AES/EBU dan S/PDIF BLASTICA Iwan B Pratama Hal 1 Digital Audio Interfaces A digital audio interface allows equipment such as audio mixers, recording devices or audio processors to be interconnected without degrading the signal quality. Unfortunately, there are different types of interface, most of which are incompatible, although in some instances you can wire a device with an AES/EBU interface to a device fitted with a S/PDIF socket. Most interfaces convey stereo (2-channel) audio over a single set of wires, although some also require a separate word clock (WC) circuit, which is often wired via a standard BNC connector. Common interfaces are described in the following sections. Sony/Philips Digital Interface (S/PDIF) S/PDIF conveys both channels of a stereo signal, usually via an RCA phono (PIN) connector, although some devices use a TOSLink fibre-optic connector. This interface is commonly used for domestic equipment, such as CD/DVD players, Digital Audio Tape (DAT) recorders, and MiniDisc (MD) machines, conveying 16, 20 or 24-bit samples at rates of up to 96 kHz. • An S/PDIF circuit can be used to convey data instead of normal audio, including AC-3 data for Dolby Digital 5.1 surround sound, or DTS surround sound data. Electrical Hardware S/PDIF in its ‘copper’ form uses a signal level of between 0.5 and 1 volts into a 75 Ω load over an unbalanced circuit. The similarity to a standard television signal makes it suitable for routing through equipment designed for video material. Since it’s meant to operate with 75 Ω cable it can also be used over long distances without equalisation, and in this regard is superior to the professional AES/EBU interface (see below). • Some devices incorporate an output transformer for improved circuit isolation. • Although a standard RCA phono (PIN) connector is normally used for ‘copper connections’, some professional devices and adaptors employ a BNC connector. • You may encounter equipment that has a fibre-optic connector, usually of the TOSLink variety, using 1 mm plastic fibre-optic cable and visible red light. The high attenuation in this kind of fibre limits the maximum cable length to 10 metres or less. In addition, you may need an adaptor box to connect other S/PDIF or AES/EBU devices. • PC-based hardware sometimes provides a digital output via a 2-pin header, also known as an HDR-2 interface. Although the data conforms to the S/PDIF standard, the 5 volt TTL signals are incompatible with S/PDIF. Connecting to devices with an AES/EBU interface is possible, although the following points should be noted:- 1. A standard AES/EBU input accepts data with minimum eye height of 200 mV. This means that it should accept the S/PDIF minimum signal level of 500 mV peak-to-peak. 2. An S/PDIF input normally requires an attenuator to protect it from being damaged by the comparatively high level signal produced by an AES/EBU output. 3. The AES/EBU interface and S/PDIF convey different data flags. This means that you may find that a connection, although electrically satisfactory, simply refuses to work. Further information about connecting different digital interfaces appears elsewhere in this article. Encoding S/PDIF is a domestic version of the AES/EBU interface (see below). The data is assembled in the same way, except for differences in the Channel Status bits (C-bits). It’s possible to interconnect AES/EBU and S/PDIF devices, either directly or via an adaptor box, the latter correcting differences in signal levels or modifying any offending C-bits. • S/PDIF is sometimes called CD/DAT, IEC 958-11 or IEC 60958, although there are some differences in settings of C-bits between the S/PDIF and IEC 958-11 standards (see below). • What appears to be an AES/EBU interface may actually be an SP/DIF interface with extra balancing circuitry, also known as the IEC 958-11 interface. Equipment with this
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AES/EBU dan S/PDIF BLASTICA
Iwan B Pratama Hal 1
Digital Audio Interfaces A digital audio interface allows equipment such as audio mixers, recording devices or audio processors to be interconnected without degrading the signal quality. Unfortunately, there are different types of interface, most of which are incompatible, although in some instances you can wire a device with an AES/EBU interface to a device fitted with a S/PDIF socket. Most interfaces convey stereo (2-channel) audio over a single set of wires, although some also require a separate word clock (WC) circuit, which is often wired via a standard BNC connector. Common interfaces are described in the following sections. Sony/Philips Digital Interface (S/PDIF) S/PDIF conveys both channels of a stereo signal, usually via an RCA phono (PIN) connector, although some devices use a TOSLink fibre-optic connector. This interface is commonly used for domestic equipment, such as CD/DVD players, Digital Audio Tape (DAT) recorders, and MiniDisc (MD) machines, conveying 16, 20 or 24-bit samples at rates of up to 96 kHz.
• An S/PDIF circuit can be used to convey data instead of normal audio, including AC-3 data for Dolby Digital 5.1 surround sound, or DTS surround sound data.
Electrical Hardware S/PDIF in its ‘copper’ form uses a signal level of between 0.5 and 1 volts into a 75 Ω load over an unbalanced circuit. The similarity to a standard television signal makes it suitable for routing through equipment designed for video material. Since it’s meant to operate with 75 Ω cable it can also be used over long distances without equalisation, and in this regard is superior to the professional AES/EBU interface (see below).
• Some devices incorporate an output transformer for improved circuit isolation. • Although a standard RCA phono (PIN) connector is normally used for ‘copper
connections’, some professional devices and adaptors employ a BNC connector. • You may encounter equipment that has a fibre-optic connector, usually of the TOSLink
variety, using 1 mm plastic fibre-optic cable and visible red light. The high attenuation in this kind of fibre limits the maximum cable length to 10 metres or less. In addition, you may need an adaptor box to connect other S/PDIF or AES/EBU devices.
• PC-based hardware sometimes provides a digital output via a 2-pin header, also known as an HDR-2 interface. Although the data conforms to the S/PDIF standard, the 5 volt TTL signals are incompatible with S/PDIF.
Connecting to devices with an AES/EBU interface is possible, although the following points should be noted:-
1. A standard AES/EBU input accepts data with minimum eye height of 200 mV. This means that it should accept the S/PDIF minimum signal level of 500 mV peak-to-peak.
2. An S/PDIF input normally requires an attenuator to protect it from being damaged by the comparatively high level signal produced by an AES/EBU output.
3. The AES/EBU interface and S/PDIF convey different data flags. This means that you may find that a connection, although electrically satisfactory, simply refuses to work.
Further information about connecting different digital interfaces appears elsewhere in this article. Encoding S/PDIF is a domestic version of the AES/EBU interface (see below). The data is assembled in the same way, except for differences in the Channel Status bits (C-bits). It’s possible to interconnect AES/EBU and S/PDIF devices, either directly or via an adaptor box, the latter correcting differences in signal levels or modifying any offending C-bits.
• S/PDIF is sometimes called CD/DAT, IEC 958-11 or IEC 60958, although there are some differences in settings of C-bits between the S/PDIF and IEC 958-11 standards (see below).
• What appears to be an AES/EBU interface may actually be an SP/DIF interface with extra balancing circuitry, also known as the IEC 958-11 interface. Equipment with this
AES/EBU dan S/PDIF BLASTICA
Iwan B Pratama Hal 2
kind of interface may not work due to incorrect C-bits, such as bit 0 which is different for domestic and professional audio material.
• More about encoding can be found in the section concerning the AES/EBU Interface.
Channel Status Bits The C-bits for S/PDIF are organised as follows:- Bit(s) Function 0 0 - Consumer source of material 1 - Professional source of material 1 0 - Sample contains audio information 1 - Sample contains data • 2 0 - Copying prohibited (Copy Protect) 1 - Copying permitted (Copy Protect 'off') 3, 4 0, 0 - No audio emphasis 1, 0 - 50/15 µs emphasis 'on' 5 0 - Two-channel audio (default) 1 - Four-channel audio 6, 7 0, 0 - Mode 0 (bits 0-25 defined) Otherwise only bits 0-7 defined 8-15 Category Code of sender (Mode 0 only) 0, 0, 0, 0, 0, 0, 0, 0 - General (Gen) 1, 0, 0, 0, 0, 0, 0, 0 - Compact Disc (CD) 0, 1, 0, 0, 0, 0, 0, 0 - PCM Adaptor (PCM) 1, 1, 0, 0, 0, 0, 0, 0 - Digital Audio Tape (DAT) 1, 1, 0, 0, 0, 0, 0, 1 - DAT-P (Copying permitted)16-23 Reserved (Mode 0 only) 24, 25 Sampling Rate (Mode 0 only) 0, 0 - 44.1 kHz 0, 1 - 48 kHz 1, 1 - 32 kHz • Data samples can include MPEG3, AC3, DTS and other special IEC 61937 formats. Bits 0 and 1 have the same function in both S/PDIF and AES/EBU data. Bits 1 to 5 and 24 to 25 (in Mode 0) are copied from the source to destination. This means that the C-bits in recorded material are identical to those contained in the original data. The most important bits are usually set as follows:- Bit Usual State Notes 0 0 (off) Can prevent connection to professional equipment1 0 Indicates interface used for audio, not data 2 1 (on) Can be ‘0’ to prohibit copying (Copy Protect 'on')3 0 Set to ‘1’ if emphasis used whilst bits 4-8 are normally at 0. Assuming an SP/DIF input ignores bit 0, bit 2 is 1 (Copy Protect set to off) and bit 3 is 0 (No audio emphasis) then transfer from an AES/EBU source is possible. Bit 2 is usually at 1 in the AES/EBU interface, allowing copying via this kind of connection, even though copying the same material via an S/PDIF circuit may be prohibited.
• When copying between DAT machines the interface conveys indexing information, also known as start IDs. You should therefore turn off auto-indexing on the receiving machine.
Copy Protection (CP) Apart from the Copy Protect bit (see above), you may not be able to copy material via an S/PDIF when:-
AES/EBU dan S/PDIF BLASTICA
Iwan B Pratama Hal 3
1. Using an early DAT recorder Older DAT machines prohibit recording at 44.1 kHz to stop piracy of CDs, although recording at 48 kHz is possible. Such machines can sometimes be modified to record at all rates.
2. Using a DAT recorder fitted with SCMS Modern domestic DAT machines use the Serial Copy Management System (SCMS), which allows the recorder to make digital copies from some sources, such as an original CD, but prevents you making a second-generation DAT recording from such a copy. Hence a recording’s owner can copy material but endless duplication is prevented. Analogue copying is unaffected but ‘second generation’ copying via S/PDIF is again prohibited.
• SCMS is part of the S/PDIF standard and is not used for data transferred via an AES/EBU interface. Some professional machines with S/PDIF connections aren’t fitted with SCMS.
• Enabling Copy Protect whilst making an original recording prevents any digital copying.
SCMS works by recording two bits of data, collectively known as ID6, which are buried within the data recorded onto DAT. These bits control Copy Protection (CP), working as follows:- ID6 Effect 0 Unlimited copying is permitted 10 No copies (Copy Protect) 11 One copy, in subsequent copies ID6 set to 10The system works in conjunction with the S/PDIF Category Codes (see above). The following table shows how this operates in an SCMS-equipped DAT recorder:-
Source Cat Code ID6 1st Copy ID6 2nd Copy ID6 3rd Copy ID6CD CD - Yes 10 No - - - CD (CP) CD - No - - - - - General Gen - Yes 11 Yes 10 No - General (CP) Gen - Yes 11 Yes 10 No - DAT* Gen 00 Yes 11 Yes 10 No - DAT (PR)* Gen 11 Yes 11 Yes 10 No - DAT (CP)* Gen 10 Yes 11 Yes 10 No - DAT* DAT 00 Yes 00 Yes 00 Yes 00 DAT (PR)* DAT 11 No - - - - - DAT (CP)* DAT 10 No - - - - - SCMS DAT DAT 00 Yes 00 Yes 00 Yes 00 SCMS DAT (PR) DAT-P 11 Yes• 10 No - - - SCMS DAT (CP) DAT 10 No - - - - - * Pre-SCMS DAT machine: can produce DAT or General Category Codes (PR) Prerecorded material, including an analogue recording made on an SCMS DAT machine (CP) Copy-protected material • During copying, the Category Code is set to DAT-P and the Copy Protect flag is sent via the S/PDIF connection. The receiver always records from a DAT-P source but the Copy Protect flag identifies the material to prevent any further copies.
• Digital copying from an SCMS DAT machine to a pre-SCMS machine is only possible if ID6 is set to 00. This means that even ‘one-copy-allowed’ tapes will be blocked.
• Some professional machines allow you to manually set the condition of ID6.
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AES/EBU dan S/PDIF BLASTICA
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Channel Status Bits Equipment responds to C-bits in various ways. Some devices ignore certain data bits, allowing the audio data to be used, whilst others ‘lock out’ the material if the C-bits aren’t as expected. The C-bits in the AES/EBU interface operate as follows:- Bit(s) Function 0 0 - Consumer material (Not used with this interface) 1 - Professional source of material 1 0 - Sample contains audio information 1 - Sample contains pure data 2, 3, 4 0, 0, 0 - Audio Emphasis is not indicated (Receiving equipment should default to 'off'. Manual switching may be used) 1, 0, 0 - No audio emphasis 1, 1, 0 - 50/15 µs emphasis 'on' 1, 1, 1 - CCIT J17 emphasis 'on' 5 0 - Sampling frequency of the source is locked (default) 1 - Sampling frequency of source is unlocked 6, 7 0, 0 - Sampling frequency is not indicated (Default 48 kHz, receiver may get rate from AES/EBU data) 0, 1 - 48 kHz 1, 0 - 44.1 kHz 1, 1 - 32 kHz 8-15 Channel mode 12-15 User bits management 16-18 Use of aux sample bits 18-23 Source word length & encoding history 24-31 Multi-channel function description 32-47 Reserved 48-63 Channel origin data - alphanumeric 64-79 Channel destination data - alphanumeric 80-95 Local sample address code (32 bit binary) 96-111 Time of day code (32 bit binary) 112 - 119 Reliability flags 120 - 127 Cyclic redundancy check character The usual states for these bits are as follows:- Bit Usual State Notes 0 1 (on) Can prevent connection to domestic equipment 1 0 (off) Indicates interface used for audio, not data 2 1 Can be ‘0’ if bits 3 and 4 do not indicate state of emphasis3 0 Set to ‘1’ if emphasis used 4 0 Set to ‘1’ only when CCIT J17 emphasis used Bits 5-7 are often set to 0 whilst many devices simply ignore bits 8 to 127. When copying between DAT machines the interface doesn’t convey indexing information, also known as start IDs. Synchronisation Techniques In a complex studio system there can be problems with timing between the different digital devices. There are two basic methods of synchronising the equipment in such a system:- 1. Each device uses the word clock extracted from its AES/EBU input This is similar to the genlock principle used in a television studio. Although perfectly adequate for joining two devices the following puzzles can be encountered in a complex system:-
AES/EBU dan S/PDIF BLASTICA
Iwan B Pratama Hal 7
1. Which of two devices, interconnected in both directions, acts as the master clock? Note that some devices only become slaves to their input clock when switched into record mode.
2. Which device do you choose as a clock to drive other devices? 3. What happens if the device used as a master clock is switched off by mistake?
2. A master clock unit is used as the reference for the entire studio This requires a separate clock input on each device, preferably the same as an AES/EBU audio input. Unfortunately many pieces of equipment have a BNC word clock (WC) input instead, which requires a suitable conversion box that accepts a standard AES/EBU master clock.
• An AES/EBU master clock signal can also convey audio material or other AES/EBU data.
In a television studio the master clock should be locked to the video frame reference frequency, as well as to the local source of SMPTE timecode. Locking all three of these signals together ensures that sound, vision and timing information are all in step. Master clocks can have two grades of accuracy, as shown below:- Grade Accuracy in Parts per Million (ppm)1 ±1 2 ±10 Time Delays Signals are synchronous if the start of the preamble is within a set margin of the reference clock. Outputs should be within 1⁄20 of the sampling frequency and inputs within 1⁄4. It may be necessary to align the timing of clock inputs by adjustments within the equipment, but once set, it shouldn’t need to be changed again, since drift isn’t usually a problem.
• Most timing delays are caused by the effects of different lengths of cable. • Any jitter in the digital waveform can cause serious problems. This blurs the edges of the
data causing an increase in noise on the audio output. Unsynchronised Devices Some sources can’t be locked to the studio system, such as a CD player that provides a nominal output of 44.1 kHz but really operates at 44.098 kHz. Unfortunately, such devices often lack a clock input, whilst others run at the wrong rate, such as a CD player supplying a 44.1 kHz signal to a studio working at 48 kHz. There are three solutions to such problems:-
1. Use Analogue Circuits This can be a very cost-effective remedy, causing little degradation in quality.
2. Use a Sample Rate Convertor (SRC) An expensive option, often best avoided, except when converting between frequencies. The device adds or deletes samples as necessary, sometimes causing a small amount of distortion.
3. Use a Synchroniser The ideal solution where rates are nominally the same. A buffer within the device fills or empties over time, depending on which frequency is highest. The buffer is reset at its mid-point, when a 10 ms cross-fade is made between the current sample and the mid-point data.
An alternative type of synchroniser, known as a sample-slip synchroniser, works by dropping or repeating samples during silent periods. Both types of synchroniser perform the special operation once every 20 minutes for a 10 parts per million (ppm) error in sample rate. Another option is a short-term SRC, which uses interpolation to fix the problem at a faster rate. Other Interfacing Problems Most of the following problems can be solved, albeit with expensive hardware:-
1. Differences in bits per sample Even when things seem normal there may be problems with signal levels or distortion when connecting devices that use a different bit counts, such as a 16 bit ADC connected
AES/EBU dan S/PDIF BLASTICA
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to 20-bit device. It may be necessary to ‘re-dither’ the material or use a digital gain control box.
2. Clash of C-bits with S/PDIF See the section in this document that gives further details on S/PDIF.
3. Some C-bits giving incorrect flags One example of this is pre-emphasised audio data which actually lacks a no-emphasis flag.
4. Pre-emphasis Digital processing can introduce or remove emphasis, although not all devices can deal with emphasised data. CCIT J17 (NICAM 728) emphasis should only be used for broadcasting purposes.
5. DC offset This problem is usually caused by a badly aligned analogue-to-digital converter (ADC). If this happens, you may have to use a special device to apply digital high-pass filtering.
6. Asynchronous timecode In this situation the timecode contained in the AES/EBU data gets out of step with real time. A special device can be used to remove or replace the timecode in the data.
Sony Digital Interface (SDIF2) SDIF2 is rarely encountered, although it’s commonly employed in early analogue-to-digital converters, as used to adapt a video recorder for digital sound recording. Electrical Hardware The interface uses separate 75 Ω BNC connectors for left (L), right (R) and word clock (WC) signals. Each unbalanced circuit uses 5 volt transistor-transistor logic (TTL) levels into a 75 Ω load with direct current (DC) coupling. In a multi-track system several SDIF2 signals, in the form of balanced RS-422 circuits, can be wired via a single 50-way D connector. Encoding SDIF2 conveys audio in 16-bit to 20-bit form, complete with Emphasis and Copy Protect flags. Unfortunately, some equipment ignores these flags, requiring the use of manual switching. Each 32-bit slot, which occupies one cycle of the word clock, is made up as follows:- Bit(s) Function 0-20 Audio data 21-28 Control Information (word ‘1’ only, start of block)21-25 00 - Fixed value 26-27 00 - Emphasis ‘off’ 01 - 50/15 µs emphasis ‘on’ 28 0 - Copying permitted (Copy Protect ‘off’) 1 - Copying prohibited (Copy Protect ‘on’) 29 0 - Not start of block 1 - Start of block (block flag) User Information (in words ‘2’ to ‘256’ only) 30-32 Divided into two ‘bits’ of 1.5 bits length Bit A followed by bit B:- Bit A: 0 - Start of block 1 - Not start of block Bit B: 0 - Not start of block 1 - Start of block The most significant bit (MSB) of the audio data is sent first, irrespective of the audio word length. Two’s complement coding is used and unused bits are blanked to 0.
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that two connectors are needed for an eight-track machine; one for the eight inputs and another for the outputs. In the same way, a 16-track machine requires a total of four A-DAT connectors. Digital audio cards that accommodate interface usually have a separate 9-pin synchronisation connector, which lets your computer control the transport mechanism of an associated A-DAT machine. Multi-channel Audio Digital Interface (MADI) MADI conveys multiple digital audio channels over a single circuit, each channel conforming to the AES/EBU standard. The 56 AES/EBU sub-frames are carried over a 75 Ω coaxial cable, up to 50 metres long, and fitted with BNC connectors. A separate synchronising clock cable is required. The interface operates at a fixed data rate of 125 Mbit/s, whatever sample rate is used, although the rate at the cable is reduced to 100 Mbit/s by using 4 to 5 bit encoding. This process breaks up each 32 bit sub-frame into 4-bit words, encoded as 5-bit words by means of a look-up table. This form of encoding reduces the direct current (DC) content of the signal. Synchronisation blocks are inserted at least once per frame. If the link isn’t used to it’s full capacity, extra synchronisation blocks are inserted to ‘fill’ the space on the bus. This is done using a device known as a Transparent Asynchronous Xmitter and Receiver Interface (TAXI). The AES/EBU sub-frames are as normal, except that the preamble bits 0-3 are replaced by:-
Bit Function 0 Frame Sync flag 1 Channel On/Off 2 A/B of stereo pair 3 Sync block
Tascam Digital Interface (TDIF) This interface is used for connecting Tascam multi-channel digital tape machines. It employs the same optical connector as the A-DAT interface, conveying the same number of channels, although it uses an entirely different data format. Yamaha 8 channel Interface This special interface consists of eight sets of RS-422 mono audio data and a clock signal, all wired via a 25-way D connector. The data itself can be in Yamaha, SDIF2 or Mitsubishi format. When used in Yamaha format this works in the same way as Yamaha’s stereo interface described above. However, if you feed stereo data into one channel of a multi-channel interface only the left-hand information is received. Also, there’s no direct method available for connecting two channels from a multi-channel device to an input that has a stereo interface. Digital Audio Connectors Most connectors for digital audio are in coaxial form, as used in radio frequency (RF) and video systems. They have a central signal pin, surrounded by a cylinder that provides screening. An appropriate coaxial cable should always be used. BNC This twist-and-lock coaxial connector is used for SDIF2 and various kinds of video interfaces. It’s also sometimes used for S/PDIF connections in professional systems. The 75 Ω type of connector is most common. Unfortunately, it’s rather too easy to plug this into a similar 50 Ω socket, which often results in jammed or damaged connectors.
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