PITCH OF VOICE PROBLEM (SOLVED)

Audacity Help Forum Windows
1

Hi

It has been noticed by other people in my studio that when I record my vocals, the pitch of my voice is higher recorded than in reality. I was told this was because the Sample Rate needed to be changed to 44100 Hz and the Sample Format needed to be changed to 16 bit so I changed it to them settings. However as I was looking through all the settings, I came across a LOT more options and I just wondered do any of these options also need to be changed to help exterminate the problem?? I have listed the options below and in your comments also include why they need to be changed

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  • Gain - 20 (db)
  • Frequency Gain - 0 (db/dec)
  • Range - 80 db
    -Minimum frequency - 0 Hz
    -Maximum frequency - 8000 Hz
  • Latency Correction - -130 milliseconds
  • Sound Activation Level - -36 db
  • Sample rate converter/dither - various settings
  • Meter/Waveform db range - -36 db shallow range for high amplitude editing (there are a few settings for this including one that says - -96db PCM range of 16 bit samples - should I use this one because my current input setting is at a sample rate of 44100 Hz at 16 Bit)

Also do any of these effects (currently enabled) need to be disabled??

  • LADSPA
  • Nyquist
  • VAMP
  • VST

I would really appreciate a fast, accurate reply and for your help to resolve this situation.

Thanks in advance

Moderator note: No need to shout fella. It’s considered bad netiquette on most forums - it only startles and annoys the forum elves and doesn’t get your query answered any quicker or better. I removed the shouting for you.

2

How far off is it?

Does this only happen when you play-back in Audacity (which has a speed control), or does it also happen with Windows Media Player, etc.?

Are you recording and playing-back on different hardware? (i.e. Such as recording with a USB mic and playing-back through your soundcard?)

Recording at a different sample rate will NOT fix your problem. You simply need to record and play-back at the same sample rate, which normally happens automatically.

Speed (and pitch) is related to sample rate (44.1kHz, etc.). As long as you record and play-back at the same sample rate, the temp and pitch will not change. It’s pretty hard to accidently record and play back at different sample rates, because the sample rate is stored in the file header and it’s set automatically when you play back.

But, the clocks (oscillators) in consumer soundcards can sometimes be off by a few percent. You won’t notice any problem if you record & play-back on with the same soundcard, but if you switch soundcards you may notice a difference.

P.S.
If you record at 44.1khz and play back at 48kHz, that’s a speed/pitch increase off about 9 percent. In musiical terms (according to my quick-and-dirty research) that’s about 1.5 semitones.

3

Are you suffering from “Microphone Effect?” Do you have a different Announcer/Presenter voice than you do when you’re saying “hello” to somebody. Many people do. There’s a comedy routine going around about “cellphone voice.” People unconsciously pitch their voice up or down depending on what they’re doing.

It’s pretty unusual for Audacity or any other program to change voice pitch. In order to get the 44.1/48 thing, you would have to force the player against its will to play at the wrong sample rate. If you straight open up a 48 KHz sample (video sound) in Audacity, it will play at 48.

We have had broken soundcards and microphones.

Koz

4

Hi

Thanks for the replies.

What do you mean - how far off is it??

And it happens in all audio players, it is something to do with the recording software I use (which is audacity) not the playback software. And I record on a USB mic - on
audacity there is options on the playing back - “Windows Directsound” “MME” and then “Primary Sound Driver” “Speakers Realtek High Def” - which ones shall I select??

And shall I change my sample rate back then(I changed it from 32 bit to 16 bit) and it is currently on 44100 Hz - shall I change this??
And how do I know if im playing back at the same sample rate I am recording at?? Is there an option or something I can see within audacity that displays this??

Which soundcard shall I use (I listed the options above) and how do I know if this is being used to record AND playback??

PS
Can I avoid doing this - how do I record AND playback at the same sample rate in audacity??

And also do any of the settings I listed in my original message need to be changed??

Thanks a lot for your help

5

Does it sound just a fraction too high, or about an octave too high, or like a chipmunk breathing helium?
Also, does it sound too fast?

6

And it happens in all audio players…

…And I record on a USB mic

Bingo! …Different hardware clocks for recording & playback!

Assuming you are singing and the pitch is slightly-off, and assuming by “all players” you are talking about all software, not multple computers, or your iPod.

Your USB mic has a built-in soundcard-chip and it’s own clock. If that clock does not match the clock in your playback hardware, the pitch & timing will be off. No clock is perfect, but usually it’s “close enough”. The worst case is if the clock in your USB mic is a little slow, and the clock in your soundcard is a little fast.

For example, if have a good USB mic, and the clock is “exactly” 44,100 Hz, but the clock in your soundcard is actually 44,200Hz (when set to 44,100), playback will be faster and at a higher pitch.

If you use an analog mic and you record and play back on the same soundcard, the record and playback speed/pitch will match and you won’t notice a problem 'till you play back on a different system.

You don’t have a software problem, although there is a very-slight chance you have a driver problem. Usually, it’s the “cheap” soundchip circuit built onto your motherboard. But, it could also be a “cheap” USB mic.

In order to track-down the problem, you’ll have to try some different hardware. For example, if the file plays-back at the wrong pitch on an iPod, your USB mic is most-likely the problem. If it’s correct on your iPod but wrong on your computer, the problem is your soundcard. You could also try burning a CD and checking the pitch on a CD player.

I changed it from 32 bit to 16 bit)

Your problem is not related to bit-depth. Bit depth is related to the precision of the amplitude (vertical height) of a sample from a waveform. The sample rate (kHz) is related to the time (horizontal location) of a sample. If the samples are moved-around in-time, that mucks-up pitch and speed. This page shows how a waveform is sampled, and how the analog waveform is re-constructed when you “connect the dots”. (Each dot is a sample.)

7

I would say that it sounds anything from a fraction to half an octave too high. Also the pitch is unaffected.

Also is there any way of changing this “clock” problem in my soundcard?? I’m pretty sure it’s not the USB mic since its an expensive and reliable make, its more likely to be the soundcard.

Can this driver problem be fixed in anyway considering your stating the software I am using (audacity) is not the route of this problem and no settings need to be changed??

And could you also reply back with which soundcard I should be using to record and playback please - “Windows Directsound” “MME” and then “Primary Sound Driver” “Speakers Realtek High Def” - which ones shall I select??

And thank you for that link however on that page it quotes “Higher sampling rates allow a digital recording to accurately record higher frequencies of sound.” wouldn’t this mean that the 32 bit sample would record more higher frequencies than the 16 bit which would affect the pitch??

Thanks

8

Also is there any way of changing this “clock” problem in my soundcard?? I’m pretty sure it’s not the USB mic since its an expensive and reliable make, its more likely to be the soundcard.

I would agree.

Can this driver problem be fixed in anyway considering your stating the software I am using (audacity) is not the route of this problem and no settings need to be changed??

I really think this is a hardware problem. There is a crystal oscillator (like in a digital watch) or some kind of less expensive “resonator”. Or if this is a soundchip on the motherboard, the clock could be “derived” from the CPU clock. If the CPU clock is not an exact multiple of 44.1kHz or 48kHz, it’s hard to make a frequency divider in hardware. But it might be considered good enough for the average computer user playing MP3s or watching YouTube.

You don’t need a super high-end soundcard. I’d probably try a $20 USB soundcard. (I have a StarTech USB soundcard that I use for “troubleshooting”, but I’ve never checked the pitch.) For about $100 - $200 USD you can get a nice USB audio interface with XLR mic inputs (which you don’t need) and line & headphone outputs.

And could you also reply back with which soundcard I should be using to record and playback please - “Windows Directsound” “MME” and then “Primary Sound Driver” “Speakers Realtek High Def” - which ones shall I select??

Sorry, I don’t know enough to answer that one, but it should not affect pitch.

And thank you for that link however on that page it quotes “Higher sampling rates allow a digital recording to accurately record higher frequencies of sound.” wouldn’t this mean that the 32 bit sample would record more higher frequencies than the 16 bit which would affect the pitch??

The number of bits is NOT the sample rate.

The sample rate is the kHz (or Hz = samples-per-second). The sample rate is your TIME resolution. In order to “connect the dots” and make an analog “wave”, you need at least one sample for the positive half of the wave, and one sample for the negative half. At, 44.100Hz, your cannot represent audio above 22,050Hz. That’s high enough. It’s higher than you can hear.

The bit depth (number of bits per sample) is your AMPLITUDE resolution. It relates to loudness, but one sample doesn’t tell you anything about loudness because it depends on where the sample “lands” on the waveform. One sample doesn’t tell you anything about frequency/pitch either… One sample is just one instant time.

If you remember geometry, you need two values to represent a point (a dot or sample) in 2-dimensions. The sample rate is related to how accurately you can get the X (horizontal or time) location, and the bit-depth is related to how accurately you can get the Y (vertical or amplitude) position.

You don’t need to know this… I always have to look it up… But, with 16-bits you can “count” from -32,768 to +32,767 (when the binary numbers are converted to decimal for us humans to understand). If you have a “maximized” 0dB waveform, the peaks will be at (or very near) those numbers. And, you will also have sample-values everywhere in-between depending on where the sample “lands” on the waveform.

At 24-bits, the sound isn’t louder (the waveform isn’t any higher) but you have more numbers in-between. (There are 256 times as many values.) It’s basically like measuring in inches instead of feet, or millimeters instead of centimeters.

There are no 32-bit analog-to-digital converters or digital-to-analog converters so there’s no point in recording in 32-bit resolution. Or, if you your USB mic is 16-bits, there is no point in recording in 24-bits. However, there ARE advantages to using 32-bit floating-point when you are doing digital signal processing. Audacity (like most audio editors) works in 32-bit floating-point internally.

16-bits is usually plenty. You cannot hear a 1-bit (1 count) change in a 16-bit signal. A 1dB change (which you can barely hear) is about 10%. At lower volume levels, a 1-bit change is a bigger percentage change, but if you get to the point where it’s a 10% change, the sound is so quiet you can’t hear anything anyway.

And, 16-bits gives you more than 90dB of dynamic range (the difference between the loudest and quietest sound). If you want to get an idea of how far “down” -90dB is, try using the Amplify effect to reduce the peaks to -70 or -80dB. (You should probably configure Audacity to turn-off dithering because that adds a tiny noise.)

9

would say that it sounds anything from a fraction to half an octave too high. Also the pitch is unaffected.

I would say the pitch is a fraction to half an octave too high.

You never answered the speed question. Find a clock not your computer and smack a fork in front of the microphone and then smack it again exactly one minute later. Does the sound file play with exactly a minute between the ting sounds? Again, Do Not use the computer to tell you the time.

Which microphone is it down to model numbers?

No you can’t change the clocks. The “clock” is a tiny physical device that sits in a system and generates ticks like a very tiny, very fast and very accurate electronic pendulum (Illustration). They’re all over the place and normally, they don’t cause any problems because they’re really close in time to each other. But if you have one lose its marbles as compared to the others, then you can get performance shifts or other odd effects or errors.

I would be going to great effort to plug the microphone into a different computer and try it. Right now you have multiple variables and there’s no good way to nail down what’s wrong. There’s no chance of adjusting your way out of this.

Koz
ClockChips.jpg

10

Ok I think I’m going to get a good USB soundcard, this will probably resolve this situation.

And ok thanks, if anyone knows which option I should pick please reply, I currently have it on Windows DirectSound and Speaker Realtek High Def.

And ok thanks I understand now, I think I’m going to leave it on 44100 Hz and 16 bit considering this is what Mic records in.

Thanks for all your help.

Ok so when I do this fork test, I should look at a stopwatch and not the computer time??

And my microphone is the editor keys sl300.

Thanks

11

And also if anyone knows names and models of really good soundcards that will help improve recording a lot, please reply with the name, model,etc even a link!!

I dont really care about the price at the moment - so any really good type will do - it needs to improve my recording excessively though

Thanks

12

The computer has little clock chips governing its behavior, too. We don’t know what’s wrong. You can use the sweep second hand on a wall clock or the Time Check or stopwatch on your cellphone. I’m not a stopwatch fan because then you have to build your reaction time into the measurement. Just do it when the kitchen clock second hand comes around.

If your vocal pitch is far enough off for someone else to notice it, then the time change may be significant. Go for two minutes or longer for better accuracy.

Koz

13

I just did the test, it’s definitely not the timing. The timing was perfect when I did the test.

Thanks

14

Cue the puzzled look. The pitch is off but not the system timing…??

Koz

15

yes, the timing is perfect.

16

I know we’re all going to laugh when we figure out what this is.

Koz

17

You posted the same question again in a new topic so I deleted it. I think no-one has answered this because it’s of no relevance to your complaint about pitch being recorded incorrectly.

If you are not sure what these preferences do, please look in the Manual: http://manual.audacityteam.org/o/man/preferences.html. From that page, each preferences pane has its own link which you can click to look at detailed information about that pane.

No. Audacity does not apply effects in real-time to incoming audio. Effects can be useful for applying finishing touches to your voice recording after you’ve made the recording, for example add some more reverberation, or apply compression (make the difference between loud and soft less). You don’t want to disable effects.


Gale

18

You asked that in another post so I deleted that as well. It would be confusing out of context.

MME is the most compatible host for a basic motherboard sound device, which is what the Realtek Speakers or the Realtek Mic/Line/stereo mix or other inputs are - your built-in audio.

If you are going to buy a USB sound card then Windows DirectSound may be beneficial. This may be better for the USB microphone too. What version of Windows are you using? Vista? XP?

If it is not stating the obvious, you cannot put a USB microphone in a USB sound card. They are still separate devices, and may or may not give you a clock difference. However, any difference is not likely to be sufficient that you can hear a pitch difference between your voice recorded on the USB mic and the playback of the recording on the USB sound card.

If you are going to record one track, play that first track then record another track while listening to the first track, then you only want to use one device for playback and recording.


Gale

19

Hi

I have read through the preference manual and there is still some things that are not explained.
1.What is the difference between Fast Sinc Interpolation (Real Time Conversion) and High Quality Sinc Interpolation (High Quality Conversion)??
2. Also what happens if I place the triangle dither on Real Time Conversion??
3. My sample rate is currently on 44100 Hz and my maximum frequency is on 8000Hz. Should I raise my Maximum frequency to 22050Hz?? (This is half of the sample rate)
4. My Meter/Waveform db range is currently on the (-36 db shallow range for high amplitude editing) setting, Should I change it to the (-96db PCM range of 16 bit samples)setting considering I am using 16 bit??
5.Should I leave the gain at 20db and the range at 80db when recording at 44100Hz 16 bit?? Also what would happen if I change this setting??

I am currently using the Windows 7 Home Premium and does the Windows DirectSound have less latency?? Will using Windows DirectSound be more beneficial to my USB mic in this case??

If you are going to record one track, play that first track then record another track while listening to the first track, then you only want to use one device for playback and recording.

What do you mean by this?? Can you please elaborate on this.

Thanks

20

Also getting a USB soundcard will be of no benefit to me since not being able to put the USB microphone into the USB soundcard indicates that the soundcard will not be able to help my recording but only playback which is useless to me.