RME Hdsp 9632
RME - Internal - 24-bit - With MIDI I/O
The RME Hammerfall HDSP 9632 PCI Card makes the dream of an all-in-one soundcard for every possible application come true. As usual, RME has not made any compromises: Latest 192kHz A/D/A with more than 110dB signal to noise ratio. Balanced stereo analog I/O, ADAT digital I/O, S/PDIF digital I/O with breakout cable for S/PDIF optical, MIDI I/O with 16 channels of hi-speed MIDI, and stereo headphone output. All inputs and outputs are simultaneously useable. Optional hi-quality analog expansion boa... Read more
Part Numbers: HDSP 9632, HDSP9632
UPC: 0003110007044, 874792002067
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RME Hdsp 9632
RME HDSP 9632
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Word Clock Module
for the HDSP 9632
Introduction.. 3 Package Contents.. 3 System Requirements.. 3 Power Supply.. 3 Brief Description and Characteristics. 3 Technical Specifications and Features.. 4 Hardware Installation.. 4 First Operation.. 5 Setup and Operation 9.1 General... 5 9.2 House Clock... 6 9.3 Extended Modes.. 6 9.4 Multi-card Operation... Word Clock 10.1 Technical Background... 7 10.2 Cabling and Termination.. Warranty... Appendix... 8 9
User's Guide HDSP 9632 WCM
Thank you for choosing the Hammerfall DSP series. This expansion board adds word clock input and output in professional quality to the HDSP 9632. A transformer-isolated input, a switchable termination and two low jitter outputs extend the powerful capabilities of the HDSP 9632.
2. Package Contents
Please ensure that all the following parts are included in the WCM's packaging box: Word Clock Module Flat ribbon cable, 10 pins Quick Info guide RME Driver CD
Note: The WCM needs no drivers and no additional software installation!
3. System requirements
One free slot in the PC's housing HDSP 9632
4. Power supply
The WCM gets its power from the HDSP 9632 via the supplied cable.
5. Brief Description and Characteristics
The 9632 Word Clock Module is a small companion board to RME's HDSP 9632, and needs no slot on the motherboard. It adds word clock input and outputs to this powerful digital interface card. All connectors are BNC jacks, so there's no hassle with adapters. The module includes a transformer-isolated input, a switchable termination with LED indication, and two low jitter outputs. HDSP 9632 includes SteadyClock, guaranteeing an excellent performance in all clock modes. Its highly efficient jitter suppression refreshes and cleans up any clock signal, and provides it as reference clock at the two outputs. The WCM is connected via one cable to the HDSP 9632, making it already fully operational. The drivers of the HDSP 9632 include support for the WCM. Thanks to several LEDs (power, termination, LOCK) and the highly integrated concept of installation, first operation and usage are simple even for the inexperienced user.
6. Technical Specifications and Features
Low jitter design: < 1 ns in PLL mode, all inputs Internal clock: 800 ps jitter, random spread spectrum Jitter suppression of external clocks: about 30 dB (2.4 kHz) Input PLL ensures zero dropout, even at more than 100 ns jitter High sensitive input stage works from 1 Vpp input level Rejects DC offsets in the word clock net Overvoltage protected input stage Short protected output stage Frequency range PLL input: 27 kHz - 200 kHz Frequency range output: 27 kHz - 200 kHz Input BNC, high impedance (> 10 kOhm) or terminated (75 Ohm) Output BNC, low impedance (10 Ohm) Power supply: from HDSP 9632 series board, 5 V DC, 20 mA Standard slot, board dimensions 95 x 50 mm
7. Hardware Installation
Important: Switch off the computer and remove the power cable from the power supply before fitting the WCM. 1. Disconnect the power cord and all other cables from the computer. 2. Remove the computer's housing; further information on how to do this can be obtained from your computers instruction manual. 3. Neutralize the static build up by touching the computer's metal-chassis before unpacking the WCM from the protective bag. 4. Connect WCM and HDSP 9632 using the supplied 10-pin flat cable. Plug one end into the connector Word Clock Module of the HDSP 9632, the other end into X102 on the WCM. Note: The cable's connectors will automatically plug in with the correct polarity. 5. Insert WCM into a free slot, press and fasten the screw (if any). As the WCM needs no slot on the motherboard we recommend to use a free slot above the last PCI- or AGP slot. 6. Replace the computer's housing and tighten the screws. 7. Reconnect the power cable and all other cables/connections.
8. First Operation
After fitting the WCM (see 7. Hardware Installation) and switching on the computer, activate the push button switch 'Term' located between the BNC jacks. Provided the module is correctly connected to the HDSP 9632, the yellow LED beside the switch must light up. * Technical note: The LED VD108, located on the WCM board, indicates whether the WCM gets the needed 5V from the HDSP or not. This LED is not visible from the outside.
9. Setup and Operation
As soon as a valid word clock signal is present the green 'Lock' LED beside the input jack lights up.
To switch from the HDSP's internal clock to the WCM's clock, activate the mode 'Word Clock' in the field 'Pref. Sync Ref' of the Settings dialog. Also select 'AutoSync' in the field 'Clock Mode'. The field 'System Clock' will indicate that the card has changed into clock mode 'Slave'. The external signal is now used as reference. At the same time, the frequency (Freq.) of the input signal is displayed.
The word clock outputs of the WCM are always active. They provide the current sample frequency of the HDSP 9632 as word clock signal. As long as the HDSP 9632 operates in 'Master' mode (field 'Clock Mode'), the word clock will be fixed to the current sample rate. In 'AutoSync' mode the sample rate is identical to the one present at the currently chosen input (Pref. Sync Ref). Without a valid input signal, the card will change between the inputs automatically. As long as no valid input signal is found, the card will stay in Master mode. The word clock signal received by the WCM be distributed to other devices by using the WCM's word clock output. With this the usual T-adapter can be avoided, and the WCM operates as Signal Refresher. This kind of operation is highly recommended, because Input and output are phase-locked and in phase (0) to each other SteadyClock removes nearly all jitter from the input signal the exceptional input of the WCM (1 Vpp sensitivity instead of the usual 2.5 Vpp, dc cut, Signal Adaptation Circuit) plus SteadyClock guarantee a secure function also with most critical word clock signals.
9.2 House Clock
The high quality of the word clock signal offered by the HDSP 9632 allow to use the HDSP 9632 as house clock generator. To simpify this application, the WCM offers two electronically decoupled word clock outputs.
9.3 Extended Modes
Due to the HDSP 9632's outstanding clock control a synchronization of the output signal to the input signal is not only possible at identical sample rates, but also at half, quarter, double and quad sample rates. A playback of 192 kHz can be easily synchronized via a 48 kHz clock source. Example 1: A playback or recording at 44.1 kHz can be synchronized via an external signal of 44.1 kHz, 88.2 kHz or 176.4 kHz. Example 2: A playback or recording at 192 kHz can be synchronized via an external signal of 48 kHz, 96 kHz or 192 kHz. The input accepts all those frequencies fully automatically. Like all ADAT ports, the word clock output operates in Single Speed mode only. At 96 kHz or 192 kHz, the word clock output will therefore be a 48 kHz signal. Later a flash update will allow to choose double and quad speed for the word clock output.
9.4 Multi-card Operation
The WCM does not support multi-card operation directly. But that's not necessary anyway, as multiple HDSP 9632, and also combinations with the HDSP 9652, can be sychronised in several ways. Also for such a system of multiple cards, only one word clock input can be used at all. The synchronization of other cards to the word clock input of the HDSP 9632 can be done by internal cabling from Sync Out (card 1) to CD/Sync/AEB In (card 2) internal cabling ADAT Out (card 1) to CD/Sync/AEB In (card 2) usage of the ADAT or SPDIF optical output (card 1) to input card 2 word clock output (card 1) to word clock input (card 2)
In all cases the cards will operate with sample-accuracy, provided the correct setting has been chosen in the Settings dialog.
10. Word Clock
10.1 Technical Background
In the analog domain one can connect any device to another device, a synchronization is not necessary. Digital audio is different. It uses a clock, the sample frequency. The signal can only be processed and transmitted when all participating devices share the same clock. If not, the signal will suffer from wrong samples, distortion, crackle sounds and drop outs. AES/EBU, SPDIF and ADAT are self-clocking, an additional word clock connection in principle isn't necessary. But when using more than one device simultaneously problems are likely to happen. For example any self-clocking will not work in a loop cabling, when there is no 'master' (main clock) inside the loop. Additionally the clock of all participating devices has to be synchronous. This is often impossible with devices limited to playback, for example CD players, as these have no SPDIF input, thus can't use the self clocking technique as clock reference. In a digital studio synchronisation is maintained by connecting all devices to a central sync source. For example the mixing desk works as master and sends a reference signal, the word clock, to all other devices. Of course this will only work as long as all other devices are equipped with a word clock or sync input, thus being able to work as slave (some professional CD players indeed have a word clock input). Then all devices get the same clock and will work in every possible combination with each other. But word clock is not only the 'great problem solver', it also has some disadvantages. The word clock is based on a fraction of the really needed clock. For example SPDIF: 44.1 kHz word clock (a simple square wave signal) has to be multiplied by 256 inside the device using a special PLL (to about 11.2 MHz). This signal then replaces the one from the quartz crystal. Big disadvantage: because of the high multiplication factor the reconstructed clock will have great deviations called jitter. The jitter of a word clock is typically 15 times higher as when using a quartz based clock. We even know a Synchronizer which generates word clock signals digitally (!) with more than 30 ns jitter, and - when used as house clock for the whole studio - lowers the reliability and audio quality of all attached devices. The end of these problems should have been the so called Superclock, which uses 256 times the word clock frequency. This equals the internal quartz frequency, so no PLL for multiplying is needed and the clock can be used directly. But reality was different, the Superclock proved to be much more critical than word clock. A square wave signal of 11 MHz distributed to several devices - this simply means to fight with high frequency technology. Reflections, cable quality, capacitive loads - at 44.1 kHz these factors may be ignored, at 11 MHz they are the end of the clock network. Additionally it was found that a PLL not only generates jitter, but also also rejects disturbances. The slow PLL works like a filter for induced and modulated frequencies above several kHz. As the Superclock is used without any filtering such a kind of jitter and noise suppression is missing. No wonder Superclock did not become a conmmonly accepted standard. The actual end of these problems is offered by the SteadyClock technology of the HDSP 9632. Combining the advantages of modern and fastest digital technology with analog filter technique, re-gaining a low jitter clock signal of 11 MHz from a slow word clock of 44.1 kHz is no problem anymore. Additionally, jitter on the input signal is highly rejected, so that even in real world usage the re-gained clock signal is of highest quality.
10.2 Cabling and Termination
Word clock signals are usually distributed in the form of a network, split with BNC T-adapters and terminated with resistors. We recommend using off-the-shelf BNC cables to connect all devices, as this type of cable is used for most computer networks. You will find all the necessary components (T-adapters, terminators, cables) in most electronics and/or computer stores. The ideal word clock signal should be a 5 Volt square wave having the same frequency as the sample rate. This signal will generate harmonics up to more than 500 kHz. To avoid voltage loss and reflections, both the cable itself and the terminating resistor should have an impedance of 75 Ohm. If the voltage is too low, synchronization will fail. High frequency reflection effects can cause both jitter and sync failure. In practice, the situation has improved in recent years. The relatively low frequency of word clock signals is not a problem for modern electronic circuits. Because of the higher voltage, big word clock networks are often more stable and reliable if cables are not terminated at all. Also, 75 Ohm cable is almost impossible to find these days. 50 Ohm cable is standard - this will also work as long as the termination resistors are 75 Ohm. The WCM's input is a high impedance type to offer highest flexibility for the user. In case a termination according to the standard is necessary (because the WCM is the last device in a chain of several devices), push the switch with a small tool, so that the yellow LED lights up. In case the WCM resides within a chain of devices receiving word clock, plug a T-adapter into the WCM'S BNC input jack, and the cable supplying the word clock signal to one end of the adapter (as above), but connect the free end to the next device in the chain via a further BNC cable. The last device in the chain should be terminated using another T-adapter and a terminator plug as described in the previous paragraph. Some devices (like the WCM) have switchable 75 Ohm resistors, which saves both T-adapter and terminator. Due to the outstanding SteadyClock technology of the HDSP 9632, we recommend not to pass the input signal via T-adapter, but to use the WCM's two word clock outputs. Thanks to SteadyClock, the input signal will both be freed from jitter and - in case of loss or drop out be held at the last valid frequency.
Each individual Word Clock Module undergoes comprehensive quality control and a complete test in a PC environment at RME before shipping. The usage of high grade components allows us to offer a full two year warranty. We accept a copy of the sales receipt as valid warranty legitimation. RMEs replacement service within this period is handled by the retailer. If you suspect that your card is faulty, please contact your local retailer. The warranty does not cover damage caused by improper installation or maltreatment - replacement or repair in such cases can only be carried out at the owners expense. RME does not accept claims for damages of any kind, especially consequential damage. Liability is limited to the value of the WCM. The general terms of business drawn up by Synthax Audio AG apply at all times.
RME news, driver updates and further product information are available on our website: http://www.rme-audio.com If you prefer to read the information off-line, you can load a complete copy of the RME website from the RME Driver CD (in the \rmeaudio.web directory) into your browser. Distributor in Germany: Synthax, Am Pfanderling 62, D-85778 Haimhausen, Tel.: (49) 08133 / 91810 Manufacturer: Ingenieurbuero Mueller, Leipziger Str. 32, D-09648 Mittweida
Trademarks All trademarks and registered trademarks belong to their respective owners. RME, DIGI96, SyncAlign, DIGICheck, SyncCheck, Hammerfall and ZLM are registered trademarks of RME Intelligent Audio Solutions. Alesis and ADAT are registered trademarks of Alesis Corp. ADAT optical is a trademark of Alesis Corp. Windowsis a trademark of Microsoft Corp. Synthax is a registered trademark of Synthax OHG.
Copyright RME, Matthias Carstens, 07/03. Version 1.0 Although the contents of this Users Guide have been thoroughly checked for errors, RME can not guarantee that it is correct throughout. RME does not accept responsibility for any misleading or incorrect information within this guide. Lending or copying any part of the guide or the RME drivers CD, or any commercial exploitation of these media without express written permission from RME Intelligent Audio Solutions is prohibited. RME reserves the right to change specifications at any time without notice.
CE This device has been tested and found to comply with the limits of the European Council Directive on the approximation of the laws of the member states relating to electromagnetic compatibility (EMVG) according to EN 55022 class B and EN50082-1.
FCC Compliance Statement Certified to comply with the limits for a Class B computing device according to subpart J or part 15 of FCC rules. See instructions if interference to radio reception is suspected.
FCC Warning This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This device complies with part 15 of FCC rules. Operation is subject to the following two conditions: 1. This device may not cause harmful interference 2. This device must accept any interference received, including interference that may cause undesired operation. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures: Reorient or relocate the receiving antenna Increase the seperation between the equipment and receiver Connect the equipment into an outlet on a circuit different from that to which the receiver is connected Consult the dealer or an experienced radio/TV technician for help. In order for an installation of this product to maintain compliance with the limits for a Class B device, shielded cables must be used for the connection of any devices external to this product.
Hammerfall DSP System HDSP 9632
TotalMix 24 Bit / 192 kHz SyncAlign
PCI Busmaster Digital I/O System 2 + 8 + 2 Channels SPDIF / ADAT / Analog Interface 24 Bit / 192 kHz Digital Audio 24 Bit / 192 kHz Stereo Analog Monitor MIDI I/O
Introduction.. 6 Package Contents.. 6 System Requirements.. 6 Brief Description and Characteristics. 6 Hardware Installation.. 7 Hardware Connectors 6.1 External Connectors.. 7 6.2 Internal Connectors.. Accessories.. Warranty... Appendix... 6
Driver Installation and Operation - Windows
10 Driver and Firmware 10.1 Driver Installation..12 10.2 Driver Update...12 10.3 Deinstalling the Drivers..12 10.4 Firmware Update..Configuring the HDSP 9632 11.1 Settings Dialog...13 11.2 Settings Dialog DDS..16 11.3 Clock Modes Synchronization.Operation and Usage 12.1 Playback..19 12.2 DVD Playback (AC-3 / DTS) under MME.20 12.3 Low Latency under MME.21 12.4 Multi-client Operation...21 12.5 Digital Recording..22 12.6 Analog Recording..Operation under ASIO 2.0 13.1 General..23 13.2 Known Problems..Operation under GSIF..Using more than one Hammerfall DSP.DIGICheck..Hotline Troubleshooting 17.1 General..26 17.2 Installation...Diagrams 18.1 Channel Routing ASIO 96 kHz..28 18.2 Channel Routing MME 96 kHz..29
User's Guide HDSP System HDSP 9632 RME
Driver Installation and Operation - Mac OS X
19 Driver and Flash Update 19.1 Driver Installation... 32 19.2 Driver Update.. 32 19.3 Flash Update... Configuring the HDSP 9632 20.1 Settings Dialog.. 33 20.2 Settings Dialog DDS.. 35 20.3 Clock Modes Synchronization.. Mac OS X FAQ 21.1 Round about Driver Installation.. 38 21.2 MIDI doesn't work... 38 21.3 Supported Sample Rates.. 39 21.4 Repairing Disk Permissions. 39 21.5 PCI Compatibility... 39 21.6 Various Information.. Hotline Troubleshooting.. Diagram: Channel Routing at 96 kHz. 41
Connections and TotalMix
24 Analog Connections 24.1 Line Inputs.. 44 24.2 Line Outputs.. 45 24.3 Phones.. Digital Connections 25.1 ADAT... 46 25.2 SPDIF... 46 25.3 MIDI... Word Clock 26.1 Word Clock Input and Output.. 48 26.2 Technical Description and Background. 49 26.3 Cables and Termination. 50 26.4 General Operation.. TotalMix: Routing and Monitoring 27.1 Overview... 51 27.2 The User Interface.. 53 27.3 Elements of a Channel.. 54 27.4 Tour de TotalMix.. 54 27.5 Submix View... 54 27.6 Mute and Solo... 56 27.7 Quick Access Panel.. 57 27.8 Presets.. 57 27.9 Monitor Panel.. 59 27.10 Preferences... 59 27.11 Editing the Names.. 60 27.12 Hotkeys... 61 27.13 Menu Options.. 62 27.14 Level Meter... 63
TotalMix: The Matrix 28.1 Overview..64 28.2 Elements of the Matrix View.64 28.3 Usage...64 28.4 Advantages of the Matrix.TotalMix Super-Features 29.1 ASIO Direct Monitoring (Windows only).65 29.2 Selection and Group based Operation.66 29.3 Copy Routings to other Channels..66 29.4 Delete Routings..66 29.5 Recording a Subgroup (Loopback).67 29.6 Using external Effects Devices..TotalMix MIDI Remote Control 30.1 Overview..69 30.2 Setup...69 30.3 Operation...69 30.4 Mapping...70 30.5 Simple MIDI Control..71 30.6 Loopback Detection..71
Tech Info...74 Technical Specifications 32.1 Analog...75 32.2 Digital...76 32.3 Digital Inputs..76 32.4 Digital Outputs...77 32.5 MIDI..77 32.6 Transfer Modes: Resolution/Bits per Sample.Technical Background 33.1 Lock and SyncCheck..78 33.2 Latency and Monitoring..79 33.3 DS Double Speed..80 33.4 QS Quad Speed...81 33.5 AES/EBU SPDIF..81 33.6 Noise Level in DS / QS Mode..82 33.7 SteadyClock..Diagrams 34.1 Block Diagram HDSP 9632..83 34.2 Connector Pinouts...84 34.3 Overview Channels and Expansion Boards.85
Thank you for choosing the RME Hammerfall DSP system. This unique audio system is capable of transferring analog and digital audio data directly to a computer from practically any device. The latest Plug and Play technology guarantees a simple installation, even for the inexperienced user. The numerous unique features and well thought-out configuration dialog puts the Hammerfall DSP at the very top of the range of computer-based audio interfaces. The package contains drivers for Windows 2000 SP4, Windows XP, and Mac OS X. Our high-performance philosophy guarantees maximum system performance by executing as many functions as possible not in the driver (i.e. the CPU), but directly within the audio hardware.
2. Package Contents
Please check your Hammerfall DSP system's package contains each of the following: HDSP 9632 PCI card Quick Info guide RME Driver CD Digital adapter cable (phono / phono to D-type 9 pin) Analog adapter cable (phono / phono / TRS / MIDI to D-type 15 pin) Internal cable (2-core) 1 optical cable (TOSLINK)
3. System Requirements
Windows 2000 SP4, Windows XP, Mac OS X (10.28 or higher) PCI Interface: a free PCI rev. 2.1 Busmaster slot
4. Brief Description and Characteristics
Hammerfall design: 0% (zero!) CPU load, even using all 32 ASIO channels All settings can be changed in real-time Analog, ADAT and SPDIF I/Os can be used simultaneously 8 buffer sizes/latencies available: 1.5 / 3 / 6 / 12 / 23 / 46 / 93 / 186 ms 4 channels 96 kHz/24 bit record/playback via ADAT optical (S/MUX) Clock modes slave and master Automatic and intelligent master/slave clock control Unsurpassed Bitclock PLL (audio synchronization) in ADAT mode TotalMix for latency-free submixes and perfect ASIO Direct Monitoring SyncAlign guarantees sample aligned and never swapping channels SyncCheck tests and reports the synchronization status of input signals 1 x MIDI I/O, 16 channels high-speed MIDI DIGICheck DSP: Level meter in hardware, peak- and RMS calculation SteadyClock: Jitter-immune, super-stable digital clock TotalMix: 512 channel mixer with 40 bit internal resolution Optional word clock input and output
5. Hardware Installation
Before installing the PCI card, please make sure the computer is switched off and the power cable is disconnected from the mains supply. Inserting or removing a PCI card while the computer is in operation can cause irreparable damage to both motherboard and card! 1. Disconnect the power cord and all other cables from the computer. 2. Remove the computer's housing. Further information on how to do this can be obtained from your computers instruction manual. 3. Important: Before removing the HDSP 9632 from its protective bag, discharge any static in your body by touching the metal chassis of the PC. 4. Prior to installation: Connect the HDSP 9632 card to any Expansion Board (if present) using the supplied flat ribbon cable. Please read the Expansion Board's manual for more details. 5. Insert the HDSP 9632 firmly into a free PCI slot, press and fasten the screw. 6. If present, insert the Expansion Board(s) and fasten the screw(s). 7. Replace the computer's housing. 8. Reconnect all cables including the power cord.
6. Hardware - Connectors
6.1 External Connectors
The bracket of the HDSP 9632 has one ADAT optical input and output, a 9-pin and a 15-pin D-type socket. The included breakout cable provides all the analog and digital connections. The ADAT I/O can also be used as optical SPDIF I/O, if set up accordingly in the Settings dialog. The 9-pin digital breakout cable has two RCA connectors as coaxial SPDIF I/O. The red phono socket is the output. The breakout cable BO968 (option) has the same RCA connectors, but adds an XLR AES/EBU input and output. The 15-pin analog breakout cable has four RCA connectors (stereo analog I/O), a 1/4" TRS jack (headphones), and two 5-pin DIN connectors (MIDI I/O). Using the optional analog XLR breakout cable BO9632-XLRMKH, the HDSP 9632 offers balanced Line inputs and outputs via female and male XLR connectors.
Common Problems Please note that Gigastudio is running unexpectedly in the background (thus blocking its assigned audio channels), as soon as the Gigastudio MIDI ports are used even when Gigastudio itself hasn't been started. This causes a lot of confusion, as the driver seems to behave completely buggy, and the user does not recognize the simple reason for it for example simultaneous operation of ASIO and GSIF on the same channels. If Gigastudio starts up properly, loads gig files too, but won't play at all even when using the virtual keyboard: Go to Hardware/Routing and select a valid MIDI input port. Note that blank is not valid, but <none> is.
15. Using more than one Hammerfall DSP
The current drivers support operation of up to three Hammerfall DSP systems. Different I/Oboxes may be used, in any combination. HDSP 9632, HDSP 9652, Digiface, Multiface and Multiface II use the same driver, therefore can be used at the same time. Please note that only one ADAT Sync In can be used (of course). All units have to be in sync, i.e. have to receive valid sync information either via word clock or by using AutoSync and feeding synchronized signals. If one of the HDSP systems is set to clock mode Master, all others have to be set to clock mode AutoSync, and have to be synced from the master, for example by feeding word clock. The clock modes of all units have to be set up correctly in their Settings dialog. If all units are fed with a synchronous clock, i.e. all units show Sync in their Settings dialog, all channels can be used at once. This is especially easy to handle under ASIO, as the ASIO driver presents all units as one.
Note: TotalMix is part of the hardware of each HDSP system. Up to three mixers are available, but these are separated and can't interchange data. Therefore a global mixer for all units is not possible.
The DIGICheck software is a unique utility developed for testing, measuring and analysing digital audio streams. Although this Windows software is fairly self-explanatory, it still includes a comprehensive online help. DIGICheck 4.4 operates as multi-client ASIO host, therefore can be used in parallel to any software, be it MME, ASIO or GSIF, with both inputs and outputs (!). The following is a short summary of the currently available functions: Level Meter. High precision 24-bit resolution, 2/8/18 channels. Application examples: Peak level measurement, RMS level measurement, over-detection, phase correlation measurement, dynamic range and signal-to-noise ratios, RMS to peak difference (loudness), long term peak measurement, input check. Oversampling mode for levels higher than 0 dBFS. Vertical and horizontal mode. Slow RMS and RLB weighting filter. Supports visualization according to the K-system. Hardware Level Meter for Input, Playback and Output. As above, received pre-calculated directly from the HDSP system hardware with near zero CPU load. Spectral Analyser. World wide unique 10-, 20- or 30-band display in analog bandpass-filter technology. 192 kHz-capable! Vector Audio Scope. World wide unique Goniometer showing the typical afterglow of an oscilloscope-tube. Includes Correlation meter and level meter. Totalyser. Spectral Analyser, Level Meter and Vector Audio Scope in a single window. Bit Statistics & Noise. Shows the true resolution of audio signals as well as errors and DC offset. Includes Signal to Noise measurement in dB and dBA, plus DC measurement. Channel Status Display. Detailled analyzis and display of SPDIF and AES/EBU Channel Status data. Completely multi-client. Open as many measurement windows as you like, on any channels and inputs or outputs! To install DIGICheck, go to the \DIGICheck directory on the RME Driver CD and run setup.exe. Follow the instructions prompted on the screen. DIGICheck is conctantly improved. The latest version is always found on our website www.rme-audio.de, section Downloads/Tools.
The dialog 'New hardware component found does not appear: Check whether the PCI interface is correctly inserted in the PCI slot. The card and drivers have been installed correctly, but playback does not work: Check whether the Hammerfall DSP appears in the Device Manager. If the ' Hammerfall DSP device has a yellow exclamation mark, then there is an address or interrupt conflict. Even if there is no yellow exclamation mark, it is worth checking the Resources tab anyway.
18.1 Channel Routing ASIO at 96 kHz
This diagram shows the signal paths in ASIO double speed mode (88.2 / 96 kHz). The devices available via the ASIO driver have been designed to avoid conflicts in normal operation. Record and playback are identical. Device: The device name in the audio application SR: Sample Rate Device name code: Channel in ASIO host, interface, HDSP 9632, card number
18.2 Channel Routing MME at 96 kHz
This diagram shows the signal paths in MME double speed mode (88.2 / 96 kHz). The devices available via the MME wave driver have been designed to avoid conflicts in normal operation, which is why channels 5, 6, 7 and 8 of the ADAT device have been omitted. Record and playback are identical. Device: The device name in the audio application SR: Sample Rate
Driver Installation and Operation Mac OS X
19. Driver and Flash Update
19.1 Driver Installation
First fit the card (see 5. Hardware Installation), then switch on the computer and install the drivers from the RME Driver CD. The driver file is located in the folder Hammerfall DSP. Installation works automatically by a double-click on the file hdsp.mpkg. RME recommends to download the latest driver version from the RME website! If done, the procedure is as follows: Double-click onto hdsp_xx.gz to expand the archive file to hdsp_xx.tar and the folder HDSP_xx, which includes the driver file hdsp.mpkg. Installation works automatically by a double-click on this file. During driver installation the programs Settings and Mixer (TotalMix) will also be installed. Both programs start automatically as soon as a HDSP system is detected. They stay in the dock when exited, and remove themselves automatically from the dock when the HDSP system is removed. Reboot the computer when installation is done.
19.2 Driver Update
In case of a driver update it's not necessary to remove the old driver first, it will be overwritten during the installation. Exception: driver updates from version <1.6. Remove the former Settings dialog and TotalMix from the Login Items, and delete both files from your hard drive! This driver version did not have the features AutoLoad, Dock Lock and AutoRemove. Therefore one has to make sure that both programs have been removed from the system, to prevent the old Settings dialog and TotalMix from being loaded.
19.3 Flash Update
The Flash Update Tool updates the HDSP 9632 card to the latest firmware version. It requires an already installed driver. Start the program HDSP Flash. The Flash Update Tool displays the current revision of the HDSP interface, and whether it needs an update or not. If so, then simply press the 'Update' button. A progress bar will indicate when the flash process is finished. The bar moves slowly first (program), then faster (verify). If more than one interface card is installed, all cards can be flashed by changing to the next tab and repeating the process. After the update the PCI card needs to be resettet. This is done by powering down and shutting off the PC. A warm boot is not enough! When the update fails (status: failure), the card's second BIOS will be used from the next cold boot on (Secure BIOS Technology). Therefore the card stays fully functional. The flash process should then be tried again on a different computer.
20. Configuring the HDSP 9632
20.1 Settings Dialog
Configuring the HDSP 9632 is done via its own settings dialog. The panel 'Settings' can be opened by clicking on the hammer icon in the dock. The mixer of the HDSP 9632, TotalMix, can be opened by clicking on the mixer icon in the dock. The Hammerfall DSPs hardware offers a number of helpful, well thought-of practical functions and options which affect how the card operates - it can be configured to suit many different requirements. The following is available in the 'Settings' dialog: Input selection Level of analog I/Os Configuration of digital I/Os Synchronization behaviour State of input and output Current sample rate Latency
Any changes performed in the Settings dialog are applied immediately - confirmation (e.g. by exiting the dialog) is not required. However, settings should not be changed during playback or record if it can be avoided, as this can cause unwanted noises. The status displays at the bottom of the dialog box give the user precise information about the current status of the system, and the status of all digital signals. SyncCheck indicates whether there is a valid signal (Lock, No Lock) for each input (Word Clock, ADAT, SPDIF), or if there is a valid and synchronous signal (Sync). The AutoSync Ref(erence) display shows the input and frequency of the current sync source. SPDIF In Defines the input for the SPDIF signal. 'Optical' relates to the optical TOSLINK input, 'Coaxial' to the RCA socket, 'Internal' to the jumper CD/AEB/SYNC IN, 'AES' to the optional XLR cable. SPDIF Out The SPDIF output signal is constantly available at the phono plug. After selecting 'Optical' it is also routed to the optical TOSLINK output. For further details about the settings Professional, Emphasis and Non-Audio, please refer to chapter 25.2. SPDIF Freq. Displays the sample rate of the signal at the SPDIF input.
Clock Mode The unit can be configured to use its internal clock source (Master), or the clock source predefined via Pref Sync Ref (AutoSync). Pref Sync Ref Used to pre-select the desired clock source. If the selected source isn't available, the unit will change to the next available one. The current clock source and sample rate is displayed in the AutoSync Ref display. The automatic clock selection checks and changes between the clock sources Word Clock, ADAT optical and SPDIF. Word Clock Out The word clock output signal usually equals the current sample rate. Selecting Single Speed causes the output signal to always stay within the range of 32 kHz to 48 kHz. So at 96 kHz sample rate, the output word clock is 48 kHz. AEB AEB activates the internal connector as ADAT input instead of the optical TOSLINK. An expansion board (AEB4-I, AEB8-I, TEB) can be connected here.
Multicard Operation OS X supports more than one audio device. Their simultaneous usage within one program had been limited to Motu's Digital Performer until 10.3.9. Since 10.4 (Tiger) Core Audio offers the function Aggregate Devices, which allows to combine several devices into one, so that a multi-device operation is now possible with any software. The Hammerfall DSP driver adds a number to each unit, so they are fully accessible in any multicard-capable software.
22. Hotline Troubleshooting
The newest information can always be found on our website www.rme-audio.com, section Support, Macintosh OS. The 8 ADAT channels dont seem to work The optical output has been switched to SPDIF. The ADAT playback devices are still usable by routing and mixing them in TotalMix to other outputs. Playback works, but record doesnt: Check that there is a valid signal at the input. Check whether the Hammerfall DSP has been selected as recording device in the audio application. Crackle during record or playback: Increase the number and size of buffers in the application. Try different cables to rule out any defects here. The card and drivers have been installed correctly, but playback does not work: Is Hammerfall DSP listed in the System Profiler/PCI? (Vendor 10EE, Device ID 3FC5). Has Hammerfall DSP been selected as current playback device in the audio application?
23. Diagram: Channel Routing at 96 kHz
This diagram shows the signal paths in double speed mode (88.2 / 96 kHz). The last four channels of the ADAT port have no function anymore in Core Audio, but are used by the hardware to transmit data at double sample rate.
24. Analog Connections
24.1 Line Inputs
The HDSP 9632 has an unbalanced stereo Line input via RCA connectors. The optional analog XLR breakout cable turns the inputs into fully balanced ones. The electronic input stage is built in a servo balanced design which handles unbalanced and balanced signals correctly, automatically adjusting the level reference. When using unbalanced cables with the XLR breakout cable: be sure to connect the 'ring' contact of a stereo TRS jack, and pin 3 of a XLR jack, to ground. Otherwise noise may occur, caused by the unconnected negative input of the balanced input. One of the main issues when working with an AD-converter is to maintain the full dynamic range within the best operating level. Therefore the HDSP 9632 internally uses hi-quality electronic switches, which allow for a perfect adaptation of all inputs to the three most often used studio levels. The 'standardized' studio levels do not result in a (often desired) full scale level, but take some additional digital headroom into consideration. The amount of headroom is different in different standards, and again differently implemented by different manufacturers. Because of this we decided to define the levels of the HDSP 9632 in the most compatible way. Reference Lo Gain +4 dBu -10 dBV 0 dBFS @ +19 dBu +13 dBu +2 dBV Headroom 15 dB 9 dB 12 dB
With +4 dBu selected, the according headroom meets the latest EBU recommendations for Broadcast usage. At -10 dBV a headroom of 12 dB is common practice, each mixing desk operating at -10 dBV is able to send and receive much higher levels. Lo Gain is best suited for professional users who prefer to work balanced and at highest levels. Lo Gain provides 15 dB headroom at +4 dBu nominal level. The above levels are also found in our ADI-8 series of AD/DA converters, the Multiface, and even in our Mic-Preamps QuadMic and OctaMic. Therefore all RME devices are fully compatible to each other.
24.2 Line Outputs
The short circuit protected, low impedance line outputs (channels 11/12) are available as unbalanced outputs via phono breakout cable. The optional analog XLR breakout cable provides XLR connectors and fully balanced operation. The electronic output stage does not operate servo balanced! When connecting unbalanced equipment, make sure pin 3 of the XLR output is not connected. A conection to ground will cause a decreased THD (higher distortion)! To maintain an optimum level for devices connected to the analog outputs, the HDSP 9632 internally uses hi-quality electronic switches, which allow for a perfect adaptation of all outputs to the three most often used studio levels. As with the analog inputs, the analog output levels are defined to maintain a problem-free operation with most other devices. The headroom of the HDSP 9632 lies between 9 and 15 dB, according to the chosen reference level: Reference Hi Gain +4 dBu -10 dBV 0 dBFS @ +19 dBu +13 dBu +2 dBV Headroom 15 dB 9 dB 12 dB
With +4 dBu selected, the according headroom meets the latest EBU recommendations for Broadcast usage. At -10 dBV a headroom of 12 dB is common practice, each mixing desk operating at -10 dBV is able to send and receive much higher levels. Hi Gain is best suited for professional users who prefer to work balanced and at highest levels. Hi Gain provides 15 dB headroom at +4 dBu nominal level. When using the analog XLR breakout cable, make sure 'Breakout Cable / XLR' is selected in the Settings dialog. Else the analog output level will be 6 dB too high! Note that not checking XLR with balanced cables will cause an output level of +25 dBu @ 0 dBFS. Reducing the output level by 1 dB within TotalMix then makes the HDSP 9632's analog outputs compatible to SMPTE (+24 dBu, 15dB headroom).
Note: Recordings with (pre-) emphasis show a treble boost (50/15 s), which has to be compensated at playback. Therefore, when selecting Emphasis all analog outputs will be processed by a treble filter based on 50/15s, which sounds like a high cut. The HDSP 9632s new output header is optimized for largest compatibility with other digital devices: 32 kHz, 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, 192 kHz depending on the current sample rate Audio use, Non-Audio No Copyright, Copy Permitted Format Consumer or Professional Category General, Generation not indicated 2-channel, No Emphasis or 50/15 s Aux bits Audio Use For AES/EBU output operation the optional digital XLR breakout cable is required, as the phono output has too low voltage for a reliable operation in AES format. Note that most consumer HiFi equipment (with optical or phono SPDIF inputs) will only accept signals in Consumer format! The audio bit in the header can be set to 'Non-Audio'. This is often necessary when Dolby AC-3 encoded data is sent to external decoders (surround-sound receivers, television sets etc. with AC-3 digital inputs), as these decoders would otherwise not recognize the data as AC-3.
The HDSP 9632 offers one MIDI I/O via two 5-pin DIN connectors. The MIDI ports are added to the system by the driver. Using MIDI capable software, these ports can be accessed under the name HDSP MIDI. Using more than one HDSP 9632, the operating system adds a consecutive number to the port name, like HDSP MIDI In (2) etc. The MIDI In port is available for both GSIF (GSIF-2 Low Latency) and standard MME MIDI simultaneously.
26. Word Clock
26.1 Word Clock Input and Output (optional Word Clock Module)
SteadyClock guarantees an excellent performance in all clock modes. Based on the highly efficient jitter suppression, the HDSP 9632 refreshes and cleans up any clock signal, and provides it as reference clock at the BNC output (see section 33.7). Input The HDSP 9632's transformer isolated word clock input is active when Pref. Sync Ref in the Settings dialog has been switched to Word Clock, the clock mode AutoSync has been activated, and a valid word clock signal is present. The signal at the BNC input can be Single, Double or Quad Speed, the HDSP 9632 automatically adapts to it. As soon as a valid signal is detected, the green LED is lit, and the Settings dialog shows either Lock or Sync (see chapter 33.1). Thanks to RME's Signal Adaptation Circuit, the word clock input still works correctly even with heavily mis-shaped, dc-prone, too small or overshoot-prone signals. Thanks to automatic signal centering, 300 mV (0.3V) input level are sufficient in principle. An additional hysteresis reduces sensitivity to 1.0 V, so that over- and undershoots and high frequency disturbances don't cause a wrong trigger. The word clock input is shipped as high impedance type (not terminated). A push switch allows to activate internal termination (75 Ohms). The switch is found beside the word clock input socket. Use a small pencil or similar and carefully push the blue switch so that it snaps into its lock position. The yellow LED will be lit when termination is active. Another push will release it again and de-activate the termination. Output The word clock output of the HDSP 9632 is constantly active, providing the current sample frequency as word clock signal. As a result, in Master mode the provided word clock is defined by the currently used software. In Slave mode the provided frequency is identical to the one present at the currently chosen clock input. When the current clock signal fails, the HDSP 9632 switches to Master mode and adjusts itself to the next, best matching frequency (44.1 kHz, 48 kHz etc.). Selecting Single Speed in the Settings dialog (Word Clock Out) causes the output signal to always stay within the range of 32 kHz to 48 kHz. So at 96 kHz and 192 kHz sample rate, the output word clock is 48 kHz. The received word clock signal can be distributed to other devices by using the word clock output. With this the usual T-adapter can be avoided, and the HDSP 9632 operates as Signal Refresher. This kind of operation is highly recommended, because input and output are phase-locked and in phase (0) to each other SteadyClock removes nearly all jitter from the input signal the exceptional input (1 Vpp sensitivity instead of the usual 2.5 Vpp, dc cut, Signal Adaptation Circuit) plus SteadyClock guarantee a secure function even with highly critical word clock signals the Word Clock Module provides two word clock outputs with separated driver stages
27.4 Tour de TotalMix
This chapter is a practical guide and introduction on how to use TotalMix, and on how TotalMix works. Starting up TotalMix the last settings are recalled automatically. When executing the application for the first time, a default file is loaded, sending all playback tracks 1:1 to the corresponding hardware outputs with 0 dB gain, and activating phones monitoring. Hold down Ctrl and click on preset button 1 to make sure that factory preset 1 is loaded. The faders in the top row are set to maximum attenuation (called m.a. in the following), so there is no monitoring of the input channels. The Submix View is active, therefore for improved overview all outputs except AN1/AN2 are greyed out. Additionally all faders are set to the routing target AN 1+2. All faders of the middle row are set to 0 dB, so no matter on which channels a playback happens, the audio will be audible via the Phones output. Just try it! We will now create a submix on the analog outputs. Please start a multitrack playback. In the third row, click on the channels of hardware output AN1 or AN2. The Submix View changes to AN1/AN2. Both the fader settings and the output levels of all other channels are still visible, but greyed out for improved orientation. As soon as AN1/AN2 became active, all faders of the second row jumped to their bottom position except those of playback channel 11/12. This is correct, because as mentioned above the factory preset includes a 1:1 routing. Click on A 3/4 and the faders above are the only active ones, same for A 5/6 and so on. Back to AN1/2. Now you can change all the faders of all inputs and playback channels just as you like, thus making any input and playback signals audible via the analog outputs. The panorama can be changed too. Click into the area above the fader and drag the green bar in order to set the panorama between left and right. The level meters of the third row display the level changes in real-time.
As shown it is very easy to set up a specific submix for whatever output: select output channel, set up fader and pans of inputs and playbacks ready! For advanced users sometimes it makes sense to work without Submix View. Example: you want to see and set up some channels of different submixes simultaneously, without the need to change between them all the time. Switch off the Submix View by a click on the green button. Now the black routing fields below the faders no longer show the same entry (AN 1+2), but completely different ones. The fader and pan position is the one of the individually shown routing destination. In playback channel 1 (middle row), labeled Out 1, click onto the routing field below the label. A list pops up, showing a checkmark in front of 'A 1+2' and 'AN 1+2'. So currently playback channel 1 is sent to these two routing destinations. Click onto 'A 7+8'. The list disappears, the routing field no longer shows 'AN1+2', but ' A 7+8'. Now move the fader with the mouse. As soon as the fader value is unequal m.a., the present state is being stored and routing is activated. Move the fader button to around 0 dB. The present gain value is displayed below the fader in green letters. In the lower row, on channel 7, you can see the level of what you are hearing from output 7. The level meter of the hardware output shows the outgoing level. Click into the area above the fader and drag the mouse in order to set the panorama, in this case the routing between channels 7 and 8. The present pan value is also being displayed below the fader. Please carry out the same steps for Out 2 now, in order to route it to output 8 as well. In short: While editing the Submix A 7+8 you have direct access to other submixes on other channels, because their routing fields are set to different destinations. And you get a direct view of how their faders and panoramas are set up. This kind of visual presentation is a mighty one, but for many users it is hard to understand, and it requires a deep understanding of complex routing visualizations. Therefore we usually re-commend to work in Submix View. Often signals are stereo, i. e. a pair of two channels. It is therefore helpful to be able to make the routing settings for two channels at once. Hold down the Ctrl-key and click into the routing field of Out 3. The routing list pops up with a checkmark at 'A 3+4'. Select 'A 7+8'. Now, Out 4 has already been set to 'A 7+8' as well. When you want to set the fader to exactly 0 dB, this can be difficult, depending on the mouse configuration. Move the fader close to the 0 position and now press the Shift-key. This activates the fine mode, which stretches the mouse movements by a factor of 8. In this mode, a gain setting accurate to 0.1 dB is no problem at all. Please set Out 4 to a gain of around -20 dB and the pan close to center. Now click onto the routing field. You'll now see three checkmarks, at 'A 3+4', 'A 7+8' and 'AN 1+2'. Click onto 'SPDIF'. The window disappears, fader and pan jump to their initial values, the signal can now be routed to the SPDIF output. You can continue like this until all entries have got a checkmark, i. e. you can send the signal to all outputs simultaneously.
TotalMix includes eight factory presets, stored within the program. The user presets can be changed at any time, because TotalMix stores and reads the changed presets from the files preset11.mix to preset81.mix, located in Windows' hidden directory Documents and Settings, <Username>, Local Settings, Application Data, RME TotalMix. On the Mac the location is in the folder User, <Username>, Library / Preferences / Hammerfall DSP. The first number indicates the current preset, the second number the current unit. This method offers two major advantages: Presets modified by the user will not be overwritten when reinstalling or updating the driver The factory presets remain unchanged, and can be reloaded any time.
Mouse: The original factory presets can be reloaded by holding down the Ctrlkey and clicking on any preset button. Alternatively the files described above can be renamed, moved to a different directory, or being deleted. Keyboard: Using Ctrl and any number between 1 and 8 (not on the numeric keypad!) will load the corresponding factory default preset. The key Alt will load the user presets instead. When loading a preset file, for example 'Main Monitor AN 1_2 plus headphone mix 3_4.mix', the file name will be displayed in the title bar of the TotalMix window. Also when loading a preset by the preset buttons, the name of the preset is displayed in the title bar. This way it is always clear what the current TotalMix state is based on. The eight factory presets offer a pretty good base to modify them to your personal needs. In all factory presets Submix View is active by default. Preset 1 Description: All channels routed 1:1, monitoring of all playback channels via analog out. Details: All inputs maximum attenuation. All playback channels 0 dB, routed to the same output. All outputs 0 dB. Submix of all playbacks to the analog output. Level display set to RMS +3 dB. View Submix active. Note: This preset is Default, offering the standard functionality of a I/O-card. Preset 2 Description: All channels routed 1:1, input and playback monitoring via analog out. As Preset 1, plus submix of all inputs (0 dB) to the analog output. Preset 3 Description: All channels routed 1:1, input and playback monitoring via analog outputs. As Preset 2, but all inputs set to 0 dB (1:1 pass through). Preset 4 Description: All channels 1:1, playback monitoring via analog outputs. As Preset 3, but all inputs muted. Preset 5 Description: All faders m.a. As Preset 1, but all outputs maximum attenuation, only analog monitor of the playbacks is active. Preset 6 Description: Submix to SPDIF. As Preset 1, plus submix of all playbacks to SPDIF. Preset 7 Description: Submix to SPDIF. As Preset 6, plus submix of all inputs to SPDIF. Preset 8 Description: Panic. As Preset 4, but playback channels muted too (no output signal).
29.6 Using external Effects Devices
With TotalMix a usage of external hardware - like effects devices - is easy and flexible. Example 1: The singer (microphone input channel 1) shall have some reverb on his headphones (outputs 11/12). A direct routing In 1 to Out 11/12 for monitoring had been set up already. The external reverb is connected to a free output, for example channel 8. In active mode Submix View click on channel 8 in the bottom row. Drag the fader of input 1 to about 0 dB and the panorama fully to the right. Adjust the input level at the reverb unit to an optimal setting. Next the output of the reverb unit is connected to a free stereo input, for example 5/6. Use the TotalMix level meters to adjust a matching output level at the reverb unit. Now click on channels 11/12 in the bottom row, and move the fader of inputs 5/6 until the reverb effect gets a bit too loud in the headphones. Now click on channel 8 in the bottom row again and drag fader 1 down a bit until the mix of original signal and reverb is perfect for the singer. The described procedure is completely identical to the one when using an analog mixing desk. There the signal of the singer is sent to an output (usually labeled Aux), from there to a reverb unit, sent back from the reverb unit as stereo wet signal (no original sound), back in through a stereo input (e.g. Effect return) and mixed to the monitoring signal. The only difference: The Aux sends on mixing desks are post-fader. Changing the level of the original signal causes a change of the effects level (here the reverb) too, so that both always have the same ratio. Tip: Such a functionality is available in TotalMix via the right mouse button! Dragging the faders by use of the right mouse button causes all routings of the current input or playback channel to be changed in a relative way. This completely equals the function Aux post fader. Example 2: Inserting an effects device can be done as above, even within the record path. Other than in the example above the reverb unit also sends the original signal, and there is no routing of input 1 directly to outputs 11/12. To insert an effects device like a Compressor/Limiter directly into the record path, the input signal of channel 1 is sent by TotalMix to any output, to the Compressor, back from the Compressor to any input. This input is now selected within the record software.
Unfortunately, very often it is not possible within the record software to assign a different input channel to an existing track 'on the fly'. The loopback mode solves this problem elegantly. The routing scheme stays the same, with the input channel 1 sent to any output via TotalMix, to the Compressor, from the Compressor back to any input. Now this input signal is routed directly to output 1, and output 1 is then switched into loopback mode via Ctrl-mouse. As explained in chapter 29.5, the hardware input of channel 1 now no longer feeds the record software, but is still connected to TotalMix (and thus to the Compressor). The record software receives the signal of submix channel 1 instead the Compressor's return path.
30. TotalMix MIDI Remote Control
TotalMix can be remote controlled via MIDI. It is compatible to the widely spread Mackie Control protocol, so TotalMix can be controlled with all hardware controllers supporting this standard. Examples are the Mackie Control, Tascam US-2400 or Behringer BCF 2000. Additionally, the stereo output faders (lowest row) which are set up as Monitor Main outputs in the Monitor panel can also be controlled by the standard Control Change Volume via MIDI channel 1. With this, the main volume of the HDSP 9632 is controlable from nearly any MIDI equipped hardware device.
Open the Preferences dialog (menu Options or F3). Select the MIDI Input and MIDI Output port where your controller is connected to. When no feedback is needed (when using only standard MIDI commands instead of Mackie Control protocol) select NONE as MIDI Output. Check Enable MIDI Control in the Options menu.
The channels being under MIDI control are indicated by a colour change of the info field below the faders, black turns to yellow. The 8-fader block can be moved horizontally and vertically, in steps of one or eight channels. Faders can be selected to gang them. In Submix View mode, the current routing destination (output bus) can be selected via REC Ch. 1 8. This equals the selection of a different output channel in the lowest row by a mouse click when in Submix View. In MIDI operation it is not necessary to jump to the lowest row to perform this selection. This way even the routing can be easily changed via MIDI. Full LC Display Support: This option in Preferences (F3) activates complete Mackie Control LCD support with eight channel names and eight volume/pan values. Attention: this feature causes heavy overload of the MIDI port when ganging more than 2 faders! In such a case, or when using the Behringer BCF2000, turn off this option.
When Full LC Display Support is turned off, only a brief information about the first fader of the block (channel and row) is sent. This brief information is also available on the LED display of the Behringer BCF2000. Tip for Mac OS X users: LC Xview (www.opuslocus.com) provides an on-screen display emulating the hardware displays of a Logic/Mackie Control, for use with controllers that can emulate a Logic/Mackie Control but do not have a display. Examples include the Behringer BCF2000 and Edirol PCR series. Deactivate MIDI in Background (menu Options) disables the MIDI control as soon as another application is in the focus, or in case TotalMix has been minimized. This way the hardware controller will control the main DAW application only, except when TotalMix is in the foreground. Often the DAW application can be set to become inactive in background too, so that MIDI control is switched between TotalMix and the application automatically when switching between both applications. TotalMix also supports the 9th fader of the Mackie Control. This fader (labeled Master) will control the stereo output faders (lowest row) which are set up as Main Monitor outputs in the Monitor panel. Always and only.
As the transmission of double rate signals is done at standard sample rate (Single Speed), the ADAT outputs still deliver 44.1 kHz or 48 kHz. The SPDIF (AES) output of the HDSP 9632 provides 96 kHz as Single Wire only.
33.4 QS Quad Speed
Due to the small number of available devices that use sample rates up to 192 kHz, but even more due to a missing real world application (CD.), Quad Speed has had no broad success so far. An implementation of the ADAT format as double S/MUX (S/MUX4) results in only two channels per optical output. Devices using this method are few. In earlier times the transmission of 192 kHz had not been possible via Single Wire, so once again sample multiplexing was used: instead of two channels, one AES line transmits only one half of a channel. A transmission of one channel requires two AES/EBU lines, stereo requires even four. This transmission mode is being called Quad Wire in the professional studio world, and is also known as S/MUX4 in connection with the ADAT format. The AES3 specification does not mention Quad Wire. The SPDIF (AES) output of the HDSP 9632 provides 192 kHz as Single Wire only.
33.5 AES/EBU - SPDIF
The most important electrical properties of 'AES' and 'SPDIF' can be seen in the table below. AES/EBU is the professional balanced connection using XLR plugs. The standard is being set by the Audio Engineering Society based on the AES3-1992. For the 'home user', SONY and Philips have omitted the balanced connection and use either Phono plugs or optical cables (TOSLINK). The format called S/P-DIF (SONY/Philips Digital Interface) is described by IEC 60958. Type Connection Mode Impedance Level Clock accuracy AES3-1992 XLR Balanced 110 Ohm 0.2 V up to 5 Vss not specified IEC 60958 RCA / Optical Un-balanced 75 Ohm 0.2 V up to 0.5 Vss I: 50ppm II: 0,1% III: Variable Pitch not specified
< 0.025 UI (4.4 ns @ 44.1 kHz)
Besides the electrical differences, both formats also have a slightly different setup. The two formats are compatible in principle, because the audio information is stored in the same place in the data stream. However, there are blocks of additional information, which are different for both standards. In the table, the meaning of the first byte (#0) is shown for both formats. The first bit already determines whether the following bits should be read as Professional or Consumer information. Byte Mode Pro Con Bit 0 P/C P/C 1 Audio? Audio? 5 Emphasis Locked Copy Emphasis 7 Sample Freq. Mode
It becomes obvious that the meaning of the following bits differs quite substantially between the two formats. If a device like a common DAT recorder only has an SPDIF input, it usually understands only this format. In most cases, it will switch off when being fed Professional-coded data. The table shows that a Professional-coded signal would lead to malfunctions for copy prohibition and emphasis, if being read as Consumer-coded data. Nowadays many devices with SPDIF input can handle Professional subcode. Devices with AES3 input almost always accept Consumer SPDIF (passive cable adapter necessary).
The RME Hammerfall HDSP 9632 PCI Card makes the dream of an all-in-one soundcard for every possible application come true. As usual, RME has not made any compromises: Latest 192kHz A/D/A with more than 110dB signal to noise ratio. Balanced stereo analog I/O, ADAT digital I/O, S/PDIF digital I/O with breakout cable for S/PDIF optical, MIDI I/O with 16 channels of hi-speed MIDI, and stereo headphone output. All inputs and outputs are simultaneously useable. Optional hi-quality analog expansion boards are easy to install, and a clock section provides maximum suppression of external clock signal jitter. An internal ADAT I/O allows for up to 14 analog inputs, 14 analog outputs, or the usage of a TDIF interface directly inside the computer.
CCD-TRV32 FDA-FM1AM PEG-T665C Filmscan 35 Chess Networks LH-D6245D 240v DC399-3 Cabrio 380 Gpsmap 76CX TF-DVD7500 CFX-9950GB Plus KV-21FT1K Ryobi 975R EP6000 RM-LJ302 777ES Laserjet 3500 Citroen C2 CW F500 Price Finepix F650 Software Yamaha P90 ANT706A PLC-XD2200 XT6200 TH-50PX8E UC8T2 Studio 10 Veriton 1000 SA-970 FVG318 Deskjet 5943 XRC350 CM507A Xtrail-NEW-2006 125 4T 800 840 Ericsson T616 Caplio R1 Quickreference Octane X2550 KD-G425 Pci DVD-A1XV EWT1016 HT-WP30 Elite EWF882 AL-1220 RM-P6 SCS145 5 112 Plus DO10PSS-5 DVD-S657 A650 IS EOS 7D KX-TG7321RU DMR-EH575 MD 4689 HTC S620 71003 Mediaconverter PSR-180-PSR-76 IN 1 Procoder 2 CF-VDL01 Impression A4- LBT-D790 CDP-XB630 HT-Z310 Treo 750V Mitsubishi SL4U Taurus X MX 700 Es80 Grafx CJ1W-nc271 P4M800pro-M Manager BDV-F700 SRC20134AC 30FS4D 130PRO TA-F240 Server 2011 Windows 7 CA-MXK3 Ncch-DL HM70-100 Xpressmusic KX-TSC11W MF 4640 QRX-5500 DS-150 LN46A750r1F Acoustic SE SRU7060 RH188S History 125
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