MSP interfaces with your computer's audio hardware via the dac~
objects and their easy-to-use equivalents ezdac~
. If you don't have any special audio hardware and have no need for inter-
application audio routing, the default driver on your system will give you stereo full-
duplex audio I/O with no special configuration on your part.
In addition to Core Audio or MME on Windows, there are a number of other ways to get
audio into and out of Max/MSP. Each of these methods involves using what we call a
driver, which is actually a special type of Max object. Some of these drivers
facilitate the use of MSP with third- party audio hardware. Also, a non real-time driver
allows you to use MSP as a disk-based audio processing and synthesis system, removing
the limit of how much processing you can do with your CPU in real time.
MSP audio driver objects are located in the ad folder located in the
Cycling '74 folder inside the Max application folder. These object files must be in this
folder called ad (which stands for audio driver), otherwise MSP will be
unable to locate them.
We will begin with a discussion of audio input/output in MSP in general. Later in this
chapter we will discuss aspects of specific audio drivers that are available to you in MSP.
First we'll discuss the DSP Status window and how to use it to get information about your
audio hardware and set parameters for how MSP handles audio input and output.
All global audio parameters in MSP are displayed in the DSP Status window. To open the
DSP Status window, just double-click on any dac~
object in a
locked Patcher window. Alternately, you can choose DSP Status... from the Options
The DSP Status window is arranged as a group of menus and checkboxes that set all of
the parameters of the audio input and output in MSP. Since all of these options can be
changed from within your patch, the DSP Status window serves as a monitor
for your current audio settings as well.
The DSP Status window is in fact a Max patch
(called DSP Status, in the interfaces
folder inside the Cycling '74 folder). Every parameter shown in
the DSP Status window is a menu or checkbox hooked up to an instance of the
object. The adstatus
object can be used inside of your MSP
patches so that you can set and restore audio parameters specifically for certain patches.
object is also useful for obtaining information current CPU load,
vector size, and sampling rate. See the adstatus
object manual pages in the MSP
Reference Manual for more details.
At the very top of the DSP Status window is a pop-up menu for
turning the audio in MSP on and off
and a set of pop-up menus that let you
select an audio driver
and configure its input source
and output destination.
The second pop-up menu allows you to view and select an audio driver for MSP.
The specific audio drivers will be discussed later in this chapter. A brief summary
will suffice for now:
: This setting shuts off MSP audio processing.
: This is the default audio driver for MSP on Macintosh.
It interfaces with the system's built-in Core Audio system and can be used with the
built-in audio of the computer, or, with the proper software support, a
third-party hardware interface, such as ASIO.
or : (Windows only) On Windows, MSP
loads the MME driver by default. If you have correctly installed external hardware
and it also supports DirectSound, it should also appear as an option on the pop-up
: This driver supports a standard developed by Propellerhead
Software that allows sound generating applications (ReWire Devices) to send multiple
channels of audio and midi to other applications (ReWire Mixers) that process and output
it. Selecting the ad_rewire driver enables Max/MSP to function as a ReWire Device to
route audio from MSP into applications that support ReWire (such as Live, Digital
Performer or Cubase). Using MSP to host ReWire devices (such as Reason) can be
accomplished with the
: (Windows only) If you have a third-party audio interface which
supports ASIO (a cross-platform audio hardware standard developed by Steinberg), and it
is installed correctly, it will be found by the MSP ASIO driver. You may have as many
ASIO devices as you wish; they will all be found by the driver and will appear in the
Driver pull-down menu in the DSP Status Window preceded by the word ASIO.
: This driver enables MSP to work in non real-time mode,
allowing you to synthesize and process audio without any real-time processor
performance limitations. Real-time audio input and output are disabled under this driver.
Only one audio driver can be selected at any given time. MSP saves the settings for each audio driver
separately and will recall the last used audio driver when you restart Max.
The next two pop-up menus are active only when using the Core Audio driver on Macintosh or ASIO drivers.
When the Core Audio driver or either the MME or DirectSound drivers on Windows are selected, the pop-up menus
allow you to change the audio input source. These settings can also be changed using the Audio MIDI Setup
application on Macintosh or the Sounds and Audio Devices Properties window (Start > Settings > Control Panel >
Sounds and Audio Devices) on Windows, but only with these menus while MSP is running.
When ASIO is in use, the pop-up menus allow you to set the clock source for your audio hardware and
whether or not to prioritize MIDI input and output over audio I/O.
The DSP Status Window lets you control the size of the blocks of samples (called signal vectors)
that MSP uses. There are two vector sizes you can control.
The I/O Vector Size (I/O stands for input/output) controls the number of samples
that are transferred to and from the audio interface at one time.
The Signal Vector Size sets the number of
samples that are calculated by MSP objects at one time. This can be less than or
equal to the I/O Vector Size, but not more. If the Signal Vector Size is less than
the I/O Vector Size, MSP calculates two or more signal vectors in succession for
each I/O vector that needs to be calculated.
With an I/O vector size of 256, and a sampling rate of 44.1 kHz, MSP calculates about 5.8 milliseconds of audio data at a time.
The I/O Vector Size may have an effect on latency and overall performance. A smaller
vector size may reduce the inherent delay between audio input and audio output, because MSP
has to perform calculations for a smaller chunk of time. On the other hand, there is an additional
computational burden each time MSP prepares to calculate another vector (the next chunk of
audio), so it is easier over-all for the processor to compute a larger vector. However,
there is another side to this story. When MSP calculates a vector of audio, it does so in
what is known as an interrupt. If MSP is running on your computer, whatever you happen to
be doing (word processing, for example) is interrupted and an I/O vector's worth of audio
is calculated and played. Then the computer returns to its normally scheduled program. If
the vector size is large enough, the computer may get a bit behind and the audio output
may start to click because the processing took longer than the computer expected. Reducing
the I/O Vector Size may solve this problem, or it may not. On the other hand, if you try to
generate too many interrupts, the computer will slow down trying to process them (saving
what you are doing and starting another task is hard work). Therefore, you'll typically
find the smaller I/O Vector Sizes consume a greater percentage of the computer's
resources. Optimizing the performance of any particular signal network when you are
close to the limit of your CPU's capability is a trial-and-error process. That's why
MSP provides you with a choice of vector sizes.
Some audio interface cards do not provide a choice of I/O Vector Sizes. There are also
some ASIO drivers whose selection of I/O Vector Sizes may not conform to the
multiple- of-a-power-of-2 limitation currently imposed by MSP's ASIO support. In some
cases, this limitation can be remedied by using the ASIO driver at a different sampling rate.
Changing the vector sizes does not affect the actual quality of the audio itself, unlike
changing the sampling rate, which affects the high frequency response. Changing the
signal vector size won't have any effect on latency, and will have only a slight effect
on overall performance (the larger the size, the more performance you can expect). However,
certain types of algorithms benefit from a small signal vector size. For instance, the
minimum delay you can get from MSP's delay line objects tapin~
is equal to the number of samples in one signal vector at the current sampling rate.
ith a signal vector size of 64 at 44.1 kHz sampling rate, this is 1.45 milliseconds,
while at a signal vector size of 1024, it is 23.22 milliseconds. The Signal Vector size
in MSP can be set as low as 2 samples, and in most cases can go as high as the
largest available I/O Vector Size for your audio driver. However, if the I/O Vector Size
is not a power of 2, the maximum signal vector size is the largest power of 2 that
divides evenly into the I/O vector size.
You can set the audio sampling rate with the Sampling Rate pop-up menu. For full-range audio,
the recommended sampling rate is 44.1 kHz. Using a lower rate will reduce the number of samples
that MSP has to calculate, thus lightening your computer's burden, but it will also reduce
the frequency range. If your computer is struggling at 44.1 kHz, you should try a lower rate.
Subpatches loaded into the poly~
object can function
at different sampling rates and vector sizes from the top-level patch. In addition, the poly~
object allows up- and down-sampling as well as different vector sizes. The DSP Status window only displays
and changes settings for the top-level patch.
The Max Scheduler in Overdrive
option enables you to turn Max's Overdrive setting on and off
from within the DSP Status window. When Overdrive is enabled, the Max event scheduler runs at interrupt
level. The event scheduler does things like trigger the from a repeating metro
as well as send out any recently received MIDI data. When overdrive is not enabled, the event scheduler
runs inside a lower-priority event handling loop that can be interrupted by doing things like pulling
down a menu. You can also enable and disable Overdrive using the Options menu. Overdrive
generally improves timing accuracy, but there may be exceptions, and some third-party software
may not work properly when Overdrive is enabled.
The Scheduler in Audio Interrupt feature is available when Overdrive is enabled. It runs
the Max event scheduler immediately before processing a signal vector's worth of audio.
Enabling Scheduler in Audio Interrupt can greatly improve the timing of audio events that are
triggered from control processes or external MIDI input. However, the improvement in timing can
be directly related to your choice of I/O Vector Size, since this determines the interval at
which events outside the scheduler (such as MIDI input and output) affect Max. When the Signal
Vector Size is 512, the scheduler will run every 512 samples. At 44.1 kHz, this is every 11.61
milliseconds, which is just at the outer limits of timing acceptability. With smaller Signal
Vector Sizes (256, 128, 64), the timing will sound ‘tighter.’ Since you can change all of these
parameters as the music is playing, you can experiment to find acceptable combination of precision
If you are not doing anything where precise synchronization between the control and audio is
important, leave Scheduler in Audio Interrupt unchecked. You'll get a bit more overall CPU
performance for signal processing
The next portion of the DSP Status helps you
monitor your system's performance.
The CPU Utilization field displays a rough estimate of how much of your computer's CPU is
being allocated for crunching audio in MSP.
The CPU Limit option allows you to set a limit (expressed in terms of a percentage of your
computer's CPU) to how much signal processing MSP is allowed to do. MSP will not go above the set CPU limit
for a sustained period, allowing your computer to perform other tasks without MSP locking them out. The
trade-off, however, is that you'll hear clicks in the audio output when the CPU goes over the specified
limit. Setting this value to either 0 or 100 will disable CPU limiting.
The number next to Signals Used shows the number of internal buffers that were needed
by MSP to connect the signal objects used in the current signal network. The number of
Function Calls gives an approximate idea of how many calculations are being required
for each sample of audio. Both of these fields will update whenever you change the
number of audio objects or how they are patched together.
Vector Optimization only applies to PowerPC computers. Vector optimization
allows four samples to be processed within the space of a single instruction. However, not all audio signal
processing algorithms can be optimized in this way (for example, recursive filter algorithms are
substantially immune from vector optimization). MSP itself no longer uses vector optimization, but third-party audio objects may still use it. In other words, unless you are using a vector-enabled third-party audio object on a PowerPC computer, this setting will have no effect.
The next portion of the DSP Status Window lets you map logical I/O channels.
The pop-up menus labeled Input Channel 1
, Input Channel 2, Output Channel Output Channel 2
allow you to map the first two logical channels of I/O in MSP (i.e. the first two outlets of the adc~
and the first two inlets of the dac~
object) to physical channels used by your audiodriver. Different audio drivers
give you different options, for example, the MME driver on Windows only supports two channels, so you will normally
use the default options. To map additional logical channels, use the I/O Mappings window, which you can view by
clicking the I/O Mappings button at the bottom of the DSP Status window (see below for more information about the
I/O Mappings window). In addition, you can use the adstatus
object from within your patch to map any
of the 512 logical audio I/O channels.
In MSP 2, you can create a dac~
object that uses a channel
number between 1 and 512.
These numbers refer to what we call logical channels and can be dynamically reassigned
to physical device channels of a particular driver using either the DSP Status window, its
I/O Mappings subwindow, or an adstatus object with an input or output keyword
objects allow you to specify arguments which
define which logical channels are mapped to their inlets and outlets, respectively. In the
example below, multiple logical channels are in use in a simple patch:
In this example, two separate adc~
objects output audio signals from logical
channel pairs 1/2 and 3/4, respectively. The output of all four channels is sent to
objects which attenuate the incoming signals and send them to the first
four logical output channels, as specified by the first dac~
object. The input
signals are also multiplied (ring modulated) and sent out logical channels 9 and 10. Up to
sixteen arguments can be typed into a single adc~
object; if you
want to use more than 16 logical channels, you can use multiple adc~
objects. The ezadc~
objects only access the
first two logical input and output channels in MSP.
The purpose of having both logical channels and physical device channels is to allow you
to create patches that use as many channels as you need without regard to the particular
hardware configuration you're using. For instance, some audio interfaces use physical
device channels 1 and 2 for S/ PDIF input and output. If you don't happen to have a
S/PDIF-compatible audio interface, you may wish to use channels 8 and 9 instead. With
MSP 1.x, you would have been required to change all instances of dac~
objects with arguments 1 and 2 to have arguments 8 and 9. With MSP 2,
this is no longer necessary.
You can simply go to the DSP Status window and choose the eighth and ninth physical
channels listed in the Input and Output pop-up menus.
Logical channels in MSP are only created if there is a dac~
object using them. In other words, if you're only using logical outputs 1 and 2, there aren't
510 unused audio streams sitting around hogging your CPU. However, since you can mix
any number of logical channels to a single physical channel if necessary, you can create a
complex multi-channel setup that will allow other people to hear all of your logical
channels when they use it on a two-channel output device.
To assign multiple logical channels to one physical channel of an output device, use the
I/O Mapping window. Click on the I/O Mappings button at the bottom of the DSP Status
The configuration shows that logical channels 1, 3, 5, and 7 have been mapped to the left
output channel of the current audio device, and logical channels 2, 4, 6, and 8 have been
mapped to the right output channel of the current audio device.
I/O Mappings are saved for each audio driver. You can also create I/O mappings within
your patch using the adstatus
object. The example patch below accomplishes the
same remapping as that shown in the I/O Mapping window above, but does so
automatically when the patch is loaded.
Core Audio provides audio I/O to Mac applications from
both the computer's built-in audio hardware as well as any external audio hardware you
If you have external audio hardware, it should come the drivers to interface with Core
Audio. When these drivers are installed and the hardware is present, Core Audio will
include the external device as a Core Audio choice in the Driver menu in the DSP Status
The Sound part of the System Preferences application can be used to set basic sound
settings for the system, such as the Output volume, left/right balance, and sound output
device, as well as the Input volume and sound input device. You can also use the Audio
MIDI Setup application (located in /Applications/Utilities) for more detailed control of
the sound I/O settings. Note that modifications you make to the Sound section of the
System Preferences application, such as changing the output volume or balance, are
reflected in the audio MIDI Setup (and vice versa). You can open the Audio MIDI Setup
application by clicking on the Open Audio Control Panel button in the lower left corner
of the DSP Status Window.
The Audio part of the Audio MIDI Setup application shows Input settings on the left side,
and Output settings on the right.
The System Settings let you choose which audio device is used for system audio
input and output, while the Selected Audio Device menu allows you to control the
various settings for the built-in and any external hardware audio devices.
When using external audio devices, the Input Volume and Output Volume
sliders can be used to set the overall input and output volumes of the selected device
(they are not available when using the built-in audio controller). The Device Mute
checkboxes allow you to mute the input and output devices, if applicable.
Play Through is available on PowerPC Macs only. Play Through checkbox just under the Input Volume slider lets you choose
whether or not the input device is 'monitored' directly through to the output. When
playthrough is enabled, the dry signal from the input source will play through to the
output mixed in with any processed signal you may be sending to the output in MSP.
Disabling playthrough will enable you to control how much (if any) dry signal from the
audio input is routed to the output.
This option can be changed in MSP by sending a message to the
object to change it. Put the following in a message box and clicking on it
will turn playthrough off:
Using an argument of 1 will turn it on.
The Input Section allows you to select the Input Source (for example Line or Mic
input for the selected device) as well as the sampling rate and bit depth in the Current
Format pop-up menu. Similarly, the Output Section also allows you to select the
sampling rate and bit-depth in its Current Format pop-up menu. The available
selections will vary, depending on your audio hardware.
You can set the volume levels for the individual audio input and output channels, mute
individual channels, and/or select them for playthrough using the controls located below
the Current Format menus. The lower part of the window is used to display the current
input and output settings.
Three types of sound card drivers are supported in Windows: MME, DirectSound
and ASIO. Your choice of driver will have a significant impact on the performance and
latency you will experience with MSP.
The MME driver (ad_mme) is the default used for output of Windows system sounds,
and are provided for almost any sound card and built-in audio system. While
compatibility with your hardware is almost guaranteed, the poor latency values you get
from an MME driver make this the least desirable option for real-time media operation.
DirectSound drivers, built on Microsoft's DirectX technology, have become
commonplace for most sound cards, and provide much better latency and performance
than MME drivers. Whenever possible, a DirectSound driver (ad_directsound) should be
used in preference to an MME driver. Occasionally, (and especially in the case of
motherboard-based audio systems) you will find the DirectSound driver performs more
poorly than the MME driver. This can happen when a hardware-specific DirectSound
driver is not available, and the system is emulating DirectSound while using the MME
driver. In these cases, it is best to use MME directly, or find an ASIO driver for your
The best performance and lowest latency will typically be achieved using ASIO drivers.
The ASIO standard, developed by Steinberg and supported by many media-oriented
sound cards, is optimized for very low latency and high performance. As with the
DirectSound driver, you need to verify that performance is actually better than other
options; occasionally, an ASIO driver will be a simple ‘wrapper’ around the MME or
DirectSound driver, and will perform more poorly than expected.
On Windows, MSP loads the MME driver by default. The MSP MME and DirectSound
drivers are located in C:\Program Files\Common Files\Cycling Ô74\ad\.
If you have correctly installed external hardware, it should support playback and
recording with the MME driver and the Direct Sound driver in the Driver Menu of the
DSP Status Window.
If an audio device only supports MME or DirectSound, the Windows OS does an
automatic mapping of one to the other. Since many audio devices initially did not support
DirectSound, Microsoft emulated DirectSound with a layer that bridged from
DirectSound to MME. Currently, there is greater support for native DirectSound drivers,
and sometimes when you use MME drivers Windows is actually running a layer to
convert from MME to DirectSound.
Note: Some devices such as the Digidesign mBox only support the ASIO driver standard.
In such cases, you will need to select the proper ASIO driver in the DSP Status Window.
See the section ‘Using ASIO Drivers on Windows’ for more information.
You can make overall changes to the basic operation of your default audio driver by
accessing the Sounds and Audio Devices Properties window (Start > Settings
> Control Panel > Sounds and Audio Devices). Here you can select Audio
devices, and create settings for balance and output volume.
MSP supports the use of different input and output devices with MME and DirectSound
drivers. Use the DSP Status Window to choose input and output devices.
When using MME or Directsound drivers, you may choose input and output devices from
the pull-down menus in the DSP Status window, which will be automatically populated
with the drivers for your audio hardware. When using the MME and Directsound drivers,
it is possible to use different audio devices for input and output simultaneously.
However, this is not recommended or supported and unless there is some external (from
Max/MSP) provision for synchronizing the devices dropouts will likely occur over time.
Both MME and Directsound drivers include settings for Thread Priority and Latency.
These are both set by default to settings which we hope will work on your computer in
the majority of situations. However, you may find that when you are working with a
patch that you have problems which you may be able to resolve by changing some of
these settings. If your audio is crackling or there are glitches in it, you may want to try
increasing the latency setting. This has the disadvantage of making your audio feel less
responsive in real time, but it will allow the audio driver more time to work on the extra
audio demands you have placed on it.
If your system is slow in other areas -- such as screen redrawing or general timing
accuracy -- you may wish to decrease the thread priority of the audio driver. This
will give your other tasks more room to get done, but may also result in you needing to
increase latency in order to give your audio driver room to breathe at the new lower
Timing between the max scheduler and MSP is best when the I/O vector size is on the
order of 1ms. We recommend setting the IO vector size to 128 samples. Having a
setting of the latency separate from the I/O vector size allows this to work without audio
glitches on most hardware.
The ad_rewire driver allows you to use MSP as a ReWire Device, where MSP audio will
be routed into a ReWire Mixer application such as Cubase. Both Max/MSP and the mixer
application must be running at the same time in order to take advantage of ReWire's
services. The mixer application should be also compatible with ReWire 2 or later for best
When the ad_rewire driver is selected, audio from MSP can be routed to any of 16 inter-
application ReWire channels which will appear as inputs in ReWire mixer host
applications. The first time ad_rewire is selected it will register itself with the ReWire
system. Subsequent launches of ReWire Mixer applications will then offer Max/MSP as a
For example, after the Max/MSP ReWire Device is registered, Cubase will have a
Max/MSP menu item in the Devices menu. When you choose it you will see a list of the
audio outputs from Max/MSP. They will default to the off state. Click on any of the
buttons to activate that channel. Once the channel is activated it will show up in the
Cubase Track Mixer.
MSP can also be used as a Mixer Application for ReWire Devices such as Reason. To do
this, you use the rewire~
object. Please see the MSP Reference Manual pages on
~ object for more information.
If you try to use rewire~
and the ad_rewire audio driver simultaneously, you
won't get any audio output. This is because each is waiting for the other: the ad_rewire
driver is waiting for the rewire~ object to ask it for an audio stream, but the
object can't do anything unless given processing time by an audio driver.
However, you can use rewire~
in conjunction with the Max Runtime or a
standalone built using Max/MSP when the runtime or standalone is using the ad_rewire
ReWire supports sending synchronization, transport, and tempo information both to and
from ReWire Devices. The hostsync~
MSP objects can work with the ad_rewire driver to provide this
information and to control the host's transport. See the MSP Reference Manual pages of
these objects for more information.
Rewire 2 also supports MIDI communication to and from ReWire Devices. Currently
both the rewire~
object and the ad_rewire driver support MIDI, although they
work in different ways. To send and receive midi using the rewire~
send message to and receive messages directly from the object. See the MSP Reference
Manual pages for the rewire~
object for more information.
The ad_rewire MIDI support is more integrated into the Max MIDI system -- Max
MIDI ports are created so you can use the standard Max MIDI objects to send and receive
MIDI via the ad_rewire driver. After you choose the ad_rewire driver in the DSP Status
Window, MIDI ports will appear in the MIDI Setup window the next time it is opened.
The number of midi ports dedicated to ReWire can be changed using the MIDI Ports
option in the DSP Status Window.
For example, you can choose one of the Max ReWire MIDI ports as a MIDI output
device in Cubase and then use standard Max MIDI objects (such as notein
control your Max/MSP created synthesizer. Likewise, you can send MIDI into Cubase
using the max MIDI objects and the ReWire MIDI ports, and recorded the results to a
track for further manipulation or playback.
When you build a standalone application using Max/MSP you can use the ad_rewire
driver in your standalone and it will create an ReWire Device that works independently
of Max/MSP and other Max/MSP-created standalone applications. By default, the
ReWire Device will take on the name of your application and will have 16 channels. You
can customize this by editing the msprewire.config file in the support/ad folder
for your standalone. Note: This file doesn't exist until the default one is created the first
time you launch your standalone and choose ad_rewire in the DSP Status window.
The msprewire.config file is located in the ad folder found in the following locations:
Macintosh: Library/Application Support/Cycling Ô74/ad/
Windows: c:\Program Files\Common Files\Cycling '74\ad\
In a Max/MSP-built standalone this is in the standalone's support/ad/ folder. The msprewire.config
contains two lines that specify the name that ReWire will use for the device and the number of
audio channels. You can edit this to change the behavior of Max/MSP or your standalone.
Selecting an ASIO driver from the DSP Status window allows MSP to talk directly to an audio
interface. To use ASIO soundcards your device needs to be correctly installed and connected;
The MSP ASIO driver will find it at startup.
All correctly installed ASIO devices should be available to you for selection in the DSP
Status window. However, MSP does not check to see if the relevant audio interface hardware is
installed correctly on your system until you explicitly switch to the ASIO driver for that
interface card. If an ASIO driver fails to load when you try to use it, an error message will
appear in the Max window (typically, an initialization error with a code of –1000) and
the menus in the rest of the DSP Status window will blank out. Switching to the MME and/or
DirectSound driver will re-enable MSP audio.
The Clock Source pop-up menu lets you to set the clock source for your audio hardware. Some
ASIO drivers do not support an external clock; if this is the case there will only be one option
in the menu, typically labeled Internal.
The Prioritize MIDI pop-up menu allows you to set the clock source for your audio hardware and
whether or not to prioritize MIDI input and output over audio I/O.
Many ASIO drivers have other settings you can edit in a separate window. Click the Open ASIO Control
Panel button at the bottom of the DSP Status window to access these settings. If your interface card
has a control panel in its ASIO driver, the documentation for the interface should cover its operation.
Version 2 of the ASIO specification allows for direct monitoring of inputs to an audio interface.
In other words, you can patch audio inputs to the interface directly to audio outputs without having
the signals go through your computer. You also have control over channel patching, volume, and pan settings.
To control direct monitoring, you send the dsp
The message takes the following arguments
message to the
Obligatory. A number specifying an input channel number (starting at 1)
Optional. A number specifying an outlet channel number, or 0 to turn the
routing for the specified input channel off. This is also what happens if the second argument is
or Optional. A number specifying the gain of the input ->
output connection, between 0 and 4. 1 represents unity gain (and is the default).
or Optional. A number specifying the panning of the output
channel. -1 is left, 0 is center, and 1 is right. 0 is the default.
Here are some examples of the
Patches input 1 to output 1 at unity gain with center pan.
Turns off input 1
patches input 1 to output 4 with 6dB gain panned to the left
Note: When using these messages, the word ‘driver’ is optional but recommended. Not all ASIO drivers
support this feature. If you send the
message and get an ASIO result error -998 message
in the Max window, then the driver does not support it.
Another feature of ASIO 2 is the ability to start, stop, and read timecode messages. To start timecode
reading, send the following message:
To stop timecode reading, send the following message:
object reports the sample position reported by the audio interface when you enable
timecode, but there isn't currently an object that reports the timecode of the interface.
The MSP ad_nonreal driver allows you to use MSP for synthesis and signal processing without worrying
about the constraints imposed by the speed of your computer's CPU. Non-real-time mode simply calculates
samples in MSP independently of any physical scheduling priority, allowing you to process a vector of
audio using a signal path that might take your computer more than one vector's worth of real time to compute.
Typically, you will want to use the dsptime~
object to see how long the audio has been turned on,
and you will pipe the output of your routine to sfrecord~
to capture the results. Hardware audio
input and output under the non-real-time driver are disabled.
A typical non-real-time signal path in MSP would look something like this:
Starting the DSP (by toggling the dac~
object) will start the dsptime~
object at 0 samples, in
sync with the playback of the audio out of sfplay~
and the recording of audio into the sfrecord~
the bottom of the patch. When five seconds have elapsed, the sfrecord~
object will stop recording the output
Audio input and on/off
Report and control audio driver settings
Audio output and on/off
Audio input and on/off button
Audio output and on/off button