Adobe Audition vs Waves
- Thread starter dacalion
- Start date
thedreampolice
A backwards poet writes inverse.
They are really not at all the same product, Do you like a toaster or a TV better?
thedreampolice
A backwards poet writes inverse.
LMAO, but audition is an audio editor and waves are effects processing plugins.
Why not use waves inside audition?
Why not use waves inside audition?
thedreampolice
A backwards poet writes inverse.
Go to the settings and point it to your VST folder, the built in effects are not nearly as good.
thedreampolice
A backwards poet writes inverse.
Dude I was up until 4am yesterday, tell me about it!
OK let me explain it this way... I've submitted several beats for the Beat This and Battle That over time. The majority of those beats have sounded "equal to or better than" the originals (the versions in my sequencer), however I noticed that 2 of them sounded worse. Now I'm on this quest to make all of my beats sound better overall. I may be going about this the wrong way but thats why I'm asking you guys. It could just be me, I don't know for sure. Now, those same 2 beats sounded great on my website which leads me to believe that the problem lies somewhere else.
I've been doing a ton of testing with different programs (Waves, Goldwave, iZotope and Audition) in hopes of getting a louder, cleaner, clearer sound. While testing, I noticed a change when I convert to mp3 but I've heard some very clean, very clear and loud mp3's so I can't contribute my problems to the degradation that naturally occurs when converting to mp3.
On another note (that could be relative...), when I load a sample in Edison (my chopping program), it increases the overall volume of the sample automatically which also increases the impurities in that sample which adds to my need to clean up my beats clarity and volume. Ya feel me?
I've been doing a ton of testing with different programs (Waves, Goldwave, iZotope and Audition) in hopes of getting a louder, cleaner, clearer sound. While testing, I noticed a change when I convert to mp3 but I've heard some very clean, very clear and loud mp3's so I can't contribute my problems to the degradation that naturally occurs when converting to mp3.
On another note (that could be relative...), when I load a sample in Edison (my chopping program), it increases the overall volume of the sample automatically which also increases the impurities in that sample which adds to my need to clean up my beats clarity and volume. Ya feel me?
Mike Chief
ILLIEN
Yeah, honestly the worst format is mp3. It loses way too much data, and if you notice, it really loses a lot of data when there are reverbs and delays on vocals and/or the beat itself. Unfortunately, everyone wants mp3 and its the most accessible. However, I found one way to help solve it slightly, and its through the sample rate on new sessions. Sample rate is how fast your computer should be processing the binary code in the digital audio. So, the higher the sample rate, the more data that has to be configured and processed, hence more numbers and data to pick up the right sound. Try a sample rate of like 92 khz or w/e the secondest highest is, convert to mp3, and then take one of your previous mp3s and compare the sound. It should sound better.
Also, the .aiff is the highest quality format for digital productions, and the most universal.
Also, the .aiff is the highest quality format for digital productions, and the most universal.
thedreampolice
A backwards poet writes inverse.
Sorry Mike, that is not accurate or correct at all. Sampling rate is the number of samples per second (or per other unit) taken from a continuous signal to make a discrete signal. For time-domain signals, the unit for sampling rate is hertz (inverse seconds, 1/s, s−1). The inverse of the sampling frequency is the sampling period or sampling interval, which is the time between samples.
Also if his original session has not data at 96k, going there well not help.
Ponder this, humans can't hear above 22k but science to the rescue with
The Nyquist–Shannon sampling theorem states that perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the maximum frequency of the signal being sampled,[2] or equivalently, when the Nyquist frequency (half the sample rate) exceeds the highest frequency of the signal being sampled. If lower sampling rates are used, the original signal's information may not be completely recoverable from the sampled signal.
For example, if a signal has an upper band limit of 100 Hz, a sampling frequency greater than 200 Hz will avoid aliasing and allow theoretically perfect reconstruction.
At any rate 96k ONLY helps the converter be more efficient, because NO HUMAN can hear above 22k, but as you can see we need to sample at DOUBLE what we want to actually reproduce, this is why CD's are 16bit 44.1lk to reproduce 22k, Make sense?
Sampling rate is VERY different from bitrate, Dac try encoding your MP3 at a higher bit rate like 256k this will make things sound much better. This compresses the file less and will leave more high-end in tact. Also try the LAME MP3 encoder.
Also if his original session has not data at 96k, going there well not help.
Ponder this, humans can't hear above 22k but science to the rescue with
The Nyquist–Shannon sampling theorem states that perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the maximum frequency of the signal being sampled,[2] or equivalently, when the Nyquist frequency (half the sample rate) exceeds the highest frequency of the signal being sampled. If lower sampling rates are used, the original signal's information may not be completely recoverable from the sampled signal.
For example, if a signal has an upper band limit of 100 Hz, a sampling frequency greater than 200 Hz will avoid aliasing and allow theoretically perfect reconstruction.
At any rate 96k ONLY helps the converter be more efficient, because NO HUMAN can hear above 22k, but as you can see we need to sample at DOUBLE what we want to actually reproduce, this is why CD's are 16bit 44.1lk to reproduce 22k, Make sense?
Sampling rate is VERY different from bitrate, Dac try encoding your MP3 at a higher bit rate like 256k this will make things sound much better. This compresses the file less and will leave more high-end in tact. Also try the LAME MP3 encoder.
thedreampolice
A backwards poet writes inverse.
LMAO, I always turn in MP3's at 192k for ill
But play with the LAME encoder and that will help.
But play with the LAME encoder and that will help.
I was going to say that, I had the same problem, with peaks and distorion being introduced by the wav > mp3 conversion.
I think I solved it by converting to a very high bit rate mp3
(256 or 320 should do) and later convert down from that to 128 for myspace and other flash mp3 players, this stops their automatic converters ruining the conversion.
Im not sure but it may have something to do with the dividability of the bitrates or some shit like that.
Haha, its good to be appreciated
thanks.As for aiff, im not sure they are as good as FLAC. http://flac.sourceforge.net/
(Free Lossless Audio Codec) but then you need a flac player to play them and they are not a standard that can be played in mp3 players unless there is some new shit out. Aiff's are not standard unless you are a Mac user. I thought it was the audio standard just for macs.
thedreampolice
A backwards poet writes inverse.
.aiff and wave are basically the same, about anything can play it. But even on macs everyone uses wave now a days.
FLAC player - http://www.foobar2000.org/