Aiff Kbps

[Julia Dodoo]

Id like to create a smart playlist in iTunes that collects all of my 24 bit recordings. What is the minimum bitrate (kbps) for 24 bit so I can set the parameter to include everything equal to or greater than that?

Thanks for the tip. Its confusing. I have an album with a bit rate of 1700 but its supposedly a 24 bit recording. I have a couple of spectrum analyzer programs, but they just say the bit rate without always specifying if its 16 or 24 bit. My ripped CD's are in AIFF, but most everything else is ALAC.

The lowest bitrate 24-bit recording in my current collection is 1016 kbps, and is a track in Angela Hewitt's 2008 recording of Bach's The Well-Tempered Clavier - purchased direct from Hyperion. My highest bitrate 16-bit recording comes in at 1233 kbps and it belongs to a track in the A State Of Trance Year Mix 2012 CD.

An MP3 file at 256 kbps is a "lossy" format. That means sound quality has already been lost, due to the compression process used to encode. You cannot get back the loss in sound quality by converting it to AIFF format.

You can convert to AIFF using iTunes but the quality is already gone for good. Converting at this stage will only result in a larger file. You will have to re-rip from the original CD. If you do that and wish to save space select ALAC Apple Lossless format.

If you burn a disc using iTunes with the setting to burn an Audio CD, iTunes converts songs (on the iTunes playlist you are burning) to AIFF. Any songs that are stored in a lossy format like MP3 will be AIFF on the disc, but sound quality will not be any better than the source MP3 file.

Although a conversion chart would be good one reason I would like to know is I am sure most of you are aware of Mixcloud, they say for music it needs to be at 320 kbps, but how does that compare to say, 4100 kHz @ 16 bit or 96 kHz @ 24 bit.

Uncompressed CD quality is more than sufficient for playback and anything else is really overkill at the expense of more space being used, IMO. If you really want to have the high quality recordings, you can still go CD quality but compress it a lossless format such as FLAC.

My digital copies are made from vinyl at 192,000 kHz @ 24 bit Wav uncompressed on a reasonable system and they sound no different to playing the record direct on my Hi-Fi which is fairy good but rather old now unfortunately.But I do scale them down (re-save) at 96,000kHz @ 24 bit for my media players, the players do play the 192,000 kHz file but the output of the players is only 96,000 kHz so no point, plus takes up a bit less memory but also they load quicker too.

This utility calculates the size of audio files (both uncompressed, PCM/IEEE FP audio, such as .WAV/ .W64/ .RF64, .AIFF/.AIF and also lossy compressed files such as MP3, WMA, AAC and OGG Vorbis), according to the recording duration and file settings you choose:

Enter the duration of your file in hours, minutes, seconds and milliseconds. Calculating the size of uncompressed files also requires the Sample Rate, Bit Depth and Channel information (but not the Bit Rate, which is automatically calculated). In addition to the duration, calculating the size of compressed files such as MP3 etc. requires only the Bit Rate information (in this case the Sample Rate, Bit Depth and Channel information is ignored). For compressed files encoded with CBR (Constant Bit Rate), the displayed file size should be as accurate as possible (notwithstanding variables such as header information etc- see below). For compressed files encoded with VBR (Variable Bit Rate), the displayed file size can be slightly less accurate because in this case the bit rate can vary depending on the programme material.

I have a 16GB sd card and would like to know how many hours of telephone quality audio, mono could be fitted into this. I read in an earlier post of yours, that when you made a blank recording lasting 74 minutes, it took up about 68mbs of memory. From that you concluded that there were 234 lots of 74 mins wav capacity on the sd card. Ie 290hours.

Hello, my question is the following: At the same sampling frequency, why is the size of an 8-bit WAV half that of a 16-bit WAV if 256 resolution positions are stored in 8 bits and in 16 bits do they keep 65536?

You just stopped my brain being fried. Working out 3 gigs this weekend with 14 tracks of differing rates, time etc. I either launched my calculator oot the widow or used this and, woo hoo, in seconds I had all answers for all 3 gigs. Pure magic. Thanks

Glad you found this site useful and that you found a solution. You are indeed correct, though your formula could be simplified for clarity. In fact, the simplest formulae for calculating either file size, duration or bit rate would be:

A sound engineer has chosen to record the background audio for an upcoming movie in stereo. To produce a high quality non-compressed digital audio file, the sound engineer will use a bit depth of 16 and a sample rate of 88kHz. What would be the approximate resulting file size for a 1.5 minute 37 second audio track?

Since you ask for my advice; I suggest that you always record to WAV files in future (which are uncompressed and non-lossy) and NOT mp3 (which are both compressed AND lossy). Mp3 is fine as a delivery format but (for several reasons too involved to go into here) far from ideal as a recording format. Personally, I would always record spoken material (music is a little different) as a standard WAV file at either 44.1kHz/16-bit or 48kHz/16-bit. You can then convert to mp3 (or whatever other format may be required) for delivery, keeping the original WAV files intact in case you might need to convert them to a different format later without any loss of audio quality.

1. Is this really an audio file? If so, 47 kbps is extremely low (even for a lossy compressed format) and would likely represent pretty poor audio quality. Reducing it further may even render it useless.

2. If you really must lower the bit rate, then you should convert the file again (but with the new settings) from the original, uncompressed audio file. Converting an already lossy compressed file again will reduce its quality even further.

I think the binary (2^10) to decimal (1000) converter is off. It just multiplies by 1.024, but that only works for KiB -> kB. For MiB -> MB it needs 1.024^2, for GiB -> GB 1.024^3, and so on, given that the binary prefix equivalent is 2^(10*n/3), where n is the power in 10^n.

Hello collin how do i calculate this

Video file has the following specifications

true colour mp4 video compression ratio at 4:1 and size of 9000MB and duration of 3 minutes ,

and audio compression ratio 2:1 at bit depth of 16bit, sampling at 96khz stereo

Audio Interchange File Format (AIFF) is an audio file format standard used for storing sound data for personal computers and other electronic audio devices. The format was developed by Apple Inc. in 1988 based on Electronic Arts' Interchange File Format (IFF, widely used on Amiga systems) and is most commonly used on Apple Macintosh computer systems.

The file extension for the standard AIFF format is .aiff or .aif. For the compressed variants it is supposed to be .aifc, but .aiff or .aif are accepted as well by audio applications supporting the format.

Because the AIFF architecture has no provision for alternative byte order, Apple used the existing AIFF-C compression architecture, and created a "pseudo-compressed" codec called sowt (twos spelled backwards). The only difference between a standard AIFF file and an AIFF-C/sowt file is the byte order; there is no compression involved at all.[5]

Apple uses this new little-endian AIFF type as its standard on macOS. When a file is imported to or exported from iTunes in "AIFF" format, it is actually AIFF-C/sowt that is being used. When audio from an audio CD is imported by dragging to the macOS Desktop, the resulting file is also an AIFF-C/sowt. In all cases, Apple refers to the files simply as "AIFF", and uses the .aiff extension.

For the vast majority of users this technical situation is completely unnoticeable and irrelevant. The sound quality of standard AIFF and AIFF-C/sowt are identical, and the data can be converted back and forth without loss. Users of older audio applications, however, may find that an AIFF-C/sowt file will not play, or will prompt the user to convert the format on opening, or will play as static.

Apple has also created another recent extension to the AIFF format in the form of Apple Loops[6] used by GarageBand and Logic Pro, which allows the inclusion of data for pitch and tempo shifting by an application in the more common variety, and MIDI-sequence data and references to GarageBand playback instruments in another variety.

AIFF files can store metadata in Name, Author, Comment, Annotation, and Copyright chunks. An ID3v2 tag chunk can also be embedded in AIFF files, as well as an Application Chunk with Extensible Metadata Platform (XMP) data in it.[8]

AIFF supports only uncompressed PCM data. AIFF-C also supports compressed audio formats, which can be specified in the "COMM" chunk. The compression type is "NONE" for PCM audio data. The compression type is accompanied by a printable name. Common compression types and names include, but are not limited to:

So, I have been presented to think about something I haven't in a long time. All prior projects were submitted to publication in WAV and in 16 bits. I don't know what the sample rates were, probably whatever was default. At some point, I read that 16 bit was the best to submit for mp3 distribution and I think that it didn't matter whether you submitted AIFF or WAV, I had been submitting all WAV.

But now, I realize that my entire project has been done under different settings, seemingly Logic Pro defaults: I checked and all the recording and mastering was done on files set to AIFF, 24 bit, 44.1 sample rate.

So except for maybe AIFF rather than WAV, you're all good.

PS. Logic Pro has or had a bug (haven't tested it recently) when importing MP3 files, where it hard truncates the contents to 16 bit. This means with Logic, specifically, it's better to use a 16 bit dithered source for MP3 conversion, but that's more of a quirk/bug than something you need to think about in general.

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