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‘Silences can be as different as sounds’Elizabeth BowenWhen working on Windows 10, it is common to encounter system volume issues. They might seem pretty grave, but rest assured there is no need to worry – you do not have to be an expert to fix them. Shown next are the problems to which we have found easy-to-implement and effective solutions:.

How to increase system volume on Windows 10?. How to fix volume control not working on Windows?. How to get the speaker icon back in the taskbar?Although our methods work best for Windows 10, feel free to try them on Windows 8, 8.1, and 7 since they may help you achieve the result you desire. So, it’s time to make your system volume drama a thing of the past. How to increase system volume on Windows 10?. Check all your volume controlsThis method might seem too plain, but there is more to it – users often forget to check the most obvious solutions to their issues and start digging too deep.

To prevent unnecessary troubleshooting, check every volume control in your system. Run your media player apps and configure their volume sliders. You should also check your Volume Mixer in Windows 10:. Move down to the taskbar. Right-click the speaker icon. Select Open Volume Mixer.Here you can raise the sliders for your devices and apps.

Enable Loudness EqualizationIf your volume level is inconsistent, it is best to turn on the Loudness Equalization feature: this option will stabilize your audio playback. Here are the instructions for you to follow:. Press the Windows logo key + S shortcut.

Type ‘audio’ (without quotes) into the Search area. Hit Enter. Select ‘Manage audio devices’ from the list of options. Select Speakers and click on the Properties button.

Navigate to the Enhancements tab. Check the Loudness Equalizer option. Select Apply and OK.Hopefully, your issue has been resolved. Check your default playback deviceThings often get mixed up when it comes to configuring one’s devices. What we are driving at is that you might have the wrong piece of hardware set as your default playback device, which means you might be tweaking the wrong settings. To check your default playback device, do the following:.

Go to the system tray. Right-click the speaker icon. Click Sound. Select Playback devices. Locate the device you wish to set as your default playback device.

From the drop-down menu, select Set as Default Device.We hope your low system volume issue is no more. Update your audio driversIf your sound is too low on Windows 10, you will do well to update your audio drivers. The point is, if they are obsolete or corrupt, it can afflict the output volume and cause you to have a hard time hearing what your speakers or headphones produce.In a case like this, you can use any of the following methods to make your audio drivers up to date: Use a dedicated toolThat is actually the easiest and the most efficient way to resolve your driver problems – a special program will save you a lot of time and effort by scanning your system and updating the drivers that are out of date or faulty. There are plenty of solutions available on the market these days, so make sure to choose the one that is safe, reliable, and effective. For instance, we recommend because this piece of software enables you to fix all your driver issues in just one click. Utilize Device ManagerAnother option implies employing the built-in Device Manager to get the job done. Here is how you can use it to get your audio drivers in tip-top shape:.

Press the Windows logo key + the letter X key shortcut on your keyboard. From the menu that appears, select Device Manager. In the Device Manager menu, locate your audio device and right-click it. Select Update driver. Select the option that implies searching for the necessary software online.Update your drivers manuallyThere is absolutely no necessity for you to do that yourself, but if you have your heart set on choosing this method, you are free to do so. Just make sure you know what you are looking for and start your browsing manoeuvres.

Search for the software needed online and install it on your PC.Whichever method seems best to you, it is essential that you restart your computer after you have installed the latest driver versions – otherwise, they may fail to become effective. Once your system has booted up and settled down, see if things have looked up. If they haven’t, do not hesitate to proceed to the following solution. Run the Windows Audio TroubleshooterIf your Win 10 audio needs fixing, the Windows Audio Troubleshooter is most likely the most efficient means of getting the job done. Feel free to call it upon to resolve your low system volume issue:. Open the Search app (the Windows logo + S keyboard shortcut).

Type ‘troubleshooter’ (no quotes needed) into Search. Press Enter. Select Troubleshoot to proceed. Click the Playing Audio troubleshooter. Select Run the troubleshooter.The tool will start scanning your system for audio issues.

If there are any, they will be fixed. Clean your speakersWe all know too well that dust is the arch-enemy of one’s computer. When our hardware and devices get clogged with dust, they start playing up. Unfortunately, your speakers are no exception.

So, if they produce suspiciously low sound, it is high time you gave them a thorough cleaning. Update WindowsYour audio level might have been reduced due to your system lacking essential updates.

JUSTINI have a question.i dont know if its a bad or unprofessional step but in the mixing stages i use a peak compressor to cut down peaks and mantain transients and keep a solo instrument in a natural sounding way if it keeps going up and own.then i use an RMS Compressor to raise the relative loudness not too much of it and then put a a limiter.without ever normalizing the audio file.i also want to understand if there are any Dynamics dangers to it.since normalization doesnt change the dynamics of a song. David Post authorThe only ‘dynamic dangers’ are the ones you can hear and don’t like!

So long as you can hear the changes you make at every step and like them then there are no problems.Using peak compression, then RMS compression, then limiting sounds OK on special occasions, but it could be overkill on a solo instrument. I would use peak compression with a long attack if you want to enhance the transient and add punch (rather than maintain the transient), RMS compression to make the sound smoother, and a limiter to ruin the sound at the end 🙂Seriously though. Don’t ever do anything to make something sound louder, only better! Compression can make things sound punchier, smoother and glue sounds together. It can also make things sound small, weak and pathetic. Make sure you compare before and after at the exact same volume. Check this video out for info on that.

David Post authorThere are a million things to do to improve vocals before and after recording, normalizing won’t affect the sound, so don’t worry about it.You can’t ‘see’ the signal in your DAW? Do you have enough gain in the pre-amp?

It should be as loud as possible WITHOUT ever clipping (going into the red). Normalize and listen, can you see the waveform now but there is a huge amount of noise in the background? That would mean your gain was far too low.Best to track the vocals in the studio, otherwise you need good quality equipment in a good room with good technique and experience!. KalypsoHi, I have to disagree about normalizing to maximize volume for numerous reasons:it’s the loudness-war non-sense, that every producer feels the need to obliterate their mixes by constantly finding ways to increase loudness, it’s bad for sound. What we need is to focus on the dynamics, that’s what music is aboutnot how “loud” can you go.Another very important reason not to raise volume to maximum is when you are doing your mix down you NEED to leave lots of headroom in the audio for mastering, you cannot master an audio track that eats up all the headroom and has clipping. If you want a louder recording then turn up the levels on the mixer a bit higher or add some gain to the mixer channels.

The first screenshots audio levels are perfectly fineI don’t understand why that needs to be boosted???If you want a little bit of loudness, the person doing your mastering will do that for you.Just try to avoid the need by normalizing by recording properly in the first place 🙂Please enjoy your day 🙂. David Post authorI agree with you 100%, but i think you are mixing up compression/limiting with normalizing. My article opens:“ It is different from compression that changes volume over time in varying amounts. It does not affect dynamics like compression, and ideally does not change the sound in any way other than purely changing its volume.”I despise the loudness war, it has ruined so many albums! Best not to digitally limit anything to gain volume, it sucks the life out of music and creates a harsh lifeless sound. You can edit the peaks down instead if there are a few offending ones. Compression on the other hand can make things sound better if done well.The main thing I use normalizing for is to make tracks all about the same volume in a mix (-18dBFS as a starting point), or for critical a/b listening tests.

I do it with the SWS loudness tool in Reaper which is non destructive and uses the r128 integrated loudness measurement. David Post authorIf it went over 0dBFS then I am afraid it is already clipped. It may not be noticeable if you are lucky, so have a close listen. To be sure not to clip it again keep it under 0dBFS at all times.When you say -0.3dBFS I think you are talking about avoiding inter-sample peaks. This is too big a subject for my reply, but I will tell you what I do (that some might argue with). If you normalize, do it to -1dBFS. You then avoid inter-sample peaks AND leave a little head room for lossy codec compression, for example, if you upload to YouTube or convert to MP3.

Would an article on inter-sample peaks be useful, or maybe you can get the info you need via Google? Hope that helps!. Prakash WadhwaniHI!!I’m a DJ & am looking to “normalize” my tracks some tracks are really very soft.

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I’m not so concerned with “perceived” loudness here just wanna ensure the track is at max volume without any clipping. Most importantly I’d like this to be performed “non-destructively” ie without any transcoding performed so as to avoid any detriment to the mp3 quality.I have mptrimPro What’s your view on that??I’ve also heard about mp3gain never tried it what’s your view on this too?Any other software you’d suggest free AND paid I’m looking for the best in both categories.Thank you so much for your time. David Post authorThere is already a perfect technology for DJing called ReplayGain. It analyses (a selection of) your music and attempts to make it the same volume, which is what you really want.

When you say ‘max volume’ it has to be compared to something else. If you just normalize everything, like this article talks about, then some tracks will sound a totally different volume due to their peak levels. We hear in RMS, but a digital normalize function is using PEAK value.ReplayGain stores the volume information it generates inside your mp3 file in a tag, so it is totally lossless. There are different programs for this that work in different ways, some use RMS and some R128 (similar to RMS, but modelled on human hearing). You can analyses the music per track, per album, or for you, probably the entire DJ set. I used Foobar, but all the major DJ software has a version of this built in.

Ernest GeorgeI am not a sound engineer Just a “listener”. Why doesn’nt some one devise a chip or circuit that would “intercept” the audio signal of a TV set just before it hits the speakers to “normalize” the volumn So that no matter what volumn the program or commercial puts out, you always get the same volumn outof the speaker, whatever you have set on your TV. Seems simple enough, even though the advertisers would probably not like it, I would happily pay for such a feature on a TV.ThanksE. GaharietHi and thanks for the informative article.I use Matlab to normalize several audio clips. I wanted them to be comparable in RMS.

But once I made them comparable, they clipped when saving them (because the amplitude values exceed the range (-1 1). I found out the only way to save them without clipping is by increasing the bit depth to 32 (from initial 16), as to allow greater values outside the (-1 1) range. Now my question: can I save the clips in bit32 depth and when they are reproduced somewhere else they won’t clip? This is my worry. I need to process this for a perceptual experiment and I am worried that the device used will need to be of certain quality (or maybe not). David Post authorI think the 32bit files will clip on some devices and not others, so I recommend that you make sure that none of the samples clip at 16bit. I suspect your 32bit audio is also float rather than fixed, so that may affect the clipping depending on what software is playing it.

It is probably reducing bit-depth somewhere depending on your sound-card output capabilities.If you want accurate comparable volume, I made a video on how to use a free tool to get great results. It’s linked here with the software:Sometimes you will need to use a limiter on the master bus if you want to RMS volume match and still have peaks all the way up to 0dB. You can also just turn down the waveform peaks in the audio editor manually, this can be the best quality way of doing it if there are only a few.There is also the issue of inter-sample peaks causing clipping. You could normalize to -1dB peak max, which is what I do for anything that is later to be converted (for example YouTube). You can use free meter to measure inter-sample peaks in the ‘true peak’ reading.Some cheap DACS distort at 0dB regardless of inter-sample peaks, I would just default to -1dB, the loudness war is over 🙂.

Geoff Lee“It is a destructive process. Performing any digital processing to a file is going to change it. Its bad reputation was mainly earned back in the days when digital files were all stored as 16 bit. If you turned the volume down you effectively reduced the bit depth. Your CD quality 16 bit file could end up 12 bit or less.

Even turning it up with PEAK normalization caused damage.”Normalising applies a fixed amount of gain to the audio and that can be removed in a none destructive way. It might help watching this as a brilliant primer to the technicalities of digital audio as to why what you said about bit depth doesn’t actually matter. David Post authorI have not used Nero for years so don’t know exactly what it is doing. What you need is to use ReplayGain technology. This adjusts the volume on playback losslessly rather than altering the original file. You scan your files and the meta-data is added in tags to them. You can scan per album and it should adjust individual tracks to playback at approximately the same level.

This will mess with the artists intention of dynamics, so you might want to try and make your entire collection of albums match better instead. There is no one-click solution to this, but ReplayGain is where you should start looking. Please let me and the other readers know how you got on. Foobar is a fantastic music player for this job on the PC, or Deadbeef on Linux. BrianI’d never tried to publish any of my home recordings until today, when I exported a file from GarageBand in uncompressed 24 bit to SoundCloud.

When I played it back it was badly distorted. I reset all the gain controls on the instruments and repeated the process.

At this stage I emailed SoundCloud with my concerns about the problems I was experiencing.Whilst waiting for a reply I fiddled about and found the Preferences settings in GarageBand, where under “Advanced” – somewhere I’d never been before – I found “Auto Normalise”, a box that was checked by default. So of course I unchecked it, repeated the exporting and uploading process – and what do you know? No more distortion. The sound was beautiful.And the moral of this story is: never, ever tick the box marked “Normalisation”. Do not ever go there.

Do not try to justify it with fancy diagrams, scientific data or statistics. Just leave the bloody thing alone!That’s my advice, anyway. MLOProblem I’m having is playing videos (movies) through my USB media player on a TV.

Volume is always too low. Have to turn up TV volume to 100% and it’s still too low. Same video plays fine volume wise on computer. To get my TV to play these videos at a volume comparable to watching normal TV, I have to convert the videos boosting the volume and then play converted result on TV. Should I boost volume on these conversions or normalize? Right now with these videos, I’m having to boost volume about +15 db to get them to play on TV at the same approx volume as watching regular TV. David Post authorI hear what you are saying, but I think it is beyond the scope of this article.

PEAK and RMS normalization mean the same thing across different software, so it is easy to find a program that can do it if you now know what to look for.If I were to go ahead and find all the software on all the different platforms that can perform these simple procedures it would require a new article. Loads of people read the comments, so if you have any tried and tested software to recommend please put it here in the comments and people should find it.The best software I have come across for all volume related functions is an extension for Reaper.

Go to extensions loudness. Amos spectorMany thanks for this post. Clarifying the issues in simple words.However, I still need some tech advice/explanation.

My audio files are mostly in WAV, and I usually use either Audacity or Audiograbber to make MP3 copies. As far as I could see, Audacity (free version) does only peak normalization. Audiograbber, on the other hand, allows average volume normalization (under advanced mode). And here are the questions: 1) what are the recommended settings for (A) use normalizing when average output is??? (seems the preset is 65)2) But only under these conditions: (A) Can you please state recommended settings?, (B) how can I impose a rule that refers to both average output and peak level?

What if they contradict?3) How do you recommend to set the compression box? Always compress? Never compress?

Only when needed? (I guess “only when needed” will be the answer), and what numbers to stick into the empty boxes?What I plan to do is create a few audio files based on you advice (without erasing the originals), try them and use the same settings for all my files.Another question: Would you expect the optimal settings to be different for different genre of music? Most of my files are jazz and classical (chamber music, such as string quartets, and orchestral music (symphonies and concerti)?Looking forward to your reply. David Post authorWhat is your exact goal?You have three separate ‘units’ of music to think about, the track, the album, and your entire collection. Normalization is going to make certain ‘units’ of music closer in volume to others. Do you want all your albums about the same volume when you listen to them? Are you aware of the lossless playback technology that does that automatically?

Maybe you are trying to do something else? If you are on PC then I highly recommend Foobar to deal with all this, it works well on Wine on Linux too. Amos spector“What is your exact goal?”First – thanks for your prompt reply.

It’s kind of you.My exact goal: I use a disk on key loaded with music in two situations: (A) I have a nice audio system in my room, and I want to use it to listen to music from my DOK. This is not a major problem, as I am home, near the computer, and can easily load the DOK with the wav files.

(B) I drive a lot, and listen to music (again – from a DOK) in my car. This is where I want to save space, and prefer to use smaller files, realizing that quality goes down a bit.

I think my car radio is not sophisticated enough to equate volume automatically at output. I believe I need equal-volume input (i.e. – equal volume files).I can use MP3, and not bother you, but I understand that M4A (or AAC, whatever the differences are) is/are better and more efficient in terms of quality-space ratio. I know how to use audacity to do it, but it allows only low-quality (96 kbs) files, and I failed to follow the instructions of how to improve sound quality.Yes – I am aware that all pieces will be about same volume, and that’s what I want (except for classical music, in which some movements are deliberately quieter than others, but I can overcome this by converting the whole piece as one unit, rather than every movement separately).I hope I am clear. I also hope there’s a good solution. David Post authorThe highest quality to size audio compression is opus 1.3, which is free and open source:Just go as high bitrate as you have the space for, and you will be getting the best possible quality. If you have more space FLAC is the standard for lossless.

I don’t know if your car supports those, if not, then LAME mp3 at 320kbps would be the next best bet.Seems you want to use Replaygain with the option to ‘hard render’ the actual output for your car. For everything you want, Foobar is the answer! You want to find a good guide on Replaygain, really get into what it can do, and then play around with the Foobar rendering.

It is very powerful and will take a bit of getting used to, but then you have the ultimate audio conversion system with a GUI. David Post authorUnfortunately it’s not as simple as choosing a level to normalize to due to the difference between RMS and Peak.

The key thing to remember is the difference between limiting and normalising. With limiting you actually chop the top off the waveforms and compress the sound, with normalizing you just bring the volume up to the highest level the peak allows.If you have a modern highly compressed track and normalize it next to a 70’s disco track with a lot of dynamics the modern track will sound massively louder. For this reason you need to use RMS to make things actually sound the same volume, but you will still have the difference between compressed and uncompressed music sounding different. As the uncompressed has such large peaks, you are in danger of limiting them if you try to match to modern tracks. This will add distortion to the transients.

You need to have software that juggles all these variables and gives you options to really get the job done. The closest thing to this is ReplayGain, see if you can learn all about how it works. It was designed to solve the exact problem you have, and to do it in a lossless way that won’t alter your music collection. TejNicely written article.

I am actually doing a small project basically identifying distortions in sounds which is turning to be quite a hard task. I have thought to generate spectrograms and then train a CNN for finding patterns based on good or bad. I have lots of audio files ranging from 10 seconds to 8 hours. Also audio has to be converted to mono format to generate spectrograms. I might lose some valuable information in here. Another issue is I have to split the audio file size into particular size chunks and analyze because I can’t bring an 6 hours to work on it.

If you have any information regarding this I would appreciate it. Thank you.1) Converting audio to mono. When i do so with librosa it cannot create with stereo only mono for mel spectrogram.2) what size or time interval chunks i should split my audio samples to?Post navigation.