6

You are most likely experiencing the non-linear frequency reception of the human hearing and the frequency response in your playback system. As listener you perceive a constant loudness of steady tones roughy following this chart (Equal loudness contours or "Fletcher-Munson" curves): In addition your headphones or speakers frequency response curve is also ...


5

There certainly isn't one single reason, and, yes, “that's just how it's always been” is probably an important point. However, I would quite agree that guitars should not be part of a standard orchestra: for the same reason pianos aren't, and shouldn't be, part of an orchestra. Orchestral music takes a lot of its attractiveness from the art to ...


4

Indeed it's easy to generate very loud noises with a speaker system whose power is only a fraction of what's actually used in PA or HiFi systems. What's not so easy is to get any sound you want to that level, without severely distorting it. Perhaps most relevantly for music: bass frequencies need way more power to be perceived as loud as a 1 kHz signal. And ...


3

Amplitude is an objective measurement. In a waveform, it is the value of the y axis at any given point in time. Loudness is a subjective measurement, based on how we perceive amplitude and other psychoacoustic dynamics like the Fletcher–Munson curves. There is no absolute way to measure loudness, there are many different options that will give you ...


3

My suggestion is to do three steps. Step one, using your EARS, adjust the volume of the various portions of your program so that they have the correct relative volume that you want. If you want them to all sound the same, adjust it until they do sound the same regardless of what your meters are telling you. An ordinary peak or even RMS meter will not be a ...


3

OK, so not a really comprehensive question from me, but really I posted this question simply so that I could forward this link into the stack, which I believe is useful to update people on more up-to-date practises for loudness normalisation and mixing for streaming and content delivery across multiple platforms.


3

It is true that the sensitivity of our ears varies based on frequency and that high pressure sound can be more damaging without being noticed, but if you are not listening too loud it shouldn't be a problem. You just need to be really careful that it isn't actually too loud. It is possible to damage your hearing without feeling any pain when you are using ...


2

Some of that chart isn't very good in my opinion. 85dB isn't that loud. A typical concert will be at 106dB to 110dB or sometimes louder and the power requirement doubles every 6dB or so because it is an exponential scale. Additionally, sound falls off very rapidly as well. You might be able to produce an 85dB sound with one watt of power at the cone of ...


2

As someone who comes from a sound company, the only way you can be sure that it would negatively effect the headset or headphones in said question is whether it would pass the specification of the speaker or subwoofer in the headphones. Depending on its use and how frequently you push it past it's limit (Mild distortion or blurred sound) is the point where ...


2

Welcome to broadcast mixing. ;) I would suggest using an EQ to carve out some space in the music track to fit it around the dialog. That way, you can pull the dialog track down a little without having to touch the music's levels, and it should fight less. You can do that and set it to just hold over the whole music track, or you can automate the ducks in ...


2

Master Reference documentation relating to EBU and ITU Recommendations that relate to loudness measurement techniques and algorithms can be found at : https://tech.ebu.ch/docs/r/r128.pdf https://www.itu.int/pub/R-REP-BS.2054-4-2014 https://www.itu.int/rec/R-REC-BS.1770-4-201510-I/en Loudness measurement techniques apply: Channel filtering (K-Weighting) ...


2

0 LU is defined by the EBU as -23 LUFS (https://tech.ebu.ch/docs/tech/tech3341.pdf) : The scale used may either be an absolute scale, using the unit ‘LUFS’, or alternatively the zero point may be mapped to some other value, such as the target loudness level (as in ITU-R BS.1771). In the latter case the unit shall be ‘LU’, indicating a ...


1

As mentioned in a comment, FFMPEG will revert to dynamic in the conditions specified in the manual: linear Normalize by linearly scaling the source audio. measured_I, measured_LRA, measured_TP, and measured_thresh must all be specified. Target LRA shouldn’t be lower than source LRA and the change in integrated loudness shouldn’t result in a true peak which ...


1

The situation you are describing is where I would use a noise gate. When there is talk going on, the gate opens. When there is no talk going on, the noise gate closes and there will be silence. This is only possible when there is a difference in volume between the sound you want to keep and ambient noise. If the ambient noise is stronger than the voice a ...


1

Inspecting the input and output files, I realized that the output was upsampled from 48 kHz to 192 kHz. I disabled that with the option -ar 48000.


1

A classical guitar (or two or three) could be a lovely component of a chamber ensemble. All that remains is to write/arrange some music for Guitar and Strings. You will notice that in concertos genrally, the ensemble often leaves "room" for the featured instrument, either because the composer wants to give emphasis to the musical theme (Mozart's piano ...


1

The argument for why it isn't now is pretty much because it has not been done that way. Orchestras are typically structured quite traditionally. Your point around volume is definitely valid - a guitar is too quiet. You'll also find that 4 guitarists are too quiet unless they are amplified - the attack when a string is hit may shine through, but the ...


1

So many possible reasons, really. It's possible that there is some setting that is not obvious and is increasing the level of the audio inside the application that is making it exceed 0 dBFS. For instance, if it's a stereo file and the left and right channels are both panned to the center, you could be adding 3 - 6 dB to the summed audio. Probably more ...


1

A simple, easy way of doing it is by using the amplify effect in audacity. Just select the audio and increase by a couple of dbs and listen to how it sounds (Remember to enable clipping). If you want it more distorted apply the effect again by different amounts until you get what you want. Really what you are looking for is distortion. Any digital ...


1

Your research is heading in the right direction, but you should firstly understand that as an absolute measurement metric, the "decibel" is meaningless. It's like asking someone "how long is that piece of string" and getting the answer "half". "Half of what?" is usually where that goes... Decibels are useful as relative measurements with reference to a ...


1

Although I am not a lawyer, I believe that the CALM act does apply to commercials, either the ones you produce in-house or the one provided by a third party (as I suppose you are in the US). For the technical aspects of achieving such compliance, the ATSC A/85:2013 seems a good starting point.


1

You're heading in the right direction. Compression is what you need, though maybe you want more of it. A 'harder' version of compression is known as Limiting. Using a limiter rather than regular compressor adds a 'ceiling' above which signals cannot get louder. This will allow quieter sounds to pass unchanged, but anything above your volume limit [...


1

Wav files are better quality but unfortunately converting a compressed .mp3 to a .wav doesn't improve quality. I would interpret the "blanket" over the voice to sound like it's muffled which means the high-mid to highs may be lost but you can try to balance it out by lowering the low as well as raising the higher frequencies. Also using a limiter or ...


1

I'm not entirely well versed in the mathematics behind acoustics, but I don't think you can map a purely linear correlation with perceived loudness. There are multiple kinds of frequency functions related to loudness, and none of them are perfect but are rather based on perceptual biases. I'd recommend giving this webpage (and the whole website really) ...


1

Try leveling, a much used compression technique. set compressor speed to the fastest setting lower threshold to the lowest setting play with the ratio between 1...2(2is kinda extreme 1.2-1.3 is recommended) At last if you want more volume try a limiter!


1

Guitars, like pianos, are percussive instruments. As such, they are severely limited in being able to contribute to a smooth sound texture. They can be employed as solo instrument or in chamber ensembles (baroque and renaissance ensembles routinely employed lutes and there was more similarity in sound, volume, use and instrumentation between the lute and ...


1

All of these options others have listed here are great. Just wanted to add that iZotope's Insight plugin is also excellent for monitoring True Peak loudness in accordance with EU broadcast standards. It's expensive, but it's also a pretty kickass metering plugin in general, it's frequency analysis is fantastic as well.


1

Welcome to broadcasting :) In Europe we have a EBU R128 regulation, which is a loudness normalisation with ITU 1770 measurement. Our loudness is -23LUFS and our true peak -1dBFS. I am doing a lot of TV commercials so I hope i can help you with the mix idea: Normalisation of audio gave me a real headache when it was introduced, but it is quite simple. ...


1

You'll need to modify the gain structure by compression/limiting or manual volume automation. You could use a brickwall limiter, but make sure by listening and comparing that it actually results to increased volume and not decreased volume as crushing dynamics can result in either one depending on how the limiting is done or how the sound reacts to it. You ...


1

Have the sound effect files been normalised? If not, I'd start by normalising them as this operation will analyse the file, find the highest level and will then increase the level of all samples so that the highest value sits at 0dB. If you are not familiar with audio software I'd suggest downloading something like Audacity and then look at the manual for ...


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