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#is lufs the same as db
fatasfunkmastering · 2 years
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What Are LUFS? - The Clearest Explanation of LUFS meaning!
What Are LUFS? – The Clearest Explanation of LUFS meaning!
What Is LUFS? Jargon-busting answer: When artists ask “How loud should my master be”, what they probably mean (technically speaking) is “What LUFS should my master be?”. But understanding the concept behind integrated LUFS vs dB (decibels) is essential, and can be confusing. What are LUFS in audio? = LUFS stands for Loudness Units relative to Full Scale (or just “Loudness Units/LU”). They are a…
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vyl3tpwny · 1 year
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loudness in music.
you may be listening to music that is lesser in quality than artists really intend.
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(pictured: a visual guide of FabFilter Pro L2's metering section)
this post is mainly aimed to talk to non-musicians and non-audio engineers, but if you're one of those and also feel for this post, cool. but i want to put this into perspective for people who aren't in music (recording and production in specific). i'm going to attempt to explain a very complicated concept as simply as i can fathom.
there's a lot of stages to the creation of a song or album. there's the writing, recording, production, mixing, and mastering (to simplify). what i'm going to be talking about has to do with "mastering", the final stage a song goes through before it's considered 'final' (in a sense).
mastering, in short, is the process of taking a fully recorded, produced, and mixed song; cleaning it up and preparing it for people to listen to it. a mastering engineer will often add some equalization adjustments to the overall song, adding some compression to bring things together and unify elements, and other stuff like that.
but the duty of establishing the "perceived loudness" is imparted on the mastering engineer.
loudness ≠ amplitude. amplitude is the true measurement of an audio's volume. but "loudness" is a measurement of how loud we actually perceive something. there are technically quiet songs that you perceive as louder than they are, and loud songs that you actually perceive as quiet. this is because loudness ≠ actual technical volume. amplitude is measured in Decibels (db) and perceived loudness is measured in Loudness Units Full Scale (LUFS).
audio processing and transmission always has a limit. for example, if you turn up the volume up too high while listening to music on your headphones, distortion and crackling may occur. this is called "clipping". the threshold for clipping varies from one device to another. cheap earbuds may distort without you even pushing the volume much. other headphones may be really hard to even get audio to clip.
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(image source: MTX)
thus, because of the limitations of how audio works — both in the digital and analog domains — the mastering process of music seeks to finalize a song to be loud enough, but not so loud that it clips in a way artists and listeners don't want.
enter: The Loudness War
in the 90's when digital music studios started to become a thing, people started mastering songs louder and louder because things didn't distort as much when you pushed them in digital spaces. you can get stuff to clip in the digital world, yes, but it was much easier to push things louder than on purely analog equipment. ever since then, songs have been getting mastered louder and louder. we've called this "The Loudness War", because people compete to have the loudest mastered songs.
something you must understand is that the louder you try to master a song (without it clipping above a digital threshold) the more squashed the song gets. the methods for mastering a song really loud often make it lose quality and fidelity. for some people this is fine. stuff like dubstep usually benefits from the squash and the distortion (although it's a personal preference). nonetheless, people have actively chosen to master songs loudly to compete with each other, but at the cost of a song's quality and dynamics.
quick side note: a song's dynamic's is a measurement of how quiet the quiet parts are, how loud the loud parts are, and what the ratio is between these are. so a VERY dynamic song will have very quiet parts and very loud parts. a NOT VERY dynamic song will have mostly everything in the same volume range at all times.
so ok. artists and engineers have been fighting in this loudness war. everyone wants to have loudly mastered songs even though the more you try to push a song to be loud the more it will lose fidelity and the more compressed it will become.
below is a video containing three versions of my song "bonnie". the first version is the original master. the second version is being pushed into a mastering limiter hard. and the third version is being pushed into a mastering limiter VERY hard. a mastering limiter is usually the last part of the signal path of a song's master. i've (very roughly) gain matched the examples so that instead of hearing the loudness differences, you will hear the quality differences. they should be roughly the same volume, but the quality differences will be what you hear. pay close attention to the waveforms getting progressively more squashed the more the song is being pushed into the limiter.
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keep in mind, this example is extreme and unscientific, only meant to demonstrate the kind of quality deterioration that occurs when you try to push a song to be potentially too loud. these quality issues often exist at least a tiny bit in songs that are mastered to be really loud.
so we know that trying to master something loud comes with tradeoffs. but people still really like to do it and sometimes it can be pulled off just fine. but not every song handles these treatments equally.
but what i really want to get at here is the fact that mastering things super loudly in the modern day is kind of pointless! the main exception i can think of is club mixes, DJ's really like to have loud stuff to work with because of the complications that come with DJing in venues. but aside from that, there kind of isn't a point to mastering that loudly anymore.
streaming services like spotify and youtube follow broadcasting standards and automatically adjust music to be roughly in the same range. this is so people listening to music on these platforms, especially through playlists, won't be bothered by a discrepancy in REALLY LOUD songs and really quiet songs being in the same space.
i've been thinking about all of this for a long time. i've been learning how to master my own music for maybe 4-5 years now? but recently an audio engineering youtuber, white sea studio, came out with this video:
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i highly recommend you watch it if you have the time and are interested in learning more on this aspect of music. but to divulge upon the points and information they talk about...
...it's unanimously recommended to master your music to be at least ABOVE -14 LUFS. there are two kinds of misconceptions. the first being that you should try to have your song reach EXACTLY -14 LUFS. wytse explains in the video how this is wrong, and that experts instead recommend you master just to be above -14 LUFS, not exactly and certainly not lower. popular and loud music is being delivered at around -5 and -4 LUFS even. this is really loud and you can imagine there are lots of compromises to quality to get songs to be that loud. but as wytse continues to explain, there's clearly not much point in doing this if streaming services are going to be turning loud songs down anyway.
we're in an age where the loudness war really shouldn't exist. not only has it caused lots of songs (including my own) to be finalized and delivered with compromises to quality and dynamics as well as unfavourable distortion, but it's established this misconception that songs need to be mastered super loudly in the first place.
if it were so simple though, i don't think we'd be talking about this at all. to speak from personal experience, i actually feel self conscious if my music isn't as loud as other songs in the same genre that i'm making. i've found myself not even checking if i'm making it above -14 LUFS, often times i'm just trying to get everything to be as loud as possible. and this is a huge problem for me because i make music in every genre. period. my album CUTIEMARKS mixed dubstep, pop, rock, acoustic, ambient, and so many other genres in one place; but i worked so hard to get everything to be consistently the same loudness. the reality is that it's so hard to pull that off and i'm pretty sure the album is lesser in quality because of it. i already have frustrations listening back to things like "nonexistent meet-cute" and "how to kill a monster" from that album because i tried to push them to close enough loudness to a dubstep song lol.
but despite knowing all of these things very transparently, i still feel very self conscious about it. and i feel like this conversation actually needs to extend beyond musicians and engineers. that's why i'm writing this. i want listeners and fans to know that there's this thing going on and it's been going on for a while and it causes a lot of stress and misconceptions. i'm especially self conscious because i'm actually going to try to actively master my next album quieter than all my other ones on purpose. to try and combat all of this. even if it makes me self conscious to listen to some of my favourite albums of this year and know how much louder they've been mastered compared to mine. but i think it's really important we let songs breathe and not try to compromise quality anymore. it would relieve a lot of stress from me personally to be able to know this is all ok.
if you ended up reading this whole thing, thanks so much. this is a thing that's actually been affecting a lot of musicians who do their own engineering a lot. and it's been affecting all audio engineers anyway. this is an extremely complex topic and to anyone who knows as much as i do about it, trust me i KNOW this post is EXTREMELY simplified. but i want to speak to listeners and fans, not musicians and engineers specifically.
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tonkizone · 1 year
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Mp3 normalizer windows 10 free
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Mp3 normalizer windows 10 free how to#
Mp3 normalizer windows 10 free movie#
Mp3 normalizer windows 10 free full#
Mp3 normalizer windows 10 free series#
Moreover, this value leaves some room for other effects you might want to apply. 1.0 dB is optimal because going beyond this value may distort your audio and make it hard to listen to. The second option given by Audacity is to Normalize peak amplitude to any value you like (as you’ve already adjusted the DC offset, you can proceed to normalize the amplitude between peaks). Just check this box to avoid offset before you edit the track (the DC offset can block some other editing options, so it’s best to do it before you apply any of the effects). The DC offset can distort the audio, so it’s essential to make sure that the waveform is on the 0.0 line. This might look complicated, but it isn’t. The first option is to Remove DC offset (center on 0.0 vertically).
Mp3 normalizer windows 10 free movie#
The first allows you to edit the sound in a movie and not just a stand-alone audio track, and the latter offers a large number of effects.
Mp3 normalizer windows 10 free how to#
Using the Normalize option is really no different from turning up the volume control.īelow, we’ll describe how to handle the two best MP3 normalizers, Movavi Video Editor Plus and Audacity. Because the same amount of gain is applied throughout the whole audio track, the signal-to-noise ratio and relative dynamics rest unchanged. Applying the Peak Normalization effect increases the dB level across an entire audio track by a constant amount. Peak Normalization, on the other hand, is the process of making sure that the loudest parts of an audio track don’t exceed a specific dB value. 14 LUFS is also the standard normalization level for many other streaming platforms. Here are the loudness normalization levels that music producers usually stick to. Loudness Normalization helps ensure that the average volume of your audio is the same from track to track. So Loudness Normalization simply refers to the process of attaining an average value. LUFS are used to measure the loudness over the entire length of an audio track (average value).
Mp3 normalizer windows 10 free full#
The first is Loudness Normalization (more accurately, LUFS, Loudness Units relative to Full Scale), and the second is Peak Normalization. Sometimes you can edit an MP3’s volume using automation, clip gain, or a plug-in.īut what does it mean to normalize audio? Precisely what does Normalize do? In fact, there exist two kinds of normalization. There are situations where it’s best to abstain from using it simply because there are better ways to get the same result. Of course, the Normalize effect should be used wisely. In fact, this used to be the problem about three decades ago, due to the processing algorithms. Some of them claim that normalization can degrade/change dramatically the way the audio sounds. There’s a myth about audio normalization that beginners bring up when they come across this topic.
Mp3 normalizer windows 10 free series#
Another purpose is matching volumes in a series of audio tracks recorded at different volumes (often the problem with podcasts episodes). But why and when would you need to normalize your audio?įirst of all, to get the maximum volume for a quiet audio file without changing the dynamic range (for example, tweak the audio in a movie with a low sound level). One team claims that normalizing your audio can degrade it the other group says sound normalization can be a handy tool. No signup required.The Audio Normalization effect has been around since the beginning of digital audio, but opinions in the music society are still contradictory. The free audio converter does not expire and includes most common audio file formats. A free version of Switch is available for non-commercial use.
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audiohut · 4 years
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Thoughts for Engineers
Here is something for the engineers. I have been working as an engineer for almost a decade and have come to find a few elements the new found engineer can implement to get them started!
When you first have your mix, it might be hard to grasp but finding balance is key. Start your mix by simply adjusting the volume faders and panning knobs to 1. Create your stereo image and 2. Keep away from any un needed processing that might over saturate or damage your mix in the long run. When you do this, imagine that this is all you have to mix with and keep an eye on your master bus to see what is peaking above -6 dB and note it down as the first things you will approach along with anything else you see fit with dynamics, frequency or busing.
Once you have a balance go ahead and briefly solo out certain elements that you prefer to start with. For me, i solo my kick and snare and sweep for clashing frequencies within each. I always start with subtractive EQ before my compression to only compress the frequencies i see as needed or as “the good ones”. Once i have removed frequencies i do not like, i will throw them back into the mix and make minor adjustments without making arbitrary changes that suck up time and that could lead to a foggy direction for the mix eventually ruining it. I repeat this process with very minor removals until i feel i have a good foundation setup.
Side Note: Always focus on your subtractive EQ. You are more likely to find good results in removing bad frequencies which allow the positive ones to show themselves and breathe especially after compression. Boosting to much low, mid or high can result in a muddy, boxy and airy end point in your mix. While boosting is beneficial in many ways you want to boost with closure and know exactly what you want to bring out with said boost. A good example is with bass. I will dual process the bass which entails one low end track and one high end track containing the grit or clank. To really grasp the clank, i will either use a saturation plugin (Fab Filter Saturn) and boost the 2khz-3khz range or simply boost this region with a multi band EQ like Fab Filter Pro Q or SSL’s Channel. If i did not know what i had been looking for i would not have boosted to begin with but considering this range is the attack and grit and i want it brought out, i chose to boost it.
Once i have gotten a foundation, i will then apply Waves L2 on my master bus with a dB and LUFS level that is fitting for the time being. As i achieve certain milestones i will adjust my limiter to a commercial volume level to sort of guide the mix as i go. This helps me focus on sounds that need relief rather than crushing my mix with a limiter after my mix which could result in thinned out sounds especially when clipping elements like snares, kicks and vocals.
Side Note: With every single move i make and more specifically EQ i implement the LDFC method. Listen, Diagnose, Fix and Compare. This will keep you grounded to your moves and make them happen with the reassurance that they are there for a reason. When i first started mixing, i found myself making moves without knowing what i was doing. This is obviously a n00b move, but without starting and actually applying these tools like EQ, Compression etc i would not have gotten better.
After i have gotten my EQ some what out of the way, i move to compression. This is usually done by instrument for me but it never hurts to change up your work flow. If you have a good dB balance paired with a good frequency balance you will be able to hear which areas need tamed dynamics and or punch and attack. Depending on your instrument and what you want to pull from it, you will want to approach your compressor a little differently.
Side Note: I want you to know that a great place to start is to find a signal chain of your own that you are comfortable with tool wise and that is logical to your work flow and use this over and over again starting from scratch every time. You do not want to simply copy and paste the same chain on every vocal as every performance is different and every recording is as well. Doing this will help you grasp the tools and how they work together and become familiar with where they should be used and how heavily they should be worked next time. Remember this: There is no magic button, signal chain or piece of gear/plugins that just make it great. It requires time and effort and real work to learn how to get a colorful and punchy mix.
These two elements (EQ and Compression) once mastered will absolutely bring out what you want from your recordings. There is no simple way to achieve a good mix other than by starting and repeating a process that hones in the skills you desire to get a colorful and bright vocal, guitar or drum track.
If this helped you in anyway i am glad! I hope you the best in your mixing journey as i know it has been fun learning and advancing every mix for myself. If you are looking to produce, write or compose songs please follow and look back at my thoughts for song writers! These posts are all first hand from my experience and i love sharing the information with you and hope you learn from it!
-Andrew Giordanengo
-Audiohut
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blankvirtue · 4 years
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April 12, 2020
- 4:03pm -
     I did manage to do something yesterday, even if it didn’t manage to get it written on yesterday’s entry lol. Mostly a lot of studying really, which I will be doing a lot more of. I see now just how much I don’t know, and knowledge is power.      After learning about creating a stereo send bus and reverb sidechaining, I began watching some youtube videos that I got clickbaited into with a video called something along the lines of “Top 5 Mixing Tips” or whatever. Anyways, it glossed over the idea of gainstaging and dynamics, which made me want to research the topic further. I digged up some good info and tools ultimately. Here’s what I learned.       Dynamics is not what I expected it to be. I’ve been using the wrong unit of measurement to determine how loud my song is this whole time. I’ve been measuring my songs loudness by RMS, or rather what the db Meters show on FL. Which is not how loudness is measured.       Loudness is measured by a unit called LUFS (Loudness Meter Full Scale), a unit of measurement that was heavily implemented from 2010 - 2015 when the loudness war was in full throttle.       Now a days artists aim for -14 to -9 LUFS when mixing due to the music streaming platforms demand to keep music at the same volume. -14 is the standard among other regulations that are put into place to keep it a leveled dynamic playing field.      I learned about tonal balance, the distribution of your frequencies for the duration of your song. When you balance your frequencies you're essentially shaping the waveform. My first time seeing this was watching old school Evoke youtube videos of him explaining some of his process. I saw him constantly checking the waveform of the sound he was working on and making changes to it to make it a balanced waveform. After he made changes and noticed that if the waveform would spike anywhere he would EQ/Compress to manipulate the frequencies, ensuring it was balanced across the spectrum.      To sum things up, overall I’ve been learning a lot about mastering, dynamics, and how much I’ve neglected to consider in all of my previous work at this point. As always though, I HAVE to put this stuff into practice and apply this new knowledge to see any results.      I have a lot of experimentation to do with compressors and saturators! So I can begin to analyze and understand what the effects that these tools have with different instruments. I'm particularly interested in experimenting with compressing drums with reverb effects along side mild saturation and distortion to create glitchy drum effects. That ought to be a fun start.
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jessicakmatt · 4 years
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What are LUFS? Loudness Metering Explained
What are LUFS? Loudness Metering Explained: via LANDR Blog
LUFS are the new way to measure loudness in audio.
This new measurement scale is an important development for many issues in music production.
But understanding LUFS can be pretty difficult at first. They’re different from the ways you’re probably used to measuring your signals.
Even so, these new units are being used all over the audio world. It’s important to know how they work to understand the role of loudness in audio production.
In this article I’ll go over everything you need to know about LUFS.
What are LUFS?
LUFS stands for Loudness Units relative to Full Scale. It’s a standardized measurement of audio loudness that factors human perception and electrical signal intensity together. LUFS are used to set targets for audio normalization in broadcast systems for cinema, TV, radio and music streaming.
If that sounds complicated, it just means that LUFS are the latest and most precise way to measure loudness in audio.
As simple as it seems, using LUFS for loudness has some important consequences that everyone who produces music should understand.
Why do we use LUFS?
You may not realize it, but most of the audio you hear in your daily life is tightly produced to sound great in the environment where you experience it.
Movies, TV, radio and streaming services all feature audio meticulously designed to work perfectly on each platform.
Movies, TV, radio and streaming services all feature audio meticulously designed to work perfectly on each platform.
But how did we get there? Someone had to decide on the audio standards for each different medium in order to make consistent sound possible.
LUFS are one of the latest tools engineers and researchers developed to help us make those decisions.
By integrating the loudness of audio signals and human perception into a single scale, LUFS acts as a kind of audio measuring tape.
The units help engineers compare different types of audio and match them to the requirements of their respective listening environments.
Loudness in music production
The biggest obstacle for consistent sound across mediums is loudness.
It seems like an easy problem, but making everything the same volume for every different playback system out there is pretty tough.
For starters, what even is loudness?
In your DAW you might think of the dB levels on your track meters. That’s a good start, but it doesn’t tell the whole story.

This type of loudness is a property of signals. But it might surprise you to learn that it doesn’t translate directly to how we experience loudness.
The reasons why aren’t exactly straightforward. It has to do with the technique used to measure the signal and the structure of our inner ears themselves.
To learn more about how loudness works, check out our overview.
When it comes to music perception and cognition, things get even more murky, but we broke down the basics in our guide to psychoacoustics.
To fix it, engineers developed a way to gauge listeners’ perceived loudness and signal intensity at the same time—LUFS!
How to use LUFS
Metering audio with LUFS is a little different from the other loudness measures you’re used to.
Metering audio with LUFS is a little different from the other loudness measures you’re used to.
First off, there are a few different ways to use it. Here are the most important ones.
Integrated loudness
Imagine you’re mixing a film soundtrack.
There are some extremely loud scenes with explosions and intense music, and others with barely any sound at all as the characters sit in silence. How loud should the mix be overall?
To make a judgement you’d need to take the entire duration of the mix into account. That measurement is called integrated loudness. It’s recorded in LUFS.
Film and TV have strict standards for integrated loudness that are set in LUFS values.
Dynamic range
Dynamics are important in any recorded audio. But how big should the difference between loud and quiet really be?
LU—or LUFS without the “full scale” part—can help answer that question. LU uses the same perception based units to evaluate how loud something seems to you.
But when you measure dynamic range in LU it’s no longer relative to full scale. Instead, it tells you the difference between the quietest and loudness sound over time like integrated LUFS.
Many standards organizations publish recommended dynamic range figures for their audio content.
Short term LUFS
Integrated LUFS tells you about the whole audio file, but you need to take a closer view of individual sections of sound to get the whole picture.
Even if your track hits the overall LUFS target, there still might be some sections that are too loud or too quiet.
Short term LUFS gives you perceived loudness over the last three seconds three seconds of audio.
Momentary LUFS
Momentary LUFS is the shortest period LUFS measurement. It’s the closest in style to the electrical Peak measurement you’d find on your DAW’s dB meter, but it’s not quite the same.
Momentary LUFS is measured across the last 400 ms of audio.
That’s the kind of fine grain level of detail you need to know exactly you loud your material sounds in the moment.
Why do LUFS matter?
At some point in the history of audio engineering, the music industry decided that recordings should be loud.
The idea was that listeners would subconsciously prefer the CD that sounded loudest on their CD player.
The evidence to support the theory was thin, but it set off a boundary-pushing race called “the loudness war.”
Eventually the trend wore out and loudness was reigned in when streaming platforms like Spotify and Apple Music took over.
Those platforms use LUFS to evaluate loudness.
Since LUFS indicates the perceived loudness, engineers are no longer racing toward the physical limit of the medium’s headroom.
Instead they’re aiming for a target that’s much more in tune with how listeners perceive loudness—and it’s not even close to the max!
Understanding this paradigm shift is important for how you work with your mix in its final stages of development.
In most workflows, these issues will come up most during mastering. Modern mastering is a highly technical art form that pushes your volume levels right to the edge—but never over it.
LUFS is the tool that makes it possible. Measuring audio correctly and hitting the right targets is a key part of any mastering process.
But if you don’t have the tools and experience to evaluate loudness this way, you should consider leaving mastering to the experts.
Whether you decide to hire a professional or try AI mastering, good mastering means getting loudness right every time.

Accurate audio metering
LUFS are an important technical standard in audio.
Loudness is a complicated subject, but with the right tools you can understand how it works and how it impacts your sound.
The post What are LUFS? Loudness Metering Explained appeared first on LANDR Blog.
from LANDR Blog https://blog.landr.com/lufs-loudness-metering/ via https://www.youtube.com/user/corporatethief/playlists from Steve Hart https://stevehartcom.tumblr.com/post/628078287468511232
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ericwollersberger · 4 years
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Mastering The Abject Sublime
Since I’m now in the final stages of mastering, I wanted to give an update on how that’s going even though there’s also a section about it in my report. Initially I just ran all of the “final” mixes through an L1 with an out ceiling of 1.5 and then through a YouLean loudness meter to make sure the songs were all approximately the same LUFS rating. For the most part they all fell between -11 and -14 LUFS, which seemed like a reasonable margin of loudness difference at first. Then, after listening to it a few more times, I decided that certain songs definitely seemed quieter, and when I put them side by side in my “mastering” project and toggled between them, it did seem like some songs were distinctly louder or quieter than they should be. In general, I don’t know if this is good practice, because you end up comparing the quiet parts of some songs to the louder parts of other songs and treating them like they’re the same, which they shouldn’t be. So I was careful about that, mainly making sure the loud, distorted parts were all at about the same volume even if the character of the sound was somewhat different. 
For example, the loud guitars in the verse of “The Abject Sublime” sounded a little thin when compared to the other guitars on the album, but that could largely be because of the lack of a full string arrangement behind the guitars and the surrounding elements of noisy, glitchy percussion elements. The loud guitars in “I Like it Down Here,” on the other hand, sounded a lot more full and powerful, but I think this is mainly because of the full string arrangement and choir filling out the sound. Even though those sounds are relatively low in the mix and not particularly distinct, they have a huge effect on how the guitars that are in the foreground are perceived. The loud guitars at the end of “My Love is Like a Desert” sounded kind of quiet in comparison to the ones in Abject Sublime, though, which they shouldn’t because the arrangement is actually quite similar. I feel like “My Love is Like a Desert” may have gotten a somewhat lower LUFS reading because the first half of the song is so quiet in comparison with the second half. I ended up boosting it by about 2 db, applying some compression, which made the part with the loud guitars sound more comparable to the ones in “the Abject Sublime.”
It was also kind of strange to master the songs that didn’t have loud guitars in them, because their LUFS scores tended to be quieter. I turned up “Follow Me Down” by 2db because it was the lowest in the bunch, with Loudness penalty actually saying that iTunes would turn it down by 0db. “Your New Life” was a little higher than that, and “No More Mirrors” was actually quite close to the songs with loud guitars in them. I ended up turning “Your New Life” up by a couple of db as well, because after the bombastic ending of “I Like it Down Here,” it sounded a little too quiet for my ears. “I Like it Down Here” was especially odd because even after turning the whole mix down by 1db, and the true peak only making it up to 2.5, it still had a higher LUFS score than any of the other songs. It probably has a more compressed instrumental mix than any of the other songs, but I think it’s one of the best-sounding tracks on the album, so I’ve decided not to worry about this anomaly. “A Better Place” was odd in that it was a song without loud guitars, but with a distorted lead that went through most of the track. I actually ended up turning it down by half a db because it felt a little too loud coming on after the long, noisy ending of “the Abject Sublime.”
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lorrainecparker · 6 years
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Build your dynamic microphone modularly: Pyle PDMIC78 with Shure accessories
Some people call the Pyle PDMIC78 a knockoff of the Shure SM57. That description is true regarding the physical shape and dimensions, but not the sound. For vocal recording, many people find the Pyle PDMIC78 to have a brighter, more desirable sound than the Shure SM57, which for decades has been the one used during presidential inaugurations in the United States. Previously, I reviewed the Shure palindromic 545 dynamic mic. Today, I am reviewing an even more interesting one, combined with two Shure accessories: the US$33 windscreen and the US$22 shockmount, which actually cost much more than the under US$20 Pyle PDMIC78 microphone, although they are essential. Ahead you’ll hear a test recording made with this under US$75 combination. I’ll also give you the details on a possible connection caveat which may affect you, depending whether you get an older or newer one and whether you use the PDMIC78 only with other of the same, or together with any other mics in the same location.
About the hardware
In my review of the Shure 545 microphone (Amazon • B&H), I went into great detail about how I love the Shure A55M shockmount (Amazon • B&H) and the enormous Shure A81WS windscreen (Amazon • B&H), so here I’ll concentrate on the Pyle PDMIC78 dynamic microphone (Amazon • B&H). It is simple, since it has no switches at all. Together with those Shure accessories, it sounds amazing for the price. Here are the pattern and frequency response according to Pyle:
Test recording
I made the following test recording using the Pyle PDMIC78 dynamic microphone (Amazon • B&H), Shure A55M shockmount (Amazon • B&H) and the enormous Shure A81WS windscreen (Amazon • B&H) which (as demonstrated by Matt Rygelski in my review of the Shure 545 microphone) sonically transforms the microphone while making it immune from pops and making it look much more interesting. The Pyle PDMIC78 dynamic microphone is one of the few third-party microphones that fit the A81WS windscreen perfectly.
https://www.provideocoalition.com/app/uploads/Pyle-PDMIC78-A55M-A81WS-16LUFS.wav
I recorded a mono 48 kHZ WAV file at 24 bit using the Zoom H5 recorder (Amazon • B&H). I had to set its preamp to about 5.8 out of 10 to get our desired -12 dB raw recording. Then I normalized it to -16 LUFS with Auphonic, trimmed it with Fission and uploaded it to the server.
Potential connection caveats
Balanced?
Even though it has an XLR plug, according to many reports online, the PDMIC78 was previously shipped from the factory wired as unbalanced and required disassembling and rewiring to achieve a proper balanced connection. See my recent Balanced audio: benefits and varieties (illustrated above) for more information. However, it now appears that Pyle has recently corrected that and begun to ship the PDMIC78 with proper balanced wiring internally.
Does the PDMIC78 comply with EIA Standard RS-297? Does it even matter?
Before 1970, some balanced microphones came wired for XLR pin 2 hot, while others came with pin 3 hot. In 1970, the Electronic Industries Alliance (EIA; until 1997 Electronic Industries Association) saved the day and set the RS-297 standard decreeing that XLR pin 2 be hot. Since then, most manufacturers have complied. Why does this matter? It matters to avoid potential phase cancellation, whereby spill (undesired sound that enters into more than one microphone in a multi-mic setup) can cause sounds to cancel themselves out and not be heard.
Several reports online indicate that (at least while Pyle was shipping the PDMIC78 wired internally as unbalanced), it was shipping with pin 3 hot, the opposite of RS-297. It is not yet clear whether when Pyle corrected the balanced signal, it also began to comply with RS-297. Of course, even if Pyle hasn’t yet corrected that, it can be fixed by disassembling the microphone and rewiring it. But you may be asking whether this even matters. It does not matter at all if there is a single microphone. It only matters if used together with other microphones that may be 180 degrees out of phase.
Recent articles and reviews you’ll enjoy
Review + comparison: iRig Pre HD cross platform audio interface
Review: iRig Mic HD 2 handheld digital microphone
Review: RØDE Broadcaster, the voice microphone that challenges conventional wisdom
Review: RØDE improves VideoMic Pro+ with several new features
Use wireless Apple AirPods’ mic for shooting standups/monologues with your smartphone?
Balanced audio: benefits and varieties
Review: MixPre-3 audio recorder/mixer from Sound Devices
My multitrack crossgating audio workflow to eliminate crosstalk
Review: AT875R shotgun mic & AT8415 shockmount from Audio Technica
Upcoming articles, reviews, radio shows, books and seminars/webinars
Stand by for upcoming articles, reviews, and books. Sign up to my free mailing list by clicking here. Most of my current books are at books.AllanTepper.com, and my personal website is AllanTepper.com.
Si deseas suscribirte a mi lista en castellano, visita aquí. Si prefieres, puedes suscribirte a ambas listas (castellano e inglés).
Listen to his CapicúaFM show at CapicúaFM.com or subscribe via Apple Podcasts, Radio Public or Stitcher.
Save US$20 on Project Fi, Google’s mobile telephony and data
Click here to save US$20 on Project Fi, Google’s mobile telephone and data service which I have covered in these articles.
Learn to speak Castilian, the most widely used Spanish language
SpeakCastilian.com
FTC disclosure
No manufacturer is specifically paying Allan Tépper or TecnoTur LLC to write this article or the mentioned books. Some of the other manufacturers listed above have contracted Tépper and/or TecnoTur LLC to carry out consulting and/or translations/localizations/transcreations. Many of the manufacturers listed above have sent Allan Tépper review units. So far, none of the manufacturers listed above is/are sponsors of the TecnoTur programs, although they are welcome to do so, and some are, may be (or may have been) sponsors of ProVideo Coalition magazine. Some links to third parties listed in this article and/or on this web page may indirectly benefit TecnoTur LLC via affiliate programs. Allan Tépper’s opinions are his own.
Copyright and use of this article
The articles contained in the TecnoTur channel in ProVideo Coalition magazine are copyright Allan Tépper/TecnoTur LLC, except where otherwise attributed. Unauthorized use is prohibited without prior approval, except for short quotes which link back to this page, which are encouraged!
The post Build your dynamic microphone modularly: Pyle PDMIC78 with Shure accessories appeared first on ProVideo Coalition.
First Found At: Build your dynamic microphone modularly: Pyle PDMIC78 with Shure accessories
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jessicakmatt · 5 years
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Fletcher-Munson Curves: How Equal Loudness Contours Affect How You Mix
Fletcher-Munson Curves: How Equal Loudness Contours Affect How You Mix: via LANDR Blog
Your experience of music is more than just vibrations travelling through the air. Fletcher-Munson Curves explain why.
As soon as a sound reaches your ears, a complex process turns the sound waves into music you can perceive and understand.
Fundamental features of music like pitch and timbre and are wired into your brain’s perception and cognition systems.
Even a psychoacoustic phenomenon as simple as loudness can be a bit mysterious.
Fletcher-Munson Curves are a tool for evaluating how you experience sound. The effect they describe has a big impact on your approach to mixing music.
In this article you’ll learn what Fletcher-Munson Curves are and how to use them to help you understand loudness.
What are Fletcher-Munson curves?
Fletcher-Munson curves are equal-loudness contours. Equal-loudness contours describe the perceived loudness of a sound in relation to its frequency for human listeners.
Equal-loudness contours describe the apparent loudness of a sound in relation to its frequency for human listeners.
This scale is important because human loudness perception is extremely sensitive to differences in frequency.
The original Fletcher-Munson curves were created from experimental measurements made by Harvey Fletcher and Milden A. Munson in the 1930s.
But they’ve since been replaced by a series of generic equal-loudness contours like the modern ISO 226:2003 which better represent human hearing as a whole.
Human hearing and the cochlea
Why does perceived loudness depend so much on frequency? Why does a 60 dB sine wave at 10 kHz seem so much louder than at 50 Hz?
Why does a 60 dB sine wave at 10 kHz seem so much louder than at 50 Hz?
The answer has to do with your ears—and your brain.
After a sound wave enters your ear canal, your eardrum transmits the vibrations to the fluid in your inner ear with small bones called ossicles.
Inside your inner ear these vibrations excite special hair cells called the stereocilia which create the electrical signals that travel through the auditory nerve to your brain.
The organ inside your ear where all this takes place is called the cochlea—and it’s shaped like a coiled up garden hose.
Here’s where it gets interesting…
The hair cells are distributed differently throughout the cochlea. They’re closer together near the centre where the tube is coiled up the tightest and the highest frequencies are handled.
In fact, if you were to unroll the cochlea, you’d find that the distribution of hair cells mimics the same kind of logarithmic scale that we use to measure sound pressure level (SPL).
That means that at least some part of the perceived loudness puzzle is physiological—it’s basically built-in to your body!
The Fletcher-Munson curves were discovered by measuring people’s subjective experience. They were created to come up with ways to compensate for the differences in loudness throughout the spectrum.
Phons and sones
The next logical step in the quest for equivalence in loudness was to use the curves to create standard units.
The decibel scale works fine for SPL, but engineers needed something better for perceived loudness.
The decibel scale works fine for SPL, but engineers needed something better for perceived loudness.
If you follow the lines, the perceived loudness at each position along the curve is the same— despite the decibel difference.
That means by using an 1000 Hz sound at 1dB as a reference, engineers could define a unit of perceived loudness called the phon.
Researchers soon realized that an increase of 10 phons was equivalent to a perceived doubling of loudness.
This gave rise to a another proposed loudness unit called the sone. On this scale, each additional doubling of perceived loudness is a doubling in value.
Equal loudness curves in mastering
All this psychoacoustic research into quantifying loudness has a big impact on your masters.
Using equal-loudness curves and the concept of phons and sones, engineers developed Loudness Units relative to Full Scale (LUFS)—the standard for the perceived loudness of signals inside your DAW.
This is what enables mastering engineers to push levels exactly the right loudness to excite your ears.
And it’s part of the technology that makes precise volume matching possible.
Matching levels accurately is one of the most important factors to consider for good mix referencing.
Matching levels accurately is one of the most important factors to consider for good mix referencing.
When comparing your original track to the master, use volume matching to help you make the most informed decision about your masters.
When mix referencing both versions with volume matching, listen for details like:
Overall EQ contour
Overall dynamics
Level and frequency range of each element
Ahead of the Fletcher-Munson curve
The Fletcher-Munson curve was one of the first music perception experiments to have an impact on music producers and engineers.
The data they gathered to create the first equal-loudness curves helped us understand our own psychoacoustic biases.
Now that you know how Fletcher-Munson curves make sense of loudness, use volume matching to make better mix decisions.
The post Fletcher-Munson Curves: How Equal Loudness Contours Affect How You Mix appeared first on LANDR Blog.
from LANDR Blog https://blog.landr.com/fletcher-munson-curves/ via https://www.youtube.com/user/corporatethief/playlists from Steve Hart https://stevehartcom.tumblr.com/post/183478658134
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lorrainecparker · 6 years
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Review: RØDE Broadcaster, the voice microphone that challenges conventional wisdom
The general consensus is that in order to reject background noise (especially with acoustically untreated rooms), it behooves you to use a dynamic microphone, not a condenser one. The RØDE Broadcaster microphone’s 1” condenser proves to be a worthy exception to that general consensus. In addition, it allows better high-frequency coverage than most dynamic mics, and has a much higher output level, so you won’t need such a demanding preamplifier or pre-preamp like a CloudLifter or FetHead. Ahead are test recordings I made in an untreated room, compared with another mic that challenges that “rule”, the Audio Technica AT875R shotgun I reviewed in July 2017. Both comparative test recordings of each also include my reading in English of part 3 of the Spain’s Constitution of 1978, which is particularly appropriate since today, Catalonia declared its independence from Spain, and the Spanish Senate just approved taking over. This is all happens in the same month when Amazon independently decided to feature and honor two of my related books. Do you believe in coincidences?
As you will hear ahead, the RØDE Broadcaster (Amazon link • B&H link), is the 1” condenser microphone that breaks the rules. It rejects background sound as well as comparable dynamic mics.
Another advantage of the RØDE broadcaster and the Audio Technica AT875R shotgun I reviewed in July (Amazon link • B&H link) is that their output is much higher than most dynamic mics, so I was able to set the input gain on my Zoom H5 recorder just under 5 in order to get a conservative -12 dB level on the raw recording. However, both do require phantom power from your preamp/interface/recorder. The raw recordings were 48 kHz (see All audio production & distribution should go 48 kHz. Learn why.) and 24-bit (see Understanding 24-bit vs 16-bit audio production & distribution). The low cut is always active with the AT875R, and was active with the RØDE Broadcaster.I used no equalization or filtering in post. I only normalized the two recordings to -16 LUFS before uploading them. The final audios are 48 kHz 16-bit mono WAV.
Even a tally light!
Another unique advantage of the RØDE Broadcaster is its inboard tally light, which requires a special mic cable with an extra conductor with a compatible radio or TV mixer. I did not test that automatic feature, since I didn’t have such a mixer with tally capability. However, in that situation, the tally light does illuminate continually when the mic receives phantom power from a standard 3-pin XLR.
Detailed specs
Acoustic Principle: Pressure Gradient
Active Electronics: JFET impedance converter with bipolar output buffer
Capsule: 1.00″
Polar Pattern: Cardioid with good rear rejection
Address Type: End
Frequency Range: 20Hz – 20kHz (with low cut filter deactivated)
Output Impedance: 40Ω
Maximum SPL: 128dBSPL
Maximum Output Level: 2.0mV (@ 1kHz, 1% THD into 1KΩ load)
Sensitivity: -34.0dB re 1 Volt/Pascal (20.00mV @ 94 dB SPL) +/- 2 dB @ 1kHz
Equivalent Noise Level (A-weighted): 14dBA
Phantom Power: 24-48 volts
Weight 577.00g
Dimensions 167.00mmH x 50.00mmW x 65.00mmD
Output: XLR
Warranty: 1 year with free extension to 10 years following registration
RØDE clarifies its situation with Amazon
Back in 2015, I published Amazon’s clarification about buying RØDE microphones from Amazon:
Amazon itself is authorised for purchases from Amazon, but other dealers ‘fulfilled by Amazon’ are not. We know the sales channel through which Amazon purchases and it is legitimate but other dealers that send their stock to Amazon and sell on their marketplace are not only grey importing (which makes it difficult to give proper support) but we’ve identified counterfeit VideoMic GOs. To tell, the user just needs to select the microphone marked as ‘sold and fulfilled by Amazon’.
Test recordings
https://www.provideocoalition.com/app/uploads/Rode-Broadcaster_leveled.wavhttps://www.provideocoalition.com/app/uploads/AT875R-versus-Rode-Broadcaster-leveled.wav
Example text at the end of each: Spanish Constitution, Part 3 in English
At the end of each recording, you’ll hear my voice read Spanish Constitution, Part 3 in English:
Castilian is the official language of the State. All Spaniards have the obligation to know it, and the right to use it.
The other Spanish languages shall also be official in their respective autonomous communities, according to their statutes.
The richness of Spain’s diverse linguistic modalities represents our national heritage and shall be the object of special respect and protection.
My related books (which Amazon “coincidentally” decided to feature during is independent author month (#MesIndie, amazon.com/MesIndie, amazon.es/MesIndie, amazon.com.mx/MesIndie) are:
La conspiración del castellano
El encubrimiento de la Real Academia
The English versions are not being featured by Amazon this month, and the Apple iBooks Author versions are not being featured by Apple, but are available at their normal everyday low price:
The Castilian Conspiracy
The Royal Spanish Coverup
For more general about Castilian and “Spanish”, visit my SpeakCastilian.com. Do you believe in coincidences, or do you think Amazon had advanced inside information when the company picked the two of my (presently) 12 titles listed at books.AllanTepper.com several months in advance?
Conclusions
Both the RØDE Broadcaster (Amazon link • B&H link) and the Audio Technica AT875R shotgun (Amazon link • B&H link) do an extraordinary job of producing high quality voice capture while minimizing background noise, and reduce the need for a strong preamp with +60db gain or more, although both do require phantom power. You will probably love the RØDE Broadcaster esthetics and tally light capability. After hearing both recordings, you may prefer the quality of the RØDE Broadcaster for doing voiceovers, however the Audio Technica AT875R shotgun can be more versatile if you need to use the same mic for field production, in those cases when a shotgun is required. However, if you have a budget for more than one mic, or need to use more than one mic in a studio environment, you might want one of each.
Upcoming articles, reviews, radio shows, books and seminars/webinars
Stand by for upcoming articles, reviews, and books. Sign up to my free mailing list by clicking here. Most of my current books are at books.AllanTepper.com, and my personal website is AllanTepper.com.
Si deseas suscribirte a mi lista en castellano, visita aquí. Si prefieres, puedes suscribirte a ambas listas (castellano e inglés).
Listen to his CapicúaFM show at CapicúaFM.com or subscribe via Apple Podcasts, Radio Public or Stitcher.
Save US$20 on Project Fi, Google’s mobile telephony and data
Click here to save US$20 on Project Fi, Google’s mobile telephone and data service which I have covered in these articles.
Learn to speak Castilian, the most widely used Spanish language
SpeakCastilian.com
FTC disclosure
No manufacturer is specifically paying Allan Tépper or TecnoTur LLC to write this article or the mentioned books. Some of the other manufacturers listed above have contracted Tépper and/or TecnoTur LLC to carry out consulting and/or translations/localizations/transcreations. Many of the manufacturers listed above have sent Allan Tépper review units. So far, none of the manufacturers listed above is/are sponsors of the TecnoTur programs, although they are welcome to do so, and some are, may be (or may have been) sponsors of ProVideo Coalition magazine. Some links to third parties listed in this article and/or on this web page may indirectly benefit TecnoTur LLC via affiliate programs. Allan Tépper’s opinions are his own.
Copyright and use of this article
The articles contained in the TecnoTur channel in ProVideo Coalition magazine are copyright Allan Tépper/TecnoTur LLC, except where otherwise attributed. Unauthorized use is prohibited without prior approval, except for short quotes which link back to this page, which are encouraged!
The post Review: RØDE Broadcaster, the voice microphone that challenges conventional wisdom appeared first on ProVideo Coalition.
First Found At: Review: RØDE Broadcaster, the voice microphone that challenges conventional wisdom
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lorrainecparker · 7 years
Text
Review: MixPre-3 audio recorder/mixer from Sound Devices
As soon as Sound Devices announced the new MixPre-3 audio recorder/mixer, I requested it for review. Ahead I’ll tell you everything that I discovered to be so great about it, and a few thinks I’d like added via future firmware updates. This review article will also include 4 audio clips to demonstrate its quality with a dynamic microphone, compared to the much lower-priced Zoom H5. Using a dynamic mic is the best way to stress a preamp to see how it sounds under the worst scenario. I’ll cover its features, and include some videos from colleagues who have also reviewed it.
In this article, you’ll find:
Listen to the MixPre-3’s superior preamps with a dynamic microphone
Why are the preamps called Kashmir?
True hardware limiter vs software limiter
Basic and Advanced modes
How the MixPre-3 records multitrack
Connect a USB keyboard to enter metadata
USB: digital audio interface and file transfer
Modern exFAT formatting system
Wingman: remote wireless control from Android or iOS
Timecode… and Does anybody really know what time it is?
Multiple power options
Four improvements I’d like to see via firmware
Conclusions and related articles
Listen to the MixPre-3’s superior preamps with a dynamic microphone
I made the following tests using the Shure palindromic 545 dynamic cardioid microphone, together with the enormous A81WS windscreen which I reviewed back in Why Shure’s own 545 palindromic mic beats the SM57 nowadays (illustrated above). The A81WS windscreen sonically transforms the 545 and SM57 microphones, as explained in detail by Matt Rygelski in his video which is embedded in the above article. The members of the now defunct palindromic group ABBA were not available to join me for this MixPre-3 review, so the following recordings were done with my own voice.
Raw WAV recording from the MixPre-3:
https://tecnotur.us/capicuafm/wp-content/uploads/2017/07/MixPre-3_raw.wav
Raw WAV recording from the Zoom H5:
https://tecnotur.us/capicuafm/wp-content/uploads/2017/07/ZOOM-H5_raw.wav
Processed WAV recording from the MixPre-3
https://tecnotur.us/capicuafm/wp-content/uploads/2017/07/MixPre-3_processed.wav
Processed WAV recording from the Zoom H5
https://tecnotur.us/capicuafm/wp-content/uploads/2017/07/ZOOM-H5_processed.wav
The processed versions were filtered and normalized to -16 LUFS by Auphonic.com. I used a semi-treated room courtesy of Francisco Javier Arbolí.
Why are the preamps called Kashmir?
The name that has given to the new preamps by Sound Devices in the MixPre-3 and MixPre-6 is Kashmir. I asked the question, and the official answer was that Sound Devices prefers not to explain the origin of the Kashmir name, and to keep it as an internal secret. The source may be related to one of the several clues listed on Wikipedia’s Kashmir disambiguation page, which include maritime references, music, films and regions throughout planet Earth. With an alternate spelling, it refers to a type of wool, and a type of goat. On the Castilian Kashmir page, there are even references to jewelry.
As time passes, the mysterious origin of Sound Devices’ Kashmir preamp name may become even more popular than the “Paul is dead” urban legend and conspiracy theory, which alleges that Paul McCartney of the English rock band The Beatles died in 1966 and was secretly replaced by a look-alike. Perhaps there will be even more documentaries and mockumentaries about Sound Devices’ Kasmir than there have been about “Paul is dead”.
True hardware limiter vs software limiter
Back in the analog recording days, we would cheat and let our recordings overmodulate a bit, since the slight distortion was favorable to the tape hiss. As we moved into the digital recording era, we learned that going past 0 dB is fatal. Most have learned to use -12 dB as our target ceiling when setting peak levels in a raw recording, in order to have some headroom in case of something unexpected. Even so, it is great to have a safety net or parachute beyond our knowledge and experience of setting peaks to -12 dB, just in case.
Although recorders like the Zoom H5 have much better preamps then the H4 (and its revamps with various suffixes), they still don’t have preamps that are nearly as good as the MixPre-3’s (as you heard earlier in this article), nor do they have a digital limiter. In the video below, Nuanced Films of the United Kingdom demonstrates the advantage of a hardware limiter vs the software one in the H5:
Thanks to Nuanced Films for doing that demonstration, so I wouldn’t have to 🙂
Basic and Advanced modes
The MixPre-3 offers both Basic and Advanced modes via the touchscreen display. The Basic mode is designed to get up and running as soon as possible; while the Advanced mode gives further options.
Despite the advantages of the Advanced mode for those who can take the time to use them, there is currently one disadvantage: Presently, when adjusting the level of a particular input using the Advanced mode, there is no way to see a level meter while adjusting. As a result, it is necessary to adjust, leave the adjustment window to see the result, and then potentially repeat the process. This is a known issue reported by several reviewers, and is listed in the section of this article called Four improvements I’d like to see via firmware.
One particular feature that requires using the Advanced mode is timecode, which will be covered in a dedicated section later in this article.
How the MixPre-3 records multitrack
As I covered in the recent My multitrack crossgating audio workflow to eliminate crosstalk article, the MixPre-3 and some other recorders from Sound Devices record a polyphonic WAV file, which contains multiple mono WAV files.
The polyphonic file can later be separated using Sound Devices’ own Wave Agent software (illustrated above), which is available free for both macOS and Windows. That software does many other things, but this particular function is called Poly Split. Here is a ±3 minute video from Sound Devices which demonstrates the use of the Wave Agent software.
At 1:41 (coincidentally palindromic) the Poly Split function is covered.
In addition the the polyphonic file, the MixPre-3 also records a stereo mix file.
Connect a USB keyboard to enter metadata
Fortunately, the MixPre-3 allows users to connect a USB (wired) keyboard to facilitate entering metadata. Unfortunately, the current firmware is limited to the US (United States) keymap. This issue is listed among the four improvements I’d like added via a firmware update.
USB: digital audio interface and file transfer
The MixPre-3 also works as a USB audio interface, offering the ability to record audio to an SD card while simultaneously streaming multiple channels of audio to a computer. This can be used both for recording or live Internet broadcast.
File transfer can also take place via USB. These two features are not unique to the MixPre-3, and exist in other brands. However, to my knowledge, Sound Devices is the first manufacturer to use the superior USB-C.
Modern exFAT formatting system
To my knowledge, Sound Devices is the first audio recorder manufacturer to allow for the more modern exFAT file system when the card is 32GB or greater.
Wingman: remote wireless control from Android or iOS
The free Wingman remote control application from Sound Devices is now available for Android and iOS users. The Wingman app works with both the Sound Devices 6-Series line of professional field mixer/recorders and the all-new MixPre Series of audio recorders, including the MixPre-3. With Wingman, users can start and stop audio recordings, monitor meter levels, enter and edit metadata, as well as arm, disarm, and rename tracks.
The above image shows the Wingman being used with the larger MixPre-6, not the MixPre-3 which is the focus of this article.
Unlike the USB keyboard covered in the prior section (which is currently limited to the US keymap), I personally verified that the Spanish keyboard —together with accented vowels and the ñ character— were respected properly within the Wingman app on Android ßeta O (8.0) on my Google Pixel XL (Amazon link).
I did not use the Wingman app during the recording you heard earlier in this article. I just used it to verify whether it was compatible with the Google GBoard Spanish keyboard and those accented letters, and to be sure it would run under the ßeta Android O (8.0). Fortunately, it worked perfectly, to the extent that as I used it.
Timecode… and Does anybody really know what time it is?
My metaphysical friends —who insist that time doesn’t actually exist— praise timecode, and describe it as: “a very accurate way of measuring something that doesn’t exist”. Perhaps that’s why the US rock band Chicago produced and released what became hit song called Does anybody really know what time it is? Does anybody really care (about time)?. Those same metaphysical friends say that timecode is extremely useful in audiovisual production, and I know that they are right about that.
The timecode feature is only available in the Advanced mode of the MixPre-3. It facilitates synchronizing high quality audio recorded in the MixPre-3 with video shot on a camcorder.
The MixPre-3 has three timecode modes:
HDMI TC In
To use the HDMI TC IN on the MixPre-3, you must have a camcorder that outputs timecode over its HDMI output connection. You connect the HDMI cable to the HDMI TC In port on the MixPre-3’s right hand panel and set Menu>Timecode>TC Mode to HDMI TC In. Set your camcorder to output HDMI timecode. When the MixPre-3 receives HDMI timecode, it is displayed on the Home screen beneath the current filename. Each time you press Record, the incoming timecode value is stored inside the MixPre-3’s recorded file. Professional video editing applications including Apple Final Cut Pro X, Adobe Premiere CC, Avid Media Composer and Blackmagic’s DaVinci Resolve (both the free and paid versions) officially support reading this timecode value and sync the audio with the camera video, and all do it well when it is done clip-by-clip. However, according to Sofi Marshall’s excellent article from May 2017 Why DaVinci Resolve is the ultimate (free) tool to sync clips in Frame.io, the only editing software that does it correctly in batch mode is DaVinci Resolve. She states that the other three (FCP X, Premiere CC and Media Composer) all screw it up in batch mode. On the other hand, she says that batch mode works great with DaVinci Resolve, and that “…you can sync an entire day’s worth of footage with a single click”.
Aux In 1, Aux In 2
If your camcorder can output LTC (SMPTE Linear Timecode), you can connect it to the MixPre-3’s Aux/Mic In 3.5mm connector.
To enable Aux In Timecode:
Set the Aux In Mode to Timecode (Menu>Inputs>Aux In Mode)
Set Menu> Timecode>TC Mode> to Aux In 1 or Aux In 2. When the MixPre-3 receives Aux In timecode, it is displayed on the home screen beneath the current filename.
Time of Day (TOD)
Use TOD mode, when you want the recorded WAV files to be timecode stamped with the time of day. The time of day is derived from the MixPre-3’s built-in date/time clock. This mode is not used for frame accurate synchronization with external devices. Rather, it is used as simple way of logging what time a recording took place.
Timecode Frame Rate Auto-Detection
With both the HDMI TC In and Aux In Timecode modes, the MixPre-3 autodetects the incoming timecode framerate and stamps the recorded files accordingly.
Timecode Stamp
Regardless of which timecode mode you use, each time you press record, the timecode value at that moment is embedded within the MixPre-3’s WAV file.
Multiple power options
The US$649 MixPre-3 (B&H link) can be powered from four (4) AA batteries, Li-Ion batteries, a laptop’s USB-C or USB-A connectors, or AC wall outlet.
Sound Devices offers the optional MX-8AA battery sled for US$70 (B&H link).
Sound Devices also offers the MX-LMount battery sled for US$65 (B&H link), which holds 2 hot-swappable L-Series type batteries.
In my brief experience with four (4) fresh AA batteries in the MixPre-3, I would highly recommend using any of these extended options. If in the future, I am able to own a MixPre-3, I will certainly use one of them, since in the little time I have been testing the MixPre-3, it was too short without them. Alternatively, I’d consider the Anker PowerCore+ 20100 USB-C listed below.
Here are some other options offered by Sound Devices:
USB-C port via optional MX-CHARGE AC power wall adapter
USB-C port via MX-USBY cable and a computer with 2 USB-A ports
USB-C port via a compatible USB-C to USB-C cable and a computer with a USB-C port
D-Tap to L-mount adapter for use with the optional MX-LMount battery sled. See here.
Hirose to L-mount adapter for use with the optional MX-LMount battery sled. See here.
Anker PowerCore+ 20100 USB-C, Ultra-High Capacity Premium Portable Charger, 20100mAh USB-C battery with USB-C to USB-C cable. (Amazon link)
To fully power the MixPre-3 (or MixPre-6) from a USB power source not listed above, you will need BOTH of the following:
USB-C power source that is compliant with the USB powering specification and is able to output 7.5W
USB-C cable that incorporates the correct pull-up resistors to identify itself as able to provide 7.5W
If either of these conditions are not met, the MixPre recorder will operate in low power mode: maximum 2x 48v phantom channels, no HDMI, no USB-A. You can still use all of the MixPre’s channels, but only a maximum of 2 can be phantom powered.
NOTE: A USB-A power source used in conjunction with a USB-A to USB-C cable cannot deliver the required 7.5W for full power operation.
Due to the complex nature of the USB-C specifications for powering requirements, Sound Devices cannot make recommendations for USB-C power supplies, batteries or cabling other than its purpose-designed accessories. Sound Devices has no control over other manufacturer’s products and their (non-)compliance with the published USB-C specifications.
Four improvements I’d like to see via firmware
Currently, when in Advanced Mode, it is impossible to see any level meter when adjusting the level for any particular input. This request has been made by me and other reviewers. No promise has yet been made by Sound Devices. However, Sound Devices support has noted the request.
Regarding the USB keyboard option, I would like to see a keymap selection. With the current firmware, the MixPre-3 presently only supports the US keymap. My first requests are to add both Latin American and Spanish (European), aka Spanish ISO. The latter one has been used by Apple for at least 15 years, both in Spain and in the Americas, in addition to Acer and Logitech. HP also began offering the European Spanish (ISO) keyboard in the ZBook Studio G4 in the United States, as indicated in this article. Other keymap requests that you may have should be sent directly to Sound Devices support.
Considering that the MixPre-3 already has Bluetooth to communicate with the WingMan apps for Android and iOS, I would like Bluetooth keyboards to be supported via a free firmware update.
Either free, or for a reasonable upgrade fee, I would like Sound Advice to add automix capabilities to the MixPre-3. As I have covered in several past articles, automix reduces or eliminates crosstalk (spill) between microphones, either live or live-to-drive. Automix is already available from other products from Sound Devices, as well as from Audio Technica, Behringer and Shure.
Conclusions and related articles
Comparing the US$649 MixPre-3 (B&H link) with the US$269 Zoom H5 (Amazon link • B&H link) is unfair yet necessary, since the MixPre-3 is the smallest recorder that Sound Devices currently offers. If someone is looking for the input capacity of the H5 (2 balanced XLR inputs + 1 “stereo” unbalanced via 3.5 mm TRS), but seeks higher quality preamps, a hardware limiter and timecode, the unfair comparison becomes a necessary comparison. The MixPre-3 is evidently superior to the H5, and its quality and features justify its higher price. It will be regretful but necessary to send it back to Sound Devices after publishing this review.
I do not know Avry or Dave Girtsman, but I couldn’t resist including this:
Please see my related articles:
All audio production & distribution should go 48 kHz. Learn why.
Understanding 24-bit vs 16-bit audio production & distribution
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