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Old 11-09-2015, 06:04 AM   #1
Hypex
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Default My rendered master is flat and lacks "oomph"

Hi guys.

So I was doing a test to see how a final master would sound. My intention was to master it for CD a listen to it in a few stereos.

My sources aren't perfect, they were 44.1 KHz 16-bit 160 Kbps Ogg stems. But sounded fine when played from Reaper. I converted them to 16-bit WAV using Audacity first.

So I rendered them in Reaper. Working under the assumption that the stems had been volume scaled I didn't alter any volume or panning settings. Nor do any FX except muting parts to isolate some tracks. The waveforms aren't full scale although there is a loud guitar through part of it. I had a drum, bass, guitars and vox+extras tracks.

I rendered them as one 16-bit WAV. I then burnt the WAV to CD using Toast. But when I listened to it the resulting sound was flat. What I mean is it lacked treble. It didn't sound distorted but it just sounded flat and the drums lacked oomph. I didn't test it in the best stereo but I could tell there was something wrong with it.

Aside from my sub-CD quality source, have I gone about it the wrong way? I've read about doing renders in the forums and come across recommendations to render as FP so there is enough headroom for dynamic range and to avoid clipping. However, I kind of expected Reaper to take care of this internally, by rendering in memory as 64-bit FP, then scaling/normalising that down to 16-bit PCM so it all fits nicely. Well that's what I thought mastering would be internally and 100-200MB of memory should be enough to render a master sample.

I've used another DAW that used 16-bit mixing routines and volume scaled the tracks before mixing and that sounded all right, the result needing to be normalised. Pretty good in fact. So I was surprised it sounded better then Reaper. But to be fair they were other tracks, not these ones. Converted from Oggs also so comparable.

So do I need to render as 24-bit/FP and then render that again as a final 16-bit mix? I did notice the output going into the red as it was rendered so definitely slightly out of range. I'd really like to just do it in one action. After all I'm using an advanced DAW so I can do it in one sweep.
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Old 11-09-2015, 09:37 AM   #2
DVDdoug
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Quote:
My sources aren't perfect, they were 44.1 KHz 16-bit 160 Kbps Ogg stems. But sounded fine when played from Reaper. I converted them to 16-bit WAV using Audacity first.
There's no need to do that unless for some reason you can't open the OGG files in REAPER.

Quote:
But when I listened to it the resulting sound was flat. What I mean is it lacked treble.
The sound quality should not be worse than the original OGG unless you are applying effects that make it worse, or unless you are clipping. Or, if you re-render to OGG (or MP3 or any lossy format) you are going through an additional lossy-compression step.

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So I rendered them in Reaper. Working under the assumption that the stems had been volume scaled I didn't alter any volume or panning settings.
Mixing is done by addition (summing) so it's possible to have peaks over 0dB (can clipping when rendered) even though you have plenty of headroom in the individual tracks.

Quote:
I've used another DAW that used 16-bit mixing routines and volume scaled the tracks before mixing and that sounded all right, the result needing to be normalised. Pretty good in fact. So I was surprised it sounded better then Reaper. But to be fair they were other tracks, not these ones. Converted from Oggs also so comparable.
With 16-bits you may be getting some clipping that "sounds better" than an un-distorted file. Or, it may be the fact they they were different tracks and/or you used slightly different mix settings, etc. It's difficult to do the exact same project on two different DAWs. DAWs don't have a "sound" other than effects (or integer clipping). Summing is summing and it's about the simplest thing you can do with a computer, The computer or DAW don't make any difference.

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So do I need to render as 24-bit/FP and then render that again as a final 16-bit mix? I did notice the output going into the red as it was rendered so definitely slightly out of range. I'd really like to just do it in one action. After all I'm using an advanced DAW so I can do it in one sweep.
Since it's difficult to predict the final level, it's best to do it in two steps.

- Render to 32-bit floating-point. (24-bit WAV is an integer format and it's hard-limited to 0dB just like 16-bit.)

- Since you have Audacity, open the file in Audacity. If your floating-point file goes over 0dB, Audacity will show red for potential clipping (if you have Audacity configured to show clipping).

- Run Audacity's Amplify effect. Audacity has already scanned your file and the Amplify effect will default to whatever change is needed (up or down) for normalized ("maximized") 0dB peaks. So, just accept the default and apply the gain change.
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Old 11-09-2015, 09:37 AM   #3
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There's no "automatic scaling" going on anywhere. You need to set your levels so that the audio doesn't clip.

But are you actually saying things sound fine when you are working inside Reaper but no longer sound fine when you have rendered the mixed result? What are you using to listen to the rendered file?
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Old 11-10-2015, 06:32 AM   #4
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Originally Posted by DVDdoug View Post
There's no need to do that unless for some reason you can't open the OGG files in REAPER.
I can open them, I just didn't want the CPU burdened with decompressing them on the fly when I play back.

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The sound quality should not be worse than the original OGG unless you are applying effects that make it worse, or unless you are clipping.
I was only applying mute so it should have made it better in the muted parts. And rendering to lossless.

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Mixing is done by addition (summing) so it's possible to have peaks over 0dB (can clipping when rendered) even though you have plenty of headroom in the individual tracks.
Yes I suppose I was expecting that all the stems themselves had been scaled down relative to the whole mix so that wouldn't happen. Looks not.

Quote:
With 16-bits you may be getting some clipping that "sounds better" than an un-distorted file. Or, it may be the fact they they were different tracks and/or you used slightly different mix settings, etc. It's difficult to do the exact same project on two different DAWs. DAWs don't have a "sound" other than effects (or integer clipping). Summing is summing and it's about the simplest thing you can do with a computer, The computer or DAW don't make any difference.
The other DAW didn't do mixing itself, it prepared all the tracks with FX and then offloaded the mixing to an audio subsystem. This then used "primitive" mixing routines that summed all the samples and divided by number of channels, not sure in which order. Sure this works but each track looses volume and dynamic range in the process. IIRC that was an 8 track render so each sample sum would be added up, then divided by 8, which would reduce each track from 16-bit to 13-bit dynamic range, if my digital math is correct.

The end result had to be upscaled by 185% to normalise it. But it still sounded okay despite all that.

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Since it's difficult to predict the final level, it's best to do it in two steps.
That's what I thought I might need to do. I was just hoping Reaper would be able to perform the two steps for me. Explain why in my next post.

Okay, so I loaded my first render into Audacity, and is has red lines all over the place! At least 3/4 red, most notably the loud parts. :-)

I then rendered another as FP32 and got Audacity to scan it in Amplify. It detected 6.4dB overflow. So I used that to bring it down and exported to 16-bit PCM.

I loaded both back in to Reaper to compare. Examining the waveform the original has a few mountain heads chopped off. And the scaled version looks like it is half the waveform of the other. If half then only one bit is lost in scaling so each track would still retain 15-bit precision so not that bad really.

Thanks for all the advice. Think I'm getting somewhere. Just need to do another "CD" test. :-)

If the "flatness" in the treble is due to clipping, then I suppose the resulting sound would reflect the flat top given it's lacking detail. I would have expected clipping to sound distorted but in this case the flatness is reflected in the sound sphere. :-)
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Old 11-10-2015, 06:45 AM   #5
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Originally Posted by Xenakios View Post
There's no "automatic scaling" going on anywhere. You need to set your levels so that the audio doesn't clip.
I expected it to be automatic for two reasons:

1.) As in my previous post, I thought the stems would have been scaled down already when exported so they were rendered at the proper volume relative to the whole mix.

2.) Being I was doing a final master, I thought Reaper would do a full mixdown in FP64 and keep stats on maximum range calculated. And then use this info to scale it down to the PCM16 I had selected.

Perhaps I am expecting computers to be a bit to automatic here.

I'm not trying to be lazy, but my expectations may be too high. :-)

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But are you actually saying things sound fine when you are working inside Reaper but no longer sound fine when you have rendered the mixed result?
Yes. At this point I wonder how Reaper compensates in realtime playback? Does it use peaks as a reference for volume scaling for playback or do the "divide by number of channels" trick?

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What are you using to listen to the rendered file?
I used a CD stereo in the bathroom. Then I used my car. Not the greatest obviously but good enough to tell me I did something wrong.

I'll do another test.
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Old 11-10-2015, 08:52 AM   #6
LightOfDay
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I think you have a complete misconception of how digital audio and audio at all work.

please, do yourself a favor, and dont have digital math in your head. its plain wrong what you thought.

go back to square one. and dont make up any theories about anything, because you will be lead into total oblivion.

you have stems. you throw them into your DAW. a DAW is for recording, processing and mixing audio. there is nothing done unless you do it to the audio.

and what DAW should that be, that does that bs-math to the tracks loaded??

YOU are the mixer. the DAW is YOUR tool. YOU tell the DAW what to do and not to do.

and your math about the bit-rate is plain wrong, too.
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Old 11-10-2015, 09:53 AM   #7
zyisrad
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Are you sure you're looking at peaks on the master and not just RMS? It's pretty obvious when you go over 0dB on the master. When you see red. You are dead.

Right-click the Master track VU meter. Make sure to select "Peak only" or "Peak+RMS". The Top Label should show "Peak".

Also, from your writings, it's hard to tell if you understand the difference between floating point and fixed point when dealing with audio (you may understand but may not be articulating this accurately). Reaper uses 64bit floating point (As do most all DAWs) when summing and routing audio. So when summing and routing between tracks and folders you won't hear any digital distortion on anything above 0dB (yes there is a limit but it's a pretty high ceiling). On the master however your DAC uses 24bit fixed point (or lower). So anything above 0dB will be audible digital distortion.

You want to learn how to mix in the modern era of the Digital Audio Workstation on hard mode? This will allow for no mistakes in your gain staging.

Go to Preferences > Audio and select "Automatically mute any track" when volume exceeds, and input "0" in the next box. Disable "Reset on playback start".

Last edited by zyisrad; 11-10-2015 at 10:21 AM.
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Old 11-10-2015, 10:12 AM   #8
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Originally Posted by Hypex View Post
I used a CD stereo in the bathroom. Then I used my car. Not the greatest obviously but good enough to tell me I did something wrong.

I'll do another test.
And you listened with completely different equipment while mixing with Reaper?

I think that would explain this :

Quote:
My rendered master is flat and lacks "oomph"
You didn't do anything while you mixed/mastered to make it not sound flat and not lack "oomph" on different equipment? However you import and export audio files in Reaper will do close to nothing to fix that. What are you comparing against to with your bathroom/car playback?

For clarity, can you also just name the "other DAW" that you've also used that you got different results out of?
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Old 11-11-2015, 06:16 AM   #9
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Originally Posted by LightOfDay View Post
I think you have a complete misconception of how digital audio and audio at all work.
I know how it works. I've been playing with waveforms, samples and making techno music 20 years ago. I once wrote my own mixing program to mix two samples. It used a scissors clip to deal with overflow.

It's only lately I've been getting into experimenting with "real" music.

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please, do yourself a favor, and dont have digital math in your head. its plain wrong what you thought.
How is it wrong?

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and what DAW should that be, that does that bs-math to the tracks loaded??
Audio Evolution 4.

But whatever you think of the math, it's a rock solid method to compensate for clipping, as it will never happen. It's also good for a live mixer where the levels cannot be known and must be dealt with in realtime. Ot course it really just does basic statistical math by calculating the mean or average among all samples.

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YOU are the mixer. the DAW is YOUR tool. YOU tell the DAW what to do and not to do.
That may be so but I wish to mix it as close to the original as I can. Which is hard without an exact reference to the volume curve I can load in.

Quote:
and your math about the bit-rate is plain wrong, too.
How do you mean? I wasn't exactly talking about a bit rate. I was just saying if the resulting sample is halved for example then each track will lose half the volume on average.

Last edited by Hypex; 11-13-2015 at 02:14 AM.
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Old 11-11-2015, 06:33 AM   #10
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Originally Posted by zyisrad View Post
Are you sure you're looking at peaks on the master and not just RMS? It's pretty obvious when you go over 0dB on the master. When you see red. You are dead.
My guide was the Render window as it was doing a render. It was certainly over. I'll check the master VU.

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Also, from your writings, it's hard to tell if you understand the difference between floating point and fixed point when dealing with audio (you may understand but may not be articulating this accurately).
I can assure you I know the difference. Even though it was only a few years ago I came across using FP in mixing routines. And it seemed quite different and strange to using direct integer arithmetic according to a set bit depth. For example using a range of -1.0 to 1.0 instead of say -32768 to 32767 for 16-bit. And how it allows freedom by allowing you to overflow without cutting your head off.

The same could be done with integer if the headroom was used but usually the entire bit depth is used which limits the application.

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You want to learn how to mix in the modern era of the Digital Audio Workstation on hard mode? This will allow for no mistakes in your gain staging.
I can see now I'm going to lose tracks. :-)
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Old 11-11-2015, 06:49 AM   #11
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Originally Posted by Xenakios View Post
And you listened with completely different equipment while mixing with Reaper?
Yes.

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You didn't do anything while you mixed/mastered to make it not sound flat and not lack "oomph" on different equipment?
Not that I could see. They had basic treble and bass controls so good for a default reference. I like to hear it on "bad" or basic equipment so that if it sounds really shocking there is a problem that I expect will be amplified on good equipment.

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What are you comparing against to with your bathroom/car playback?
First are my ears. Which have been good at picking up issues. I trust them. For example, I once accidently took some songs i bought in iTunes, that I had burnt to CD, and imported them back into iTunes. Then wrote them to another CD and played back. I knew immediately something was wrong. I could tell the percussion was more tinny and other things lost quality. IIRC once track was burnt twice so I could flip and conform that iTunes is not CD quality and as good as an MP3, IOW lossy.

Second is listening from a comparable source. That would share a similar level of quality on bitrate.

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For clarity, can you also just name the "other DAW" that you've also used that you got different results out of?
Yes, AE4 as told above.

Last edited by Hypex; 11-12-2015 at 02:41 AM.
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