Hi All,
I have been playing with the audio channels in Realband. I have a backing track on one channel which uses audio and I have my microphone connected to a second channel.

I am using a Novation USB audio interface
Using the Novation ASIO driver
The audio channel is set to mono.


There is very good quality output from the backing channel

The microphone channel meters indicate that I am just below clipping during the record

When I playback the recorded audio is quite low

The microphone is a Behringer C1 and provides good quality input on Sonar and Ableton.

Any ideas please?

Regards

John Lundrigan

When you play back the VOCAL is low or both are low?

I suggest you mix the prerecorded stuff way back, make sure your master volume (which for the life of me I can't understand why that matters) is up, and record your vocals again. That may or may not help, but that's all I got. Also, as far as vocals go, it is better to record lower and gain change to boot rather than record too loud and flat top the channel. You can't negate the clipping no matter how many db you reduce teh gain.

I once found that when I added another rhythm guitar track on an empty track, the recorded level of the new track I added was somehow affected by where the master volume fader was. Logic says it should not matter, that the master volume is just OUTPUT, but it did. T test it I pushed that master volume up almost 100% and generated the same Real Track again on another blank and it generated louder.

One of the PG techies please do explain that and tell me if that matters or if I was just off my meds that day. As I said, I can't for the life of me think why the output level would have anything to do with track generation level, but in that test, it did.

Thanks Eddie. I moved the rest of the mix back and this seems to have helped considerably. I also had to boost the volume of the record track (I also had to do this using the Thru channel to record audio in BIAB.

I have to be honest and say that I have found getting good quality recorded audio from the the microphone quite a challenge, even in other software. I am going to ask a general question to others on the forum as to what their setups are and how they configured them for ASIO

Regards

John

Is the phantom power turned on?

Yes, thank you. I get the same results with both types of microphone, but I'm getting closer to it with some of the other suggestions.

Regards

John

I think you are WAY too deep into the "which driver should I use" debate. The rule I always went with was "If the sound card hardware supports ASIO, use it. If not, don't." ASIO has lower latency possible than MME. That's pretty much the difference. That has become such a conversational buzzword that it gets tossed around by people who really don't know what it means. Just like when people talk about "drivers" as a fix for everything. A software driver is the bridge between your operating system and any piece of hardware. Using your video card as an example, I have actually seen people on these forums tell users that if their PG software isn't working that they need to update their video card driver. Well, did the driver work okay YESTERDAY? Were there any video issues YESTERDAY? The video card gremlins don't magically come through the ether overnight and mess with your software drivers. So I said that to say this. You are going to have people who use ASIO tell you that nothing on the PLANET is better than ASIO. You will then have others tell you that MME works just fine. The difference is latency, and the hardware either does or does not support ASIO. Think about this in the same way you'd think of a guy who has never had a Ford in his life telling you how Chevy is so much better than Ford. Because it works FOR HIM.

Find your space, find your sound, do what is best for you. Try ASIO, then change to MME and reload the song and ask yourself if you notice any difference at all when you play along. There are users here who barely play instruments that are making songs that are close to 100% BIAB/RB, and for them, while not introducing live instruments, latency is a non-issue. So depending on how much you actually play in manually, a latency issue may or may not affect you. Not to insult you but I don't know your level of expertise in this arena. Latency is the time in milliseconds that elapses between the time you play a note, correctly stated "enter a note on event", and the time you actually hear that note. That lower the latency, obviously, the better. You can't have a 1/2 second delay from key press to sound, right? Processing power of your computer eventually works its way into the equation, but as always, the faster you can afford, the better. Also the more RAM you can afford the better, because it cuts down on the operating system having to swap files from RAM to hard drive, which is what your swap file is. I just paid what to me was a fortune for a new computer for the studio. I did not need a 8th computer in my house (3 are not even plugged into power), but I did it anyway. I'm a nerd and actually look for reasons to buy more stuff!

I agree with Eddie.

If your interface/soundcard supports ASIO, use it. If not, use whatever else works.

You won't hear any sonic/audible difference using various drivers. The difference is in their latency and compatibility with the hardware. ASIO tends to run the better interfaces smoothly.

Eddie, I think you put a lot of work into your reply and I really appreciate.

My expertise is very high on the actual technicalities of digital hardware and software so I am quite at home with latency issues and ASIO vs MME. I'm not so good on some of the issues that affect sound reproduction through the microphone. Somebody else pinpointed one item that I think was at the heart of the problem and that was how close I was to the mic. Howver, even that did not give me a really good volume on playback, but boosting the volume of the tarck sorted that. This requirement for boost was also something I came across in BIAB when recording to the audio track. I'm not sure why this should be such a problem in BIAB and RB because it does not arise in SONAR or Ableton. Anyway, I can get it to work now and I'm grateful to members for that.

I am now retired from all that stuff and hence my attention being turned to something I always wanted to do. It is also where I am more at the basic level. I am amazed at how helpful members such as yourself are prepared to put time into lengthy replies.


You also mentioned about a lot of people who will simply be using BIAB and RB to generate songs without external input from instruments or voice. Well my initial intention was to do just that but when I saw the possibilities of playing keyboard, guitar, harmonica, sing (all very sub standard you understand) then I couldn't help but push my luck.

I have a whole lot to learn and find the whole thing very exciting.

Regards

John

I just went around and around with low vocal recordings in this thread:

http://www.pgmusic.com/forums/ubbthreads.php?ubb=showflat&Number=373992#Post373992

The thread starts out with me asking how to globally turn down the default midi synth but I soon found out the real question was about the vocals being too low.

Bob

There are many ways and methods contained in RB to help bring your audio up. As others have noted, it's not likely your hardware, mic, driver or latency. It is your audio signal chain where to focus.

The bad news is, if you think there are a lot of opinions about ASIO vs MME, drivers and latency, audio signal chain opinions dwarf that discussion. The good news, it doesn't matter. Regardless whether you record 16 bit, 24 bit or floating bit; whether you aim to have -6 db level or -12 db level, the successful goal is to capture a clean recording that is above the noise floor and below clipping.

Capturing a clear recording. Memorize that line. There are many pathways, methods and techniques. All of the recording bit rates, db levels, distance you place yourself from the mic are all important things to know but many times they become more important to the outcome use of your recording than the recording itself. For instance, if you will be adding 30 or so VST's, effects,and other high end processing, using 24 bit for the additional headroom it provides is important.

RB is more than adequate to meet your needs to obtain a clear audio recording. Memorize that line too. You don't need to use another DAW for recording unless the workflow is more comfortable to you. RB has stock tools to bring up the level of low audio and the method to use depends somewhat on the level the audio is at in relation to noise.

Some suggestions, to consider - there are no hard, fast rules.

Before beginning to mix, balance your audio tracks. Lower louder tracks and bring up the quieter tracks. Between -12 -- -3 db for an overall level. I try to use -6db on my tracks. Again, the amount of post processing will have an impact on what the correct level of your tracks will be.

RB offers gain change, compression, limiting as options to raise a low audio recording. RB has a good gate to eliminate noise (hiss or low level hum) if necessary.

If recording clean audio continues to be an obstacle to having fun and having useable tracks to work with, consider recording to another media other than your computer.

Companies such as zoom and Tascam and others offer inexpensive digital recorders that require very little setup to record tracks. These are easily found used on Ebay or Craigslist at very affordable prices. Most record a minimum of two tracks but there are affordable models recording as many as 8 simultaneous tracks. These are portable so they can be used anywhere. They save the recordings to removable media such as SD cards, so the tracks are easily imported into RB or another DAW to process your project. Be sure if you go that route to choose a model that records at least 44.1/16 wav. Most, if not all do.

I have a Tascam DP-24 that records up to 8 tracks simultaneously and is up and recording in seconds. It is as easy to set up and record audio as the cassette recorders we are all familiar with. I have owned several models since the 1980's and have never suffered a crash or major glitch.

My workflow is to create a backing track in BIAB, export a stereo master and import the master into the DP-24 and add my additional live tracks. I import these additional tracks into RB or another DAW for final processing.

There are many other options, this is just one way. Others may present suggestions that suit you better. Just choose the way that best fits your workflow and use it. Hope this helps.

Charlie

Hello John - I see there are quite a few great responses to your question already.

I had a similar problem when I first started using a USB recording input to my PC. I finally figured out my PC USB interface for the microphone wasn't set correctly for the mixer USB output that I was using.

On a Win 7 PC, right click on the speaker Icon at the bottom right of the task bar, and select 'Recording Devices' find the USB input you are using and right click that and select 'Properties'. On the microphone properties window, select 'Levels' and move the slider to the right. This boosts the USB input from your microphone or mixer circuit to the PC. The same setting are available through Control Panel of course. I've found that I have to change this level when I use a different USB mixer.

Just as an example I use Shure SM48 and a Phonic UM-99 (cheap dynamic mics)as my microphones and run these through a Behringer X1204 USB mixer into a Windows 7 PC using Realband as my one and only DAW. My only problem with mic levels now are when I get too close and start clipping.

I hope you find whatever the issue is with your setup. I agree with Charlie that you don't need anything other than RealBand as your DAW to make clear, clean audio recordings.

FWIW, if RB uses the same interface, thus the same convertors, it records as good a quality as any other DAW; the sound is converted to digital (numbers) before it ever gets to RB.

That's one aspect of selecting drivers; they affect convertors used. Example; I could use the line in on my laptop (yes it is has a Line In) but the onboard sound card would then doing to do the conversion to digital (yuck). Whether I use ASIO or MME, wouldn't matter.

It's true that MME will in the long run give the same sound as ASIO (IF the same device driver is used for both). However sometimes switching to MME changes the driver selection (for instance I have a decent USB interface that only works with ASIO, so if I switch to MME my convertors have to change) ..

Just another reason why driver selection is important, and can affect the quality of a recording.

As Del pointed out, sometimes the Windows mixer can come into play and add another layer of complexity. That was a good point. Especially when multiple options may be in play.

Thanks all for help in this area. This has got me up and running with the audio. I didn't seem to be close enough to the mic. Mind you, when I do get close enough I still have to be careful that I now don't cause my audio to go into the red.

Regards

John

After my experience in that other thread I linked here, I think that's all it is. Few people here on these forums can be considered professional vocalists with years of vocal coach training and experience recording in studios. The trick is to control your voice so you're NOT moving in and out from the mic. A little bit yes but not too much.

Singing is pure emotion or should be. Certain words are sung softly and others can be belted out. It's that dynamic range that causes the problem because the loud parts can't clip so you must turn down the input gain to avoid that. But, the soft parts are now recorded so low as to be almost inaudible. A trained singer is able to get the full emotion out of the loud parts without actually belting it out thereby lowering those peaks and to also sing a bit louder on the softer parts but still keep the emotion.

That's the difference between pro and amateur. Singing the quiet parts a bit louder and the louder parts a bit softer results is you being able to turn up the input gain. Or like Rharv said, we can use the Input Monitoring function and insert a compressor onto the vocal track while we're recording it. My issue with that is most vocalists including my sister like to have some verb in their monitor mix and that will also get recorded onto the track and I don't want that. I need full control of the verb later during mixdown.

Obviously this is where us putting on the recording engineer hat have to work with what we got. You can't try to give singing lessons in the middle of a recording session. Once I realized what what was happening based on the discussion in my other thread it's really not that big of a deal. My signal chain is very quiet, it's no problem to put the compressor on the vocal track after it's been recorded and to also lower all the other tracks and then start mixing.

Bob

Thank you bob, you explain it very well and much appreciated. You've have also got me in a nutshell as an amateur singer. But you have certainly given me some ideas.

Regards

John