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Old 02-05-2014, 02:23 PM   #1
Jenny
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Default Effect/Amplify

Hello everyone. I just recently purchased Reaper and think it's a great program for it's price and size. Lots of fun.

There is one thing I can't seem to figure out. Is there anyway I can get the spikes bigger while I'm recording. Audacity has this feature called Effect/Amplify which normalizes recorded output. Is there a feature like this in Reaper.

Thanks fof helping.

J
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Old 02-05-2014, 02:24 PM   #2
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Hello everyone. I just recently purchased Reaper and think it's a great program for it's price and size. Lots of fun.

There is one thing I can't seem to figure out. Is there anyway I can get the spikes bigger while I'm recording. Audacity has this feature called Effect/Amplify which normalizes recorded output. Is there a feature like this in Reaper.

Thanks fof helping.

J
If you truly want to normalize the item, right-click it and look for normalize in the properties window that pops up. Otherwise if you only want to change the appearance not the actual volume, there is a key combo that I am currently forgetting.

btw... welcome to the forum.
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Old 02-05-2014, 02:27 PM   #3
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For the waveform appearance: Shift + Up/Down
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Old 02-05-2014, 02:35 PM   #4
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For the waveform appearance: Shift + Up/Down
^Yea that. Thanks DarkStar.
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Old 02-05-2014, 02:37 PM   #5
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If you truly want to normalize the item, right-click it and look for normalize in the properties window that pops up. Otherwise if you only want to change the appearance not the actual volume, there is a key combo that I am currently forgetting.

btw... welcome to the forum.
Thank you for such a quick response. Awesome! Do you lose any quality when you do this? I don't know that much about audio recording.

Thanks again.

J
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Old 02-05-2014, 03:05 PM   #6
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I'd suggest searching for another way to get higher peaks than by normalizing or changing the zoom of the waveform. In fact, normalize is very rarely of real use and in most cases I wouldn't recommended using it. It makes your signal louder but sound quality suffers.

Imagine a small digital photo. You can make it bigger in size but you end up with more apparent artifacts (perceptible square pixels) as well. You rather want to work on a picture having more pixels right from the start that would have more "headroom" while enlarging it before artifacts get visible.

In audio, dpi or pixels translate into volume (and bit resolution). The higher the volume (and the bit rate) the lesser artifacts you will percieve/get raising the volume subsequently. Signals with a decent level that are converted to digital have smaller "steps" (if any) after the conversion process than low level input signals.

Figure out how to record a louder signal right from the start. Normalize is not the cure for low level a/d conversion. It's only one more unnecessary DSP calculation that reduces the resolution of your signal.
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Old 02-05-2014, 03:36 PM   #7
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Thanks for your help. When I asked if it would effect the quality, I was feeling something of what you just said but would never have been able to put it in words. If that makes any sense. LOL.
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Old 02-05-2014, 03:41 PM   #8
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I think the digital photo analogy makes it easier to understand what's going on :-) How do you set the recording levels in your system? Mic pre, interface, other method?
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Old 02-05-2014, 04:00 PM   #9
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Gotta go have supper but I will tell you exactly what I'm trying to do in about half an hour if you are still around. Thank you.
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Old 02-05-2014, 04:04 PM   #10
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midnight here but studio nights can get pretty long :-)
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Old 02-05-2014, 04:06 PM   #11
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It makes your signal louder but sound quality suffers.
The above is a can of worms (actually it sounds just plain wrong but maybe I misread it) because it doesn't do anything to the actual sound quality of what you are normalizing. The closet you can get is intersample overs but normalize doesn't need to mean normalize to zero.

Now I'm 100% with you in the overall sound quality of the mix suffers if you normalize everything but that isn't the integrity of the 1s and 0s but rather the overall quality of the result from having silly levels. Again can of worms maybe another thread since the last thing I want to do is ruin this new user's thread with something that has been hashed over already a million times.

As far as zooming the waveform it's up to user preference and their role in using a DAW. AKA that absolutely doesn't alter the bits or quality and only comes down again to the role the person plays in the mix.

The only thing bad about the photo analogy is it is off a little, you are comparing resolution as in frequency with resolution as in DR/bits. A better analogy would be normalizing the dynamic range (distance between lightest/darkest pixel in steps) not pixilation due to an image that is too small because that is impossible in normalizing audio and would be a gross misinterpretation of what normalizing is.

Please though, lets do it in another thread and not ruin this one unless the OP wants it here.
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Old 02-05-2014, 04:27 PM   #12
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let me put it this way: if you feel the need of using normalization on a track it means that the recording level of that track had probably been set to low while recording, thus A/D conversion has taken place under less than optimal conditions. The track will not make use of the optimal bit resolution. Perhaps it will only use 16 out of 24 bits.

Normalizing this track let's say to -1 dB (or simply applying the respective amount of gain without normalizing) will make the artifacts (steps in waveform) already present in the weak digitized signal more perceptible. The signal might not be degraded, OK, but its flaws will likely become more apparent - just as with a digital photo of poor resolution that does reveal its flaws of insufficient resolution after making it bigger.

Normalization is not altering the dynamic rage, it's not compressing the signal. It's a simple linear volume change.

peace :-)
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Old 02-05-2014, 04:40 PM   #13
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I'm out of my league here me thinks. I can write out scores for 5 or so instruments in Finale or Sibelius but when it comes to recording I have a lot to learn.

Anyway what I'm trying to do is record an acoustic guitar using a
L1® Compact system:

http://www.bose.com/controller?url=/...pact/index.jsp

and a Bose T1 ToneMatch® audio engine:

http://www.bose.com/controller?url=/...ematch_acc.jsp

The whole system is plugged into my computer and the spikes in Reaper are affected by the volume level of the T1. I have to turn all my volume controls up almost to maximum to get a recording that is not quite loud enough unless I normalize it. The same thing happens with Audacity. Does this make any sense? I have tried many different combinations of different scenarios with no luck. Maybe I'm missing the simplest thing. Probably not many people are familiar with the Bose system here so I'm probably out of luck.

Thanks

J
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Old 02-05-2014, 04:42 PM   #14
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Quote:
Originally Posted by SonicAxiom View Post
let me put it this way: if you feel the need of using normalization on a track it means that the recording level of that track had probably been set to low while recording, thus A/D conversion has taken place under less than optimal conditions. The track will not make use of the optimal bit resolution. Perhaps it will only use 16 out of 24 bits.

Normalizing this track let's say to -1 dB (or simply applying the respective amount of gain without normalizing) will make the artifacts (steps in waveform) already present in the weak digitized signal more perceptible. The signal might not be degraded, OK, but its flaws will likely become more apparent - just as with a digital photo of poor resolution that does reveal its flaws of insufficient resolution after making it bigger.

Normalization is not altering the dynamic rage, it's not compressing the signal. It's a simple linear volume change.

peace :-)
That make sense even to me!
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Old 02-05-2014, 05:06 PM   #15
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Jenny, do you record the guitar with a mic or does it have a line out?

The manual of the T1 states that it can only operate at 48 kHz/24 bit. After connecting it to your computer via USB you have to configure reaper to use it as an audio input device and follow that sampling rate. All level settings for recording have to take place before the signal arrives in reaper. First thing would be setting the level of your guitar (if it outputs a line signal) or the level of the mic plugged into the T1 (Master, trim and channel volume set to 0 to start).

Raise trim level until the led flashes yellow on very loud peaks occasionally (never red!). Set Master to 12 the o'clock position and then raise the channel level until you hit the desired level (ca. -9 dB to -6 dB on very loud peaks). No need to normalize tracks that peak at -15 db or above.

I just had a quick read in the manual. Hope this is correct.
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Old 02-05-2014, 05:08 PM   #16
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let me put it this way: if you feel the need of using normalization on a track it means that the recording level of that track had probably been set to low while recording, thus A/D conversion has taken place under less than optimal conditions. The track will not make use of the optimal bit resolution. Perhaps it will only use 16 out of 24 bits.
16 out of 24 bits would be 48dB down!!!! (6dB per additional bit.) Yes, that would be WAY too low!

It's quite common to record at -12 to -18dB at 24-bits (to allow headroom for mixing & effects as well as unexpected peaks). And most 24-bit A/D converters are only accurate to about 20 bits anyway.

Quote:
Normalizing this track let's say to -1 dB (or simply applying the respective amount of gain without normalizing) will make the artifacts (steps in waveform) already present in the weak digitized signal more perceptible. The signal might not be degraded, OK, but its flaws will likely become more apparent -
Yes, but that's no different from analog amplification. In the end, it's going to get played-back at whatever level the listener wants to play it back. The 1st "defect" you'll probably notice is that the background noise increases when you increase the volume.

If you amplify digitally, there are some rounding errors when you quantize to integer for rendering or playback through the DAC, but these are teeny-tiny and you can't hear a 1-bit change (in the least significant bit) at 16-bits or higher, because it's more than 90dB down at 16-bits. Digital amplification certainly does less "damage" than analog amplification.

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...will make the artifacts (steps in waveform) already present in the weak digitized signal more perceptible.
"Steps in the waveform" are quaitization noise, which you can hear at 8-bits, but you can't hear at 16-bits (and probably can't hear at 12-bits).

Volume adjustments/changes (up and down) are the most basic part of mixing and it's done every day with no problems!

Last edited by DVDdoug; 02-05-2014 at 05:13 PM.
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Old 02-05-2014, 05:17 PM   #17
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There are no "steps" in the waveform, the actual waveform is recreated, that is a misconception from seeing drawings LOL.
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Old 02-05-2014, 05:19 PM   #18
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Originally Posted by SonicAxiom View Post
Jenny, do you record the guitar with a mic or does it have a line out?

The manual of the T1 states that it can only operate at 48 kHz/24 bit. After connecting it to your computer via USB you have to configure reaper to use it as an audio input device and follow that sampling rate. All level settings for recording have to take place before the signal arrives in reaper. First thing would be setting the level of your guitar (if it outputs a line signal) or the level of the mic plugged into the T1 (Master, trim and channel volume set to 0 to start).

Raise trim level until the led flashes yellow on very loud peaks occasionally (never red!). Set Master to 12 the o'clock position and then raise the channel level until you hit the desired level (ca. -9 dB to -6 dB on very loud peaks). No need to normalize tracks that peak at -15 db or above.

I just had a quick read in the manual. Hope this is correct.
Thanks for going to all this trouble.

My guitar has a line out and I know all about the level settings for the guitar but you said something about "you have to configure reaper to use it as an audio input device and follow that sampling rate". Would you know how do do that?

Thanks

J
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Old 02-05-2014, 05:23 PM   #19
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let me put it this way: if you feel the need of using normalization on a track it means that the recording level of that track had probably been set to low while recording, thus A/D conversion has taken place under less than optimal conditions.
I can agree with that completely. I might would have said "If you think your levels are too low" but that's just nitpicking. Context is everything here and it's extremely important users understand that getting it through the ADC converters properly is of utmost importance.

The only thing I'll add is that to the uninitiated, the waveforms (when they are correct per the specs of the sound card being used), will likely "appear" too small when in reality they are not. In most cases -18 dbFS is the nominal area for the incoming signal. I say area because it is dependent on the soundcard and the best way to know is to just look it up in that card's spec and use that.
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Old 02-05-2014, 05:32 PM   #20
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As much as I love a technical debate, jenny still has some unanswered questions/issues.

Jenny, the quiet sound thing is extremely common (more like guaranteed) when you start out recording. The good news is that you aren't doing anything wrong by having your levels lower than you are used to hearing on commercial CDs. In fact, it's usually preferred to have your tracks at a significantly lower volume that you are used to hearing on CDs.

The easiest thing you can do to get your levels up to a "normal" volume is to put a limiter on your master output track. That will let you boost the level of the whole mix and will prevent it from distorting. Of course, there's a bit more to it than just that if you don't want some unwanted side effects, but it will get you 85% of the way there and at least point you in the right direction.

There is a plugin that comes with reaper called LOSER/masterLimiter. Put that on your master track and adjust the Threshold (dB) slider. Your song will get louder and it won't distort.
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Old 02-05-2014, 05:46 PM   #21
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sorry about this normalization discussion going on, Jenny, but why not clarifying a subject that seems to need some clarification. It does no harm to anybody and we will all be able to and make better recordings afterwards :-)

DVDdoug, please try this: play back a track of mastered music with the fader set to -90 dB and record it with your DAW set to 16 bit resolution. Then normalize this recording to -1 dB ad take a look at the waveform. Make shure to turn down your speakers/phones levels before listening to this!!!

Then set your recording software to 24 bit resolution and rerun the same test. While the first recording is plain torture to listen to (square waves all over the place), the second recording should be of almost studio quality. 1st recording: ca. 2 bits of resolution, 2nd recording: 10 bits.

There are significant differences in analog and digital level changing! Digital audio comes to its limits at very low levels and/or with low bit resolution and low sample rates (under 44.1 kHz/24 bit). Analog audio gives superb resolution at any level beneath clipping, even at very low levels. It the has to struggle against the noise floor obviously but its resolution is way better at low levels than that of digital signals unless you use plenty of bits (24 or more). Digital has discrete/finite level resolution, analog has infinite level resolution. Low level is the weak point of digital audio and raising the level subsequently will shift these flaws into the audible range.

You won't notice the quantisation steps of a digital 16 bit low level signal peaking at -40 dB as long as it stays at -40 dB in the mix. Raise it (normalize it) to peak at -6 dB and you will probably hear its harshness. Subsequent, unnecessary dsp calculations and rounding errors are the cause of digital audio being described as cold and harsh. If you keep levels reasonably high (especially during conversion but also afterwards during the whole mix and mastering process) and avoid unnecessary dsp calculations your signal will sound much better - more analog :-)
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Old 02-05-2014, 05:54 PM   #22
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Digital has discrete/finite level resolution, analog has infinite level resolution.
well crap, now I can't help but be dragged into this. Analog certainly does not have infinite resolution. The resolution is determined by the noise floor. Try amplifying any analog signal by 90dB and you'll just get a bunch of noise.

Unless you are purposefully trying to screw up your recording, there's no reason that quantization noise should ever be an issue anywhere. Even at 16 bits, quantization noise won't be an issue with any half competent tracking level.
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Old 02-05-2014, 06:11 PM   #23
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well crap, now I can't help but be dragged into this. Analog certainly does not have infinite resolution. The resolution is determined by the noise floor.
I apologize, bozmillar, but you're wrong. The resolution of an analog signal has nothing to do with a noise floor that may masking it. Digital audio "degrades by design" as it drops to lower levels. It's obvious, it is represented by fewer bits and approaches a pure square wave approaching the level of the LSB. An analog audio wave can be of great resolution/sound quality even while "bathing" in a noise floor. The analog audio system adds unwanted noise to the signal at low levels but it doesn't force the waveform to alter quite as much as in digital.
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Old 02-05-2014, 06:15 PM   #24
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Originally Posted by bozmillar View Post
As much as I love a technical debate, jenny still has some unanswered questions/issues.

Jenny, the quiet sound thing is extremely common (more like guaranteed) when you start out recording. The good news is that you aren't doing anything wrong by having your levels lower than you are used to hearing on commercial CDs. In fact, it's usually preferred to have your tracks at a significantly lower volume that you are used to hearing on CDs.

The easiest thing you can do to get your levels up to a "normal" volume is to put a limiter on your master output track. That will let you boost the level of the whole mix and will prevent it from distorting. Of course, there's a bit more to it than just that if you don't want some unwanted side effects, but it will get you 85% of the way there and at least point you in the right direction.

There is a plugin that comes with reaper called LOSER/masterLimiter. Put that on your master track and adjust the Threshold (dB) slider. Your song will get louder and it won't distort.
Thanks for that info. Actually my recordings sound fairly good with those lower levels. I tried what you said and it seems to work really well. Would you like to hear my one of my recordings then maybe you could have some more tips for me. I play acoustic fingerstyle instrumentals and occasionally sing if I've had a couple of drinks. LOL

P.S. I don't mind at all if you discuss or debate. I'm learning a little bit.
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Old 02-05-2014, 06:19 PM   #25
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Jenny, could you tell me at what level your raw guitar track (pure recording without fx, gain, etc.) is peaking at.
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Old 02-05-2014, 06:28 PM   #26
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Jenny, in reaper's preferences -> Audio -> you need to select the desired sound device (and its driver) to work with. Then you need to specify audio ins and outs as well (obviously, you've already done this).

Select to get the sample rate from the audio device. In the project settings (in file menu) set the project sample rate to 48.000 Hz (and check "save as standard" option). Does the T1 come with a dedicated software configure it?
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Old 02-05-2014, 06:37 PM   #27
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Quote:
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Jenny, could you tell me at what level your raw guitar track (pure recording without fx, gain, etc.) is peaking at.
Sorry no. This is another little problem I'm having. I can't seem to record a dry signal using the T1. I've tried somewhat to figure that out but havn't come across a solution yet.
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Old 02-05-2014, 06:43 PM   #28
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did you try to follow the steps I indicated in my last post to configure your device?
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Old 02-05-2014, 06:53 PM   #29
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Jenny, in reaper's preferences -> Audio -> you need to select the desired sound device (and its driver) to work with. Then you need to specify audio ins and outs as well (obviously, you've already done this).

Select to get the sample rate from the audio device. In the project settings (in file menu) set the project sample rate to 48.000 Hz (and check "save as standard" option). Does the T1 come with a dedicated software configure it?
What do you mean 'Select to get the sample rate from the audio device'
I went to Project Settings and changed the sample rate to 48.000 but noticed little difference.
The T1 doesn't come with dedicated software but there are many different options and settings for different situations.
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Old 02-05-2014, 07:00 PM   #30
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on the audio device/driver selection page you can check an option so that reaper takes over the sample rate of the external device. I'm using the German version of reaper and probably mis-translated the option's name :-)
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Old 02-05-2014, 07:08 PM   #31
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on the audio device/driver selection page you can check an option so that reaper takes over the sample rate of the external device. I'm using the German version of reaper and probably mis-translated the option's name :-)
Ah. Yes I did what you said but it doesn't seem to affect the spikes. Sorry. Am I driving you crazy yet. Thank you for helping but don't worry if we can't find a solution. I'll live. LOL.
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Old 02-05-2014, 07:35 PM   #32
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Sorry no. This is another little problem I'm having. I can't seem to record a dry signal using the T1. I've tried somewhat to figure that out but havn't come across a solution yet.
You initially said that you can record but that the small waveforms (or the low levels instead?) are bothering you. How come you can record but not a dry signal?? What are you recording if not a dry signal?
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Old 02-05-2014, 07:44 PM   #33
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You initially said that you can record but that the small waveforms (or the low levels instead?) are bothering you. How come you can record but not a dry signal?? What are you recording if not a dry signal?
What I mean by a dry signal is just the signal from the guitar without any effects on it. When I start recording the preset on my T1 Tonematch has reverb, chorus and delay and Reaper picks all that up. Have I got it wrong about what I dry signal is?
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Old 02-05-2014, 07:59 PM   #34
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Ah, I see. I would recommend, however, that you record your guitar without any fx applied by the T1. ReaFX and a lot of other free (or commercial) plugs will offer a far better quality, I suppose. And on top of that, you are more flexible and can tailor the sound of your guitar later in every way. You might try out different things/settings non-destructively. Recording with fx limits/steals a lot of these options.
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Old 02-05-2014, 08:16 PM   #35
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Originally Posted by SonicAxiom View Post
Ah, I see. I would recommend, however, that you record your guitar without any fx applied by the T1. ReaFX and a lot of other free (or commercial) plugs will offer a far better quality, I suppose. And on top of that, you are more flexible and can tailor the sound of your guitar later in every way. You might try out different things/settings non-destructively. Recording with fx limits/steals a lot of these options.
I understand what you are saying. The trouble is some of my songs don't sound as effective without fx applied to them. So when playing without any fx I am not hearing how they are REALLY supposed to sound and I can't get totally emotionally and spiritually into the song. So the finished result [no matter how well I treat it with fx afterwards] will not sound as good as if I had originally recorded it with effects. Does this make sense. What I have been trying to do with no success is: send a dry signal to USB-Computer and any Presets and EQ and Effects I have chosen could still be available on the T1 Master Output that I'm sending to the Compact.

You would have to hear the song to understand how fx makes it come alive.

Im off to bed. Really have enjoyed talking to and appreciate your help. Very kind. I'd like to send you a couple of songs I did with reaper and you could give me your opinion, if you like.

Jen
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Old 02-05-2014, 08:30 PM   #36
SonicAxiom
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I understand, it makes sense. It seems to be a monitoring/routing issue. Perhaps the interface is capable of monitoring your guitar incl. fx but reaper only gets the dry signal (and passes it back into the interface adding the desired fx while mixing).

Is the USB output of the T1 limited to sent only its master signal to reaper or can you choose other sources?
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Old 02-06-2014, 05:04 AM   #37
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Quote:
Originally Posted by Jenny View Post
Woke up early and as I was lying in bed I solved how to play with effects but only send a dry-signal to Reaper:

Now I just have to figure how to raise the spikes in Reaper while lowering the volume on the T1.
I don't use the Bose system you are using but on mine I have to do just the opposite. If my recording levels are too low in Reaper I have to raise them in my mixer so it feeds a hotter signal to Reaper. I don't quite understand how you could lower the output from the T1 and look for higher peaks in Reaper. Maybe I am misunderstanding...


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Old 02-06-2014, 05:09 AM   #38
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For some reason I have to set the volume levels on the T1 almost to maximum to get reasonable peaks in Reaper. This doesn't seem right to me. It seems more logical if the volume controls were set in the middle and I got good peaks. Then there would be more room to make adjustments.
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Old 02-06-2014, 05:25 AM   #39
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there's nothing wrong with raising the gain to get a decent peak level. It depends on the source's output (guitar). It's important to hit the converter inside the T1 with a high level so that the conversion makes use of every possible bit of its resolution.

Please tell me what max. peak level a raw track hits after the recording (with channel fader and reaper master fader at 0 dB). It should peak between -12 dB and -6 dB. You wouldn't need normalization then.

You should always gain stage levels so that outputs are set relatively high in level (before distorting), allowing you to turn down the following input gain (avoiding to add amp noise to the signal).
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Old 02-06-2014, 05:29 AM   #40
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Originally Posted by bozmillar View Post

The easiest thing you can do to get your levels up to a "normal" volume is to put a limiter on your master output track. That will let you boost the level of the whole mix and will prevent it from distorting. Of course, there's a bit more to it than just that if you don't want some unwanted side effects, but it will get you 85% of the way there and at least point you in the right direction.

There is a plugin that comes with reaper called LOSER/masterLimiter. Put that on your master track and adjust the Threshold (dB) slider. Your song will get louder and it won't distort.
Is this OK to do? I mean, I suppose, is it good practice that experienced engineers would do?
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