Desktop audio is extremely low and audio monitoring doesn't work

I'm entirely new to OBS and I'm testing the settings with videos. I've set it to optimise for recording, not streaming.

I'm on Windows 10 and have turned off audio ducking, stopped applications from taking control and messed with the decibel meter in the mixer. I can turn the decibels up but that reduces the audio quality too much.

The audio from my headphones (which has been set as the default in the obs settings) is at max - to the point where it is too loud for me to have them on - but it is barely audible when I replay the recording.

I don't know what to do to make the audio any louder and I've done everything that I've found online.

Another problem is that the audio monitoring only works for my mic. I change it to monitor and output and when I move the volume slider, the mic volume changes. If I do the same for the desktop audio, nothing changes. I feel like it's still playing the audio directly from my desktop and not from obs. Is there a fix for this? Are these two problems linked?
 

srantum

New Member
I don't know is it the right answer but for me when I recording and the sound from the recording a little bit lower than when you hear from your headset,speaker, or anything, then you should change the desktop audio output capture from default to the correct device. For me I change from default to speaker realtek R(audio).

You can change it by accessing OBS -> sources -> add audio output capture -> right click on it -> properties -> change the default to the correct device like Realtek R(Audio) or anything just test it one by one and determine which is the right one to fix your low audio output.

Hopes it helps
 

Andy Miira

New Member
In my case, installing the win-capture-audio OBS plugin and using its new source "Application Audio Capture" fixed my low desktop audio volume issues!
 
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I'm not sure if it's related, but here is my audio issue. We use a Focusrite Scarlette Solo to take audio from a matrix send from the board into a Camera computer. In OBS it comes in through the Mic/Aux audio. With the slider set to where the signal is just at the end of the yellow and pops into the red occasionally, the recording and livestream audio is still very low. Is there a fix? I can give more details about our setup, but I don't know enough to know what to give you so you can better answer. Thanks for any help.
 

Edwo

New Member
i have the same thing, after putting in my audio sources i have to give everything the gain filter and put +30 dB on everything.
it doesnt reduce audio quality for me tho. im not a audio guy, so i dont know if by "reduced audio quality" you mean something, that i dont even notice.
 

AaronD

Active Member
Broadcast audio and live audio are NOT the same thing! And of course, when you broadcast live, you have to translate in real-time, from one to the other. You can't just plug the two together and expect both the volume and the quality that you're used to from a commercial broadcast.

For live audio, you don't know what's coming, so you have to keep some "headroom", as we call it - turn it down a bit, right up front at the mic itself, or the physical thing the directly receives the raw mic - so that the natural spikes and variation don't overload it. If you mess up here, there's no fixing it later...but it also sounds quiet if you just push it out to your audience as-is.

For broadcast audio, you need to completely fill the meter. Historically, this was done to overcome the limitations of the distribution medium - vinyl, cassette tape, radio, etc. - and it's become the standard that everyone's used to and what their listening controls are set for. Again, if you did that with a raw mic, you'd absolutely kill the quality. Practically clipping *all the time*.

In between, to do that translation, you need a compressor. Often followed by a limiter. OBS has both as filters. Both are like fast automatic volume controls, so you can keep the average high, but it turns down the peaks and comes right back up again for the rest. No way you could do all that by hand! OBS has terrible visibility for its tools, so here's a screenshot of a much better one:
1692148062984.png

The graph on the right is what OBS is sorely missing! It's a plot of the input volume horizontally vs. the output volume vertically. As you can see, it's a normal boost at low volume, but then it flattens out at high volume. The transition is called the Knee.

In order of OBS's settings:
  • Ratio controls how flat the high-volume part of the curve is.
    • A higher ratio squashes more (flatter line), and produces a more constant output volume regardless of the input volume, but only if the input is above the Threshold.
  • Threshold controls the input volume where the compressor starts working at all.
    • Below this, it's just a normal boost as set by the Output Gain.
    • Above this, it progressively turns itself down as set by the Ratio.
  • Attack is how fast it turns itself down for a louder input.
  • Release is how fast it turns itself up for a softer input.
  • Output Gain (called Makeup in the screenshotted one) is the amount of boost that it has to start with. The compressor action itself turns down from here.
    • If you're keeping a "live level" at this point in the processing chain, you'd watch the meter that shows how much it's turning down, and set the Gain to match that...except that OBS doesn't have any of those meters! Which is why I'm using the screenshot of a different one.
    • If you're translating at this point from "live" to "broadcast", you'd use the final output meter to set the Gain to *exactly* fill it. This is annoyingly the *only* meter that OBS has.
  • Sidechain selects which audio signal controls the compressor, if it's different from the one that it's processing.
    • "None" uses the same signal (called Feed-forward in the screenshot, because that one also has a Feed-back option and a few others)
    • Anything else allows *that* signal to drop the volume of *this* one according to this one's settings, without affecting that other one, and without actually compressing this one. This might be used to have some background music at full volume, for example, without interfering with your voice, all automatically. In that case, you'd put the compressor on the music, and set its sidechain to your voice.
    • If you want to both compress and "duck", as it's called, you'd need *two* compressors:
      • The first in the chain set to "None", as the compressor.
      • The second set to the other signal, as the ducker.
Other significant controls shown here, that OBS does not have:
  • Knee is how gradually the two lines connect:
    • Hard knee (0dB radius) is an instant transition. Very audible if you "ride the knee", not so much for incidental spikes.
    • Soft knee (non-zero radius) is a more gradual transition.
    • A *very* soft knee as shown here, combined with a high ratio, is somewhat like an "automatic ratio". At low volume, it might do "something", but you'd be hard-pressed to notice. As the input volume increases, it slowly clamps down harder and harder until it becomes practically a limiter. I like that because it keeps most of the transparency for normal speaking, and gradually transitions into "definitely being controlled" for yelling. But OBS doesn't have that soft of a knee, or even a control for it at all.
    • To approximate a soft knee, you might have several compressors, one after another, with progressively higher ratios...
  • The Mix settings allow for "parallel compression" by mixing the original input signal with what the compressor comes up with.
    • This effectively boosts softer signals via the Wet or compressed path, without squashing the louder sounds because the original Dry path takes over then. OBS doesn't do that...unless you have multiple copies of the same source, one with the compressor and one without.
Anyway, to send a live mic to broadcast, all in OBS, you might want something like this:
  1. Noise Suppressor, as the FIRST thing in the chain, so it's not trying to chase a varying noise floor after the compressors.
  2. Other processing as desired.
  3. Compressor
    • Ratio between 2:1 and 6:1
    • Threshold to make it work *most* of the time but not all the time
    • Attack and Release set by ear to sound natural or transparent
    • Output Gain to get back to the original "live" volume with normal material
  4. Compressor
    • Ratio 10:1
    • Threshold to make it work about half as much as the first one
    • Attack and Release set by ear to sound natural or transparent
    • Output Gain to exactly fill the meter
  5. Limiter
    • Threshold at -1dB
    • Release set by ear to sound natural or transparent
#2 and #3 might be swapped or interleaved as desired, but the last two need to be in that order and LAST. Likewise, the first one does need to be FIRST, or at the very least, before any compressors, limiters, gates, or other dynamic things.

Record yourself dead raw, then play that through your speaking chain. Set each thing in order, with everything after it either not added yet or disabled. Turn your speakers up as needed, until you get to the point where the processing chain does that for you.

---

For one of my rigs, that has the audio path entirely *outside* of OBS and only gives it the finished soundtrack to pass through unchanged, I have these two compressors, preceeded by a Noise Suppressor and a 2nd order highpass at 150Hz:
1692150806877.png

Highpass -> Noise Suppressor (order doesn't really matter for those two specifically) -> Left Compressor here -> Right Compressor here.

I'm using the right compressor as my Limiter, because that's all a limiter is: just a compressor with the Ratio all the way up, and usually a hard Knee and instant Attack, which I've also set here. If you take the vertical axis of the left one and put it on the horizontal axis of the right one, you might see that a typical raw input level of -18dBFS or so, "rides the knee" of the left (first) one, which puts it right at the knee of the second but doesn't quite get there, which puts the final output right at full-scale.

Whispering (farther left on the left graph) doesn't reduce the final output by much, but it does a little bit. Much more importantly though, it's not compressed as much (lower ratio because of the really soft knee that it's riding), so it sounds like a bigger difference than it really is.

Yelling will never go over full-scale either. That'll ride up higher on the super soft knee, which is a high ratio and thus more compressed. Again, the aggressive compression is what makes it sound louder, not the actual transmitted volume that barely changed at all.

You might also note that the Attack time is not quite instant for the first one that does most of the work. 10 milliseconds, or 1/100th of a second, which corresponds to 1/2-wave at 50Hz. Low frequencies will be distorted, starting slightly higher than that. So pretty much all of what the highpass allows through, stays clean at this stage, and the more "peaky" parts of words stay "peaky" too, which makes it sound a little bit more natural.

The second one, as a Limiter, *is* instant. Its output will *never* exceed the set level, even if it has to distort to do it. But because it's just a momentary thing - clipping only the single first wave, and all the rest stay clean at a low enough volume that fits in the meter - you're not going to notice.
 
Last edited:

emirena

New Member
Broadcast audio and live audio are NOT the same thing! And of course, when you broadcast live, you have to translate in real-time, from one to the other. You can't just plug the two together and expect both the volume and the quality that you're used to from a commercial broadcast.

For live audio, you don't know what's coming, so you have to keep some "headroom", as we call it - turn it down a bit, right up front at the mic itself, or the physical thing the directly receives the raw mic - so that the natural spikes and variation don't overload it. If you mess up here, there's no fixing it later...but it also sounds quiet if you just push it out to your audience as-is.

For broadcast audio, you need to completely fill the meter. Historically, this was done to overcome the limitations of the distribution medium - vinyl, cassette tape, radio, etc. - and it's become the standard that everyone's used to and what their listening controls are set for. Again, if you did that with a raw mic, you'd absolutely kill the quality. Practically clipping *all the time*.

In between, to do that translation, you need a compressor. Often followed by a limiter. OBS has both as filters. Both are like fast automatic volume controls, so you can keep the average high, but it turns down the peaks and comes right back up again for the rest. No way you could do all that by hand! OBS has terrible visibility for its tools, so here's a screenshot of a much better one:
View attachment 96826
The graph on the right is what OBS is sorely missing! It's a plot of the input volume horizontally vs. the output volume vertically. As you can see, it's a normal boost at low volume, but then it flattens out at high volume. The transition is called the Knee.

In order of OBS's settings:
  • Ratio controls how flat the high-volume part of the curve is.
    • A higher ratio squashes more (flatter line), and produces a more constant output volume regardless of the input volume, but only if the input is above the Threshold.
  • Threshold controls the input volume where the compressor starts working at all.
    • Below this, it's just a normal boost as set by the Output Gain.
    • Above this, it progressively turns itself down as set by the Ratio.
  • Attack is how fast it turns itself down for a louder input.
  • Release is how fast it turns itself up for a softer input.
  • Output Gain (called Makeup in the screenshotted one) is the amount of boost that it has to start with. The compressor action itself turns down from here.
    • If you're keeping a "live level" at this point in the processing chain, you'd watch the meter that shows how much it's turning down, and set the Gain to match that...except that OBS doesn't have any of those meters! Which is why I'm using the screenshot of a different one.
    • If you're translating at this point from "live" to "broadcast", you'd use the final output meter to set the Gain to *exactly* fill it. This is annoyingly the *only* meter that OBS has.
  • Sidechain selects which audio signal controls the compressor, if it's different from the one that it's processing.
    • "None" uses the same signal (called Feed-forward in the screenshot, because that one also has a Feed-back option and a few others)
    • Anything else allows *that* signal to drop the volume of *this* one according to this one's settings, without affecting that other one, and without actually compressing this one. This might be used to have some background music at full volume, for example, without interfering with your voice, all automatically. In that case, you'd put the compressor on the music, and set its sidechain to your voice.
    • If you want to both compress and "duck", as it's called, you'd need *two* compressors:
      • The first in the chain set to "None", as the compressor.
      • The second set to the other signal, as the ducker.
Other significant controls shown here, that OBS does not have:
  • Knee is how gradually the two lines connect:
    • Hard knee (0dB radius) is an instant transition. Very audible if you "ride the knee", not so much for incidental spikes.
    • Soft knee (non-zero radius) is a more gradual transition.
    • A *very* soft knee as shown here, combined with a high ratio, is somewhat like an "automatic ratio". At low volume, it might do "something", but you'd be hard-pressed to notice. As the input volume increases, it slowly clamps down harder and harder until it becomes practically a limiter. I like that because it keeps most of the transparency for normal speaking, and gradually transitions into "definitely being controlled" for yelling. But OBS doesn't have that soft of a knee, or even a control for it at all.
    • To approximate a soft knee, you might have several compressors, one after another, with progressively higher ratios...
  • The Mix settings allow for "parallel compression" by mixing the original input signal with what the compressor comes up with.
    • This effectively boosts softer signals via the Wet or compressed path, without squashing the louder sounds because the original Dry path takes over then. OBS doesn't do that...unless you have multiple copies of the same source, one with the compressor and one without.
Anyway, to send a live mic to broadcast, all in OBS, you might want something like this:
  1. Noise Suppressor, as the FIRST thing in the chain, so it's not trying to chase a varying noise floor after the compressors.
  2. Other processing as desired.
  3. Compressor
    • Ratio between 2:1 and 6:1
    • Threshold to make it work *most* of the time but not all the time
    • Attack and Release set by ear to sound natural or transparent
    • Output Gain to get back to the original "live" volume with normal material
  4. Compressor
    • Ratio 10:1
    • Threshold to make it work about half as much as the first one
    • Attack and Release set by ear to sound natural or transparent
    • Output Gain to exactly fill the meter
  5. Limiter
    • Threshold at -1dB
    • Release set by ear to sound natural or transparent
#2 and #3 might be swapped or interleaved as desired, but the last two need to be in that order and LAST. Likewise, the first one does need to be FIRST, or at the very least, before any compressors, limiters, gates, or other dynamic things.

Record yourself dead raw, then play that through your speaking chain. Set each thing in order, with everything after it either not added yet or disabled. Turn your speakers up as needed, until you get to the point where the processing chain does that for you.

---

For one of my rigs, that has the audio path entirely *outside* of OBS and only gives it the finished soundtrack to pass through unchanged, I have these two compressors, preceeded by a Noise Suppressor and a 2nd order highpass at 150Hz:
View attachment 96828
Highpass -> Noise Suppressor (order doesn't really matter for those two specifically) -> Left Compressor here -> Right Compressor here.

I'm using the right compressor as my Limiter, because that's all a limiter is: just a compressor with the Ratio all the way up, and usually a hard Knee and instant Attack, which I've also set here. If you take the vertical axis of the left one and put it on the horizontal axis of the right one, you might see that a typical raw input level of -18dBFS or so, "rides the knee" of the left (first) one, which puts it right at the knee of the second but doesn't quite get there, which puts the final output right at full-scale.

Whispering (farther left on the left graph) doesn't reduce the final output by much, but it does a little bit. Much more importantly though, it's not compressed as much (lower ratio because of the really soft knee that it's riding), so it sounds like a bigger difference than it really is.

Yelling will never go over full-scale either. That'll ride up higher on the super soft knee, which is a high ratio and thus more compressed. Again, the aggressive compression is what makes it sound louder, not the actual transmitted volume that barely changed at all.

You might also note that the Attack time is not quite instant for the first one that does most of the work. 10 milliseconds, or 1/100th of a second, which corresponds to 1/2-wave at 50Hz. Low frequencies will be distorted, starting slightly higher than that. So pretty much all of what the highpass allows through, stays clean at this stage, and the more "peaky" parts of words stay "peaky" too, which makes it sound a little bit more natural.

The second one, as a Limiter, *is* instant. Its output will *never* exceed the set level, even if it has to distort to do it. But because it's just a momentary thing - clipping only the single first wave, and all the rest stay clean at a low enough volume that fits in the meter - you're not going to notice.
Thanks so much for this detailed response! I looked at filters in OBS and I see that there's no visual representation to set the options from, as you stated.

I'm absolutely uneducated in audio engineering and sound tech. Just trying to improve my boyfriend's YouTube video volume, which is extremely low compared to other videos.

We're using a MacBook pro for OBS at the moment. What LSP software are you using to visually determine the different levels for the filters in OBS?

Would you ever be willing to help walk us through this? Or perhaps there's a video online that demonstrates....

He also happens to be using an aux via TRS input for a piano audio into a focusrite scarlet but we couldn't figure out how to get a separate audio device in OBS mixer for that versus the microphone. The point you make about setting a compressor for BG volume versus voice sounds relevant here.

Thanks again for your input!!!
 

AaronD

Active Member
We're using a MacBook pro for OBS at the moment. What LSP software are you using to visually determine the different levels for the filters in OBS?
LSP = Linux Studio Plugins
It's not a tool to determine the settings in something else. It's a set of processors itself that has all of that.

There are TONS of tools available if you use a DAW instead of OBS like I do. I just happened to find LSP myself, already using Lubuntu Linux, maybe a year before I switched to Ubuntu Studio Linux, and I was VERY pleasantly surprised to see that UStudio already has the LSP toolset preinstalled and working! (and a bunch of others too) Yay!
That "preinstalled and already working" theme continues for a bunch of other things too, including OBS, albeit an old version. If you go that way, you'll want to update OBS to the latest official release before you build a rig on it, but that's easy.

He also happens to be using an aux via TRS input for a piano audio into a focusrite scarlet but we couldn't figure out how to get a separate audio device in OBS mixer for that versus the microphone. The point you make about setting a compressor for BG volume versus voice sounds relevant here.
Sidechain compression ("ducking") is not that. And you've found another annoying problem with OBS by itself: it does not like multiple things coming from the same physical device.

OBS started as a bedroom game stream, and not much more than that. Thus, each source grabbed the entire device - the game or a single mic - and that was it. A multichannel device must therefore be surround sound, and downmixed accordingly, with no option to change that behavior. Fast-forward to today: none of that has changed, and OBS's audio is such a mess under the hood that it's probably not going to change *at all* until it's completely replaced, and that's going to take a while too.

So that's another reason to *get the audio processing out of OBS*, and into something that is actually capable without being annoyingly naive or blind. OBS then, becomes a straight-wire dumb-passthrough of exactly one audio source, which comes from that external tool. Absolutely all of the audio processing is done by that tool, and OBS has nothing to do with it whatsoever except to receive the result already finished.

On my rig, that external tool is a DAW called Ardour, which is also preinstalled and working in Ubuntu Studio:
It takes a *little* bit of work to make the connection between it and OBS, but it's not too bad. It's supposed to be even easier now with the latest LTS release of UStudio, but I haven't actually tried it for myself yet. I'm still on the previous LTS, which is more convoluted but very well documented, and does work well once you get it all right.

I'm absolutely uneducated in audio engineering and sound tech. Just trying to...
Buckle up and get ready for a ride! Media Production is inherently technical. In fact, it's right on the border between art and tech, and so you need to wear *both* hats well.

Fortunately, the tools are all the same, no matter where they appear or who makes them or how their controls are laid out or how they work internally or whatever. Some have fancier features than others, but the basic principles are all the same. So go watch a bunch of YouTubes, read articles, whatever works for you, about Compressors, EQ's, mixing tips and tricks, etc. and so on, and look especially for the underlying principles and not just "which knobs to turn".

Don't shy away from a sea of knobs just because all of those controls are intimidating. All of that boils down to almost nothing, really. Most of it is just the exact same thing, repeated over and over and over again. A 40-channel analog console, for example (I happened to inherit the linked one from someone's upgrade to digital, and something tells me I'm going to use it again somewhere), with 8 auxiliary mixes and a 4-band semi-parametric EQ, has 40*(8+6+2) = 640 knobs just for the input processing! (each input channel has the 8 aux volumes, the 4-band EQ has 6 knobs because the middle 2 bands also have a knob for what frequency they work at, and there's also a preamp gain at the top and a stereo pan control at the bottom of each channel strip) But once you learn the 16 knobs on just one strip, and realize that there's really only 7 or 9 unique functions depending on how you count the middle EQ bands, you've suddenly got almost the entire board!

Same with media production in general. It may seem intimidating at first, but there's really only a handful of critical things to understand. It can be difficult to get there, starting from scratch, but once you have that, the whole world suddenly opens up! Sure, there's always something more to learn, but those things quickly become minor. (that's when it becomes fun!)
 
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