Box Contents

Along with your Saffire PRO 24 DSP you should have: 1 - 6 pin FireWire cable (also known as an IEEE1394 cable) 1 - Universal DC power supply unit (PSU). 1 - Installer CD: Contains combined installer for drivers and Saffire MixControl software for Mac and Windows. Focusrite VST and AU Plug-in Suite for Mac and Windows - Includes: Compressor EQ Gate Reverb 4 - Self adhesive rubber feet - to be fitted to base of the Saffire PRO 24 DSP 1 - Registration Card 1- Excite Bundle - Includes: Ableton Live LE version 7 Bass Station AU and VST synthesiser plug-in with Serial number card Over a Gigabyte of Loopmasters and Mike the Drummer royalty free Samples 1 - Focusrite Product Brochure 1 - Novation Product Brochure

Getting Started

The Saffire PRO 24 DSP has a single 6 pin FireWire port on the back. This FireWire connection will work with any of the current FireWire standards and connectors; FireWire 400 (6 pin or 4 pin connection); and FireWire 800. BEFORE YOU CONNECT THE Saffire PRO 24 DSP TO YOUR COMPUTER, PLEASE RUN THE INSTALLER. This will ensure that the correct drivers are used, thus preventing any unexpected behaviour The Saffire PRO 24 DSP ships with a 6 pin FireWire cable. However on Windows laptops, the FireWire connection may be 4 pin FireWire. In this case you will need to purchase a 6 pin to 4 pin cable. You may have a FireWire 800 connector on your computer. In this case you will need to purchase a 6 pin FireWire 400 to 800 cable or adapter. The Saffire PRO 24 DSP can be powered via the FireWire connection when using 6 pin FireWire 400 or FireWire 800. When connected to a 4 pin FireWire you will need to use the supplied PSU. Note: FireWire considerations - Most computers will generally be equipped with 1 FireWire bus. You may have multiple FireWire ports (connectors) on your computer, but these are all connected to 1 physical chip which controls the bus. The FireWire bus is limited in the amount of data bandwidth it can handle. Each additional FireWire device connected to the FireWire bus demands additional bandwidth resources, thus increasing the chances of exceeding the total available bandwidth. Consequently, it is recommended that Saffire PRO 24 DSP DSP run on its own FireWire bus for best performance. The Saffire PRO 24 DSP should be able to work alongside other devices connected to the same FireWire bus. However, problems may arise depending on which FireWire devices are connected. For example, a FireWire disk used for back-up or a digital camera should not cause any problems, but when streaming audio from a FireWire hard disk or a FireWire box such as Focusrite Liquid Mix, the total FireWire bandwidth may be reached. This will result in audio drop outs, or reduced performance on either the Saffire PRO 24 DSP or the other connected FireWire device. For this reason, we would recommend that you use a separate FireWire bus for each FireWire device. This may be a PCI / PCIe card in your desktop, or a PCMCIA or Express card in your Laptop. In order to maximise the life of your FireWire ports, we recommend that you do not connect or disconnect your Saffire PRO 24 DSP whilst bus powered.

Mac OS

1. Insert your installer disk into your computers CD-ROM drive. 2. You should see a window pop up showing the following installer icons: Install Saffire MixControl.pkg 3. Double click on the installer icon to begin the installation process. 4. Follow on screen instructions to complete the installation process. 5. 6. Restart your computer. Connect the Saffire PRO 24 DSP to your computer.
Once the installation has been completed, your Computer OS should automatically switch its default audio outputs to be the Saffire PRO 24 DSP. To make sure this is the case: Go to System Preferences > Sound > Set the input and output to Saffire. For more detailed setup options on a Mac, go to Applications > Utilities > Audio Midi Set-up. Note that the included VST and AU plug-in suite can be installed by running Focusrite Plug-in Suite.pkg from the same installer disk.

Audio Set-up in your DAW

The Saffire PRO 24 DSP is compatible with any DAW that uses ASIO drivers on Windows, and any DAW that uses Core Audio on Mac. Your DAW software may not automatically switch the device it uses to input and output audio. You must ensure that you select Saffire as the ASIO driver (Windows) or Core Audio driver (Mac) in your DAWs audio set-up page. Please refer to your DAWs documentation if you are unsure where to select the ASIO / Core Audio driver.
Saffire PRO 24 DSP Architecture
Saffire PRO 24 DSP offers more than just simple input and output routing to and from your computer. Saffire MixControl software also allows you to re-route audio signals to any output, and to create custom mixes to be sent to your recording artists, outboard processing equipment, or mixing console. The following diagrams give an overview of the Saffire PRO 24 DSP audio paths when set at different sample rates and Optical input configurations. The Hardware Inputs are routed directly to the DAW Inputs. The table below each diagram shows the routing configuration.
44.1kHz / 48kHz Optical as ADAT
Saffire PRO 24 DSP HARDWARE IN
Analog S/PDIF ADAT 1234567 8

Saffire Mix Control

Music Software DAW IN

HARDWARE OUT

Analog 123456 S/PDIF 1 2

ROUTER

DAW OUT

12345678

Loop back
HARDWARE IN Analogue 1-4 S/PDIF 1-2 (RCA) ADAT 1-8 (Opt) Loop back 1-2
DAW IN 1-4 5-6 7-14 15-16
44.1kHz / 48kHz Optical as S/PDIF
Analog S/PDIF (RCA) S/PDIF (Opt) 1 2

Analog S/PDIF 1 2

1234567 8
HARDWARE IN Analogue 1-4 S/PDIF 1-2 (RCA) S/PDIF 3-4 (Opt) Loop back 1-2 10
DAW IN 1-4 5-6 7-8 15-16 *DAW IN 9-14 are redundant

88.2kHz / 96kHz Optical as ADAT

Analog S/PDIF ADAT 1234

HARDWARE IN Analogue 1-4 S/PDIF 1-2 (RCA) ADAT 1-4 (Opt) Loop back 1-2
DAW IN 1-4 5-6 7-10 11-12
88.2kHz / 96kHz Optical as S/PDIF
Analog 1234 S/PDIF (RCA) S/PDIF (Opt) 3 4
HARDWARE IN Analogue 1-4 S/PDIF 1-2 (RCA) S/PDIF 3-4 (Opt) Loop back 1-2
DAW IN 1-4 5-6 7-8 11-12 *DAW IN 9-10 are redundant

Saffire MixControl

The Saffire MixControl software allows flexible mixing and routing of all audio signals to the physical audio outputs, as well as control of output monitor levels. All sample rate selection, digital syncing and buffer size settings (Windows only) are available from Saffire MixControl. To open Saffire MixControl. Windows Start > Programs > Focusrite > Saffire MixControl. Mac Open Finder > Applications > Saffire MixControl. This is how the Saffire MixControl Graphical Interface will appear on your computer. 1 2
5 1. 2. 3. 4. 5. 6. 7. Mixer Selected Mix Tab Mixer Input Channel Selected Mix Output Channel Routing and Pre-amp / Input FX / VRM Section Monitor Section Device Status Section

Pre-amp Section

When connecting a microphone to the Saffire PRO 24 DSP, you must connect an XLR cable to the combi inputs on the front panel. When connecting either Line level signal or Instrument signal to the combi inputs, you must select either Line or Inst from the preamp section. The pre-amp settings can be accessed by selecting Router from the panel selection drop down menu.
The panel display is split to display the routing output settings in the top half (see page 21), and the pre-amp settings in the bottom half.
Additional Line level signals can be connected to inputs 3 and 4 on the rear of the unit. The Preamp gain for Inputs 3 and 4 can be set to either Lo (low) or Hi (high) gain. The maximum input level before the signal will clip for each setting is as follows: Low Gain: 0dBFS = +16dBu High Gain: 0dBFS = -10dBV (~-6dBu)

Mixer Section

The Saffire MixControl software includes a total of 8 mixes, each with a maximum of 16 channels in the mix. Up to 8 mono mixes or 4 stereo mixes, (or any combination of mono and stereo mixes) are available, making up a total of 8 mixers. Each Mix can contain up to 16 of the possible 24 inputs signals, and each mix can be sent to any number of outputs. Each mix shares the same source inputs, but all other mixer controls are independent in each mix. The Mixer Section is used to create mixes for monitoring purposes. The mixes you create do not affect how the audio inputs are routed to the DAW, nor does it affect the audio level of the signal to be recorded. What you set up in the mixer section of Saffire MixControl only affects what is heard in the outputs of the mix. The recording levels that are sent to the DAW are therefore only affected by the input gain settings on the Saffire PRO 24 DSP, not the mixer. The mixer section is useful for creating different simultaneous mixes. For example, you may wish to provide a headphone monitor mix for the recording artist which is different to the mix heard in the monitor speakers. The artist needs to hear mainly the backing track, and a little of the recorded input signal. The engineer needs to hear mainly the recording artists signal and a little of the backing track. A separate mix can be created for both artist and engineer with the exact levels desired. Each independent mix is created on a different mix tab.

Mixer Tab

Each mix can be selected by clicking on the corresponding mixer tab.

Input Channel

Here is a picture of 2 mixer input channels. Below is a description of every component of a mixer channel.

Audio source select

When there is no input to a mixer channel, it will display Off. Clicking on the Off region brings up a list of all available inputs that can be fed to the channel. All analogue (labelled Anlg) and digital inputs, and DAW outputs are available. When selecting a source on a stereo channel, if an odd numbered input is chosen for the left channel, the next even number input is auto selected for right, and Vice Versa Note that if an input has already been taken, it will be greyed out and you cannot select it again. The input needs to be deselected from the track where it is already taken, and then re-selected on the desired track.
To get your sounds from your DAW or other computer applications into the mixer, you should select DAW 1 and DAW 2 on a stereo input track.

Pan Slider

A Pan Slider is used to position the audio signal anywhere between the left and right speakers. Moving the horizontal slider from left to right will move the audio signal from left to right within the stereo field, i.e., the signal is faded between two audio outputs such as Line out 1 and 2. Hold down the Shift key whilst moving the slider for fine control.
When used with a stereo track, the slider affects the audio signal such that when fully left, only the left channel is heard, and when fully right, only the right channel is heard
Use the fader to set the level of your audio signal for monitoring within the current mixer. Click the fader with your mouse and drag to any position. Double click on the fader to set it to 0. Hold the Shift Key to make fine adjustments to the fader value. Fader range is from - to +6 dB with the current fader level displayed in the box below.
The meter shows the signal level of the source input to the channel. The recent maximum signal level is displayed in the box below. The metering is always pre-fade, showing the level of the signal at the input. Therefore, the fader level has no effect on the metering. Clip light If the red portion at the top of the fader lights up, then the signal level is too high. You will need to turn down the signal level by either using the gain knobs on the front panel for the analogue inputs, using the gain on the external devices connected to the digital inputs, or using the gain within the DAW. Once the gains are lowered, click on the red portion to reset the clip light.
Pressing this button mutes the signal. Red indicates that Mute is active.
Pressing this button solos the signal. The level of the fader will affect the level of the soloed signal. Yellow Indicates that Solo is active.

PFL (Pre-fade listen)

THRESHOLD Knob Sets the level at which compression begins. The lower this value is set, the more of the signal will be compressed as the audio will compress when the threshold is reached. Rotate the TRSHLD dial anticlockwise to lower the threshold and so increase the compression. RATIO Knob Determines how much the signal is reduced by when it exceeds the threshold. For example, a ratio of 10:1 means that when the level of the uncompressed signal exceeds the threshold by 10dB, the compressed signal will only increase by 1dB. The higher the ratio therefore (the further the dial is rotated in a clockwise direction) the more heavily the signal is compressed. GAIN REDUCTION Meter Indicates when compression is occurring by showing the gain reduction applied to the signal. ATTACK Knob Defines how quickly the compressor kicks in, e.g. how fast the signal is turned down when it exceeds the threshold. In other words, setting a slower/longer attack time by rotating the dial clockwise will mean more of the loud part of the signal gets through uncompressed, which makes the signal much more punchy but also more likely to clip. RELEASE Knob Defines how quickly the compressor stops acting on the signal after it has begun to compress. Setting a quicker/shorter release time by rotating the dial anticlockwise will normally make the signal louder overall, however this is dependant on the attack time and amount of time the signal is above the threshold. OUTPUT Knob Defines how much the level of the compressed signal is increased after compression. This means that a heavily compressed signal can be turned up loud to give greater perceived loudness without the risk of clipping.
Equalisation of sound is an essential part of the recording process, necessary to remove or boost specific sections of the audible frequency spectrum. The Focusrite EQ is 4-band, with 2 fully parametric mid bands and the option of shelving or high-pass / lowpass on bands 1 and 4, and exhibits the same curves as the classic Focusrite EQ. The two central bands have the same 3 knobs for modifying parameters: Frequency, Gain and Q. When high and lows bands are in high-pass or low-pass mode (switch in lower position), the Gain knob changes into Q. This is because there is no cut or boost option with a high-pass or low-pass filter, just a slope of variable Q at a selected cut-off frequency. The shelving mode doesnt require a Q control because the slope is fixed. The controls are: EQ ACTIVE Button - Click to turn the EQ on or off.
FREQUENCY Knobs Set the frequency that the band affects, e.g. the centre frequency in bell mode, the cut-off frequency in high-pass /low-pass mode or the start of the shelf in shelving mode.
GAIN Knobs Increase or decrease the level of the corresponding band of EQ. The band will have no effect with the knob in a central position. Rotate clockwise to increase and anticlockwise to decrease the level by up to 18dB.

Q Knobs Set the level of resonance of the band, making the band have a more pronounced effect. Increasing Q decreases the bandwidth so that the bell acts on less of the frequency range.
SHELVING/HIGH-PASS or LOW-PASS Switch Selects a low-shelf (upper) or high-pass (lower) for band 1, and a highshelf (upper) or low-pass (lower) for band 4.
OUTPUT Knob Increases or decreases the level of the signal at the EQ output. No gain modification occurs with the knob in a central position. Rotate clockwise to increase and anticlockwise to decrease the level by up to 36dB.

Additional Controls

FX Chain Order Button - Click to change the order in which Compressor and EQ are chained. If the Compressor is positioned above the EQ, the signal is compressed before EQ. With EQ positioned above the Compressor, EQ is applied before compression. FX Mono / Stereo Button - Click to switch between 2 channel Mono and 2 channel Stereo operation. When in Stereo mode, FX controls for input 2 on the right are unavailable. All parameter settings are set on the Left channel but are applied to both Left and Right channels. Note that when in Stereo Mode, the Compressor works in True Stereo - i.e. equal compression is applied to both channels. Input / Output Meters - Display the signal level directly before and after Compression and EQ.
Monitoring with or without FX
Saffire MixControl allows monitoring of Inputs 1 and 2 with or without FX.
When selecting an Input source in the Mixer or Routing section, the dry signal is represented by Anlg In 1 and Anlg In 2, and the wet signal as FX(Anlg 1) and FX(Anlg 2).
Recording with or without FX
The signal routing can be setup so that analogue inputs 1 and 2 can be recorded to the DAW with or without the DSP FX. When you select input 1 and 2 in your DAW software, the input signal with DSP is recorded (when the DSP FX are active). To record inputs 1 and 2 without DSP FX, you must use the Loopback feature which is available from the Routing Panel (see page 21). The loopback inputs should be set to Anlg In 1 and Anlg In 2
When Anlg In 1 and Anlg In 2 are routed to Loopback, these will be available to record in your DAW via the Loopback input channels. As defined in the diagrams on pages 10 and 11, the Loopback channels will appear in your DAW as inputs 15, 16 11, 12 @ sample rates of 44,1kHz, 48kHz @ sample rates of 88,2kHz, 96kHz

Reverb Section

Reverberation is an effect that enables a sound source to be placed in an environment. It does this by adding a reverberant tail of reflected sound, the properties of which correspond to a space of varying dimensions. Click on the Reverb Send Mixer tab to access Reverb send settings and Reverb Controls. The level of the faders of each of the mixer channels determines the signal level being sent to the reverb unit.

In the above picture, only channel 1 - FX(Anlg1) is being sent to the reverb. The Reverb controls are: REVERB ACTIVE Button - Click to turn the reverb on or off. SIZE Knob Defines the size of the reverberant space. Rotate clockwise to increase. Increasing the size makes the reverberation greater, e.g. more time between initial sound and early reflections, plus longer decay time
PRE-FILTER Knob Acts as a high- or low- pass filter for the reflected sound (removes the bass or treble, respectively). Rotate anti-clockwise to produce a low-pass filter effect, where the maximum cutoff (lowest frequency) is set in the extreme anticlockwise position. Rotate clockwise to create a high-pass filter effect, with the maximum cutoff (highest frequency) is set in the extreme clockwise position. In the centre, no filtering of the reflected sound occurs AIR Knob Sets the amount of absorption (or damping) of reflected sound (the more absorption, the less airiness). In a fully anticlockwise position, absorption is at a maximum so there is very little air. Rotate clockwise to decrease absorption and increase air

Reverb Return

Each Mixer (Mix 1 - 8) includes a Reverb Return channel. Set the fader level of the reverb return channel to determine how much of the reverberation signal is to be sent to the output of that mix. 20

Routing Section

The routing section allows you to set up which audio sources are routed directly to which physical outputs.
The routing section displays every physical output on the Saffire PRO 24 DSP and the audio stream to be sent to that output is available for selection in a drop down menu to the left of that output.
Clicking on the box to the left of the output label will bring up a list of all available audio output sources. The available sources include: All input streams (Analogue 1-4, S/PDIF, ADAT) All DAW playback stream (DAW 1 - 8) All mixes from the mixer (Mix 1 - 8) If you have named the mix (by clicking in the track name section see previous chapter) then this name is displayed as the mix source name. Note that the routing section is linked to the selection made for the output channel destination set-up in the mixer. If you have outputs pre-assigned from when you created your Mix, you will see the routing selections have been set up. Similarly, if you change the audio source from the routing section, the output of the mix will change automatically. Headphones 1 is a copy of line outputs 3 and 4. All audio routed to Line outputs 3 and 4 will be heard in Headphone 1 Left and Right. Headphones 2 is a copy of line outputs 5 and 6. All audio routed to Line outputs 5 and 6 will be heard in Headphone 2 Left and Right. When working at sample rates of 88.2kHz or 96kHz, the total number of ADAT channels available drops to 4 channels ADAT SMUX. At these sample rates, ADAT channels 5-8 for the ADAT connection are greyed out.

Routing Presets

Routing Presets are provided as a starting point for you to create your own routing and mixer set-ups. These will enable you to quickly set up your routing for recording (monitoring your inputs); mixing (sending signals out to outboard processors or external mixer); or internal looping (routing audio between software applications without leaving the computer).
This turns off all output routing. This can be used to re-set the routing for when you need to completely restart a configuration, meaning you do not have to manually turn everything off first.

DAW Tracking

DAW Tracking is used for the initial recording process. It will automatically output DAW 1 and 2 to be sent to all Line outputs and hence your main monitors (1+2) and to Headphone outputs 1 and 2. All input channels must be monitored from within the DAW application.

Zero Latency Tracking

Zero Latency Tracking is used for the recording process. It will automatically output Mix 1 and 2 to all your Line outputs simultaneously, and hence will route to the main monitors (1+2), and to the Headphone outputs 1 and 2. Line inputs and DAW outputs must be set up in Mix 1 so that you are able to monitor these sources with zero latency. You must also ensure that you are not monitoring the same signals from within your DAW at the same time, otherwise you will be monitoring the same signal twice (once direct from Saffire MixControl AND a second time (with latency) from your DAW.)

Mixing

Mixing is used for the mixing process. When sending signals out to a mixer or to external processing hardware, hardware outputs are typically set exactly as they are set in the DAW software. DAW outputs route directly to the line output of the same number. (DAW outputs 1-6 to Line outputs 1-6.)

Loopback

Use when recording from one software program to another. You can use loopback routing to record audio from e.g. your internet browser into your DAW, or to record from one DAW to another.
To prevent any audio feedback, make sure that the DAW you are recording into is not set to monitor its inputs. Alternatively, set the outputs of the DAW into which you are recording to 3 and 4; this allows you to monitor the inputs without feeding the signal back into the record stream.
VRM Section - Virtual Reference Monitoring

Technical Information

VRM is a system that simulates the acoustics of loudspeakers and listening rooms for reproduction over headphones. A correct sense of stereo image, perspective, and environment is normally absent in headphone reproduction. VRM provides these aural cues for effective reference monitoring for mixing on headphones. The VRM system works by combining three separate modeling and measurement processes: The speaker emulations were achieved by sending test signals through the original speaker and then taking many measurements through a reference microphone placed at varying positions and distances from the speaker. This process allows the creation of a three-dimensional model of how each speaker radiates sound. The listening environments were not obtained from real rooms, but built using computer models. The model includes not just room dimensions but also the information about how sound is absorbed and reflected by the floor, walls, and ceiling. Also included in the model is the furniture found in each environment. This computer model allows the fine-tuning of the environment to produce the most comfortable listening experience. To complete the simulation, we obtained a set of head-related transfer functions models of how sound arrives and is heard in both ears of a human head. Thousands of measurements were combined to create a model of how direct and reverberated sound arrive at the two ears. The three listening environments available are a Professional Studio, Living Room, and Bedroom Studio. The following table shows which speaker models and listening positions are available in each environment.

Room Model

Speaker Models
Listening Positions (all of these speakers)
Professional Studio Living Room Bedroom Studio
Japanese White Classic centre (at 1.65m from speakers); US Yellow Cone 7.5cm left; 15cm left; Vintage Wooden Cube 7.5cm right;15cm right; US Passive Nearfield 80cm back. British Studio Finnish Studio US Yellow Cone Pro German Studio Ribbon Vintage Broadcast Modern Broadcast British 90s Hi-Fi British 80s Hi-Fi Flat-screen Television Finnish Studio New Broadcast centre (at 1.8m from speakers); 45cm left; 90cm left; 45cm right; 90cm right; 1.2m back and 45cm left; 1.2m back and 45cm right.
US Yellow Cone British 90s Hi-Fi British 80s Hi-Fi Computer Desktop Budget Micro System Flat-screen Television Finnish Studio US Yellow Cone Pro New Broadcast
centre (at 1.4m from speakers); 30cm left; 30cm right; 50cm back; 50cm back and 50cm left; 50cm back and 50cm right.
Listening Environment Data
Listening Environment Professional Studio Living room Bedroom Studio Loudspeaker Emulation Data Emulation Name German Studio Ribbon Based On: ADAM S2.5A
Dimensions 6.10 x 6.48 x 3.53 m 5.48 x 4.66 x 2.79 m 3.28 x 3.69 x 2.47 m
Volume 139.40 m 71.27 m 29.90 m
Reverb Time 0.38 s 0.36 s 0.47 s
Size (cm) 45H, 28W, 30D 38H, 22W, 24D
Tweeter Ribbon 1 silk dome tweeter w/ ferrofluid cooling [none] 1 polyester weave dome (none) 1 metal dome 1 dome 1
Woofer 8 6.5 mineral-filled polypropylene cone bass/midrange driver 2.poly composite driver 3.5 5
System Active 2-way bass reflex Passive 2-way rear firing port Passive single driver closed box Passive 2-way closed box Active single driver rear firing port Active 2-way vented box Passive 2-way ported box Passive 2-way rear ported box with passive radiator Active 2 way front firing ported box Active 2 way front firing ported box Active single driver Active 2-way bass reflex Passive 2 way closed box Passive 2 way closed box Passive 2 way closed bookshelf
US Passive Near- Alesis Monitor One field Vintage Wooden Cube British 80s Hi-Fi B&W DM12 Auratone 5C
17H, 17W, 14D 34H, 22W, 26D
Computer Desktop Finnish Studio Budget Micro System British 90s Hi-Fi
Creative S8S35 Genelec 1031A Goodmans MS188 KEF Q55.2
15H, 8W, 10D 49H, 25W, 29D 28H, 18W, 19D 85H, 21W, 25D

US Yellow Cone

KRK RP6 G2

33H, 22W, 27D

1 neodymium soft dome with ferrofluid 1 silk dome ferrite [none] 1 soft dome 0.75 0.75 1.5
6 glass aramid composite 8 woven kevlar 2 x 4 oval-shaped driver 8 cone 5 KEF BKEF Bcone

US Yellow Cone Pro Flat-Screen Television British Studio Vintage Broadcast New Broadcast Japanese White Classic
KRK VXT8 Phocus LCD 26 TV Quested S8 Rogers LS3/5a Stirling LS3/5a Yamaha NS-10M Pro
44H, 32W, 30D 45H 87W 10D (Stereo TV) 42H, 30W, 35D 30H, 19W, 16D 30H, 19W, 16D 38H, 22W, 18D
* IMPORTANT INFORMATION: FOCUSRITE, the FF logo, VRM Virtual Reference monitoring and the VRM logo are trademarks of Focusrite Audio Engineering Ltd.
All other product names, trademarks, or trade names are the names of their respective owners, which are in no way associated, connected nor affiliated with Focusrite or its Saffire PRO 24 DSP product and which have not endorsed Focusrites Saffire PRO 24 DSP. These other product names, trademarks, and trade names are used solely to identify and describe the third party loudspeaker systems, the sonic behaviour of which was studied for the VRM technology incorporated within the Saffire PRO 24DSP, and to accurately describe an element of functionality within the Saffire PRO 24DSP.
The Saffire PRO 24DSP is an independently engineered technology which utilises Focusrites VRM Virtual Reference Monitoring (Patent applied for) to actually measure examples of the sonic impact of original loudspeaker systems upon an audio stream, so as to electronically emulate the performance of the original product studied. The result of this process is subjective and may not be perceived by a user as producing the same effects as the original products studied.

Using VRM

Select VRM from the panel select drop down list
Once selected, the following panel will be shown:
Click on the power button to turn VRM On or Off.
Click on the arrow to select the listening environment
The green dot indicates the current listening position. Click in any empty circle to select a new listening position. Note that only certain listening positions are available for each listening environment (see table above)
Click on the down arrow to select the monitor speaker. Note that only certain monitor speakers are available in each environment (as shown in the table above.)
Click on the i button to display information on the listening environment, listening postion and monitor speakers. The following information will be displayed.
Environment and Speaker information is the same as found in the tables on page 21 and 22. Coordinates information is described as follows: Spkr-sep: X-Offset: Y-Offset: L- Angle: R- Angle: L- Dist: R- Dist: The distance between the two speakers. The distance from the Sweetspot listening position. A + value is to the Right, a - value is to the Left. The distance from the Sweetspot listening position. A + value is to the Front, a - value is to the Rear. The angle between the listening position and the Left speaker The angle between the listening position and the Right speaker The distance between the listening position and the Left speaker The distance between the listening position and the Right speaker

If you are experiencing clicks and pops or audio dropouts, this may be due to certain hardware in your computer that is affecting the performance of audio devices connected via FireWire. Rather than removing and replacing hardware (e.g. your graphics card or wireless internet card) trying a longer Firewire Driver Latency setting may solve the problem.
Disable WDM Audio in Windows (Windows Only)
Tick this option to ensure that only audio from your DAW is played through the Saffire PRO 24 DSP. Windows sounds will not be played through the Saffire PRO 24 DSP. Sounds from other software will not be played through the Saffire PRO 24 DSP. This is useful to prevent any unwanted audio being heard when you are working in your DAW. It is especially useful when other applications output audio at a sample rate different from the sample rate at which your DAW is operating.
ASIO Buffer Size (Windows Only)
Set the buffer size of your ASIO driver here.
A small buffer size will in result in lower latency at the expense of increased CPU usage. A high buffer size will result in a higher latency but with lower CPU usage. If you are using lots of virtual instruments and effects processing in your DAW project, and the CPU usage is high, then increase the buffer size to permit lower CPU usage.

File Menu

Mac Windows
Open - opens a File open dialog allowing selection of any pre-saved Saffire MixControl set-ups. Save - opens a File save dialog allowing selection of a location into which your Saffire MixControl set-up can be saved. Subsequent saves overwrite the original file. Save As - opens a File save dialog allowing selection of a location into which your Saffire MixControl set-up can be saved. Use this option if you want to keep your original saved set-up and create a new one with a different name. Restore Factory Default - Causes the Saffire PRO 24 DSP to revert to the original default state in which it left the factory. This can be used to globally reset all mixer, routing, and monitor settings, allowing creation of a new set-up from scratch. Clear all Settings - Causes the Saffire PRO 24 DSP to reset all settings. Save to Hardware - This saves the current Saffire MixControl set-up to the Saffire PRO 24 DSP hardware. If you are moving the Saffire PRO 24 DSP from one computer to another and want to retain the set-up, then choose this option. Note that Saffire MixControl does not automatically load from hardware (as this would overwrite a current set-up); it must be loaded manually. Load from Hardware - This loads the saved set-up from the Saffire PRO 24 DSP hardware into the Saffire MixControl software. As you can see from the above screen shots, Open, Save and Save as all have keyboard shortcuts. These are standard shortcuts for their respective functions, so if you are regularly changing settings for your various sessions, then the shortcuts will reduce your set-up time.

Performance Specifications
Microphone Inputs 1-2 Frequency Response: 20Hz - 20kHz +/- 0.1 dB Gain Range: +13dB to +60dB THD+N: 0.001% (measured at 1kHz with a 20Hz/22kHz bandpass filter) Noise EIN: 124dB analogue to digital (measured at 60dB of gain with 150 Ohm termination (20Hz/22kHz bandpass filter) Input Impedance: 2k Ohm
Line Inputs (Inputs 1-2) Frequency Response: 20Hz - 20kHz +/- 0.1dB Gain Range: -10dB to +36dB THD+N: < 0.001% (measured with 0dBFS input and 22Hz/22kHz bandpass filter) Noise: -90dBu (22Hz/22kHz bandpass filter) Input Impedance: >10k Ohm
Line Inputs 3-4 Frequency Response: 20Hz - 20kHz +/- 0.1 dB Gain Range: Switch-able between +16dBu or -10dBV for 0dBFS (balanced inputs) THD+N: 0.003% (measured at 1kHz with a 20Hz/22kHz bandpass filter) Noise: -100dBu (22Hz/22kHz bandpass filter) Input Impedance: >10k Ohm
Instrument Inputs (Inputs 1 and 2) Frequency Response: 20Hz - 20kHz +/- 0.1dB Gain Range: +13dB to +60dB THD+N: 0.004% (measured with 0dBu input and 20Hz/22kHz bandpass filter) Noise: -87dBu (20Hz/22kHz bandpass filter)
Analogue Audio Outputs (Outputs 1-6) 6 Electronically Balanced Outputs Maximum Output Level (0dBFS): +16dBu THD+N: 0.001% (0dBFS input, 20Hz/22kHz bandpass filter)

Digital Performance

A/D Dynamic Range = 105dB (A-weighted), all analogue inputs D/A Dynamic Range = 105dB (A-weighted), all analogue outputs Clock Sources: Internal Clock Sync to Word Clock on SPDIF Input (RCA) Sync to Word Clock on ADAT input Sync to Word Clock on optical SPDIF Input (when enabled) JetPLL PLL technology providing superb jitter reduction for class leading converter performance. Clock jitter < 250 picoseconds Supported Sample Rates: 44.1kHz, 48kHz, 88.2kHz, 96kHz 16 input channels to computer: Analogue (4), SPDIF (2), ADAT (8) and Mix Loop-back (2). 8 output channels from computer: Analogue (6), SPDIF (2). Assignable 16 input by 8 output mixer.
Front and Rear Connectivity
Analogue Channel Inputs (Inputs 1-4) 2 Mic XLR Combo (channels 1-2) on front panel 2 Line TRS (channels 3-4) on rear panel Automatic switching between Mic / Line (channels 1-2) Switching between Line / Instrument Inputs (channels 1-2) via Saffire MixControl Application Switching between +16dBu (low) and -10dBV (high) gain on inputs 3-4 via Saffire MixControl Application

The Focusrite Saffire 6 USB is a 2 In / 4 Out USB audio interface with a focus on just two things audio quality and reliability. Whether youre a singer song-writer who wants to record, or a laptop DJ, Saffire 6 USB will make the most of your sound. Two Focusrite mic-pres, professional digital conversion and custom drivers with super-low latency and rock-solid stability, make Saffire 6 USB the best sounding interface in its class. Flexible monitoring, four phono outputs and a professional high-level headphone output make it ideal for DJs too. If your sound matters to you, Saffire 6 USB is the only interface that will do it justice. Two award-winning Focusrite mic-pres with mic / line / instrument input capture every detail of your sound. High Quality 24-bit/48kHz USB Interface with custom drivers for super low latency, rock-solid stability and class-leading conversion specs. 4 Outputs on RCA Phono make Saffire 6 USB the ideal interface for the laptop DJs performing live. Xcite+ pack, including Ableton Live Lite (software to create,produce and perform) Novations Bass Station (legendary software synthesizer) and 1 gig of royalty-free loops. Focusrite Compression, Reverb, Gating and EQ Plug-ins provide the perfect upgrade from your standard sequencer effects. Monitor either the main mix or cue mix via headphones, all switchable from the front panel. Dedicated input/playback mix dial lets you blend your monitor mix between what youre recording, and the audio from the computer.
ADDITIONAL FEATURES: Dedicated front-panel headphone bus Bus-powering via USB Two Hi-Z instrument inputs with -10dB pads MIDI In and Out Tactile Main Monitor Dial Switchable phantom power on front fascia
KEY PERFORMANCE DATA: A-D Digital Conversion Dynamic Range: 105dB Noise (mic input): THD+N = 0.0025%