A great list of High Definition sites can be found here

It is our hope that High Definition audio will someday be all you can purchase when you go to buy music and thank goodness for that.  Whether you listen to vinyl or CD you are not enjoying the openness and resolution available on the original master tape.  In fact, most of us have never heard a master tape or a master digital recording.  Vinyl lovers have to tolerate the innacuaracies of their cartridges  and phono preamplifier s. Digital audio listeners have had to tolerate truncated resolutions and bit rates when the master tapes are forced into the low sample and bit rates required by CD’s.

Not until the advent of PS Audio’s PerfectWave Transport could we  really enjoy all the benefits of listening to bit-for-bit perfect copies of the original master tapes of these recordings.  But not everyone can own a PerfectWave Transport and PerfectWave DAC so how do “the rest of us” get a taste for High Definition audio?  Through a computer.  No, it’s not the best source and never will be.  But it’s an excellent way to get started and hear the difference of what you’ve been missing.

What’s not mentioned in this video is the software requirements for playing the various formats that High Definition audio comes in, such as FLAC , AIFF , AAC and WAV .


FLAC stands for Free Lossless Audio Codec, an audio format similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player just like you would an MP3 file.

FLAC is the preferred method of storing music in uncompromised form on your hard drive to playback later on your music server, PerfectWave DAC (through the Bridge) or eventually burn to a DVD and playback on a PerfectWave Transport.

FLAC stands out as the fastest and most widely supported lossless audio codec, and the only one that is non-proprietary, is unencumbered by patents, has an open-source reference implementation, has a well documented format and API, and has several other independent implementations.


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 co-developed by Apple Computer 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 audio data in a standard AIFF file is a lossless uncompressed big-endian pulse-code modulation (PCM). There is also a compressed variant of AIFF known as AIFF-C or AIFC, with various defined compression codecs.

Standard AIFF is a leading format (along with SDII and WAV) used by professional-level audio and video applications, and unlike the better-known lossy MP3 format, it is non-compressed (which aids rapid streaming of multiple audio files from disk to the application), and lossless. Like any non-compressed, lossless format, it uses much more disk space than MP3—about 10MB for one minute of stereo audio at a sample rate of 44.1k and a bit depth of 16 bits. In addition to audio data, AIFF can include loop point data and the musical note of a sample, for use by hardware samplers and musical applications.

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.

With the development of the Mac OS X operating system, Apple quietly created a new type of AIFF which is, in effect, an alternative little-endian byte order format.

Because the AIFF architecture has no provision for alternate byte order, Apple used the existing AIFF-C compression architecture, and created a “pseudo-compressed” codec called sowt. 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.

Apple uses this new little-endian AIFF type as its standard on Mac OS X. 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 disc is imported by dragging to the Mac OS X 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.

What meaning sowt may have as an acronym or abbreviation does not appear to be documented, but it is probably the reverse of “twos”, the big-endian designation for twos-complement format.

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.

All traditional AIFF and AIFF-C files continue to work normally on Mac OS X (including on the new Intel-based hardware), and many third-party audio applications as well as hardware continue to use the standard AIFF big-endian byte order.

Note: As of Mac OS X version 10.4.9, the system will sometimes incorrectly displays the AIFC icon for files with the .aif extension, whether or not the actual file format is AIFF or AIFF-C. This can be verified by opening the files in hex editor and checking the FORM chunk’s form type. This can sometimes happen when exporting files from QuickTime, and frequently happens when sending and receiving files between Windows and Mac computers or extracting files from an archive


Advanced Audio Coding (AAC) is a standardized, lossy compression and encoding scheme for digital audio. Designed to be the successor of the MP3 format, AAC generally achieves better sound quality than MP3 at many bit rates.

AAC has been standardized by ISO and IEC, as part of the MPEG-2 & MPEG-4 specifications. The MPEG-2 standard contains several audio coding methods, including the MP3 coding scheme. AAC is able to include 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 16 low frequency effects (LFE, limited to 120 Hz) channels, up to 16 “coupling” or dialog channels, and up to 16 data streams. The quality for stereo is satisfactory to modest requirements at 96 kbit/s in joint stereo mode, however hi-fi transparency demands data rates of at least 128kbit/s (VBR). The MPEG-2 Audio tests showed that AAC meets the requirements referred to as “transparent” for the ITU at 128 kbit/s for stereo, and 320kbit/s for 5.1 audio.

AAC’s best known use is as the default audio format of Apple’s iPhone, iPod, iTunes, and the format used for all iTunes Store audio.

AAC is also the standard audio format for Sony’s PlayStation 3 and is supported by Sony’s Playstation Portable, latest generation of Sony Walkman, Walkman Phones from Sony Ericsson, Nseries Phones from Nokia, Nintendo’s Wii (with the Photo Channel 1.1 update installed for Wii consoles purchased before late 2007), the Nintendo DSi, and the MPEG-4 video standard.

High-Efficiency AAC is part of digital radio standards like DAB+ and Digital Radio Mondiale.


AAC was developed with the cooperation and contributions of companies including Fraunhofer IIS, AT&T Bell Laboratories, Dolby, Sony Corporation and Nokia, and was officially declared an international standard by the Moving Pictures Experts Group in April 1997. MPEG-2 AAC-LC profile consists of a base format very much like AT&T’s PAC coding format[1],[2][3] with the addition of TNS,[4] the Dolby Kaiser Window described below, a nonuniform quantizer, and a reworking of the bitstream format to handle up to 16 stereo, 16 mono, 16 LFE, and 16 commentary channels in one bitstream. The Main profile adds a set of recursive predictors that are calculated on each tap of the filterbank. The SSR uses a 4-band PQMF filterbank, with four shorter filterbanks following, in order to allow for scalable sampling rates.


It is specified both as Part 7 of the MPEG-2 standard, and Part 3 of the MPEG-4 standard. As such, it can be referred to as MPEG-2 Part 7 and MPEG-4 Part 3 depending on its implementation, however it is most often referred to as MPEG-4 AAC, or AAC for short.

AAC was first specified in the standard MPEG-2 Part 7 (known formally as ISO/IEC 13818-7:1997) in 1997 as a new “part” (distinct from ISO/IEC 13818-3) in the MPEG-2 family of international standards.

It was updated in MPEG-4 Part 3 (known formally as ISO/IEC 14496-3:1999) in 1999. The reference software is specified in MPEG-4 Part 4 and the conformance bit-streams are specified in MPEG-4 Part 5. A notable addition in this version of the standard is Perceptual Noise Substitution (PNS).

HE-AAC (AAC with SBR) was first standardized in ISO/IEC 14496-3:2001/Amd.1. HE-AAC v2 (AAC with Parametric Stereo) was first specified in ISO/IEC 14496-3:2001/Amd.4.[5]

The current version of the AAC standard is ISO/IEC 14496-3:2005 (with 14496-3:2005/Amd.2. for HE-AAC v2[6])

AAC+ v2 is also standardized by ETSI (European Telecommunications Standards Institute) as TS 102005.[5]

The MPEG-4 standard also contains other ways of compressing sound. These are low bit-rate and generally used for speech.

AAC’s improvements over MP3

AAC was designed to improve on the MP3 format (which was specified in MPEG-1 and MPEG-2) by the ISO/IEC in 11172-3 and 13818-3.

Advanced Audio Coding is designed to be the successor of the MP3 format and demonstrates greater sound quality and transparency than MP3 files coded at the same bit rate[citation needed].

Improvements include:

* More sample frequencies (from 8 kHz to 96 kHz) than MP3 (16 kHz to 48 kHz)
* Up to 48 channels (MP3 supports up to two channels in MPEG-1 mode and up to 5.1 channels in MPEG-2 mode)
* Arbitrary bit-rates and variable frame length. Standardized constant bit rate with bit reservoir.
* Higher efficiency and simpler filterbank (rather than MP3’s hybrid coding, AAC uses a pure MDCT)
* Higher coding efficiency for stationary signals (AAC uses a blocksize of 1024 samples, allowing more efficient coding than MP3’s 576 sample blocks)
* Higher coding accuracy for transient signals (AAC uses a blocksize of 128 samples, allowing more accurate coding than MP3’s 192 sample blocks)
* Can use Kaiser-Bessel derived window function to eliminate spectral leakage at the expense of widening the main lobe
* Much better handling of audio frequencies above 16 kHz
* More flexible joint stereo (different methods can be used in different frequency ranges)
* Adds additional modules (tools) to increase compression efficiency: TNS, Backwards Prediction, PNS etc… These modules can be combined to constitute different encoding profiles.

Overall, the AAC format allows developers more flexibility to design codecs than MP3 does, and corrects many of the design choices made in the original MPEG-1 audio specification. This increased flexibility often leads to more concurrent encoding strategies and, as a result, to more efficient compression. However, in terms of whether AAC is better than MP3, the advantages of AAC are not entirely decisive, and the MP3 specification, although antiquated, has proven surprisingly robust in spite of considerable flaws. AAC and HE-AAC are better than MP3 at low bit rates (typically less than 128 kilobits per second)[citation needed]. This is especially true at very low bit rates where the superior stereo coding, pure MDCT, and more optimal transform window sizes leave MP3 unable to compete. However, as bit rate increases, the efficiency of an audio format becomes less important relative to the efficiency of the encoder’s implementation, and the intrinsic advantage AAC holds over MP3 no longer dominates audio quality.


WAV (or WAVE), short for Waveform Audio Format, is a Microsoft and IBM audio file format standard for storing an audio bitstream on PCs and is now used as the open format of choice for High Definition audio from a growing number of companies such as Reference Recordings and Chesky.

It is an application of the RIFF bitstream format method for storing data in “chunks”, and thus also close to the IFF and the AIFF format used on Amiga and Macintosh computers, respectively. It is the main format used on Windows systems for raw and typically uncompressed audio. The usual bitstream encoding is the Pulse Code Modulation (PCM) format.

Both WAVs and AIFFs are compatible with Windows and Macintosh operating systems. The format takes into account some differences of the Intel CPU such as little-endian byte order. The RIFF format acts as a “wrapper” for various audio compression codecs.

Though a WAV file can hold compressed audio, the most common WAV format contains uncompressed audio in the linear pulse code modulation (LPCM) format. The standard audio file format for CDs, for example, is LPCM-encoded, containing two channels of 44,100 samples per second, 16 bits per sample. Since LPCM uses an uncompressed, lossless storage method, which keeps all the samples of an audio track, professional users or audio experts may use the WAV format for maximum audio quality. WAV audio can also be edited and manipulated with relative ease using software. The WAV format supports compressed audio, using, on Windows, the Audio Compression Manager. Any ACM codec can be used to compress a WAV file. The UI for Audio Compression Manager is accessible by default through Sound Recorder.

Beginning with Windows 2000, a WAVE_FORMAT_EXTENSIBLE header was defined which specifies multiple audio channel data along with speaker positions, eliminates ambiguity regarding sample types and container sizes in the standard WAV format and supports defining custom extensions to the format chunk.


Uncompressed WAV files are quite large in size, so, as file sharing over the Internet has become popular, the WAV format has declined in popularity. However, it is still a commonly used, relatively “pure”, i.e. lossless, file type, suitable for retaining “first generation” archived files of high quality, or use on a system where high fidelity sound is required and disk space is not restricted.

More frequently, the smaller file sizes of compressed but lossy formats such as MP3, ATRAC, AAC, (Ogg)Vorbis and WMA are used to store and transfer audio. Their small file sizes allow faster Internet transmission, as well as lower consumption of space on memory media. However, lossy formats trade off smaller file size against loss of audio quality, as all such compression algorithms compromise available signal detail. There are also more efficient lossless codecs available, such as FLAC, Shorten, Monkey’s Audio, ATRAC Advanced Lossless Apple Lossless, WMA Lossless, TTA, and WavPack, but none of these is yet a ubiquitous standard for both professional and home audio.

The usage of the WAV format has more to do with its familiarity, its simplicity and simple structure, which is heavily based on the IFF file format. Because of this, it continues to enjoy widespread use with a variety of software applications, often functioning as a ‘lowest common denominator’ when it comes to exchanging sound files between different programs.

In spite of their large size, uncompressed WAV (though that format can be different from the Microsoft WAV) files are sometimes used by some radio broadcasters, especially those that have adopted the tapeless system. BBC Radio in the UK uses 44.1 kHz 16 bit two channel .wav audio as standard in their VCS system. The ABC “D-Cart” system, which was developed by the Australian broadcaster, also uses a non-compressed format to preserve sound quality, and it has become more economical as the cost of data storage has dropped. In the system of “D-Cart”, the sampling rate of WAV files is usually at a 48 kHz 16 bit two channel, which is identical to that of the Digital Audio Tape.


The WAV format is limited to files that are less than 4 GB in size, because of its use of a 32-bit unsigned integer to record the file size header (some programs limit the file size to 2–4 GB).[2] Although this is equivalent to about 6·6 hours of CD-quality audio (44.1 KHz, 16-bit stereo), it is sometimes necessary to exceed this limit, especially when greater sampling rates or bit resolutions are required. The W64 format was therefore created for use in Sound Forge. Its 64-bit header allows for much longer recording times. The RF64 format specified by the European Broadcasting Union has also been created to solve this problem.

Audio CDs

Audio CDs do not use WAV as their sound format, using instead Red Book audio. The commonality is that both audio CDs and WAV files have the audio data encoded in PCM. WAV is a data file format for a computer to use that can’t be understood by CD players directly. To record WAV files to an Audio CD the file headers must be stripped and the remaining PCM data written directly to the disc as individual tracks with zero padding added to match the CD’s sector size.