What db should i normalize to

This is where the normalization effect comes into play. It helps to eliminate clipping by reducing the dynamic range between the quieter and louder sounds.

Normalization was first developed in hardware systems such as loudspeakers to increase the volume at the end of delivery to make it sound loud. Later it started to make its way towards production systems. This all changed the moment software instruments and audio production software came into the spotlight. They completely changed the way mixing is done and put normalization to rest. This has been the case over the last twenty years. Still, normalization is used to make individual tracks sound loud and crystal clear without clipping inside digital production systems.

Normalization techniques are used in hardware systems and are not on par with the systems of limiting and clipping. Limiting and clipping offer more to the mixing and mastering engineer in terms of making the song sound loud when compared to normalization.

The only place where normalization makes more sense is when there is a little place to improve the loudness of the track without clipping and causing distortion.

This is one place where you can apply normalization to boost the quieter parts of the song and get it to be par with the other areas in the song. Another disadvantage is that, whenever you have a song that has very low head space, you cannot boost it with a limiter as you will lose detail. This is where you would expect a normalization is a tool can come to the rescue. This is not always the case in all situations, because normalization will take away the ability to increase RMS and reduce the dynamic range in the song making it less interesting.

In terms of loudness, the value is around LUFS. Best practise these days when considering content normalization is not to normalize to an arbitrary peak level, but to consider programme content and overall loudness when conducting normalization activities. You should consider normalizing content according to the content standards of the platform where you will be delivering the material. Use of these loudness measurement techniques allows content to be easily altered to match surrounding content through simple gain changes.

You will find also that music content has different normalization conventions when considering loudness. It's also worth investigating how ReplayGain fits into this model in a music context. Sign up to join this community. The best answers are voted up and rise to the top. Stack Overflow for Teams — Collaborate and share knowledge with a private group. Create a free Team What is Teams? Learn more. In general, what's a good dB level at which to normalize my audio files?

Ask Question. Asked 5 years, 2 months ago. Active 1 year, 9 months ago. Viewed 20k times. Why would I choose anything different than " In general, what are some best practices to consider when normalizing audio?

Improve this question. Pup Pup 1 1 gold badge 1 1 silver badge 5 5 bronze badges. Add a comment. So the loudest part of your audio gets turned up to as loud as it can be before clipping.

Let's say the loudest part of your audio, which is a vocal recording in our example, is a part where you shout something. See Figure 2 for an example. It wants to turn up that shouted audio to 0 dB. So it needs to know the difference between how loud the shout is, and the loudest it could possibly be before distorting. The simple math well, that is if negative numbers didn't freak you out too much is that 0 minus -6 equals 6.

So the difference is 6 dB. Remember, all the program is doing is changing the level of the loudest bit of audio to a target you choose, and changing all the rest of the audio by that same amount. So that means if you place the setting to a target that is LOWER than the loudest part the shout in our example above , the normalizing will turn everything down by the amount it takes for the loudest part to meet the target.

So let's say the loudest part of your audio - the shout - is at -2 dB. If you set your normalization target to -3 dB, then the effect will LOWER everything by 1 dB, which is the amount of reduction you need to get your -2 dB peak down to -3 dB.

So why would you want to do this? Certain services have maximum loudness standards. For example, if you are recording an audiobook for Audible using their ACX marketplace , they will not accept audio that has a peak level above -3 dB. That means our audio with the shout that goes as high as -2 dB breaks their rule because that is louder than -3 dB. So you can use normalization to reduce your loudest peak by setting the target to just under -3 dB, like say A normalization effect might offer percentages as targets as well as specific dB targets.

Yeah, another thing about measuring audio is that it isn't linear. The point here is that if you use a percentage, might be setting the max loudness to lower than you think. If you have a choice to specify the actual dB level, do that. It will make things much easier to understand. That's it. Remember I said that audio normalization was really just turning it up? I know it took a fair amount of explaining, but yeah. The only reason to normalize your audio is to make sure that it is loud enough to be heard.

That could be for whatever reason you want. As I have preached again and again, noise is the enemy of good audio. This is shown on the Fletcher-Munson curve below. If one sound file has many frequencies between — Hz as shown in the diagram, it will sound louder.

Luckily there is a recent solution, the new standard in broadcast audio, the catchily titled EBU R This is a similar way to measure volume as RMS, but can be thought of as emulating a human ear. It listens to the volume intelligently and thinks how we will hear it.

It understands that we hear frequencies between — Hz as louder and takes that into account. We still have the same 0 dBFS problem mentioned for RMS, but now the different normalized audio files should sound much more consistent in volume. Normalization can be performed in a standalone program, usually an audio editor like Sound Forge , or also inside your DAW. For the sake of this section we are assuming you are using an audio editor. Nowadays audio editing software works internally at a much higher bit depth often bit floating point.

This means that calculations are done much more accurately, and therefore affect the sound quality far less. This is only the case if we keep the file at the higher resolution once it has been processed! To take advantage of the high quality of high bit depth inside audio editing software make sure all your temporary files are stored as bit floating point.