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What really erks me about Hollywood is the fact they could quite easily support any number of speakers by using 4 channel encoding - more efficient, and vastly superior results. Of course this doesn't really make commercial sense for them though - they can't move the goalposts every 6 months!

The decoder would change depending on the number of speakers, but you could support any system this way, in any positions - ie 8 speakers could create a spherical sound stage, 3 or 4 to create conventional 360 degree surround.

See http://www.ambisonic.net/ for details

true, but you don't need more speakers to have more percieve-able directions. with delay and minor EQ changes you can produce a surround effect with a pair of headphones...

check out some of the articles on ambisonic.net.

Large, fast flash cards like this are good for high-quality (no lossy compression) portable audio recording too.

Even at 24/96 stereo, live audio needs less than 600 KB/s sustained write speed. Recording in 3D Ambisonic surround takes only double that. This page claims that a CF-compatible Microdrive cartridge can write at over 4 MB/s, so it should have no problem with data rates typical of live audio capture.

You do still have a point about durability however.

That's what DVD-A/SACD are for.
DVD-A is up to 192kHz/24bit, not 192kHz/32bit. Still, Above about 22-24 bits, there is no point in adding bits, as the noise floor of the system is below the threshold of thermal noise and other effects. A true 24-bit converter, for example, would give a noise floor of -144 dB, and apart from the fact that you can't realistically do anything with a -144 dB noise floor, no one can make components that quiet anyway: most 24-bit converters are hard pressed to reach -120dB. So 24 bits is more than enough. So I guess DVD-A is ok, but it is only 5.1, not 7.1.

Most 5.1 channel mixes are done using simple pairwise panning between two adjacent speakers to place the sound sources around you. This may be OK for movie effects but not for capturing the spatial nuances of a recording venue.

Ambisonics is a true 3D audio recording format. It is composed of 4 components: X, Y, Z and W that may be captured by the Soundfield Microphone or synthesized by audio ray tracing of the virtual venue.

The four components of the Ambisonics B format are a mathematical decomposition of the 3D sound wavefront at a point in space and are not directly related to any particular speaker placement. It may be decoded using simple linear operations into any speaker configuration. The 3D fidelity of the playback will depend on the number and placement of the speakers.

Note that 5.1 audio is still just 2D. The equivalent Ambisonics format would require only the W, X and Y components. With an additional top speaker you could feel the height of the concert hall in an Ambisonics recording.

One of the problems with Ambisonics is the chicken-and-egg problem - lack of enough media and playback equipment.

The significance of this is that AC3 on CD-R could let more people experiment with Ambisonics - the W, X and Y channels will be pre-decoded to a typical 5.1 speaker placement configuration. The AC3 should probably be recorded at the maximum quality setting of 640kbps. The resulting disk can be played back on any home theater system.

The Z channel can be somehow also stored on the disk so an Ambisonics-aware decoder could get full 3D audio. 3 of the 5 channels can be linearly combined to get back the W, X and Y channel and together with the Z channel you can decode it to any speaker configuration.

There is one particular speaker configuration that makes Ambisonics much easier to understand: imagine 8 speakers at the points of a cube. The W channel is fed to all speakers in the same polarity. The X channel is fed to the 4 right speakers with positive polarity and 4 left speakers with negative polarity. The Y channel is fed to the 4 front speakers with positive polarity and 4 back speakers with negative polarity. By now you can probably guess how the Z channel is connected.

Virtually all audio A/D and D/A converters today use sigma-delta, also commonly referred to as "one-bit" conversion.

In a sigma-delta A/D converter the audio signal is sampled with a high sampling frequency (typically a few MHz) and low sample resolution (1 bit). An error feedback mechanism is used to ensure that most of the energy of the quantization noise is "shaped" into high frequencies, giving excellent fidelity in the audio band. One bit is inherently linear - no need for carefully matched resistor networks such as those used on older A/D converters. This stream is then filtered and decimated using digital signal processing techniques to a lower sampling rate (e.g. 44100Hz) while gaining sample depth on the way (16 bits and higher).

For D/A conversion the process is reversed: the 44100Hz signal is interpolated up to a high sampling rate and then the sample depth is reduced down to one bit. Again, error feedback is used to ensure that the quantization noise resulting from the low resolution is shaped to high frequencies. This bitstream is then low-pass filtered and used as the audio signal. Again, with much better linearity than D/A converters based on carefully adjusted resistor networks.

The Sony SACD skips the decimation and interpolation stage. It stores the noise-shaped bitstream directly on the disc. The beauty of this idea is in its simplicity: it performs much less transformations on the sigma-delta signal and therefore should offer inherently higher fidelity and wider bandwidth.

If sigma-delta converters were available 20 years ago when the CD was invented they would probably have chosen this method for its simplicity. But at that time the analog conversion technique known was resistor networks so PCM was used.

Remember that at the time the CD was really stretching the limits of consumer technology. No other consumer product prior to the CD player used so many new and advanced technologies: lasers, error correction, digital signal processing. If they could have used this technique it would have reduced the cost of CD players significantly. For example, this bitstream is much more tolerant to bit errors because unlike PCM there is no "most significant bit" that can cause a large error if corrupted.

Using this technique today, though, is insane. There is no real savings in simplicity when a million digital transistors cost close to nothing. If you want higher fidelity, 96kHz and 24 bits is more than enough.

Let's say you want something simple like a graphic equalizer on your SACD player. If it's analog such a complex circuit will introduce lots of noise. If you implement it digitally it would take insanely large amounts of CPU power to process a signal sampled at over 2mHz. Manufacturers will probably end up downconverting it to PCM at 96kHz or lower, doing the signal processing and then converting back to sigma delta for playback. This will lose all of DASD's alleged advantages.

BTW, for the purpose of preserving analog masters DASD is really a good idea because they contain useful information at very high frequencies such as the tape bias signal and the intermodulations it creates. Preserving this information will allow future signal processing techniques to create accurate models of the nonlinearities of the magnetic medium and use this high frequency information to reconstruct the original recording with better fidelity down in the audio band. For home use SACD is a very bad idea. Just about the only good thing I can see about it is that it can be marketed effectively because it's such a "radical new concept".

The DVD audio uses conventional, well proven PCM with somewhat higher sampling frequency and bit depth than CD. Why use a higher sampling frequency when we can't hear over 20kHz? It turns out that while we can't hear a sinewave at frequencies higher than 20kHz the high frequency components of complex waveforms make a noticable difference even up to 26kHz. To take a good safety margin and maintain integer ration a 96kHz sampling rate was used. This does not significantly hurt the data rate required because non-lossy compression is used on DVD audio. A compressed 96kHz signal takes about 30% more space than a compressed 48kHz signal. 16 bits is, again, almost enough. In fact, with proper in-band noise shaping the noise floor is inaudible in all but very extreme circumstances. 24 bits is therefore a very good safety margin.

Another reason why DVD-audio is superior is because it supports Ambisonics. Ambisonics is a surround sound system. It was not crated for cinematic effects. Ambisonics was designed for music and for reconstructing the subtle spatial cues of the ambience of the recording venue. With a proper arrangement of speakers it can create true 3D sound - including the height dimension. Imagine listening to a recording and feeling the height of the concert hall!

Please never ask "how many channels does Ambisonics use" because it's not a relevent question. Ambisonics deconstructs the 3D sound field mathematically using a four component representation (XYZW). This representation can be processed with a simple linear matrix for playback on different speaker configurations and numbers of channles with varying levels fidelity of 3D soundfield reconstruction. This includes the popular 5.1 setup used on home theaters (it's probably going to be the default settings for DVD-Audio players) .

DVD-Audio is also backward compatible with DVD players although a DVD-audio player will be required to take advantage of all the features and full quality.

More information about DVD-Audio here

----

Virtually all audio A/D and D/A converters today use sigma-delta, also commonly referred to as "one-bit" conversion.

In a sigma-delta A/D converter the audio signal is sampled with a high sampling frequency (typically a few MHz) and low sample resolution (1 bit). An error feedback mechanism is used to ensure that most of the energy of the quantization noise is "shaped" into high frequencies, giving excellent fidelity in the audio band. One bit is inherently linear - no need for carefully matched resistor networks such as those used on older A/D converters. This stream is then filtered and decimated using digital signal processing techniques to a lower sampling rate (e.g. 44100Hz) while gaining sample depth on the way (16 bits and higher).

For D/A conversion the process is reversed: the 44100Hz signal is interpolated up to a high sampling rate and then the sample depth is reduced down to one bit. Again, error feedback is used to ensure that the quantization noise resulting from the low resolution is shaped to high frequencies. This bitstream is then low-pass filtered and used as the audio signal. Again, with much better linearity than D/A converters based on carefully adjusted resistor networks.

The Sony SACD skips the decimation and interpolation stage. It stores the noise-shaped bitstream directly on the disc. The beauty of this idea is in its simplicity: it performs much less transformations on the sigma-delta signal and therefore should offer inherently higher fidelity and wider bandwidth.

If sigma-delta converters were available 20 years ago when the CD was invented they would probably have chosen this method for its simplicity. But at that time the analog conversion technique known was resistor networks so PCM was used.

Remember that at the time the CD was really stretching the limits of consumer technology. No other consumer product prior to the CD player used so many new and advanced technologies: lasers, error correction, digital signal processing. If they could have used this technique it would have reduced the cost of CD players significantly. For example, this bitstream is much more tolerant to bit errors because unlike PCM there is no "most significant bit" that can cause a large error if corrupted.

Using this technique today, though, is insane. There is no real savings in simplicity when a million digital transistors cost close to nothing. If you want higher fidelity, 96kHz and 24 bits is more than enough.

Let's say you want something simple like a graphic equalizer on your SACD player. If it's analog such a complex circuit will introduce lots of noise. If you implement it digitally it would take insanely large amounts of CPU power to process a signal sampled at over 2mHz. Manufacturers will probably end up downconverting it to PCM at 96kHz or lower, doing the signal processing and then converting back to sigma delta for playback. This will lose all of DASD's alleged advantages.

BTW, for the purpose of preserving analog masters DASD is really a good idea because they contain useful information at very high frequencies such as the tape bias signal and the intermodulations it creates. Preserving this information will allow future signal processing techniques to create accurate models of the nonlinearities of the magnetic medium and use this high frequency information to reconstruct the original recording with better fidelity down in the audio band. For home use SACD is a very bad idea. Just about the only good thing I can see about it is that it can be marketed effectively because it's such a "radical new concept".

The DVD audio uses conventional, well proven PCM with somewhat higher sampling frequency and bit depth than CD. Why use a higher sampling frequency when we can't hear over 20kHz? It turns out that while we can't hear a sinewave at frequencies higher than 20kHz the high frequency components of complex waveforms make a noticable difference even up to 26kHz. To take a good safety margin and maintain integer ration a 96kHz sampling rate was used. This does not significantly hurt the data rate required because non-lossy compression is used on DVD audio. A compressed 96kHz signal takes about 30% more space than a compressed 48kHz signal. 16 bits is, again, almost enough. In fact, with proper in-band noise shaping the noise floor is inaudible in all but very extreme circumstances. 24 bits is therefore a very good safety margin.

Another reason why DVD-audio is superior is because it supports Ambisonics. Ambisonics is a surround sound system. It was not crated for cinematic effects. Ambisonics was designed for music and for reconstructing the subtle spatial cues of the ambience of the recording venue. With a proper arrangement of speakers it can create true 3D sound - including the height dimension. Imagine listening to a recording and feeling the height of the concert hall!

Please never ask "how many channels does Ambisonics use" because it's not a relevent question. Ambisonics deconstructs the 3D sound field mathematically using a four component representation (XYZW). This representation can be processed with a simple linear matrix for playback on different speaker configurations and numbers of channles with varying levels fidelity of 3D soundfield reconstruction. This includes the popular 5.1 setup used on home theaters (it's probably going to be the default settings for DVD-Audio players) .

DVD-Audio is also backward compatible with DVD players although a DVD-audio player will be required to take advantage of all the features and full quality.

More information about DVD-Audio here

----

http://www.sonymusic.com/sacd/

http://www.superaudio-cd.com/

Something that doesn't gush like a press release

Another more objective link

It appears they're using a dual layer method for backwards compatibility. The details about copy protection methods are vague, but they do mention visible and invisible watermarks aimed against both pirates and counterfeiters. But I can't seem to find a decent explanation of how the encoding DSD encoding scheme works.