to around 40Hz. The salesman is telling you that his little subwoofer with a single 10" woofer will extend your system down to 20Hz. If you've been paying attention, you know that his woofer will have to move eight times as much air as each of your 10" woofers, not likely. Adding that subwoofer to your system might give you more apparent bass energy, and in fact may help a little with movie special effects, but it is unlikely to extend bass response significantly.
Low-Pass Filter: A circuit that allows low frequencies to pass but rolls off the high frequencies. Most subwoofers have low-pass filters built in and many surround sound decoders have subwoofer outputs that have been low-pass filtered.
Loudness: Perceived volume. Loudness can be deceiving. For example, adding distortion will make a given volume level seem louder than it actually is.
Magnetic-Planar Speakers: A type of speaker that uses a flat diaphragm with a voice coil etched or bonded to it to radiate sound. If the magnets are both in front of and behind the diaphragm, it becomes a push-pull magnetic-planar.
Midrange: A speaker, (driver), used to reproduce the middle range of frequencies. A midrange is combined with a woofer for low frequencies and a tweeter for high frequencies to form a complete, full-range system.
Monopole: Any speaker that encloses the backwave of the speaker device even though part of this backwave may be released via. a port or duct. The primary radiation at most frequencies will be from the driver front. If the driver is not enclosed it becomes a dipole.
Muddy: Listening term. A sound that is poorly defined, sloppy or vague. For example, a "muddy" bass is often boomy with all the notes tending to run together.
Muting: To greatly decrease the volume level. Many receivers and pre-amplifiers have a muting control which allows the volume level to be cut way down without changing the master volume control. Great for when the phone rings.
Nonlinearity: What goes into a system comes out changed by its passage through that system-in other words, distorted. The ideal of an audio component and an audio system is to be linear, or nondistorting, with the image on one side of the mirror identical to the image on the other side.
Octave: An octave is a doubling or halving of frequency. 20Hz-40Hz is often considered the bottom octave. Each octave you add on the bottom requires that your speakers move four times as much air!
Overload: A condition in which a system is given too high of an input level. A common cause of distortion or product failure.
Overtones:See Harmonics.
PCM: Pulse Code Modulation. A means of digital encoding.
Planar Source: Most electrostatics and magnetic planars have a large surface area. Think of a wide board dropped flat onto the water surface. The sound can be extremely coherent, but the listening window is effectively limited to being directly on-axis of both the left and right planar speaker.
Point-Source: Most multi-unit loudspeakers try to approximate a point-source. Think of a pebble dropped into the water and the expanding wave pattern away from impact. Obviously it is difficult to integrate multiple point-sources into a truly coherent expanding wave. The best designs do quite well with careful driver engineering and crossover development.
Polarity: A speaker, for example, has a positive and a negative input terminal. Connecting a battery directly to the speaker will result in the diaphragm moving outward. If you reverse the battery leads, the diaphragm will move inward. Caution: Too high of a voltage battery will also burn out the speaker!
Push-pull: Most common type of amplification that amplifies the negative and positive sides of the waveform separately. Allows for much higher power output than single-ended.
Pre-amplifier: Or Pre-amp is a device that takes a source signal, such as from a turntable, tape-deck or CD player, and passes this signal on to a power-amplifier(s). The pre-amp may have a number of controls such as source selector switches, balance, volume and possibly tone-controls. Radio-frequency interference (RFI): Radio-frequency sound waves can be caused by many sources including; shortwave radio equipment, household electrical line, computers and many other electronic devices. RFI sometimes interferes with audio signals, causing noise and other distortions.
Q or Quality Factor: Is a measure of damping. Modern home speaker systems have Q values ranging from < .5 to approx. 2.0. Q values < .7 have no peak in the response. Q values around .5 are considered to be optimally damped, having a Bessel response. A Q of 1.0 is a Butterworth response. The lower the Q value, the better the transient response of the system, (less or no ringing), but the tradeoff is a larger required box size and the response begins to rolloff at a higher frequency. Another way to consider it is that the lower the Q, the more gradual the rolloff but the rolloff begins at a higher frequency.
RCA Connector: "Phono" plugs, used primarily as low-level connections between Phonographs/CD players/Tuners/Recievers/Amplifiers
Receiver: An audio component that combines a pre-amplifier, amplifier(s) and tuner in one chassis. A Dolby Prologic Receiver also contains a Dolby Prologic decoder for surround sound.
Resonant frequency: Any system has a resonance at some particular frequency. At that frequency, even a slight amount of energy can cause the system to vibrate. A stretched piano string, when plucked, will vibrate for a while at a certain fundamental frequency. Plucked again, it will again vibrate at that same frequency. This is its natural or resonant frequency. While this is the basis of musical instruments, it is undesirable in music-reproducing instruments like audio equipment.
Ribbon Speaker: A type of speaker that uses a pleated conductor suspended between magnets. Most true ribbons are tweeters only. Sometimes confused with mmagnetic-planar speakers.
RMS (root-mean-square): The square root of the mean of the sum of the squares. Commonly used as the effective value of measuring a sine wave's electrical power. A standard in amplifier measurements.
Satellite: A satellite speaker is usually fairly small, and does not reproduce the lowest frequencies. Usually meant to be used with a woofer or subwoofer.
Sensitivity: A measurement of how much power is required for a loudspeaker to achieve a certain output level. The general standard used is on-axis SPL(Sound Pressure Level) at 1 watt input, 1 meter distance.
Signal-to-noise (SN) Ratio: The range or distance between the noise floor (the noise level of the equipment itself) and the music signal.
Single-ended: Type of amplification often, (but not always), using vacuum tubes. Typically low power output, low damping factor and relatively high distortion. Single-ended enthusiasts claim that the sound quality is more "real".
Sound Pressure Level (Spl): Given in decibels (DB) is an expression of loudness or volume. A 10db increase in SPL represents a doubling in volume. Live orchestral music reaches brief peaks in the 105db range and live rock easily goes over 120db.
Soundstage: A listening term the refers to the placement of a stereo image in a fashion that replicates the original performance. A realistic soundstage has proportional width, depth and height.
Sound Waves: Sound waves can be thought of like the waves in water. Frequency determines the length of the waves; amplitude or volume determines the height of the waves. At 20Hz, the wavelength is 56 feet long! These long waves give bass its penetrating ability, (why you can hear car boomers blocks away).
Speaker Level: Taken from the speaker terminals. This signal has already been amplified.
Spectral balance: Balance across the entire frequency spectrum of the audio range.
Stereo: From the Greek meaning solid. The purpose of stereo is not to give you separate right and left channels, but to provide the illusion of a three-dimensional, holographic image between the speakers.
Subwoofer: A speaker designed exclusively for low-frequency reproduction. A true subwoofer should be able to at least reach into the bottom octave (20-40Hz). There are many "subwoofers" on the market that would be more accurately termed "woofers".
THX: Refers to a series of specifications for surround sound systems. Professional THX is used in commercial movie theaters. Home THX specifications are not published and manufacturers must sign non-disclosure waivers before submitting their products for THX certification. Manufacturers that receive certification for their products must pay a royalty on units sold.
Timbre: The quality of a sound that distinguishes it from other sounds of the same pitch and volume. The distinctive tone of an instrument or a singing voice.
Timbral: Refers to the overall frequency balance of a system. In a perfect world, all systems would have complete tonal neutrality. With current technology, this ideal is approached but not met. Listening to many equally "good" speakers will reveal that some sound warmer than others, some sound brighter etc. In a surround sound system it is important that all speakers have a close timbral match for the highest degree of sonic realism.
Total harmonic distortion (THD): Refers to a device adding harmonics that were not in the original signal. For example: a device that is fed a 20Hz sine wave that is also putting out 40Hz, 80Hz etc. Not usually a factor in most modern electronics, but still a significant design problem in loudspeakers.
Transducer. A device that converts one form of energy to another. Playback transducers are the phono cartridge, which changes mechanical vibrations into electrical energy, and the loudspeakers, which change it back, from electrical energy coming from the amp to mechanical movement of the diaphragm, causing audible pressure changes in the air.
Transmission Line: Also referred to as a T-line. A type of bass cabinet in which the back wave follows a relatively long, usually damped path before being ported to the outside. T-lines are usually rather large and costly cabinets to manufacture. Opinions vary widely over the "best" type of bass cabinet, but much has to do with how well a given design, such as a transmission line is implemented.
Transient response: The ability of a component to respond quickly and accurately to transients. Transient response affects reproduction of the attack and decay characteristics of a sound.
Transparency: Listening term. An analog that can be best "pictured" in photography. The more "transparent" the sound, the clearer the auditory picture.
Transients: Instantaneous changes in dynamics, producing steep wave fronts.
Tri-wiring: The use of three pairs of speaker wire from the same amplifier to separate bass, midrange and treble inputs on the speakers.
Tuning Frequency: The helmholtz resonant frequency of a box. Also refers to the resonant frequency of other types of systems.
Tweeter: A speaker, (driver), used to reproduce the higher range of frequencies. To form a full-range system, a tweeter needs to be combined with a woofer, (2-way system), or a woofer and midrange, (3-way system).
Unity gain: A circuit with unity gain will not increase or decrease the volume level.
Warmth: A listening term. The opposite of cool or cold. In terms of frequency, generally considered the range from approx. 150Hz-400Hz. A system with the "proper" warmth will sound natural within this range.
Wattage: Is the unit of power used to rate the output of audio amplifiers. For a wattage number to have meaning the distortion level and impedance must also be specified.
Wavelength The distance the sound wave travels to complete one cycle. The distance between one peak or crest of a sine wave and the next corresponding peak or crest. The wavelength of any frequency may be found by dividing the speed of sound by the frequency. (Speed of sound at sea level is 331.4 meters/second or 1087.42 feet/second).
Woofer: A speaker, (driver), used for low-frequency reproduction. Usually larger and heavier than a midrange or tweeter.
XLR: A type of connector used for balanced lines. Used for microphones, balanced audio components and the AES/EBU digital connection.
Y-Adapter: Any type of connection that splits a signal into two parts. An example would be a connector with one male RCA jack on one end, and two female RCA jacks on the other end.
Zobel Filter: A series circuit consisting of a resitance and capacitance. This filter is placed in parallel with a speaker driver to flatten what would otherwise be a rising impedance with frequency.
GM General Midi.
Piano:
1 Acoustic piano
2 Bright piano
3 Grand piano
4 Honky-tonk piano
5 Rhodes piano 1
6 Chorused piano 2
7 Harpsichord
8 Clavinet
Chromatic Percussion:
9 Celesta
10 Glockenspiel
11 Music box
12 Vibraphone
13 Marimba
14 Xylophone
15 Tubular bell
16 Dulcimer
Organ:
17 Hammond organ
18 Percussive organ
19 Rock organ
20 Church organ
21 Reed organ
22 Accordion
23 Harmonica
24 Tango accordion
Guitar:
25 Acoustic nylon guitar
26 Acoustic steel guitar
27 Jazz guitar
28 Clean guitar
29 Muted guitar
30 Overdriven guitar
31 Distortion guitar
32 Guitar harmonics
Bass:
33 Acoustic bass
34 Finger bass
35 Picked bass
36 Fretless bass
37 Slap bass 1
38 Slap bass 2
39 Synth bass 1
40 Synth bass 2
Strings:
41 Violin
42 Viola
43 Cello
44 Double bass
45 Tremolo strings
46 Pizzicato strings
47 Orchestral harp
48 Timpani
Ensemble:
49 Strings 1
50 Strings 2
51 Synth strings 1
52 Synth strings 2
53 Voice aahs
54 Voice oohs
55 Synth voice
56 Orchestra hit
Brass:
57 Trumpet
58 Trombone
59 Tuba
60 Muted trumpet
61 French horn
62 Brass
63 Synth brass 1
64 Synth brass 2
Reed:
65 Soprano sax
66 Alto sax
67 Tenor sax
68 Baritone sax
69 Oboe
70 English horn
71 Bassoon
72 Clarinet
Pipe:
73 Piccolo
74 Flute
75 Recorder
76 Pan flute
77 Bottle blow
78 Shakuhachi
79 Whistle
80 Ocarina
Synth Lead:
81 Square wave
82 Sawtooth
83 Calliope
84 Chiff lead
85 Charang
86 Solo synth lead
87 Bright saw
88 Bass and lead
Synth Pad:
89 Fantasia
90 Warm pad
91 Poly synth
92 Space pad
93 Bowed glass
94 Metal
95 Halo pad
96 Sweep pad
Synth Effects:
97 Ice rain
98 Soundtrack
99 Crystal
100 Atmosphere
101 Brightness
102 Goblin
103 Echo drops
104 Star theme
Ethnic:
105 Sitar
106 Banjo
107 Shamisen
108 Koto
109 Kalimba
110 Bagpipe
111 Fiddle
112 Shanai
Percussive:
113 Tinkle bell
114 Agogô
115 Steel drums
116 Woodblock
117 Taiko drum
118 Melodic tom
119 Synth drum
120 Reverse cymbal
Sound effects:
121 Guitar fret
122 Breath
123 Seashore
124 Bird tweet
125 Telephone Ring
126 Helicopter
127 Applause
128 Gunshot
Percussion.
Channel 10 is reserved for percussion under General MIDI, this channel always sounds as percussion regardless of whatever program change numbers it may be sent, and different note numbers are interpreted as different instruments.
35 Bass Drum 2
36 Bass Drum 1
37 Side Stick
38 Snare Drum 1
39 Hand Clap
40 Snare Drum 2
41 Low Tom 2
42 Closed Hi-hat
43 Low Tom 1
44 Pedal Hi-hat
45 Mid Tom 2
46 Open Hi-hat
47 Mid Tom 1
48 High Tom 2
49 Crash Cymbal 1
50 High Tom 1
51 Ride Cymbal 1
52 Chinese Cymbal
53 Ride Bell
54 Tambourine
55 Splash Cymbal
56 Cowbell
57 Crash Cymbal 2
58 Vibra Slap
59 Ride Cymbal 2
60 High Bongo
61 Low Bongo
62 Mute High Conga
63 Open High Conga
64 Low Conga
65 High Timbale
66 Low Timbale
67 High Agogo
68 Low Agogo
69 Cabasa
70 Maracas
71 Short Whistle
72 Long Whistle
73 Short Guiro
74 Long Guiro
75 Claves
76 High Wood Block
77 Low Wood Block
78 Mute Cuica
79 Open Cuica
80 Mute Triangle
81 Open Triangle
Controllers.
GM also specifies which operations should be performed by several controllers.
1 Modulation
6 Data Entry MSB
7 Volume
10 Pan
11 Expression
38 Data Entry LSB
64 Sustain
100 RPN LSB
101 RPN MSB
121 Reset all controllers
123 All notes off
RPN.
Setting Registered Parameters requires sending (numbers are decimal):
1) two Control Change messages using Control Numbers 101 and 100 to select the parameter, followed by
2) any number of Data Entry messages of one or two bytes (MSB = Controller #6, LSB = Controller #38), and finally 3) an End of RPN message. The following global Registered Parameter Numbers (RPNs) are standardized [1] (the parameter is specified by RPN LSB/MSB pair and the value is set by Data Entry LSB/MSB pair).
0,0 Pitch bend range
1,0 Channel Fine tuning
2,0 Channel Coarse tuning
3,0 Tuning Program Change
4,0 Tuning Bank Select
5,0 Modulation Depth Range
127,127 RPN Null
For example: RPN control sequence to set coarse tuning to A440 (parm 2, value 64):
101:0, 100:2, 6:64, 101:127, 100:127.
System Exclusive messages.
Two GM System Exclusive ("SysEx") messages are defined: one to enable and disable General MIDI compatibility, on devices which also allow modes which are not GM-compatible; and the other to modify an instrument's master volume.
GS extensions.
The first GM synthesizer in Roland Sound Canvas line featured a set of extensions to General MIDI standard. The most apparent addition was the ability to address multiple banks of sounds by using additional pair of controllers, cc#0 (Bank Select MSB) and cc#32 (Bank Select LSB), to specify up to 65536 'variation' sounds. Other most notable features were 9 Drum kits with 14 additional drum sounds each, Control Change messages for controlling the send level of sound effect blocks (cc#91-94), entering additional parameters (cc#98-101), portamento, sostenuto, soft pedal (cc#65-67), and model-specific SysEx messages for setting various parameters of the synth engine.
General MIDI Level 2.
In 1999, the standard was once again updated to include more controllers, patches, RPNs and SysEx messages. Here's a quick overview of the changes in comparison to GM/GS:
Number of Notes - minimum 32 simultaneous notes
Simultaneous Percussion Kits - up to 2 (Channels 10/11)
Additional 128 melodic sounds are included in variation banks, for a total of 256
9 GS Drum kits are included
Additional Control Change messages
Filter Resonance (Timbre/Harmonic Intensity) (cc#71)
Release Time (cc#72)
Brightness/Cutoff Frequency (cc#74)
Decay Time (cc#75)
Vibrato Rate (cc#76)
Vibrato Depth (cc#77)
Vibrato Delay (cc#78)
Registered Parameter Numbers (RPNs)
Modulation Depth Range (Vibrato Depth Range)
Universal SysEx messages
Master Volume, Fine Tuning, Coarse Tuning
Reverb Type, Time
Chorus Type, Mod Rate, Mod Depth, Feedback, Send to Reverb
Controller Destination Setting
Scale/Octave Tuning Adjust
Key-Based Instrument Controllers
GM2 System On
Additional melodic instruments can be accessed by setting CC#32 to 121 and then using CC#0 to select the bank before a Program Change. The most expanded group is Acoustic Pianos.
Denis van der Velde
AAMS Auto Audio Mastering System
www.curioza.com
The home recording market has grown exponentially in the last decade. Advancements in computer technology and the development of music recording software have brought professional quality recording capabilities to the home studio.
The focus of the articles that follow is to help you to understand how professional recording studios are designed and used. Each concept will then be translated to the home recording environment. You will learn how to best implement these concepts into your own home studio setup.
If you are building a home recording studio, it is important to understand how you create music. Write it down on a piece of paper if necessary. Then design a logical step by step process that allows you to set up and work quickly when a creative idea strikes.
Home music production and home studio design are the largest growing areas of the recording industry. Advancements in computer technology coupled with a less profitable recording industry have led the pro audio manufacturers to go where the money is… Home.
Everybody has a different idea of what they want or need out of their studio. What you want will determine the home studio design, software, interfaces, microphones and gear that you buy. It is critically important to have a goal for your home studio. Before you set your goal, answer the following questions. The answers to these basic questions will help determine the design of your home studio
Goals should always be set with two things in mind. What is my long term vision, and what is the first step I need to take. Basically this is a long term goal and a short term goal. Without the long term goal, your short term decisions may be ill-advised, unfocussed and lead to a less than satisfying outcome.
1. State of the Art Studios
2. Mid-level studios
3. Small studios
1. Highest quality Acoustic design
2. Great microphone selection
3. Extensive array of pro audio gear
4. High end monitoring systems
5. Large private lounge area
6. Attractive aesthetic
1. Good to fair quality acoustic design
2. Good to great Microphone selection
3. Good to great selection of pro audio gear
4. Mid to high end monitoring systems
5. Small or semi-private lounge
6. Less attractive aesthetic
Denis van der Velde
AAMS Auto Audio Mastering System
www.curioza.com
Speakers
Everyone wants great, high-quality sound from their audio system.
Usually people want a sound that fills the room and has a deep bass, a clear treble, and a rich middle range.
The sound quality should not deteriorate when you crank up the volume.
And you certainly don’t want insane vibrations, static hiss, or smoke to come out of the speakers!
In your quest for quality sound, speaker watts are one figure to understand and consider.
Other important values are the speakers’ sensitivity and total harmonic distortion (THD).
This article will help you interpret the manufacturer’s specifications to understand what a sound system will deliver.
Loudness and Power Explained
Decibels are a measure of loudness.
This number is important when choosing speakers, especially if you like to listen at a high volume.
For every 10 decibel increase, the noise is twice as loud, so small increases in decibel levels mean big impact on your ears.
A watt is a measure of electrical power.
As an amplifier processes sound, the output is measured in watts.
All speakers have a maximum number of watts that they can cope with and the manufacturer will tell you what this is.
Make sure that the amp you use does not put out more power than your speakers can handle, or the speakers could be damaged.
Manufacturers provide two power figures for both amplifiers and loudspeakers :
Amplifiers RMS = the power an amplifier can put out over a long period
Amplifiers Peak = the power an amplifier can put out in short bursts.
Speakers Nominal Power = what a speaker can handle long term without being damaged
Speakers Peak Power = what a speaker can handle in short bursts without being damaged
Very good speakers are more sensitive than mid-quality speakers and can deliver a lot of sound with only a little power from the amplifier.
Mid-priced speakers need more power to provide the same volume.
Speaker sensitivity is expressed in terms of the number of decibels (dB) of sound pressure level (SPL) per watt of amplifier power measured at one meter from the speaker.
To simplify this, manufacturers usually drop the SPL/W/M and just say dB.
Most speaker sensitivities are in the 85 to 91 dB range, so anything less than 85dB is not so hot.
Translate Watts to Decibels
Watts Decibels
2 93
4 96
8 99
16 102
32 105
64 108
128 111
256 114
512 117
1,024 120
Total Harmonic Distortion
TDH is a measure of how faithfully speakers translate what is on a disc or hard drive into sound.
The lower the figure, the less distortion, so lower numbers are better.
Usually values between 0.05% and 0.08% THD mean a quality "clean" system, but any figure below 0.1% THD is pretty good.
Speaker Impedance
This number tells you how much current a speaker will draw.
Eight ohms is standard. Four ohms is very good but usually a lot more expensive.
If you are buying four-ohms speakers you will need a very good amplifier to get the most out of them.
Headroom
This figure is a measure of what a system can deliver in short bursts.
A large headroom figure is important if you have a home cinema system and want to get a jolt from the explosions in action movies.
Audio Formats
The list below showcases audio formats that are able to encode audio and compress it in a lossless way ensuring your music is perfectly preserved in digital form.
WAV (WAVeform Audio Format)
The main format for AAMS is WAV format audio files, basically internally AAMS is build with it´s 16 bit 32 Bit audio drivers and 64 Bit internal processing, so WAV format is very compatible.
And musically 16 bit wav is great for normal audio and will achive good sound, the 32 bit wav floating point is really exact and is the main format for more demanding users.
For audio files to be written in a lossless fashion, wav is a good choise, wav can handle also different types of samplerates, like 44.1 Khz or higher.
The WAV format isn't thought of as the ideal choice when choosing a digital audio system for preserving your audio CDs, but still remains a lossless option. However, the files produced will be larger than the other formats in this article because there isn't any compression involved. That said, if storage space isn't an issue then the WAV format has some clear advantages. It has widespread support with both hardware and software. Much lower CPU processing time is required when converting to other formats because WAV files are already uncompressed -- they don't need to be uncompressed before conversion. You can also directly manipulating WAV files (using audio editing software for instance) without having to wait for a de-compression/re-compression cycle in order to update your changes. Short for WAVeform Audio Format, it is normally used in an uncompressed format on the Microsoft Windows platform. This raw audio format, which was developed jointly by IBM and Microsoft, stores audio data in blocks. On the digital music scene, its usefulness has diminished over time with the development of better lossless audio formats, such as FLAC and Apple lossless. It is a standard that will probably be used for some time yet due to its widespread use in professional music recording and is still a very popular format for audio/video applications. The file extension associated with WAV is: .WAV
FLAC (Free Lossless Audio Codec)
The FLAC format (short for Free Lossless Audio Codec) is probably the most popular lossless encoding system which is becoming more widely supported on hardware devices such as MP3 players, smartphones, tablets, and home entertainment systems. It is developed by the non-profit Xiph.Org Foundation and is also open source. Music stored in this format is typically reduced between 30 to 50% of its original size. Common routes to rip audio CDs to FLAC include software media players (like Winamp for Windows) or dedicated utilities Max for example is a good one for Mac OS X.
ALAC (Apple Lossless Audio Codec)
Apple initially developed their ALAC format as a proprietary project, but since 2011 has made it open source. Audio is encoded using a lossless algorithm which is stored in an MP4 container. Incidentally, ALAC files have the same .m4a file extension as AAC, so this naming convention can lead to confusion. ALAC isn't as popular as FLAC, but could be the ideal choice if your preferred software media player is iTunes and you use Apple hardware such as the iPhone, iPod, iPad, etc.
Monkey's Audio
The Monkey's Audio format isn't as well supported as other competing lossless systems such as FLAC and ALAC, but on average has better compression resulting in smaller file sizes. It isn't an open source project, but is still free to use. Files that are encoded in the Monkey's Audio format have the humorous .ape extension! Methods used to rip CDs to Ape files include: downloading the Windows program from the official Monkey's Audio website, or using standalone CD ripping software that outputs to this format. Even though most software media players don't have out-of-the-box support for playing files in the Monkey's Audio format, there is a good selection of plug-ins now available for: Windows Media Player, Foobar2000, Winamp, Media Player Classic, and others. More »
WMA Lossless (Windows Media Audio Lossless) WMA Lossless which is developed by Microsoft is a propriety format that can be used to rip your original music CDs without any loss of audio definition. Depending on various factors, a typical audio CD will be compressed between 206 - 411 MB using a spread of bit rates in the range of 470 - 940 kbps. The resultant file that is produced confusingly has the .WMA extension which is identical to files that are also in the standard (lossy) WMA format. WMA Lossless is probably the least well supported of the formats in this toplist, but could still be the one you choose especially if you use Windows Media Player and have a hardware device that supports it such as a Windows phone for example.
Uncompressed Formats
WAV and AIFF: Both WAV and AIFF are uncompressed formats, which means they are exact copies of the original source audio. The two formats are essentially the same quality; they just store the data a bit differently. AIFF is made by Apple, so you may see it a bit more often in Apple products, but WAV is pretty much universal. However, since they're uncompressed, they take up a lot of unnecessary space. Unless you're editing the audio, you don't need to store the audio in these formats.
Lossless Formats
FLAC: The Free Lossless Audio Codec (FLAC) is the most popular lossless format, making it a good choice if you want to store your music in lossless. Unlike WAV and AIFF, it's been compressed, so it takes up a lot less space. However, it's still a lossless format, which means the audio quality is still the same as the original source, so it's much better for listening than WAV and AIFF. It's also free and open source, which is handy if you're into that sort of thing.
Apple Lossless: Also known as ALAC, Apple Lossless is similar to FLAC. It's a compressed lossless file, although it's made by Apple. Its compression isn't quite as efficient as FLAC, so your files may be a bit bigger, but it's fully supported by iTunes and iOS (while FLAC is not). Thus, you'd want to use this if you use iTunes and iOS as your primary music listening software.
Lossy Formats
APE: APE is a very highly compressed lossless file, meaning you'll get the most space savings. Its audio quality is the same as FLAC, ALAC, and other lossless files, but it isn't compatible with nearly as many players. They also work your processor harder to decode, since they're so highly compressed. Generally, I wouldn't recommend using this unless you're very starved for space and have a player that supports it.
For regular listening, it's more likely that you'll be using a lossy format. They save a ton of space, leaving you with more room for songs on your portable player, and—if they're high enough bitrate—they'll be indistinguishable from the original source. Here are the formats you'll probably run into:
MP3: MPEG Audio Layer III, or MP3 for short, is the most common lossy format around. So much so that it's become synonymous with downloaded music. MP3 isn't the most efficient format of them all, but its definitely the most well-supported, making it our #1 choice for lossy audio. You really can't go wrong with MP3.
AAC: Advanced Audio Coding, also known as AAC, is similar to MP3, although it's a bit more efficient. That means that you can have files that take up less space, but with the same sound quality as MP3. And, with Apple's iTunes making AAC so popular, it's almost as widely compatible with MP3. I've only ever had one device that couldn't play AACs properly, and that was a few years ago, so it's pretty hard to go wrong with AAC either.
Ogg Vorbis: The Vorbis format, often known as Ogg Vorbis due to its use of the Ogg container, is a free and open source alternative to MP3 and AAC. Its main draw is that it isn't restricted by patents, but that doesn't affect you as a user—in fact, despite its open nature and similar quality, it's much less popular than MP3 and AAC, meaning fewer players are going to support it. As such, we don't really recommend it unless you feel very strongly about open source.
WMA: Windows Media Audio is Microsoft's own proprietary format, similar to MP3 or AAC. It doesn't really offer any advantages over the other formats, and it's also not as well supported. There's very little reason to rip your CDs into this format.
Alfabetical Order
3gp = multimedia container format can contain proprietary formats as AMR, AMR-WB or AMR-WB+, but also some open formats.
act = ACT is a lossy ADPCM 8 kbit/s compressed audio format recorded by most Chinese MP3 and MP4 players with a recording function, and voice recorders.
aiff = Apple standard audio file format used by Apple. It could be considered the Apple equivalent of wav.
aac the Advanced Audio Coding format is based on the MPEG-2 and MPEG-4 standards. Are usually ADTS or ADIF containers amr AMR-NB audio, used primarily for speech.
au = Sun Microsystems the standard audio file format used by Sun, Unix and Java. The audio in au files can be PCM or compressed with the µ-law, a-law or G729 codecs.
awb = AMR-WB audio, used primarily for speech, same as the ITU-T's G.722.2 specification.
dct = NCH Software. A variable codec format designed for dictation. It has dictation header information and can be encrypted (as may be required by medical confidentiality laws). A proprietary format of NCH Software.
dss = Olympus. Files are an Olympus proprietary format. It is a fairly old and poor codec. Gsm or mp3 are generally preferred where the recorder allows. It allows additional data to be held in the file header.
dvf = A Sony proprietary format for compressed voice files; commonly used by Sony dictation recorders.
flac = File format for the Free Lossless Audio Codec, a lossless compression codec.
gsm = Designed for telephony use in Europe. Is a very practical format for telephone quality voice. It makes a good compromise between file size and quality. Note that wav files can also be encoded with the gsm codec.
iklax = An iKlax Media proprietary format, the iKlax format is a multi-track digital audio format allowing various actions on musical data, for instance on mixing and volumes arrangements.
ivs = 3D Solar UK Ltd. A proprietary version with Digital Rights Management developed by 3D Solar UK Ltd for use in music downloaded from their Tronme Music Store and interactive music and video player.
m4a = An audio only MPEG4 file. Used by Apple for unprotected music downloaded from their iTunes Music Store. Audio within the m4a file is typically encoded with AAC, although lossless ALAC may also be used.
m4p = Apple. A version of AAC with proprietary Digital Rights Management developed by Apple for use in music downloaded from their iTunes Music Store.
mmf = Samsung. a Samsung audio format that is used in ringtones.
mp3 = MPEG Layer III Audio. Is the most common sound file format used today.
mpc = Musepack or MPC. Formerly known as MPEGplus, MPEG+ or MP+. Is an open source lossy audio codec, specifically optimized for transparent compression of stereo audio at bitrates of 160–180 kbits.
msv = Sony. A Sony proprietary format for Memory Stick compressed voice files.
ogg / oga = Xiph.Org Foundation. A free, open source container format supporting a variety of formats, the most popular of which is the audio format Vorbis. Vorbis offers compression similar to MP3 but is less popular.
opus = Internet Engineering Task Force. A lossy audio compression format developed by the Internet Engineering Task Force (IETF) and made especially suitable for interactive real-time applications over the Internet. As an open format standardised through RFC 6716, a reference implementation is provided under the 3-clause BSD license.
ra/rm = RealNetworks. A RealAudio format designed for streaming audio over the Internet. The ra format allows files to be stored in a self contained fashion on a computer, with all of the audio data contained inside the file itself.
raw = a raw file can contain audio in any format but is usually used with PCM audio data. It is rarely used except for technical tests.
sln = S Linear format used by Asterisk.
tta = The True Audio, real-time lossless audio codec.
vox = The vox format most commonly uses the Dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. Similar to other ADPCM formats, it compresses to 4-bits. Vox format files are similar to wave files except that the vox files contain no information about the file itself so the codec sample rate and number of channels must first be specified in order to play a vox file.
wav = Standard audio file container format used mainly in Windows PCs. Commonly used for storing uncompressed (PCM), CD-quality sound files, which means that they can be large in size—around 10 MB per minute. Wave files can also contain data encoded with a variety of (lossy) codecs to reduce the file size (for example the GSM or MP3 formats). Wav files use a RIFF structure.
wma = Microsoft. Windows Media Audio format, created by Microsoft. Designed with Digital Rights Management (DRM) abilities for copy protection.
wv = Format for wavpack file.
Denis van der Velde
AAMS Auto Audio Mastering System
www.curioza.com
The users that do want their music mixed or mastered by Aplus Audio Mastering Services should consider to use AAMS first. When you are in need of Aplus Audio Mastering Services by our Professionals, you can read about our excellent audio mixing and mastering services further here.
Aplus Audio Mastering Services
Welcome to APlus Audio Mastering Services!
High quality audio mastering services with low prices.
The users that do want their music mixed or mastered with more needs then the AAMS program supplies directly, or users that want their mastering done by a human ear or just want more equipment or mastering tricks to be used, can visit our mastering services. But also be sure to rty out AAMS V3 first and only when your needs are not met, maybe consider your masterings to be done with the human factor in place, and you can supply us with the information about you want your sound to be like. You can consider mastering your audio by Aplus! When you are in need of Aplus Audio Mastering Services by our Professionals, you can read about our excellent audio mixing and mastering services further here.
AAMS Auto Audio Mastering System For windows users we provide AAMS complete audio mastering software package for Audio Mastering, free of use. Process your Mix to a commercial great sounding Master.
AAMS Auto Audio Mastering System
For windows users we provide AAMS complete audio mastering software package for Audio Mastering, free of use. Process your Mix to a commercial great sounding Master. With AAMS you get a clean transfer of your mix to a good sounding master. AAMS will do this lifting up the mix, but trying to preserve the same sound. Therefore when your mix has a particular sound, AAMS will not hurt it but will only try to compare source and reference and apply that with mastering techniques available. That is great for most users who do their own mixing and creating the sound is mostly done at the mixing stage. AAMS improves the mix towards quality commercial levels for al kinds of musical styles.
Audio Mastering
Mastering is audio post production, is the process of preparing and transferring recorded audio containing the final mix towards a data storage device the master. From which all copies will be produced via methods such as pressing, duplication or replication. Mastering is a crucial gateway between production and consumption and. It involves technical knowledge as well as specific aesthetics.Results still depend upon the accuracy of speaker monitors and the listening environment. Mastering engineers may also need to apply corrective equalization and dynamic compression in order to optimise sound translation on all playback systems. It is standard practice to make a copy of a master recording, known as a safety copy, in case the master is lost, damaged or stolen.
Welcome to APlus Audio Mastering Services!
High quality audio mastering services with low prices.
We maximize your audio material to clear crisp audio mastered versions. Giving detail, clarity, definition, warm low end punch, stereo depth and prisitine sound using Analog and Digital equipment. Audio Mastering is most important part of the audio chain and we give your sound high definition and clarity. Every single track and every song of your album collection wil sound complete next to each other, with taking care from start to end. Ready for distribution and radio. That is what Aplus Audio Mastering Services stands for, excellence! Your audio material or mixes will become an adequate to commercial radio, CD or MP4/MP3 streaming services, just to fit in correctly. We do not attend the Loudness War, you need appropiate levels and professional quality!
Aplus Mastering will make single tracks or songs stand out in excellence.
Aplus Mastering will make your whole Album Sound perfect and give it togetherness.
Aplus Mastering will create a professional sound for all of you.
We make your music shine!
Supply us with your audio material, mixed down to stereo in one of the formats below.
We prefer to accept the following formats:
Sending or submit your mixes to Aplus Mastering over the Internet or post, from yours to us and get it back in no time.
Upload your final mix with our easy to use upload system.
Goto the 'Shop' and decide what option you need :
- For single audio track mastering, choose Single Track.
- For multiple tracks that need mastered as single tracks, choose Multiple Tracks.
- Finally when you have multiple tracks that need to be sounding as an album, choose Full Album.
Instructions will follow afther you have chosen.
On your own User Account page, you can sign up for your account. Or go directly to our Shop and chose. Then Upload your songs / tracks with our easy webbased upload system. We will send you back a mastered sample of your song, if you do not pay upfront. When you have paid and accept the our high quality masterings when we are finished with your content, we will send you an email and place all mastered song / tracks on your personal w web account. And you can download the finished product from your own user directory!
There are a number of audio mixing and editting tips that will help you prepare your mixes before submitting to Aplus Audio Mastering studio.
It is important to know how to prepare your mix, so you can get the best sound for your songs!
When quality is at stake, be sure to read this page and spend some time to get your mixes right.
Audio mastering is a process that stands far from mixing, it is the next stage afther mixing and it is the final stage for sound quality. Actually while mixing we do not attend the loudness much, we mix. What everybody is thinking of 'How to get our mix sound loud'! That is what Aplus Audio Mastering stands for, most likely preferred that your mix will become an adequate to commercial radio, CD or MP3 streaming levels, just to fit in correctly. We do not attend the Loudness War, but we need appropiate levels and professional quality. Also when Mastering a Full Album, Aplus Mastering will make the whole Album Sound as an Album. We name it 'the album sound'. So Aplus Mastering can do single tracks as well as full albums, and create a good quality professional sound for you. But however, mixing is an important stage before Aplus Mastering can be done. So we ask you to attend some time and thought before sending your mixes to Aplus Mastering Engineers.
Check, Check, Double Check!
0. You should do these mix check steps before you plan to hand your project to our Aplus Mastering Engineer.
1. Eliminate any noise or pops that may be in each single track. Apply fades or cuts or mutes to spots containing recorded noise, pops or clicks.
2. Keep Your Mix Clean And Dynamic. Unless there is a specific sound you need, do not put compressing or processing on the master out of the mixing bus. It is best to keep the master buss free of outboard processing or plugins. Dont add any processing to the overall mix, just to individual channels.There should never be a limiter or loudness maximiser set on the master out mix bus!
3. The loudest part in a mix should peak at no more that -3db on the master bus, leaving headroom. It does not matter How Loud your mix sounds at this time, mixing means mixing.
4. Does your mix Work In Mono? As a final reality check, switch the master buss output to mono and make sure that there is no weakening or thinning out of the sound. In any event, do not forget to switch the bussing back to stereo afther this check.
5. Only when a mix is completed and finished off, and your are happy with the overall mixing sound and quality, then the next fase is Aplus Mastering to do their work.
6. Normalising a track is not necessarily a good idea.
7. Dont add any fades or crossfades, anywhere. Dont fade beginning or end.
8. Do not dither individual mixes.
9. You can output the mix on a stereo track before sending to Aplus Mastering, save your mix in Stereo. Use a lossless format! Using digital equipment Wav 32Bit Float Stereo is a good output format.
10. Do not try to output your mix to a mp3 file, this can mean loss of information! If you do want to send in MP3 files, be sure they are of quality, prefer a bitrate higher than > 192kbps, 320kbps is quite good.
11. Export your mix out of your sequencer or audio setup in a correct and quality unharming format;
12. Finally, always back up your original mixed files! If the song is later remastered for any reason for a re-release, a compilation or for use in any other context you will want a mix thats as easy to remaster as possible.
13. Submitting reference tracks or example songs alongside your mix submission that have a similar sound desired is for a good point of view how your music must sound. Giving Aplus Audio Mastering Engineers an idea of your musical vision. This could be a reference to bands who inspire you or have a similar sound that you like.
14. Put all your files of a single mix (the stereo file, reference songs, text documents or pictures or any file that you need to send) in one single directory.
15. Use a packing program like ZIP, RAR, 7z and pack all files in that directory to one single packed file. Name this file correctly, preferably the track number and name of the track.
16. Now you can send your mix files to Aplus Audio Mastering Services!
Mastering Stems
Mastering from stems is becoming little by little more common practice. This is where the mix is consolidated into a number of stereo stems subgroups to be submitted individually. Instead of submitting a Stereo output of your mix, you can send the mixing tracks seperately. For example you might have different tracks for Drums, Bass, Keys, Guitars, Vocal, and Background Vocals. This will give Aplus Mastering more control over the mix and master. If a master from stems is desired, following the same steps listed above is best for each stem. When submitting stems each file track must start at the beginning and must durate though the end, most mixing sequencers will output this way exactly to the sample. Each stem file should be exactly the same length.
We master all Musical Styles :
- Accoustic
- Amine
- Blues
- Commercials
- Classical Music
- Chrildrens Music
- Holiday, Christmass
- Conferences
- Country Music
- Disco
- Dubstep
- Easy Listening
- Electronic Music
- Fusion
- Folk Music
- Funk
- Gospel, Inspirational
- Hardrock
- Heavy Metal
- House, EDM, Electro, Trance, etc.
- Industrial
- Instrumental
- Karaoke
- Live Performances
- New Wave, New Age
- Rap
- Opera
- Pop Music
- Reggae
- R&B
- Rock
- Singer Songwriter
- Soundtrack
- Soul
- Latin, Spanish
- Trance Music
- TV Themes
- Vocal
- World Music
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