dave@onfcanim.UUCP (08/16/87)
I'm looking for information on how to extract the data from the
signal that is fed to the "digital output" jack of my Philips (Magnavox)
CDB 650 CD player.
Looking at the signal on an oscilloscope, it appears to consist of a framing
pulse, and then a block of bits transmitted using a simple modulation
technique. I want to know:
1) how to build a circuit that will find the framing signal and then
demodulate the data into an N-bit parallel word plus strobe ready
to be connected to a computer parallel interface
2) how to interpret the various bits that I get from this. Most of them,
of course, are audio data, but there should also be an "uncorrected
error" flag, and maybe subcode information.
3) how to build a simple FIFO buffer to sit between the decoder
circuitry and the "host" computer, since the computer
will accept data in bursts.
The idea is to obtain high-quality digital audio to experiment with
(e.g. using the Fourier transform to look at the frequency spectrum)
without the cost of analog-to-digital conversion hardware.
I'd appreciate pointers to code standards, circuitry, or anything else
that seems relevant.
Dave Martindale
{watmath,musocs,micomvax}!onfcanim!daveczei@osupyr.UUCP (Michael S Czeiszperger) (08/18/87)
In article <15368@onfcanim.UUCP> dave@onfcanim.UUCP (Dave Martindale) writes: >I'm looking for information on how to extract the data from the >signal that is fed to the "digital output" jack of my Philips (Magnavox) >CDB 650 CD player. > >Looking at the signal on an oscilloscope, it appears to consist of a framing >pulse, and then a block of bits transmitted using a simple modulation >technique. I want to know: > >I'd appreciate pointers to code standards, circuitry, or anything else >that seems relevant. > Try looking up the AES/EBU digital audio standard. Your CD player probably is using that format, and it can be found at your closest Audio Engineering Society reference source. Michael S. Czeiszperger | Disclaimer: "Sorry, I'm all out of pith" Sound Synthesis Studios | Snail: Room 406 Baker Phone: (614) College of the Arts Computer Lab | 1971 Neil Avenue 292- The Ohio State University | Columbus, OH 43210 0895 UUCP : {decvax,ucbvax}!cbosgd!osupyr!czei
henkp@nikhefk.UUCP (Henk Peek) (08/23/87)
In article <15368@onfcanim.UUCP> dave@onfcanim.UUCP (Dave Martindale) writes: |I'm looking for information on how to extract the data from the |signal that is fed to the "digital output" jack of my Philips (Magnavox) |CDB 650 CD player. | |Looking at the signal on an oscilloscope, it appears to consist of a framing |pulse, and then a block of bits transmitted using a simple modulation |technique. I want to know: | | 2) how to interpret the various bits that I get from this. Most of them, | of course, are audio data, but there should also be an "uncorrected | error" flag, and maybe subcode information. The digital audio output consists of 32 bit words transmitted in biphase-mark code. That is, two transitions for a logic 1 and one transition for a logic 0. The 32 bits words are transmtted in blocks of 384 words. Composition of the 32 bit digital audio word. bit number description information 1 to 4 sync 5 to 8 auxiliary not used (always zero) 9 to 28 audio sample bits 9 to 12 not used (always zero) bit 13 (LSB) to bit 20 (MSB) two's complement audio 29 audio valid copy of the error flag 30 user data used for subcode data 31 channel status indication of control bits and catagory code 32 parity bit even parity for all word bits excluding the sync pattern The sync word is formed by violation pf the biphase rule and therefore does not contain any data. Its length is equivalent to 4 data bits. There are 3 different sync patterns: Sync B; start of a block of 384 words, contains left sample (11101000) Sync M; word contains left sample, but is not a start (11100010) Sync W; word contains right sample (11100100) The left and the right samples are transmitted alternately. The channel status is the same for both left and right words. Therefore a block of 384 words contains 192 channel status bits. bit number description subcode provided no subcode provided 1 to 4 control copy of Q channel bits 1 and 2 zero bit 3 image of SCAB bit 4 image of SDAB 5 to 8 reserved always zero always zero 9 to 16 category code CD category general category bit 9 logic 1 17 to 192 always zero always zero When there is no subcode the channel status wil switch over to general format. No subcode is identified by the subcode detector detector when SCAB is a continuous high or low. The subcode is transmitted via the user bit and asynchronous with the block rate. | |The idea is to obtain high-quality digital audio to experiment with |(e.g. using the Fourier transform to look at the frequency spectrum) |without the cost of analog-to-digital conversion hardware. | Dave Martindale | {watmath,musocs,micomvax}!onfcanim!dave henk peek ..!seismo!mcvax!nikhefk!henkp.UUCP