Import the audio WAV generated by BIAB software into STUDIO ONE. Do I need to use After using it, I found that the waveform and sound were both louder. I personally prefer audio normalization because the sound becomes louder and the waveform becomes larger. However, I don’t know if it is right to do so? So, let me ask everyone. What do you usually do? Do you mix after audio normalization, or do you not need audio normalization? Are there any videos introducing these?
There is a problem, that is, I am not really good at mixing. But I like the sound to be louder and the waveform to be larger.
I used to normalise my audio files to 0 dB using BIAB as I imported them into Reaper. (It sounds like you've already found the setting, if you haven't, though, and you want BIAB to normalise each file individually, you can find the setting under "Audio | Export song as audio file". Doing this worked fine for me.
Then I read somewhere that it's better to leave a little bit of 'headroom' in audio signals so that audio effects could be applied without causing distortion or clipping. "Headroom' means to normalise to a value that is slightly less than 0dB. The book I read at the time recommended –0.3 dB. As a consequence, these days, I export all tracks from BIAB as they are and usually normalise to –3.0 dB in Reaper (which is well below the –0.3 dB I read about).
What normalising does is to increase every aspect of the sound by the same amount. When you normalise to 0 dB, the software scans the audio and looks for the loudest sound (this is the one with the highest peak). For example... Let's say this highest peak has a reading of –2.5 dB. To normalise this peak to 0 dB means that the wave will need to be lifted up by 2.5 dB. The normalise function then uses this information to increase every component of the wave in this example by 2.5 dB. This keeps all aspects of the wave in the same relationship with one another, it is just that the overall volume has increased.
I hope this helps. I'm pretty sure that there will be a number of replies to this question as everyone will have their own views on the topic.
Babymusic, you will quickly run into those who will advise you never to use normalization. I disagree; use whatever tool works best in each situation. But if I normalize, I set the target at -3 and the reason is what Noel said, to leave headroom for processing and effects during mixing, and then mastering. However, it’s been a long time since I used normalization since I discovered ‘smart’ limiters. They do some compression to accomplish raising the gain but unequally by compressing the total dynamic range. The first tools to do this somewhat that I’m aware of were T-Racks then Ozone. Now I use a separate smart limiter by Sonible called Smart Limit. I think it’s ‘smart’ because it makes different choices depending on the genre you choose and the desired amount of limiting (hard or soft).
If some of the terms I’ve used are unfamiliar, they will give you a start in your reading.
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Noel, good explanation. I suspect you changed your math example while writing. Normalizing from a peak of -2.5 up to 0 means raising all sounds by 2.5.
Let me understand what you are saying. See if I am right.
Let me understand what you said. See if I am right. I can use audio normalization, but I have to set the normalized audio volume to -3db to leave room for later mixing. (I want to insert a sentence here. I think it is still necessary to normalize, because some sounds are really too small. If I don’t normalize, I always have to turn up the computer speakers)
Now there is a software called "Smart Limit", which can increase the volume, and it is very smart, so you can hang "Smart Limit" on each track to increase the loudness. So there is no need for normalization?
So, can I do this, for some audio files with very low loudness, I still use normalization, and then I lower the volume to -3db, and then I use "Smart Limit" to intelligently increase the brightness. Is this okay?
Finally, let me talk about my specific situation. For example, I use NEXUS software to edit a drum, the sound is loud enough, I can see the waveform and the sound is loud enough, so I don’t need to normalize. Or I click normalization in DAW, and I can’t see any changes.
However, the audio I export using BIAB or KONTAKT guitar is very small. If I don’t standardize it, the sound and waveform will be small, so I must standardize it.
To the O/P, it is important to understand that sound levels are heard by humans on a logarithmic scale. The decibel unit is the 'intensity' of the sound. In practice, increasing a sound level by 10db will actually make it twice as loud as it was previously. Reducing a sound level by 10db will halve the intensity of the sound. Leaving -3.0db 'headroom' is somewhat of an 'industry standard' to allow for fine adjustments (filters, compressors, etc) without compromising the overall sound level.
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For what it is worth I will add one other option. In Studio One Pro you can click on an audio track then while holding down the alt key you can raise or lower the track's volume by also pressing the + or - sign; + increased the volume while - lowers it. The track's wav also gets larger or smaller. This is an option that I use a lot.
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To the O/P, it is important to understand that sound levels are heard by humans on a logarithmic scale. The decibel unit is the 'intensity' of the sound. In practice, increasing a sound level by 10db will actually make it twice as loud as it was previously. Reducing a sound level by 10db will halve the intensity of the sound. Leaving -3.0db 'headroom' is somewhat of an 'industry standard' to allow for fine adjustments (filters, compressors, etc) without compromising the overall sound level.
Technically that isn't really correct for all sorts of reasons.
I think what you're saying is that an increase of 10dB makes the sound subjectively twice as loud, or reducing it 10dB make is subjectively half as loud.
A decibel has a specific definition, which is somewhat different from that. An actual "twice the level" is only +3dB and, as you indicate, it doesn't sound twice as loud.
Jazz relative beginner, starting at a much older age than was helpful. AVL:MXE Linux; Windows 11 BIAB2025 Audiophile, a bunch of other software. Kawai MP6, Ui24R, Focusrite Saffire Pro40 and Scarletts .
. However, it’s been a long time since I used normalization since I discovered ‘smart’ limiters. They do some compression to accomplish raising the gain but unequally by compressing the total dynamic range. The first tools to do this somewhat that I’m aware of were T-Racks then Ozone. Now I use a separate smart limiter by Sonible called Smart Limit..
I used a pressure limiter to increase the volume. Sometimes the sound is even too loud. I will reduce it a bit. I don't know if this is the right operation. But the sound has indeed increased, but it's not overloaded yet.
. . .it is important to understand that sound levels are heard by humans on a logarithmic scale.
Yes, we do hear on a logarithmic scale but that’s half the story. The other half is that we hear on a highly non-linear basis as a function of frequency. Some of the first research on how the ear hears different frequencies at different levels was done by Fletcher and Munson in 1933; brilliant work and the design of our auditory system is truly astonishing.
A simplified description of the several transduction steps begins with the physics of a pressure wave. The acoustic energy is converted to mechanical energy in the eardrum, which causes the fluid-filled cochlea to vibrate. This fluid motion causes the hair sensory cells to bend, causing electrical signals to be generated which are fed to the auditory cortex which processes the signals allowing us to perceive, understand and interpret the information.
I’m not aware of a single unified equation that encapsulates both the logarithmic aspect and the frequency-based equal loudness curves. If I had such an equation, I might be able to produce a 3-D surface plot showing the complete 2 input and 1 output relation. If no such relation exists (which would surprise me) I’m guessing that hearing aid designers use nonlinear regression or table look-up methods. Perhaps someone that wears them or has worked in that industry may know more about how they do their DSP in this regard.
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I typically mix songs so that the peaks are maximum -6dBFS, and I don't normalize since mastering (specifically the master limiter) would effectively do that for me. At least some headroom is necessary to prevent clipping.
Originally Posted by Noel96
What normalising does is to increase every aspect of the sound by the same amount. When you normalise to 0 dB, the software scans the audio and looks for the loudest sound (this is the one with the highest peak.
You're talking about peak normalization, however there are some apps that can normalize to an average or RMS value. When doing this, it's possible that the volume level could be reduced to hit the target RMS, or could be increased to the point of clipping.
Originally Posted by babymusic
I personally feel that louder sounds inspire me more.
Loud music typically is perceived to sound "better". Part of the reason is because our ears don't respond linearly to sounds of all frequencies, and as the volume increases the ears become more sensitive to low and high frequencies and less sensitive to mids (the Fletcher-Munson curves) - and because music is often mixed at a higher volume level (typically around 80-86dB) that's where the mix usually sounds "best". Everything is subjective though.
Last edited by Simon - PG Music; 01/09/2512:11 PM.
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