13

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 ...


6

Audio CDs are encoded with 16-bit values. Higher bitrates are generally used for editing, not for playback. You can write 24-bit WAV files to an optical disc, of course, but it will not be a standards complying audio CD.


5

Even floating point numbers can clip and degrade. If I'm mixing, I prefer 64bit floats if available, but I use mostly analogue gear now .. so the mixing happens outside of the digital domain. I do have some software I wrote to automagically remix music.. the jazz-o-tron 1000. Internally it uses 80bit IEEE samples (YES, 80.. its not a typo).. but I only use ...


5

Lower bit depths are not easily available simply because they are not looked for commercially, not because of some intrinsic technical difficulty. Audio interfaces or other capturing devices have their ADCs (Analog do Digital converters) based on integrated chips optimized for the features that are most requested commercially. In principle it would not be ...


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 ...


3

Technically it does reduce the quality, yes. It should not add any perceivable noise. Will you notice the difference? Try it and see. I don't expect you will. 16bit is CD quality, ie pretty good considering how most people listen to music these days. Some reading if you want to understand more: http://www.presonus.com/news/articles/sample-rate-and-bit-...


3

In Adobe Audition there's a built-in batch processor: Edit > Batch Process


3

Using SoX (Windows, Linux, Mac): cd /where/your/wav/files/are/ mkdir converted for i in *.wav; do sox -S "$i" -b 32 converted/"$i"; done


3

First, I'd like to take a moment to point out that that article only argues that 24bit sampling is useless for end user playback. I'm also not 100% convinced by the article personally, but that's neither here nor there. As for the dynamic range, yes, the 24 bits expands the dynamic range in the same way that it does for a camera. For both audio and ...


3

You are right! The quality of the audio will be decreased if the volume lowered by software means. As you have correctly assumed, reducing volume in software is actually similar to reducing the bit depth. Generally, every 6 dB of attenuation is equivalent to reducing the bit depth by one. Brief explanation: Max volume level of your sound card output, ...


3

With Floating point all recordings are saved usually using fractional values only between -1 and +1 so like when you say -10 DB that will be related to 1.00 as zero Db. If you want to know the numerical value of -10 DB, it is like Log-1(-10 / 20), or by a calculator the answer is + or - 0.31622776. NOW, since there are no A/D converters with greater ...


3

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

The article you read has either been misinterpreted or was monstrously misleading. It is possible that what they were referring to was the fact that the frequency response required from an analogue medium was only enough to record 100kB/s of digital information. That does not mean that the recording quality of the original recording was equivalent to a ...


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 ...


2

Having a session in a higher bit rate will allow a larger dynamic range, allowing you to get away with more gain staging with less chance of clipping in your channel. Higher samples rates in your session allow you to time stretch your sounds while retaining more high end frequencies (as the ones that are in audible then roll down into the audible range). You ...


2

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

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. http://store.rodemic.com/products/ixy Not for iPhone 5 yet, however.


2

All Digital Audio Articles are Flawed. I've been doing a bit of reading on 16bit vs 24bit audio and came across this article that claims 24bit/192kHz playback is basically pointless (but, that's not what I'm hear to discuss). There is endless online debates on virtually all aspects of digital audio, and articles as the one you have cited are common ...


2

The only error in your question is this as far as I can tell: - What I found surprising was the fact that moving to a 24bit recording process doesn't result in a higher resolution per sample, but rather a wider dynamic range. In fact 24 bits provides more resolution in the same analogue voltage range AND this means more dynamic range. Yes, a 24bit ...


2

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.


2

See this and scroll down to "Converting and Using Floating Point Samples". One good reason to convert integer samples to floating point samples in [-1,1] is so that you can mix bit-depths of integer samples easily. For instance, if you have a 24-bit file with a sample that is half of positive full scale, and an 8-bit file with a sample that is half of ...


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 ...


2

Fi: 44.1khz/16bit (1411kbps) Master: 96khz/24bit (4608kbps) Pitchfork Article Note that only ~30,000 tracks are MASTER quality right now You can tell that Master is available by the M symbol and the playback bar in bottom right will change from HI-FI to MASTER


2

Taking the 2 MSBs should be just what you're after. The way I think of it is that 1 bit (1 or 0) simply tells it the signal is above or below the middle value - and since we're talking audio, that would be the zero line. So, 2 bits would do the same, but subdivide the range into 4 'zones'. 3 bits into 4 'zones' and generally, x bits = 2^x 'zones'.


2

Partial answer : Many Analog to Digital converters are actually implemented using a one bit ADC at a high sampling frequency (several MHz). The bitstream is then decimated to more common formats like 24 bits / 48 kHz). This has the advantage that you can have a cheap analog low pass filter with a cut-off frequency at half the high sampling frequency (given ...


2

SoX (Sound Exchange), the self-described Swiss Army knife of sound processing programs is super-useful for these kind tasks. soxi is the specific tool to list audio file information and on Linux (or MacOS) the command to recursively list all the information for all the audio files would be: find . -name "*" -exec soxi {} 2> /dev/null \; (the 2&...


1

In my opinion, the audible difference between 16 and 24 bit audio is minimal at best. I'll do some bullet points because i'm tired: 24 bit does enable the file to carry more data, and therefore have a lower noise floor, but the noise floor of 16 bit audio is approximately the same (or lower) as that of most amps and preamps. It's definitely lower than the ...


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