[linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions

I read an interesting article on Direct Stream Digital (DSD) / Pulse Density Modulation (PDM) entitled "A Better Mousetrap" by Brian Smithers in the May 2003 issue of Electronic Musician. Since, Brian did a good job explaining PDM/DSD in quasi-layman terms, I'll just quote snippets from his article to set the stage for my questions. <quote a="EM: A Better Mousetrap" p="96"> In DSD, the amplitude of the waveform is represented by the relative density of the ones and zeros within the bitstream; for that reason the technique is also known as Pulse Density Modulation (PDM). A full positive waveform results in all ones and, whereas a full negative waveform results in all zeros. Silence would yield a string of alternating ones and zeros (Alternating ones and zeros reflect the "neutral" density of silence.) Sony claims a resulting dynamic range of 120db for DSD. In the simplest terms, DSD compares the energy of a waveform at each sample point to the accumulated energy of previous sample periods. If the energy is higher, a value of 1 is recorded; if the energy is lower, a value of 0 is recorded. Because the bitstream represents the change (delta) in energy compared to the sum (sigma) of the previous energies, another name for this technique is Sigma-Delta Modulation (SDM). </quote> The article goes on to explain how DSD uses a sampling rate of 2.8224 MHz resulting in a frequency response that extends to 100kHz. The higher sample rate can also improve stereo/surround imaging. <quote a="EM: A Better Mousetrap" p="96"> One more justification for DSD is that the steep filters required for PCM to conform to the Nyquist theorem can themselves degrade the sound due to their sharp cutoff. In fact, it has become common for PCM analog-to-digital (A/D) converters to start with a 1-bit conversion stage and then apply a digital decimation filter to convert the bitstream to PCM format. DSD essentially bypasses the decimation filter and stores the 1-bit data directly. </quote> DSD/PDM appears to be a superiour technique for recording and playing audio material. Granted, this technology may never catch on because of all the hardware and software changes that would be required to mirror what a typical PCM based DAW currently does. But, if DSD/PDM does catch on, and DAWs start being produced, how will this effect current audio DSP techniques? How would JACK's current design be effected by a 1-bit stream format? The article mentions a program called Pyramix (Windows) which features DSD support. However, for Pyramix to do EQ, dynamics, reverb processing, and to display waveforms and vu levels, it converts DSD to a "high quality" PCM format. I suspect that if DSD/PDM has any hope of succeeding in the current DAW world, there will have to be some real-time lossless converter constructed to seemlessly bridge the formats. But that of course begs the question of why use DSD in the first place if you just are forced to "decimate" it PCM to process it? Sony is currently using DSD/PDM in their SACD format, and are positioning themselves to be the competing format to PCM based DVD-A. The fact that something like this shows up in EM means that at some level the format is being given some credence and is being watched by others then just the audiophile recording crowd. Is DSD/PDM just a flash in the pan, or does it really represent a new wave of thinking regarding storing/processing audio data? Dustin _________________________________________________________________ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus