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Basic Sound Engineering PP 271 01 Classroom Door Code: 16333 Basic Sound Engineering PP 271 01 http://web.me.com/dtostilane/PP271/PP271_Basic_Sound_Engineering.html Classroom Door Code: 16333 Wednesday, October 19, Digital Signal Details AES3 (formerly AES/EBU) and S/PDIF are the two most common formats: AES3 calls for a balanced connection, using 110Ω cable S/PDIF is unbalanced, 75Ω cable, and uses either coaxial “copper” connection (RCA) or optical “ADAT” (or “Toslink”) connector. Data format is interchangeable - AES3 signal works with S/PDIF hardware and vice-versa. Wednesday, October 19, There are other formats - for instance MADI Multichannel Audio Digital Interface - also called AES10 - but this is a transport format that allows multiple channels of AES3 information to be carried over optical fiber or copper. So, What About That Data Format? Pulse Code Modulation (PCM) Developed in 1937 by Alec Reeves at ITT Labs in France Input signal is sampled, quantized, and coded. Binary numbers representing amplitude of the signal at the sampling point are transmitted by a pulsed code. Wednesday, October 19, Pulse Code Modulation (other schemes include Pulse Width Modulation, Pulse Amplitude Modulation, Pulse Number Modulation). Produces a SERIAL data stream with the two channels multiplexed together into a single stream that is arranged via TDM - Time Domain Multiplexing - so that it can be decoded at the other end. Sample Rates Multiple rates are used for different applictions “CD” rate of 44,100 Hz (44.1 kHz) [sample every 22.7µs] “Video” rate of 48 kHz (AES Standard) [20.8 µs] 96 kHz rate [10.4 µs] 192 kHz rate [5.2 µs] 88.2 kHz (thought to improve conversion from 44.1 kHz) [11.3 µs] Regardless of rate selected each “sample” makes up a digital “word” Number of quantization levels - resolution of signal determined by number of bits: 16 bit - 65,536 levels, 96.32 dB max Dynamic Range (DR) 20 bit - 1,048,576 levels, 120.41 dB max DR 24 bit - 16,777,216 levels, 144.5 dB max DR Wednesday, October 19, Recall that diferent sample rates afect frequency range, while diferent bit counts afect dynamic range. 88.2 kHz comes about because it is an even doubling of 44.1 kHz - which is thought to have less difcult mathematical conversion than 44.1 kHz to 48 kHz. Common wisdom is the least conversions you need to make, the more accurate the signal will remain. How to Keep it All Straight This is what the code part is all about We need to be able to handle multiple bit rates We need to be able to handle multiple sampling frequencies We need to be able to maintain regular clock sync We need to move a lot of bits and interpret them accurately 24 bit signal would need 48 bits for a stereo recording at each sample interval plus some additional bits to encode data needed to interpret the signal First Hat Trick - Biphase Mark Coding (BMC) Wednesday, October 19, Biphase Mark Coding Need to identify “0” and “1” One way would be to just say 1=+5v and 0=-5v, and go for it. But - you might wind up with long strings of the same values, making it difficult to identify changes. BCM identifies TWO bits as a “time slot” and uses both to identify a single data bit. When the value changes within the time slot it’s a “1” When the value remains the same within the time slot it’s a “0” There is ALWAYS a change from one time slot to the next Wednesday, October 19, Brilliant solution - BUT - we’ve doubled the necessary speed of the system. Now that hardware has advanced to the place it has, this is no longer an issue. Advantages: Works exactly the same regardless of polarity of connection Sample rate is communicated as 1/2 the bit rate of the signal Clock can be taken from the bit rate of the signal Always is a change from one time slot to another - never can have string of same-value bits. Disadvantage System must be able to handle twice the number of bits - i.e. for 44,100 Hz sample rate, system operates at 88,200 Hz Wednesday, October 19, AES3 Divided into Blocks, Frames, Subframes A Subframe is 32 bits long. Four Preamble bits 24 Audio Data bits (one channel of audio data) 4 Status bits A Frame is two subframes (two channels of audio data) A block is 192 Frames 4.3 ms at 44.1 kHz (229.7 blocks per second) 4.0 ms at 48 kHz (250 blocks per second) 2 ms at 96 kHz (500 blocks per second) 1 ms at 192 kHz (1,000 blocks per second) Wednesday, October 19, AES3 Subframe Wednesday, October 19, Four options for a subframe - 24 bit audio, 20 bit with zeros, 20 bit with AUX bits, 16 bits with AUX bits and zeros. LSB is at the start, MSB at the end. Preamble defines where the subframe sits - Z preamble is the start of a block, X preamble is the start of a Frame, Y preamble is the second subframe in a frame. V, U, C, and P bits combine to make 192 bit data per channel per block. AES3 Block Structure Wednesday, October 19, Preamble bits determine where in the block structure one is at. X preamble is always the start of a frame except for the first frame in a block, which uses the Z preamble. Y preamble is always the second subframe. Under no circumstance will the same preamble occur in two successive subframes. Note frames are counted starting with zero, so they go zero to 191, as are frames counted 0 and 1 rather than 1 and 2. Preamble Bits Wednesday, October 19, Preamble bits do not conform to the BMC format - there are three +5v pulses in sequence - the only place in the protocol that this can happen, and the only place that the time slot conventions are broken. So, if the receiver sees a bit sequence of either 1 or 0, it knows it is at the start of a block or subframe. Even though these don’t conform to the usual format, they still are read as four bits - always 1010, but with diferent time structure identifying which kind of preamble we have. Channel Status Blocks Possibly the most clever part of the scheme Four bits in each subframe for channel status 192 blocks give us 192 bits, or 24 bytes for each status type for each channel Validity bits used to confirm data words are good in each subframe User bits can be used to store user defined data Channel Status bits are used to describe system and format details Parity bit is used as part of error correction Wednesday, October 19, Channel Status Byte 0 Wednesday, October 19, Status Byte 0 bit 0 identifies S/PDIF or AES3 If the bit is set to “0” then the system knows that the following bits and bytes have diferent meanings. For example, in S/PDIF, Byte 0 bit 2 is the copy protection bit, but in AES3, bits 2,3 and 4 together determine emphasis (frequency shading) applied to the signal. (there is no option for copy protection in an AES3 stream) Channel Status in S/PDIF Wednesday, October 19, Byte 0 for S/PDIF - bits 0 and 1 define details about the format Bit 2 is the infamous copy protection bit - 1 is no copy, 0 is copy 3 is a single fixed pre-emphasis choice The rest of the channel status blocks are “reserved” except for some additional info in Byte 1, while in AES3, most of the channel status block slots are used for various definitions of format options. Networks for Digital Audio Standard Ethernet Networking Asynchronous - not synchronized - not predictable Data assembled into “packets” and timing negotiated for send and receive to avoid “collisions” - having a send interfere with a receive. This works for word processors, not so much for audio But, the equipment - the “physical layer” of Ethernet is quite adaptable for audio transmission, and is used by many of the audio specific protocols. Wednesday, October 19, Network Terms MAC Address: Unique address for every physical device that communicates via networks. 12 digit alpha-numeric allows 281 Trillion possible addresses. Look like “e0:f4:49:32:cf:fc” Codes define network capacity - 10base5, 10base2, 100baseTX and commonly now 1000BaseT “Gigabyte Ethernet” Cable designation such as CAT-5e and CAT-6 define rating CAT-5 - 100MHz capacity CAT-5e - tighter twist can handle slightly higher speed signal (basic minimum) CAT-6 - 250 MHz capacity CAT-6a - 500 MHz capacity Wednesday, October 19, 100baseTX type numbers define speed and size of cable - 100baseTX uses 2 twisted pairs of cable together referred to as CAT5 cable. Network Terms Network Switch a device for interconnecting network components May be managed or unmanaged - “Hub” is a typical unmanaged switch. May be defined by “Layer” - layer 1 is hub, layer 2 called bridge, layer 3 could be a router, layers 4-7 add increasing sophistication and functions. Wireless - rarely workable for multichannel audio transmission. 802.11n about 300Mbits/sec close by, down to 70Mbits/sec with distance (or less). Generally too slow for digital audio data stream. (But can often be used for control signals) Wednesday, October 19, Typical wireless routers that you buy for a home system are partially managed layer 2 or 3 devices. You can control some features with a log-on url in most cases. More sophisticated (and expensive) routers are needed for audio networks in most cases. Audio Networks BSS SoundWeb Cobranet Aviom Ethersound Dante AVB - Audio Visual Bridging set of technical standards of the IEEE In process of development by industry council Idea to provide one set of documents that all manufacturers can use so equipment will interoperate.
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