Reason 13.1 Operation Manual
DAW | Rack Plugin
Reason receives, generates and plays back digital audio - a stream of numerical values in the form of ones and zeroes. For you to be able to record and play back anything, the audio must be converted from analog to digital when recording, and from digital to analog when playing back through some kind of listening equipment (a set of speakers, headphones, etc.).
This conversion is most often handled by the audio card installed in your computer, or by an external audio interface connected via USB. To achieve the best possible performance, Reason requires that the audio card uses an ASIO driver on Windows systems. On macOS systems, Reason supports Core Audio.
To receive and deliver digital audio to the computer’s audio hardware, Reason uses the driver you have selected in the Preferences dialog. In the Rack on screen, this connection is represented by the Hardware Interface (also known as the Hardware Device):
The Hardware Interface is always located at the top of the rack.
The Hardware Interface contains 64 input and 64 output “sockets”, each with an indicator and a level meter. There are also two Sampling Inputs that can be used for sampling audio to sampler devices. 16 input and 16 output sockets are shown on the main panel, and an additional 48+48 sockets are shown if the “More Audio” button is activated on the main panel. Each one of these indicators represents a connection to an input or output on your hardware audio interface.
However, the number of available inputs and outputs depends on the number of inputs and outputs on your hardware audio interface. For example, if you are using a standard sound card with stereo inputs and outputs, only the first two inputs and outputs will be available. In the Hardware Interface, the indicators are lit green for all currently active and connected inputs and outputs. Activation of inputs and outputs on your hardware audio interface is done on the Audio page in the Preferences dialog (see “Active input and output channels”).
Inputs and outputs that are currently connected have green indicators. Available but unconnected inputs and outputs have yellow indicators and any connections made to unavailable inputs and outputs have red indicators.
In this case, Inputs 1 and 2 are available, but not connected, Outputs 1 and 2 are available and
connected, whereas Output 3 is unavailable, but connected on the back of the Hardware Interface.
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You never have to connect any cables to the Audio In jacks of the Hardware Interface to be able to record audio on sequencer tracks. This routing is made internally “in the background”, which means you just have to select audio interface inputs from the Audio Input drop-down list - see “Selecting audio input(s) and defining mono or stereo”.
#Manual audio routing
In most cases, you will want to have the Main Mixer Master Section device connected to outputs 1 and 2 of the Hardware Interface. This connection is made automatically as soon as you create a new Song document. However, there might be situations where you want to manually route audio to other outputs of the Hardware Interface. For example if you want to use the Control Room Outputs of the Main Mixer (see “Control Room output section”).
To send the sound of a device in the Rack to a specific output, you route the device output to the corresponding Output jack on the Hardware Interface. This is done by using the patch cables on the back of the rack, as described in “Manual routing”. If we flip the rack around, by pressing the [Tab] key, the Hardware Interface looks like this:
On the rear of the Hardware Interface, Inputs 1 and 2 are available but not connected,
Outputs 1 and 2 are available and connected, whereas Output 3 is connected but unavailable.
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You never have to connect any cables to the Audio In jacks of the Hardware Interface to be able to record audio on sequencer tracks. This routing is made internally “in the background”, which means you just have to select audio interface inputs from the Audio Input drop-down list in the sequencer Track List - see “Selecting audio input(s) and defining mono or stereo”.
#Audio quality
The audio quality in a computer based recording system depends on two things:
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The quality of the software calculating the audio.
In our case, this is the Reason DSP (Digital Signal Processing) code.
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The quality of the hardware audio interface used for recording and playing back the sound.
#Software
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Reason uses 32-bit floating point arithmetic for all internal audio operations, with 64-bit summing in the mix bus in the Main Mixer Master Section.
This ensures the highest possible audio quality throughout the entire signal chain.
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Reason supports 16, 20, and 24 bit resolution for input and output audio.
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Reason supports all standard sample rates between 44.1 kHz and 192 kHz.
Reason also supports lower sampling frequencies, but using a sample rate of less than 44.1 kHz is not recommended since it might affect the audio quality negatively.
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A number of digital audio techniques that reduce the risk of “aliasing”, background noise, unwanted distortion and “zipper noise” are implemented in Reason.
#Audio hardware
How good a hardware audio interface actually sounds depends on a number of things; its frequency range and frequency response curve, the resolution (bit depth), the signal to noise ratio, the distortion under various circumstances, etc. Furthermore, some designs are more prone to disturbance from the other electronics in the computer than others. Such disturbance might add hum or high pitched noise to the signal.
The only advice we can give is that if you are serious about sound, choose your audio hardware carefully!
#Audio settings
Sample rate and resolution are properties of digital audio which determine the quality of the sound. Generally, higher sample rate and resolution result in better audio quality (but also larger audio files and higher demands on computer performance and audio hardware). The table below shows some common sample rate/resolution combinations:
| Sample rate:
| Resolution:
| Comment:
44.1 kHz
16 bit
This is the format used on standard audio CDs.
44.1 kHz –
192 kHz
24 bit
These are formats used in professional studios and high-end recording equipment.
To cater for all different situations, Reason supports multiple sample rates and resolutions.
#Sample Rate settings for recording and playback
Reason handles all internal audio processing in 32-bit floating point resolution, with 64-bit summing in the mix bus in the Main Mixer Master Section. However, the resolution of the input and output audio is determined by the hardware audio interface. That is, if you have a 24-bit audio card, Reason will record and output audio in 24-bit resolution, and if you have a 20-bit audio card, audio will be recorded and played back in 20-bit resolution.
The recording and playback sample rate can be specified on the Audio tab in the Preferences dialog (accessed from the Edit menu (Win) or Reason menu (Mac)):
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Select the desired sample rate from the drop-down menu.
Note that the available options on this pop-up menu depend on which sample rates are supported by the audio hardware.
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Reason supports multiple sample rates in the same song!
Reason allows import (or recording) of audio of any sample rate. If the original sample rate of a recording is different than the rate currently set for the audio card, Reason will automatically do a sample rate conversion.
First, a real-time sample rate conversion algorithm is used (allowing the audio to be played back immediately). Meanwhile, in the background, the program calculates a sample rate conversion of the highest quality, which will be used as soon as it is calculated. The CALC progress indicator on the transport panel lights up whenever the program is doing high quality calculations in the background:
The CALC progress indicator on the Transport Panel appears when Reason performs high quality audio calculations.
#Buffer Size settings
The Buffer Size can be adjusted on the Audio tab in the Preferences dialog (accessed from the Edit menu (Win) or Reason menu (Mac)):
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Select Buffer Size by clicking and dragging the slider sideways.
The trick here is to find the optimum relationship between audio quality, DSP Load and latency. Experiment with different Sample Rate settings in combination with different Buffer Size settings to get the best result.
A professional audio interface used together with a state-of-the-art computer should normally be able to handle a combination of a high sample rate (96 kHz) and a small Buffer Size (64-128 samples) without problems. A budget priced audio interface normally requires a lower sampling frequency (44.1 kHz) in combination with a little higher Buffer Size (256-512 samples).
See “About latency” for more information about buffer size and latency. Also see “About audio rendering using the audio card buffer size setting” below for information on how to improve the DSP performance.
#About Max Audio Threads
Reason fully supports multi-core audio rendering. This means that if your computer has multiple CPU Cores (Quad Core, for example), or multiple CPUs, Reason takes advantage of this to significantly enhance the performance. A higher system performance allows for more tracks and devices in your songs.
Here you can select the maximum number of audio threads Reason should be allowed use. The “Default” setting equals the total number of physical cores in your computer’s CPU (corresponding to the old "Use multi-core audio rendering" option, but without the "Use hyper-threading audio rendering" option in older Reason versions). On Apple M1 the Max audio threads setting defaults to the number of high performance cores.
This preference can also be used to limit the number of threads if you want to leave room for other apps or perhaps multi-core VSTs.
See “Max audio threads” for more details.
#About audio rendering using the audio card buffer size setting
The “Render audio using audio card buffer size setting” function should be selected (checked) for best plugin performance. When selected, the audio batches are rendered internally according to the set Buffer size (see “Buffer Size settings” above). For example, if you have a Buffer size of 512 Samples, each audio batch will be 512 samples internally. Raising the Buffer size will let Reason process larger audio batches in one go, which is often more efficient. Many plugins are also more efficient when doing larger audio batches. if you are using DSP-heavy VSTs (mastering effects, for example), these will run a lot smoother with this function selected.
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Note that old songs might sound different with this function selected, if the songs use feedback routings and CV connections.
If unchecked (off), all audio batches are rendered internally at a fixed size of 64 samples - regardless of the Buffer size setting. This might be desirable if you are using feedback signal routings and CV connections in your songs, and want the internal latency of those connections to be fixed at a short value all the time. This might result in performance problems for DSP-heavy VSTs, though.
#Master Tune setting
By default, Reason plays back a “middle A” at 440 Hz, which is the standard tuning in most instruments. However, if you are playing Reason together with other instruments, you may want to adjust the tuning:
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Select “Preferences...” from the Edit menu (Win) or Reason menu (Mac).
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Click the “Audio” tab.
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Adjust the global tuning with the Master Tune slider or button controls.
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If you like, you can also adjust the Master Tune during playback.
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The Master Tune setting affects the tuning of all sound sources in Reason, including the Tuner function on the Audio Tracks. It also affects the tuning of the Redrum and Dr. Octo Rex loop player.
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Note that the Master Tune setting does NOT affect the pitch of the audio on audio tracks!
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Note that the Master Tune setting does NOT affect the pitch of any VST plugins used in the song!
#About audio levels
When recording and playing back in Reason, you should keep an eye on the Audio In and Audio Out Clip indicators on the Transport Panel, or on the Hardware Interface and the Big Meter. You should also keep an eye on the clip indicators on the Main Mixer Master Section. If any of the clip indicators light up, the audio level is too high, resulting in clipping (digital distortion).
The Clip indicators on the Audio In and Audio Out meters on the Transport Panel.
The Clip indicators in the Master Section of the Main Mixer.
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Note that the Main Mixer Master Section Clip indicators will only work if there are no other devices connected between the Master Section device and the Hardware Interface!
The Big Meter on the Hardware Interface.
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To remedy Audio In clipping, adjust the level at the input source, i.e. on the hardware audio interface or on the preamp connected to the hardware audio interface.
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When recording or sampling external audio signals, clipping cannot be adjusted in the Reason application - it must be adjusted at the audio input source!
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When sampling audio internally from devices in the rack, you have to adjust the Output Level on the source device to make sure clipping doesn’t occur in the Hardware Interface.
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To remedy Audio Out clipping, lower the master level on the Mixer (or other device) that is connected to the Hardware Interface, until Audio Out clipping doesn’t light up on playback.
You could also use the MClass Maximizer as an insert effect on the Master Section in the Main Mixer to ensure that clipping never occurs - see “The MClass Maximizer”.
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To get a better overview of the levels, bring up the Big Meter on the Hardware Interface, by clicking the “Big Meter” button on the front panel. Then, select the input or output pairs to view in the Big Meter by clicking on the corresponding channel selection button below each input or output pairs. Alternatively, select channel by turning the channel selection knob.
If the audio level is, or has been, too high, the Clip indicators on the Big Meter will stay lit until you click the Reset button, or select new audio channels for the Big Meter.
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In some situations, the Audio Out Clip indicator on the Transport Panel and the Output Clip indicators on the Reason Hardware Interface might light up if the metronome Click is active in the sequencer during playback (see “Click and Pre-count”). This is nothing to worry about and won’t cause any distortion in your song. To determine if the metronome is causing the clipping indication, just disable Click and see if the clipping stops.
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Note that it doesn’t matter if the level meters on the individual devices (effects, MIxer channels, etc.) “hit the red”. Output clipping can only occur in the Hardware Interface.
The technical reason for this is that internally, Reason uses high resolution floating point processing, which ensures highest audio quality and virtually limitless headroom. In the Hardware Interface, the floating point audio is converted to the resolution used by the computer’s audio interface, and that’s where audio out clipping might occur.
#If you are using multiple outputs
If you are using an audio interface with more than two outputs, you might want to have different devices connected to different outputs. If the Audio Out Clip indicator on the Transport Panel lights up, you should play back the section again while checking the Reason Hardware Interface. Each output socket has a level meter - if the red meter segment lights up, the output is clipping.
The indicator for Output 7 on the Hardware Interface indicates clipping.
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If necessary, bring up the Big Meter and select the output pair where the clipping occurs. Lower the output level of the device connected to the clipping output, until no clipping occurs.
Depending on the track types in the sequencer, the default signal chain varies. In this section we’re going to describe the default audio signal paths for Audio Tracks and Instrument Tracks.
#Audio Track signal paths
When you’re recording and playing back audio that originates from an external source, like a guitar or a vocalist, the audio signal must first travel from the source, via a hardware audio interface, into the Reason application. Then, when played back, the audio travels from the Reason application, via the hardware audio interface, to a speaker system or similar. The figure below shows a schematic overview of the audio signal paths for a “standard” auto-routed Audio Track in Reason:
An Audio Track signal path in Reason.
#Instrument Track signal paths
When you’re recording and playing back audio from an instrument device, like the ID8 Instrument device, the audio signal only has to travel only in one direction: from the Instrument device, via the hardware audio interface, to a speaker system or similar. The figure below shows a schematic overview of the audio signal paths for an auto-routed Instrument Track in Reason:
An Instrument Track signal path in Reason.
#About latency
On any personal computer system, there is a delay between the moment you input a sound, or “tell” the hardware to play a sound, and when you actually hear it. This delay is referred to as the “latency” of the design. This imposes a problem for any system where you want real-time user input to affect the sound.
#Why is there latency?
All audio applications receive and generate their audio in chunks. These chunks are then passed on to the audio card where they are temporarily stored before being converted into regular audio signals. The storage place for these chunks are called “buffers” (an analogy would be a bucket brigade, where a number of people each have a bucket, and water is poured from one bucket to another to reach its final destination).
The smaller the buffers and the fewer they are, the more responsive the system will be (lower latency). The general rules regarding the buffer size are these:
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A small buffer size reduces the latency (the time it takes for the audio to “travel” from the audio interface input(s) to the application and from the application to the audio interface output(s)).
However, a small buffer size also increases the DSP Load. Too small a buffer size setting could also make the sound crackle and distort.
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A large buffer size reduces the DSP Load (allowing for more tracks to be played back simultaneously) and also ensures good audio quality.
However, a large buffer size also increases the latency.
A high sample rate will also reduce the latency. However, this will also raise the demands on the computer and its software. If the system can’t cope with moving the data to and from the buffers fast enough, there will be problems that manifest themselves as glitches in audio playback.
To make things worse, audio playback is always competing with other activities on your computer. For example, a buffer size that works perfectly under normal circumstances might be too small when you try to open files during playback, switch over to another program while Reason is playing or simply play back a very demanding song.
#What is acceptable?
On a regular PC, the latency can vary quite a lot. This is an effect of the fact that computers and their operating systems were created for many purposes, not just for recording and playing back audio. For multimedia and games, a latency of a 100 ms might be perfectly acceptable, but for recording and playing back audio it is definitely not!
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PC audio cards running under Windows with a MME driver might at best give you a latency of around 160ms.
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The same card with a DirectX driver running under Windows provides at best around 40ms.
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A card specifically designed for low latency, with an ASIO driver under Windows, or a Core Audio driver under macOS, can usually give you figures as low as 2-3 ms. This is definitely good enough for audio applications. That’s also why ASIO or Core Audio drivers are required to run Reason.
#Reducing latency
There are a few general methods for making sure latency is as low as possible:
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Make sure you are using the latest version of the drivers for your hardware audio interface.
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Adjust the Sample Rate and Buffer Size parameters as described in “Audio settings”.
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Remove unnecessary background tasks on your computer.
This might be any background utility you have installed as well as networking, background Internet activities etc.
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Optimize your songs.
You might run into situations where you have to raise the Output Latency setting to be able to play back a very demanding song on your computer. Another option would be to actually optimize the song. See “Optimizing Performance” for details.
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Get a better audio interface.
This is only required if you find that you need to increase Output Latency because your audio card can’t really cope with the songs you try to play.
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Get a faster computer.
This is only required if you find that you need to increase Output Latency because your computer can’t really cope with the songs you try to play.
#About processors
When you run Reason, the numbers of physical CPU cores and the CPU clock speed are major factors determining how many audio tracks and devices you can use at the same time.
If you plan to buy a computer specifically for Reason, you could play it safe and choose a computer with at least a quad-core Intel i7 processor, or equivalent AMD processor, or an Apple Silicon processor. Processors with more cores will give better performance and are highly recommended.
#About RAM
Another important performance factor is the amount of installed RAM in the computer. Generally, one could say: the more the better, especially if you’re running several applications simultaneously. To run Reason, a minimum of 4 GB RAM is required, but more is highly recommended for better performance.