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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 reason ...
10
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-...
8
Download Audacity here. Also download the LAME MP3 library here
Install Audacity on your system. Install the LAME MP3 library.
Open Audacity
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 ...
6
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
5
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.
For example:
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 ...
5
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 ...
5
Nevermind, I found something. The command line tool sox seems to do the trick:
find . -name '*.wav' | xargs -I {} basename {} .wav | xargs -I {} sox -r 44100 {}.wav {}.edited.wav
find . -name '*.edited.wav' | xargs -I {} basename {} .edited.wav | xargs -I {} mv {}.edited.wav {}.wav
First line takes all of the WAV files in the current directory and saves a ...
4
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
calculation ...
4
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 ...
4
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 ...
4
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 ...
3
One way i do this, using Adobe Audition, is to "Open as" raw PCM data (one has to type an * in the name bar of the opening window to show all files), and then define the sample rate and mono/stereo information in the pop up box that comes up.
This has proved useful for opening files that were stopped while recording in programs like Ableton Live.
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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 ...
3
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 ...
3
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 ...
3
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~), ...
3
If this wikipedia page is to be believed, what you call mp3 and is actually a MPEG-1 audio layer 3 encoded audio file only supports three sample rates :
32 kHz
44.1 kHz
48 kHz
EDIT
As @PkP has mentioned, besides MPEG-1 audio layer 3, there are more recent versions of the encoding, which allow other sample rates :
MPEG2 later (1995) added 24 kHz, 22050 ...
3
OK you are getting confused between frames as in "SMPTE" frames and audio buffers. The frames you are referring to are actually buffers. They are not 'vision' frames. 25fps refers to vision frames.
44100 samples/second is correct.
1024 samples per 'buffer'. (Note these will be 'stereo' samples - possibly interleaved depending on your input format)
Strictly ...
3
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), ...
3
You will only get benefit from sampling to the next highest sampling rate you are able to mix at. For instance, if you have a 32kHz stream and a 48kHz stream, resample the 32kHz stream to 48kHz. You won't achieve any additional benefit from resampling to a multiple and additionally you won't find an audio device that will support a sampling rate that high.
...
3
It's an MP3 file. You can't deduce anything from that other than the fact that it's probably got a bit depth of 16 bits. Try the same exercise but using a WAV file. You will have better luck. MP3 is a lossy compression format. Lossy means that by compressing, you 'lose' data.
Also with a WAV file the bit-depth is contained in the header. Like it just ...
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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 ...
2
You need to think about your end goal and what platform your project is to be viewed/heard.
44.1 kHz is the exact sampling rate of a CD. Therefore, it's the go-to sampling rate when recording/editing/mixing music or anything else that is being created with the intent of being played back on a CD.
48 kHz was chosen as the standard digital audio sample ...
2
On Linux systems like Ubuntu, you can use ffmpeg:
ffmpeg -i input.mp3 -c:a libmp3lame -q:a 2 -ac 1 output.mp3
where:
-c:a libmp3lame: The audio codec to use
-q:a 2: The audio quality (bitrate), see LAME Bitrate Overview
-ac 1: One audio channel
2
Your understanding of lossy compression is close, but a little bit off. An MP3 (or any lossy compression) doesn't actually reduce the number of samples per second, but rather, it alters the values of those samples.
Lossy and lossless in terms of compression refers to the ability of the compression to reproduce the input to the compression algorithm exactly ...
2
Firstly it's nice to see someone making use of RS5000! Sadly, though, it works in the same way as most samplers in that it achieves the pitch changes simply by playing back the sample data at different speeds.
Within Reaper you could do the work using ReaPitch which would also allow you to do formant correction to compensate for the vocal sounding ...
2
I am no expert, but could your guitar have been tuned to Just temperament, instead of Equal temperament?
Correct me if I am wrong, but if one tunes the guitar by the ear, it does not necessarily sound completely in-tune from all chords? One does need to use tuning meters to check nice tuning against equal temperament instruments (like pianos or more ...
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There are several reasons for sampling with 96kHz: this allows for more natural results when doing non-linear processing (all kinds of distortion, modulation, pitch shifting) since that causes frequencies outside of the hearing range to have an effect into the hearing range.
It also makes digital IIR filters more comparable to their analog counterparts by ...
2
Your assumption that bitrate = bitdepth * channels * samplerate is true for raw PCM encoded audio. It is not true for bitrate reduction encodings, be it lossless (as flac) or lossy (as mp3).
The samplerate and bitdepth given to an mp3 encoder are used to set up appropriately the encoder, and will be used at decoding stage to output a decoded audio with the ...
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