The 44.1kHz it's the sampling frequency, i.e. the frequency at wich the encoder samples the audio data. It has nothing to do to the frequency of the audio data.
You can generate a 1000Hz sine wave, sample it at 44kHz and play it back: what you hear is still the 1000Hz sound. Take a look here.
Actually there is a reason to why CD audio data is sampled at ...
The sample rate of audio and video are two different things.
Video's Frames Per Second (fps)
In video, a frame (sample) rate of 24fps is required to prevent flickering. The common frame rates (25 and 30) has to do with the fact that early televisions used the mains frequency for the purpose of syncing (50Hz in the UK, 60HZ in the States). The primary ...
Download Audacity here. Also download the LAME MP3 library here
Install Audacity on your system. Install the LAME MP3 library.
Click on File > Open... and select the mp3 file in question
Click on Track > Stereo track to Mono
Click on File > Export. Choose "MP3 Files" as Format on the dropdown menu and click save.
Note 1: If you want to save ...
The most direct, if somewhat "hands-dirty", way of doing this is to manipulate the header with a hex editor. There are two entries in the RIFF header that need to be modified: the sample rate itself, which is 80 BB 00 00 in 32-bit little-endian,
and the adjacent byte rate, 192k or 00 EE 02 00 for a 48k 16-bit stereo file. (96k for mono-16bit, 288k for 24bit-...
Well, it seems like you have to forget everything you know about sampling frequency and frequencies in general.
The frequency in terms of audio is the number of times the speaker membrane moves in and out per second. What makes it vibrate is that the power that is sent through the audio cable makes electromagnetism that pushes and pulls to the magnet inside ...
is ABSOLUTELY necessary to record at 48 kHz that's not for film where the final audio will be bounced to 44.1 format anyway?
We know that at 44.1kHz we can accurately record and playback the frequencies that live in the human hearing threshold, so oversampling might seem an overkill. Most of the time this is the case, but some scenarios can benefit from ...
sample rate convert on import... soundminer or Pro Tools will ask when importing as the system can't work with mixed sample rates. 96 to 48 is fine, it's half data, like 88.2 to 44.1.
record sfx at 24/96 for future proofing library.
There are any number of things that could be contributing to it, but the most likely culprit is the frame rate...24 is a freakin' minefield! The biggest issue with 24fps, is that there are multiple formats: true 24, 24P, 24PsF, and 23.976 (sometimes abbreviated as 23.98). It's important to note that these are all slightly different...they are not ...
You simply re-sample them from 48 kHz to 44.1 kHz.
I believe in Sound Forge you go to the Process menu and select re-sample.
Or you can do it by using one of the many free audio editing software such as these:
Don't apply pitch-shifting or time warp, only plain and simple re-sampling.
You won't ...
With 96kHz sampling, the acoustic spectrum that can be "recorded" without error (nyquist criteria etc..) is from 0Hz to about 44kHz. The spectral range that a human can hear is the text-book "20Hz to 20kHz" so, one should be able to say that scientifically 96kHz is pretty good. In fact 44kHz (CDs etc.) should be able to cope with 20kHz as well so no problem ...
As a possible side bar to the excellent answer provided by JCPedroza above,
If human ears are the only listeners (i.e., signals between 20 and 20,000 Hz) then it is never* absolutely necessary to record at 48kHz.
[H]e used an SM57 whose frequency range is 40 Hz to 15 kHz so it
wouldn't really make a difference because in the oversampling
Very easy. Just use SoX:
sox -r 44.1k infile.wav outfile.wav
The -r 44.1k overrides the header information in infile.wav and will not perform any effects on the samples themselves.
More info: sox.sf.net
To get the best audio quality in your final lossy-eccoded audio file output at the smallest possible file size, you have to do 3 steps:
sample rate convert from 96kHz to 44.1kHz, because the lossy formats do not support 96kHz
dither the 24-bit file to 16-bits because the lossy formats do not support 24-bits, and if you don’t dither, all you are doing is ...
The standard for sound-to-picture tends to be 48kHz.
The only real advantage I know of 44.1kHz over 48kHz is that it produces a smaller file size. However, when sound is put to picture, the file size is far more influenced by the video than the audio. I'm not much of an expert on exactly why 48kHz is favoured over 44.1 for video (it could be because 48kHz ...
Be nice if you can post a sample or way we can hear this - can I run the Mathematica code easily?. 2000hz is really low for smooth audio, are you sure it's not just extra frequencies created by aliasing through the playback rate. In fact, I bet it's that.
edit. I managed to recreate that in Max/msp using an object which reduces the sample rate (degrade~), ...
The point of over sampling is not about hearing frequencies above 20k. The point of over sampling is to make it easier for the filters to filter everything out above the nyquist frequency. When you sample at lower sampling rates such as 44.1 or 48 the quality of the filter matters more. So if you record with an amazing ad converter with a really expensive ...
It seem to me that your code does not any interpolation with further addition of samples between each pair of existing ones (in that case the size of the file would double, but pitch and speed should remain the same). More likely it "reinterprets" the original file, changing its header and telling the player to read the existing samples 96000 times per ...
A sample rate is the rate at which samples are taking from the source sound. It says nothing about how much information is stored in those samples. Whether an audio codec is considered lossy or lossless is dependant on how much information is carried over from the original recorded medium usually (for the sake of argument) based on the 1,412 kbit/s bitrate ...
After some more searching, I have found that ffmpeg is able to do this by converting the WAV to raw PCM format, and then converting back to WAV with the correct sample rate in the header.
ffmpeg -i input.wav -f s16le -acodec pcm_s16le output.pcm
to convert input.wav to raw PCM data. Then
ffmpeg -f s16le -ar 44.1k -ac 1 -i output.pcm ...
I think the way you currently have the pipeline set-up is the right way to do it. Conversion should be the last block in the chain for sure as this will ensure that any processing artifacts are minimised during down-mix and resampling.
Performing conversion at the start may provide some small CPU benefits, but has the possibility of introducing artifacts ...
frequencies above what microphone allows
Microphones have a frequency response curve. This is a random example:
The exact shape of the diagram is different for each microphone, but they all have one thing in common: the response will be as close to flat as possible in the range we're usually interested in (20 Hz- 20 kHz for full-range audio microphones), ...
For pitch shifting down you need more than a 0 - ∼22.05kHz bandlimited signal to try to maintain some highs in the downpitched version.
For digital recording you need more than 44.1kHz total sampling rate for 1. Making all gear-related conversion and input/output stage signal processing so that audible problems aren't induced to the audible range or the ...
I'm 99.9% sure it's 48kHz, 16-bit Mono (or Stereo if you have a plug-in mic of some sort which supports Stereo). I use the FiRe app which is one of the most comprehensive and professional-geared recording apps, and that's the max it supports. On older iPhones like the 3G I believe it was only 44.1kHz.
To extrapolate upon Shaun's solid answer, have you checked for pulldown? A skew of that much within that duration time has me guessing theres either an 0.10$ pullup/pulldown which needs to be corrected for. Hit CRLT+2 in PT and check the bottom section for Pullup/Pulldown and see if any of the NTSC presets (+/- 0.10%) seem to fix the problem, at least ...
Well, rode has created the solution to this problem.
The iXY iPhone/iPad microphone, with its own a/d converter can record at 96kHz/24bit, bypassing iOS's 48kHz/16bit limit.
Great move from Rode.
Not for iPhone 5 yet, however.
The point is to have more resolution than the typical final output so that there is less rounding error. 96khz is chosen because it is exactly twice 48khz which is the standard audio sampling rate for video. This way, you can cut the audio from 96khz to 48khz by cutting the number of samples in half, so there aren't aliasing problems. Similarly, 24 bit ...
On Linux systems like Ubuntu, you can use ffmpeg:
ffmpeg -i input.mp3 -c:a libmp3lame -q:a 2 -ac 1 output.mp3
-c:a libmp3lame: The audio codec to use
-q:a 2: The audio quality (bitrate), see LAME Bitrate Overview
-ac 1: One audio channel