Posted in the other thread for you, I'll say it here too:

I just received A [[url=http://[/URL]="http://rodemic.com/"]RODE[/]="http://rodemic.com/"]RODE[/] Podcaster and its better than I expected. I'm actually blown away on how cool USB mics are, at least this one. I underestimated them. This plugs directly into my Mac ( haven't tried it on my studio PC yet but I'm sure its just as impressive) and you start recording. So simple. The Podcaster is definitely well made, sounds great. Its ideal for what its designed for, Podcasting. It has a headphone insert and a volume control on it too. If you have the budget, 2 thumbs up on this mic.

If you only have $50 for an interface, I have know idea what to recommend, that is a skinny budget.

Whatever you want to do is based upon your budget, bottom line. However, your desire for a mixer with a computer audio interface is really the most prudent choice you can make. Everybody starts with a basic idea of what they want to do. Before long and in no time short, you're going to want to do something that exceeds your current technical capabilities. So you are already a foreword thinking person. That's more like myself coming from a live broadcast environment. Plus I usually have twice what I usually need when I unusually need it. Again though, it's all based upon what your startup budget is. None of us started with huge startup budgets when we were first getting into the business. But even when people told me they wanted things most definitely in Mono, monaural, I always engineered everything so when asked after the fact if it could be converted to stereo, I was always ready.

This is also where microphones such as SM58's make a lot more sense than the AT condenser anything. And that's because the SM58 is more like the Radio Preferred SM 7's and all you'll need is an extra foam pop filter actually offered for that particular microphone through the manufacturer. And then when the band comes in, you'll be better prepared with the right microphones not the condenser microphone thingies which really aren't important for podcasting anyhow. Much less trying to figure out what to do about all of the distortion from the loud rock 'n roll instruments you'll get with entry level condenser thingies. SM58 is the de rigueur standard.

In that little Beringer USB mixer I would highly approve of for your applications. Not finding it online? Perhaps you should check the www for GC Pro, Sweetwater, Chuck Levin's Washington Music Center, plenty of others. Why not check the Beringer site? But really, all you need is an analog mixer and a USB line level audio interface such as the EDIROL UA 1EX, by Roland for around $85 US. But what we are talking about here is a budget just shy of $500 US. But you'll be ready for anything that comes your way when it starts to come your way. This is the basic difference between live broadcast sound & produced in the studio, sound. It's sort of like run and gun, PA work that you don't have to worry about powerful amplifiers & ear killing feedback. But you are trying to produce some kind of quality programming on a first take, without microphone checks. That's living on the edge in the exciting land of live engineering.

You can do it!
Mx. Remy Ann David

Thank you for your advice!

I will look into the SM58 and the other products you mentioned.

spiritgear, post: 379583 wrote: Hello,

I'm doing some research on microphones for vocals and podcasting.

What would you recommend as a good starter kit for around ~100.
I will also consider things 'over budget' as well.

Thanks for reading.

Hi SpiritGear,

Have you looked into the Samson C01U? It's about 75 bucks and you can usually get them at Best Buy. If you can afford to go up to about $150, you could move into the world of 24-bit conversion with a USB mic with the MXL Studio 24 USB Microphone . This will allow you to record with much less hissy noise since the noise floor is much lower in 24-bit recording.

Hope that helps!

Ken

I might add that 24-bit recording while offering a 140 DB processing dynamic range, there isn't any analog audio electronics that exceeds much beyond 100 DB. So the previous poster is not exactly 100% correct in their assumption of lower noise floor. Your noise floor will only be as good as the analog front end which will be 100 DB (approximately). So if you are recording 20 DB too low, your analog front end dynamic range will only offer 80 DB before you get to total noise floor. It doesn't matter if 24-bit provides 140 DB of peak level to noise floor if your electronics are only delivering 80 DB. And this confuses a lot of people due to simple misunderstandings. 24-bit is also not exactly higher in resolution than 16-bit. However, 88.2 kHz/96 kHz sample rate is twice the resolution of 44.1 kHz/48 kHz sample rates. So if you want improved resolution and you know how to properly set your levels, higher resolution will be achieved at 16-bit, 88.2 kHz/96 kHz than 24 bit at 44.1 kHz/48 kHz. So do want improved resolution or improved crazy processing dynamic range? Most folks don't know what they're doing when they add various effects and as a result, levels and up too high or too low. So a lot of this 24-bit stuff simply applies to sloppy inexperienced lack of technique & knowledge. It's more of a safety net than anything else. Some of my newer digital multitrack machines natively record at 24-bit. So frequently, I'll just let it do that. But at times, I'll reduce it to 16-bit recording since 16-bit dynamic range is already 96 DB and the computer can handle it much faster. But you have to understand how to properly adjust, set & manipulate your gains, trims & level. We had to know how to do that back in the analog days and for broadcasting. We also had to know how to record & mix without compressors and/or limiters on every track. That was called " riding the gain " with your fingers on the faders not on your crotch or leaning on the console bolster. This all requires time and practice to become professional at this. You don't become a doctor by taking a one day Internet course. And you don't necessarily become a psychiatrist when you have felt suicidal. That's not exactly a DIY project even though suicide is.

So don't commit audio suicide because we love your lousy engineering, we really do.
Mx. Remy Ann David

Sorry if I'm taking this farther off topic, but I wanted to make a clarification about audio resolution as it pertains to sample rate and bit depth (not bit rate; that's a measure of data used for audio compression algorithms like mp3). I find it makes a lot of sense if you think of it like a digital photo. If you take a picture at low resolution, that means that you are trying to recreate the image using very few colored dots, so your picture can end up looking like an old Nintendo game where you can make out the person but he just looks like a pile of blocks. Increasing the resolution means you squeeze more and more color dots onto the same space, so the person looks more and more real until the dots are so small that you don't notice them any more, and the picture looks like real life. Now, if you take that high resolution picture and blow it up to poster size, the same number of dots are taking up a much larger area, meaning the dots are bigger too and you can start to see the blockyness again.

SAMPLE RATE is measured in Hz (number per second) and describes how many times each second the machine measures the +/- voltage (the distance your waveform is above or below the center line) of your audio signal. The common values are 44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz and 192kHz. Obviously, the more often the machine measures the waveform, the clearer our picture of the sound will be. However, more measurements means more work for the computer. Truth be told, the resolution of all of these is good enough that the differences will only be noticeable on very good monitoring systems. A standard CD uses a sample rate of 44.1kHz, so many people don't see much use in recording a project at a better resolution when they just have to degrade it at the end to put it on a CD.

BIT DEPTH is measured in bits (number of 1's and 0's in a string that makes up a single piece of digital information) and describes how accurately the machine measures the +/- voltage of your audio signal. In a 16 bit measurement, the computer measures the momentary +/- value of the waveform on a scale of 1 to 65536. In 24 bit values, the measurement is made on a scale of 1 to 16777216. (for the math geeks, that's 2^16 and 2^24) That means that the measurements taken in 24bit audio are 256 times as precise as in 16bit audio. So again, we would get a clearer picture of the waveform with "more smaller dots". However, standard CDs are encoded in 16bit audio, and it's plenty good.

Remember when we blew up that digital picture too much and it started to look grainy? The same thing can happen if you take a 16bit track you have recorded and increase the volume too much. People tend to record in digital audio at much lower volumes because it sounds absolutely terrible if the volume exceeds the limits of the machine's measurement area, which wasn't quite the case with tape. Then, people want their CDs to be extraordinarily loud. All this means you are likely to increase the volume quite a lot, and with compression you might sometimes be increasing the volume to 10 or 20 times as loud as it originally was recorded. That means your sound only has the resolution of 1/10th or 1/20th of the original, and can begin to sound quite "low rez". Recording at 24bit will allow you to crank the volume to stupid levels and never get resolution lower than a CD would have.

As for noise floor, digital noise is created by inaccuracies in the measurements the machine makes. This noise is usually only a couple of numbers above or below the real measurement, and accounts for VERY little noise. The hiss from the mic, preamp, EM noise and everything else in the analog realm is where the real noise will come from. That will be the same percentage of your audio signal whether you record to 16bit or 24bit.

Yes as [[url=http://[/URL]="http://recording.or…"]bicasaur[/]="http://recording.or…"]bicasaur[/] indicated, that is a concern for many when it comes to resolution. Conversely, most of my stuff is done live and so, most of my processing is still in the analog realm when it goes to digital capture. But here is what some people also don't understand. When working with digital multitrack machines/computers we don't play back our tracks all at the same level. So some tracks are mixed lower, some higher so even with 16-bit capture, the end result utilizing all of your tracks at different levels creates a higher bit resolution like recording when you do a multitrack mix down.

Yup, I'm one of those people that generally will stay at 16 bit 44.1 kHz since that is the way it's going to be released. Conversely, when we had analog tape, we wouldn't run our 2 inch 24 track machines at 1 7/8 even though that's how it would be released on cassettes. Heck, we wouldn't even roll them at 7.5 IPS (although I did that once on my MM 1200 when I had only one, LOL. It worked but it wasn't impressive), since vinyl actually sounded better than cassettes. And most everybody rolled at 15 IPS when 30 IPS was not financially feasible as it is for most local band recordings. Most hits were rolled at 15 IPS instead of the higher resolution 30 IPS even when they had decent budgets. Because it was perfectly adequate. Recording engineers you see, are not audiophiles generally speaking, we are engineers. We want to advance the state of the recording arts but we are not necessarily out for the ultimate decibel. One is a matter of fixation the other is a matter of fix nation and even though those two things sound very similar, they are worlds apart.

This was not my original planet.
Mx. Remy Ann David