Allright,
I'm puzzled with mastering. My mixed down files don't sound loud! even though I've put my master limiter and my VU meter is practically in the roof.
Okay I admit, first of all.. I'M still mastering from headphones... I Know... don't scream at me...I'll have speakers/monitors in some time.
But still. Here's my wierd question:
How does your frequency spectrum affects your VU meter!! I mean, let's use a narrow band of frequency up til the vu meter is at 0. How high should it be? Which width should it be?
Is there a general principle? like width of frequency x amplitude = vu meter ?
I mean...I guess I would need to narrow the frequency range of every track of my songs so that they don't overlap too much. Would it ehance my loudness? Is this the way I should go about it?
thanks to anyone who takes the time of answering.
weird question
What you should be looking at and thinking about is energy.
The end result, the output to your amplifier, is indeed energy and that means the VU is only averaging (even when in peak mode) the amount of energy the file is creating.
Frequency band is starkly limited in audio, we talk of "20 to 20" -- meaning 20Hz to 20KHz of the "CD Quality" recording -- but the reality of the musical tones is even more truncated than that, consider that the 88 key piano covers the entire range of musical instruments and its lowest A is 27.5Hz and its highest C is 4186.0Hz. Never mind all the talk about harmonics, this is where the majority of the energy is taking place.
The content of your tracks has a lot to do with it, too. Without being able to hear and analyze what you've currently got going on it is difficult if not impossible to tell you where to place interest or what to do. If you could post a "before" mixdown and an "after" -- unMastered and Mastered, it might be possible to give you more definitive answers concerning your problem at the present.
I can tell you this with authority, though:
Headphones are not going to cut it.
You must be able to hear what you have on a good set of reference monitors that you KNOW. That means getting the monitors and then doing a lot of reference listening to recordings that have been mixed and mastered well, until you can hear the difference immediately on your monitors.
MTS has excellent Mastering stuff built into it, but one must know the theory behind the plugins and also the care and feeding of them plus develop those ears along with the visual aids in order to bring them out.
Comparing your mixdowns with pro recordings of the same exact genre and type of music using a good Spectrum Analyzer can often pinpoint the frequency areas where yours differs from the pro stuff, too. There are also other factors to consider. Noise from poor converters can be a problem, so can JITTER of a poor clock. These things will lead you to have to turn amplitude down in order not to emphasize these bad things. Function of soundcard type and converters and clock.
What are you using for a sound device?
--Mac
The end result, the output to your amplifier, is indeed energy and that means the VU is only averaging (even when in peak mode) the amount of energy the file is creating.
Frequency band is starkly limited in audio, we talk of "20 to 20" -- meaning 20Hz to 20KHz of the "CD Quality" recording -- but the reality of the musical tones is even more truncated than that, consider that the 88 key piano covers the entire range of musical instruments and its lowest A is 27.5Hz and its highest C is 4186.0Hz. Never mind all the talk about harmonics, this is where the majority of the energy is taking place.
The content of your tracks has a lot to do with it, too. Without being able to hear and analyze what you've currently got going on it is difficult if not impossible to tell you where to place interest or what to do. If you could post a "before" mixdown and an "after" -- unMastered and Mastered, it might be possible to give you more definitive answers concerning your problem at the present.
I can tell you this with authority, though:
Headphones are not going to cut it.
You must be able to hear what you have on a good set of reference monitors that you KNOW. That means getting the monitors and then doing a lot of reference listening to recordings that have been mixed and mastered well, until you can hear the difference immediately on your monitors.
MTS has excellent Mastering stuff built into it, but one must know the theory behind the plugins and also the care and feeding of them plus develop those ears along with the visual aids in order to bring them out.
Comparing your mixdowns with pro recordings of the same exact genre and type of music using a good Spectrum Analyzer can often pinpoint the frequency areas where yours differs from the pro stuff, too. There are also other factors to consider. Noise from poor converters can be a problem, so can JITTER of a poor clock. These things will lead you to have to turn amplitude down in order not to emphasize these bad things. Function of soundcard type and converters and clock.
What are you using for a sound device?
--Mac
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mcairenius
- Posts: 44
- Joined: Wed Oct 27, 2004 5:34 pm
I have allways thought how ever loud you want it however quite you want it do it!, It all depends on you. I think. Also before I use to do a mixdown with only head phones, I thought hmmm, if I do the mixdown with my speakers hearing it mabey it will sound better, it dose in some cases.
See, how ever you want the music to sound, how loud or how quite you want it will affect how the music sound, it's your creation not mine, unless ofcourse you want someone else to do it. : )
Thats my answer.. peace!
Marlon
See, how ever you want the music to sound, how loud or how quite you want it will affect how the music sound, it's your creation not mine, unless ofcourse you want someone else to do it. : )
Thats my answer.. peace!
Marlon
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NystagmusE
Remember that a VU meter is different from a Peak indicator or a computer graph. A VU meter has a needle that follows the average level. A Peak indicator is an LED that lights up when the volume goes past a certiain level. A computer bargraph indicates the instantaneous levels as if going through a series of progressively higher Peak indicators. There are other types of volume indicators too, but I won't explain them.
A typical computer bargraph indicator goes from -infinity to 0 dBfs, while a VU meter goes up to some positive level above 0 dBVU. VU meters respond more slowly than computer bargraph indicators most of the time.
On the same source material, transient high frequency signals can light up a Peak indicator or computer bargraph rather quickly and at high levels while the VU meter moves slower and at a lower dB level.
VU meters can be calibrated to different levels. Not all VU meters are alike.
The graph in MultitrackStudio seems to go from -infinity to 0 dBfs and then continues to some positve level above 0 dBfs, even though it is really a computer bargraph indicator. It is not a VU meter.
I have found that I can get a higher overall volume in songs by not using the Master Limiter and instead letting the signal go into the positive red zone of the bargraph up to the very top. This means that the signal is clipping when saved to a file, but there is a trick to it: If you can't hear the clipping distortion, then it is possible to use this .WAV file.
One issue with this technique is that you have to use 24 bit files. Bad distortion happens with the 32 bit formats when you save clipped files. But if the clipping is inaudible on a 24 bit file, it can be saved just fine as a 24 bit or 16 bit file. This is a technical issue of the file formats.
This technique probably only works on hard percussive material, and not ambient music or soft music. It works when the beats have higher amplitude than everything else.
Many people will tell you that this is a bad technique and that you should never allow clipping, but in my experience, if you can't hear the clipping, then you can get away with it. It's possible that the clipping puts some stress on the speakers when played back, but this is only theoretical. Typically, inaudible clipping happens very fast so the speakers wouldn't be exposed to very much DC current from the squarewaves of the clipped waveform. I could be wrong about this, but I play my songs on a variety of playback systems and nothing overheats; it just plays like regular music.
I do this with a most of my songs. I don't use compression or limiting, and I also monitor with headphones still. (I will master my songs when I get monitors someday in the future).
If you want to hear some songs of mine that were recorded this way with clipping, here you go: (128 kBps MP3's)
Transcender: http://www.soundclick.com/util/getplaye ... 33304&q=hi
Tribulation: http://www.soundclick.com/util/getplaye ... 14880&q=hi
Primary: http://www.soundclick.com/util/getplaye ... 73252&q=hi
Jasmine: http://www.soundclick.com/util/getplaye ... 87678&q=hi (you can hear the clipping on the beats in this song if you listen carefully)
Amazon: http://www.soundclick.com/util/getplaye ... 25117&q=hi
All of those songs were recorded using 24 bit tracks and 16 bit drum tracks. They were then directly mixed down with dither to .WAV. They were then encoded to 128 kBps MP3 using iTunes.
A typical computer bargraph indicator goes from -infinity to 0 dBfs, while a VU meter goes up to some positive level above 0 dBVU. VU meters respond more slowly than computer bargraph indicators most of the time.
On the same source material, transient high frequency signals can light up a Peak indicator or computer bargraph rather quickly and at high levels while the VU meter moves slower and at a lower dB level.
VU meters can be calibrated to different levels. Not all VU meters are alike.
The graph in MultitrackStudio seems to go from -infinity to 0 dBfs and then continues to some positve level above 0 dBfs, even though it is really a computer bargraph indicator. It is not a VU meter.
I have found that I can get a higher overall volume in songs by not using the Master Limiter and instead letting the signal go into the positive red zone of the bargraph up to the very top. This means that the signal is clipping when saved to a file, but there is a trick to it: If you can't hear the clipping distortion, then it is possible to use this .WAV file.
One issue with this technique is that you have to use 24 bit files. Bad distortion happens with the 32 bit formats when you save clipped files. But if the clipping is inaudible on a 24 bit file, it can be saved just fine as a 24 bit or 16 bit file. This is a technical issue of the file formats.
This technique probably only works on hard percussive material, and not ambient music or soft music. It works when the beats have higher amplitude than everything else.
Many people will tell you that this is a bad technique and that you should never allow clipping, but in my experience, if you can't hear the clipping, then you can get away with it. It's possible that the clipping puts some stress on the speakers when played back, but this is only theoretical. Typically, inaudible clipping happens very fast so the speakers wouldn't be exposed to very much DC current from the squarewaves of the clipped waveform. I could be wrong about this, but I play my songs on a variety of playback systems and nothing overheats; it just plays like regular music.
I do this with a most of my songs. I don't use compression or limiting, and I also monitor with headphones still. (I will master my songs when I get monitors someday in the future).
If you want to hear some songs of mine that were recorded this way with clipping, here you go: (128 kBps MP3's)
Transcender: http://www.soundclick.com/util/getplaye ... 33304&q=hi
Tribulation: http://www.soundclick.com/util/getplaye ... 14880&q=hi
Primary: http://www.soundclick.com/util/getplaye ... 73252&q=hi
Jasmine: http://www.soundclick.com/util/getplaye ... 87678&q=hi (you can hear the clipping on the beats in this song if you listen carefully)
Amazon: http://www.soundclick.com/util/getplaye ... 25117&q=hi
All of those songs were recorded using 24 bit tracks and 16 bit drum tracks. They were then directly mixed down with dither to .WAV. They were then encoded to 128 kBps MP3 using iTunes.
-
NystagmusE
Another thing I forgot to mention...
If you high pass filter your sound files to remove everything below 20 Hz, sometimes you can get a louder sound because you gain back headroom from the extra low bass frequencies. I used to high pass filter everything below 40 Hz until I noticed that some of that 20-40 Hz sub bass is nice on some songs. On my synthesizer, some of the waveforms can be reduced by 50 % in amplitude with this type of filtering.
I prefer to do this with the parametric EQ in CoolEdit instead of with a VST plug in, because I can use a steeper filter and I'm guessing that it is a higher quality filtering than with a VST.
I make it a habit to process all of my drum sounds this way so that I don't end up with too much sub bass. It also helps to get rid of DC offset which can cause clicks and also takes up headroom.
By the way, don't feel bad about mastering with headphones, the January 2007 issue of Sound On Sound was about mixing on headphones.
If you high pass filter your sound files to remove everything below 20 Hz, sometimes you can get a louder sound because you gain back headroom from the extra low bass frequencies. I used to high pass filter everything below 40 Hz until I noticed that some of that 20-40 Hz sub bass is nice on some songs. On my synthesizer, some of the waveforms can be reduced by 50 % in amplitude with this type of filtering.
I prefer to do this with the parametric EQ in CoolEdit instead of with a VST plug in, because I can use a steeper filter and I'm guessing that it is a higher quality filtering than with a VST.
I make it a habit to process all of my drum sounds this way so that I don't end up with too much sub bass. It also helps to get rid of DC offset which can cause clicks and also takes up headroom.
By the way, don't feel bad about mastering with headphones, the January 2007 issue of Sound On Sound was about mixing on headphones.
-
mcairenius
- Posts: 44
- Joined: Wed Oct 27, 2004 5:34 pm