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 ...


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 ...


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

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

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

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

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

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

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

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

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

There is no way to compare them because one is analog and one is digital. The way to calculate kbps is: bit depth * sample rate * channels Neither vinyl nor cassettes use bit depth or sample rates. The best you can try to compare is the similarity between bit depth of the CD and dynamic range of the cassette or vinyl


1

It depends on the application I guess. For example, a lot of speech processing (and transmission) nowadays is happening in the range that reaches up to 12 KHz or 16 KHz at maximum. Whether you intend to use it for speech or musical (or any other kind of audio) application you should make sure that aliasing will not occur (in considerable levels). This means ...


1

There doesn't seem to be an already existing tool for your purpose. (At least, I couldn't find one). You might have to build your own tool in your language of choice. Some basic building blocks could be : Export your audio as unsigned 8 bits 8 kHz raw. Before exporting, apply a low-pass filter with a cut-off frequency below 2 kHz with a steep attenuation. ...


1

Let's assume that what you call 5.1 Dolby is Dolby Digital (aka ac3 codec). That is what your receiver expects. 1 - Any s/pdif interface can output ac3 bitstream (it is part of the standard). 2 - As you want to output the ac3 bitstream, you shouldn't worry about the decoding in the interface. 3 - No 4 - I don't know You must in any case check that the ...


1

TL;DR: It appears the iPhone 6 supports 24-bit audio and 48 kHz playback. So I found a way to empirically test playback. This first screenshot shows a recording of playback over iPhone USB into Quicktime. You can see from the spectrogram that there is data in the 22-24k range, which means that playback is at least 48 kHz. This next screenshot shows ...


1

Unlike some earlier models of iOS devices, the DACs and ADCs in the iPhone 6s and 7 support native sample rates of 48000. You can experimentally test for this by writing an iOS app that configures the RemoteIO Audio Unit for a 48k sample rate, then synthesize sine waves at frequencies a bit above 22.050kHz and observing the audio output on an oscilloscope. ...


1

Bit depth is specific to certain file types, most notably Wav. Because the standard procedure of a music application is to play a compressed version of a file such as Mp3 or Aiff it throws bit depth out the window and instead outputs signal at a kbps rate instead such as 192 or 320kbps. As far as DAC goes I don't think it will improve the quality of an ...


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 ...


1

I love compressor for this! You can even create Drag and drop areas to Speed up the process of file conversion!


1

The batch processing feature of GoldWave will do the trick.


1

If the FLAC files are not meant to be used for further sound design, and are recorded at reasonable levels, you don't lose anything important. The dynamic range of 16-bit sound is very high - roughly 100 dB (how dither affects this makes the answer not quite as straightforward as you'd thing it is). If you are truly concerned about the audiblity of bit ...


1

As long as you dither, you'll be fine; it can overcome the dynamic range limitations of if you had recorded 16-bit. Scroll down this page for more info on bit depth/dithering (I suggest reading the whole page): http://people.xiph.org/~xiphmont/demo/neil-young.html


1

Good question, Zach. No sample rate nor any bit depth can make any recording indistinguishable from natural sound. This is due to both the simple fact that you will loose some vital information in any recording, and also due to imperfections in any replay system. Did you ever do a stereo mic recording in a room where you where present yourself, and after ...


1

16bit 44,1KHz is alaready supposed to be like natural audio. My thought are that the resolution isn't so important compared to the choice of microphone, microphone placement, preamplifier, way of working and mixing your record and speakers in which you are listening. Working in 96kHz 24 bits surely allows to have more headroom and to process sound easily ...


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