What is the possible purpose of these large audio formats?

I learn more about the amount of data is actually necessary for high fidelity audio. For as far as I can see, the data files more high quality fit to be worked with and exported to Adobe Audition (CS6) are WAV - PCM and BWF - PCM files as described below:

Format: Wave PCM (*.wav, *.bwf)

Type of sample: 192000 Hz, 32-bit

Bitrate: 24576 Kbps

Format settings: wave non-compressed, Point 64-bit floating (IEEE)

Markers and other metadata included

Sizes of large file estimated, such as 6 min = 1 GB for audio stereo

Multichannel audio available

44.1 kHz to 192 Kbps (usually from 16 - bit) is usually the maximum level of audio fidelity. I prefer a minimum of 256 Kbps and 24-bit (32-bit if possible). What is the purpose of these complex audio file formats and?

Steve.KR wrote:

44.1 kHz to 192 Kbps (usually from 16 - bit) is usually the maximum level of audio fidelity. I prefer a minimum of 256 Kbps and 24-bit (32-bit if possible). What is the purpose of these large and complex audio file formats?

Forget speeds - they are irrelevant. You don't need to taste everything faster that 44.1 k because this rate will help you understand all the information present in the signal, up to about 22 kHz. Here's several kHz more than any adult hearing extends. Your ears have a dynamic range which is potentially greater than 16-bit signal, but not for a go. If you start to 0dB (using this as the threshold of hearing), then the subject + 85dB acoustic reflex mechanism kicks in and stops gradually the sensitivity, but allows you to hear sounds with higher levels without damage. But it desensitizes you at lower levels when this happens, then the offer 96dB a linear 16-bit system is more than enough - as has been widely demonstrated by research.

The point of greater depths of bits has nothing to do with the hearing, but everything to do with the treatment. If you use the 32-bit floating-point system, you can manipulate the levels up and down and even save the results between the two without loss. It will not work with an entire system according to - you lose resolution to the point that save you if attenuate you your signal. With the floating-point system, you can whack it up and down without loss, because the changes are stored in the bits scaled, and data (24 bits) are unchanged.

The history of all these extended formats is a bit creepy and based on a lot of misinformation propagated by people who should have better, but didn't. It was at this stage a technically sound reason, why some records of highest example sounded "better" - which is doubtful-wall filters of bricks that were used to creating important phase errors in the range of audible frequencies. But now that oversampling is the norm, 44.1 k systems work quite well, as they should do, and there is absolutely no justification whatever the massively high sampling rate unless you save bats. There is no significant musical harmonic content that exceeds about 14kHz, and if you include a lot of harmonic content at higher levels, a large number of listeners is quite disturbing.

Tags: Audition

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