uncompressed 16-bit pcm audio Archives

Uncompressed audio formats

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Uncompressed audio formats

Below, we list the various types of uncompressed linear audio formats. The first three are the ones that have become the main ones supported by most of today’s professional audio applications and are: audio swap file format :

-AIF, AIFF The audio exchange file format is of Apple (1985) origin and allows the storage of mono or multichannel samples of 8 or 16 bits and various sampling rates. Being a format designed to be portable, it can also be easily converted (as we will see, Microsoft RIFF is similar) and is therefore often used. Extensions RIFF WAVE :

-WAV Developed by Microsoft and IBM in 1992, the RIFF WAVE (whose full name is Resource Sharing File Format Waveform Audio Format) follows the specifications of the more general rich information file format. It was introduced starting with Windows 3.1 and quickly became the most popular format for PC applications. WAV files support various types of sampling: they are single or multi-channel, 8 or 16 bits at different sampling rates with various encoding systems, even if the most common are PCM and ADPCM. Extensions TRANSMISSION WAVE FORMAT (BWF) :

-WAV It is an extension of the popular WAVE format and was created by the European Broadcasting Union (EBU) in 1997 and updated in 2001 and 2003. The purpose of this format is to add to the normal .WAV metadata format to facilitate the exchange of data between different platforms and different audio applications, allowing the files thus encoded to identify themselves autonomously and allow synchronization with other recordings. Since the only difference from “normal” WAVE files is in the extended information written in the file header area, the two formats are absolutely compatible and a particular player is not required for playback. In order to overcome the limitation on the maximum size of the WAVE file (2 Gb) in 2006, it was specified as an extension of the BWF l! RF-64, SD2 Sound Designer II Format SD2f file extension The Sound Designer 2 format is proprietary to Digidesign and is the evolution of the original Sound Designer 1 format, unlike which it structures the data so that all audio samples are stored in the file’s data fork and all parameters in place . resource holder. This is extremely convenient in files where the data fork can become hundreds of Mb and more, because it is possible to modify, add, cut the parameters of the audio file without having to modify the sample data, a feature that saves a lot of time, especially at a time when computers and hard drives were vastly smaller in capacity and slower in data processing and writing. It can also be monophonic or multichannel (interleaved), Sampling depth of up to 24 bits at different sampling frequencies of up to 192 KHz, as in the case of Pro Tools HD. It should be noted that in multitrack applications such as Pro Tools, the standard professional market software created for a DSP based card system, each track is recorded in a separate mono file, even for stereo or multi-channel tracks. In the case of a stereo track, the “split stereo” file is used, which consists of two monophonic files with the same name but with the two suffixes .L and .R (or in previous versions (L) and (R) ), that remain physically separated but that the application treats as a single stereophonic file, operating all the editing operations on both files simultaneously in perfect phase coherence; In native applications (ie fully dependent on CPU processing power), all those in practice outside of Pro Tools software (not Pro Tools LE or Pro Tools M-Powered who are also “native” even if they depend of specific Digidesign software or M-Audio production hardware), instead, the use of “interleaved” files is generalized, that is, stereo or multichannel files in which all the channels are stored in a single file. The stereo track is represented by a single file containing the two channels, which are “written” in blocks (first a number n of blocks from the left channel followed by an equal number n of blocks from the right channel, and so on). This type of file, created to dominate applications, used in multitrack applications still generates a bit of confusion, because although it is more practical (in theory) to be used within the native application, it needs to become a “split” file when imported into Pro Tools.

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24 bits vs. 16 bits are there audible differences?

The experiment

The author of the experiment plans to determine if there are audible differences between 24 and 16 bit audio trying to use an objectivist approach.

For this, it selects 3 audio fragments at 96/24, 2 of them edited by the renowned 2L label and a third belonging to the well-known Goldberg Variations performed by Kimiko Ishizaka. Particularly difficult passages are selected for their musical content or dynamic characteristics.

16-bit versions are generated from the 3 original samples, maintaining the sampling rate of 96 khz. Simple dithering is used in the process. In addition, special measures are taken (addition of noise at -140dB, offsets, use of flac compression with more or less efficiency) to avoid that the characteristics of the files themselves could reveal their nature.

The resulting audio files maintain a DR of between 12 and 14, with no evidence of clipping or compression of the dynamics.

The 6 files are arranged so that both versions of each of them are sequentially reproduced, but it is not known a priori if the 16 or 24 bit version sounds first.

In total 140 people participate, who must:

– Determine when one or the other version of each fragment sounds.
– Indicate the degree of certainty with which they have responded.
– Explain if they have used an ABX type comparison tool or similar.
– Provide a series of data, let’s say demographic, including whether they are musicians, have professional experience or not in audio editing and if they regularly analyze equipment.
– Describe the audio equipment used in the tests and indicate their cost.

The test runs for 2 months. Participants are encouraged to take their time.

Participants

Most of the test participants:

– They are men.
– Between 41 and 50 years old.
– They are not musicians (although 22% say they are).
– They have no experience in music production (although 24% do claim to have it).
– They use equipment whose cost is between $ 1,000 and $ 3,000.

The results

– The hit rate is practically 50%. Or what is the same, pure fluke.
– Participants who claimed to be more sure of their answers were not more correct than the rest.
– The group of “musicians” has failed more than the average.
– The group of “audio engineers” has made it a tad better than the average, but without exceeding 55% of hits in any fragment.
– The group of audio equipment analysts has also done worse than the average, but the size of the sub-sample is considered low (8 participants).
– Participants with expensive equipment (44 people with equipment over $ 6,000) did no better than average.
– The use of headphones has not been an advantage.

The conclusions

I translate more or less literally:

In a real study carried out over 2 months in which 140 people have participated and in which high-quality sound samples obtained from digital audio at 96/24 have been used, no evidence has been found that Digital audio with a 24-bit depth could be appreciably differentiated from its corresponding 16-bit versions obtained through Adobe Audition 3 using a basic algorithm (1/2 bit flat triangular dither).