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Continuing from previous article, outline specification sent to a de-correlation expert:-

A system exists for transferring data. This system allows data to be transferred reliably outside of the parameters of TCP/IP with Window Scaling and DSACK. Development of the system began as an independent project but has subsequently been funded by [redacted]. At one point, the system was being developed by five programmers, one mathematician, one graphic designer and two administrative staff.

Regardless, the system does not process data as it is received but it is desirable to add such functionality as a library. This capability has been identified as a financially viable market. (See Janus Friis and Niklas Zennström's development of Kazaa, Skype, Joost and other ventures.) Furthermore, the market can be segmented into real-time streaming of low-quality audio and delayed delivery of high quality audio. In each case, it is possible to exceed the perceptual quality of Compact Discs and Long Playing Records respectively.

Many systems provide streaming below this quality (FM radio, digital radio, NICAM, MP3 delivery, RealAudio) but the ability to provide higher quality audio will improve as bandwidth increases. Even if this market is not broadly viable, it remains a lucrative niche for audiophiles with large disposable income. Even if this market is unviable, there remain applications for higher quality audio or a more compact representation of lossless audio. This may be with or without an accompanying video codec.

The benchmark for sound quality is Compact Disc "Red Book" audio; introduced in 1980. This specifies 2KB sectors with Reed-Solomon regenerative checksums. Each sector provides 1,024 16 bit PCM samples for one of two audio channels at a sample rate of 44.1kHz. Techniques to smooth the 1% of dropped sectors create a frequency floor of approximately 43Hz irrespective of data encoded on a disc. This creates a notable absence of bass frequencies and a perceptual comb of frequencies which are integer multiples of 43Hz.

Although modern audio codecs have good perceptual response and raise the nominal sampling frequency from 44.1kHz to 48kHz or higher, the encoded datarate of MP3, AAC and similar codecs is often less than 25KB/s over two channels. This is opposed to 176KB/s for CD audio.

For an extended period, one of the system developers has been aware of efforts by a company in Cambridge, England which may be regarded as a direct competitor. Meridian first came to our attention around 1982 when the development of [MLP] Meridian Lossless Packing was demonstrated in a particularly rigged manner on the BBC [British Broadcasting Corporation]'s programme Tomorrow's World. The demo involved the chassis of a CD player and two digit LED display. When playing a conventional Compact Disc, the display read "16" to indicate 16 bit PCM [Pulse Code Modulation] data. When playing a MLP Disc, the display read "4" to indicate the bitrate of MLP. However, a cursory investigation of MLP, such as http://www.meridian-audio.com/w_paper/mlp_jap_new.PDF via http://en.wikipedia.org/wiki/Meridian_Lossless_Packing, reveals that the lossless encoding technique removes an *average* of 11 bits of data per sample - leaving approximately 5 bits per sample. Furthermore, the switch from 16 bit to 4 bit was performed during a cutaway shot and therefore the BBC may have been complicit in this rigged demo. Finally, any attempt to exhibit the perceptual quality of MLP over a television broadcast was futile because the channel capacity of television audio is lower than Compact Disc quality.

Further involvement with Meridian came in the form of a declined interview after a long discussion about digital speaker synchronisation and the computational requirements of wireless digital headphones.

One of the system developers is also aware of SACD [Super Audio Compact Disc], an audio encoding standard devised by Professor Jamie Angus and included in the first and second revision of the Sony PlayStation 3. Attempts to re-implement this technique in a digital system required an inordinate amount of bandwidth. Regardless, experiments led to a novel encoding technique and a greater appreciation of analog circuitry.

Regardless, Meridian's technique for reducing datarate is an ideal template for further development. Although care has to be taken to avoid known patents. The basic MLP specification is that:-

• An encoding technique should allow multiple, arbitrary 24 bit streams to be interleaved without change to the bit sequences.
• An encoding technique may make assumptions about the presence of return-to-zero data and therefore compression may only occur when suitable audio data is presented.
• Significant compression may be obtained by observing correlations between channels.
• Finite Impulse Response filters can be used to reduce the volume of data to be encoded.
• Lossy techniques and predictive systems may be deployed within an encoder and a decoder. A residual channel of data makes lossy, deterministic techniques into a lossless system.

Further requirements for streaming are as follows:-

• It is desirable to split streams into components so that they may be prioritised to maximise perceptual quality.
• Therefore, it is desirable to extract signal from streams prior to the application of techniques such as FIR.

In addition to streaming and otherwise exceeding the MLP specification, the following is desirable:-

• It should be possible to decode sound on legacy hardware. An embedded device should be able to decode one channel of sound and output 6 bit quality or better. A legacy desktop should be able to decode two or more channels of sound at 8 bit quality.
• It should also be possible to encode sound of limited quality on legacy hardware.
• When data is missing, it should be possible to re-construct sound with frequencies below 43Hz. Ideally, it should be an option to encode and re-construct sound with frequencies which are multiples of 1nHz or less. (At data rates exceeding 44.1kHz, this may require block sizes exceeding 44 billion samples. For 24 bit samples, this requires 120GB or more per channel. It may also require handling of RMS [Root Mean Square] errors with a magnitude of 82 bits or more.)
• Entropy encoding may be more effective if there is a bias in the symbol frequencies. Therefore, blocks may be a multiple of x^y where x is odd rather than even.

Finally, all techniques are valid. A direct clone of SACD or MLP is not desirable due to licensing issues. However, simplified or novel techniques around SACD and MLP are highly desirable.

Further mention of Meridian, sigma-delta encoding and related topics.