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Configuring Audio & Tuning Audio Performance

Selecting Audio Devices and Drivers

Csound relates to the various inputs and outputs of sound devices installed on your computer as a numbered list. If you wish to send or receive audio to or from a specific audio connection you will need to know the number by which Csound knows it. If you are not sure of what that is you can trick Csound into providing you with a list of available devices by trying to run Csound using an obviously out of range device number, like this:

   EXAMPLE 02D01_GetDeviceList.csd

-iadc999 -odac999
;Example by Andrés Cabrera
instr 1


The input and output devices will be listed seperately.1  Specify your input device with the -iadc flag and the number of your input device, and your output device with the -odac flag and the number of your output device. For instance, if you select one of the devices from the list above both, for input and output, you may include something like

 -iadc2 -odac3

in the <CsOptions> section of your .csd file.

The RT (= real-time) output module can be set with the -+rtaudio flag. If you don't use this flag, the PortAudio driver will be used. Other possible drivers are jack and alsa (Linux), mme (Windows) or CoreAudio (Mac). So, this sets your audio driver to mme instead of Port Audio:


Tuning Performance and Latency

Live performance and latency depend mainly on the sizes of the software and the hardware buffers. They can be set in the <CsOptions> using the -B flag for the hardware buffer, and the -b flag for the software buffer.2  For instance, this statement sets the hardware buffer size to 512 samples and the software buffer size to 128 sample:

-B512 -b128

The other factor which affects Csound's live performance is the ksmps value which is set in the header of the <CsInstruments> section. By this value, you define how many samples are processed every Csound control cycle.

Try your realtime performance with -B512, -b128 and ksmps=32.3  With a software buffer of 128 samples, a hardware buffer of 512 and a sample rate of 44100 you will have around 12ms latency, which is usable for live keyboard playing. If you have problems with either the latency or the performance, tweak the values as described here.

The "--realtime" Option

When you have instruments that have substantial sections that could block out execution, for instance with code that loads buffers from files or creates big tables, you can try the option --realtime.

This option will give your audio processing the priority over other tasks to be done. It places all initialisation code on a separate thread, and does not block the audio thread. Instruments start performing only after all the initialisation is done. That can have a side-effect on scheduling if your audio input and output buffers are not small enough, because the audio processing thread may “run ahead” of the initialisation one, taking advantage of any slack in the buffering.

Given that this option is intrinsically linked to low-latency, realtime audio performance, and also to reduce the effect on scheduling these other tasks, it is recommended that small ksmps and buffer sizes, for example ksmps=16, 32, or 64, -b32 or 64, and -B256 or 512.


To define the audio hardware used for realtime performance, open the configuration dialog. In the "Run" Tab, you can choose your audio interface, and the preferred driver. You can select input and output devices from a list if you press the buttons to the right of the text boxes for input and output names. Software and hardware buffer sizes can be set at the top of this dialogue box.


Csound Can Produce Extreme Dynamic Range!


Csound can produce extreme dynamic range, so keep an eye on the level you are sending to your output. The number which describes the level of 0 dB, can be set in Csound by the 0dbfs assignment in the <CsInstruments> header. There is no limitation, if you set 0dbfs = 1 and send a value of 32000, this can damage your ears and speakers!

Using Live Audio Input and Output

To process audio from an external source (for example a microphone), use the inch opcode to access any of the inputs of your audio input device. For the output, outch gives you all necessary flexibility. The following example takes a live audio input and transforms its sound using ring modulation. The Csound Console should output five times per second the input amplitude level.

   EXAMPLE 02D02_LiveInput.csd

-iadc0 -odac0 -B512 -b128
;Example by Joachim Heintz
sr = 44100 ;set sample rate to 44100 Hz
ksmps = 32 ;number of samples per control cycle
nchnls = 2 ;use two audio channels
0dbfs = 1 ;set maximum level as 1

instr 1
aIn       inch      1   ;take input from channel 1
kInLev    downsamp  aIn ;convert audio input to control signal
          printk    .2, abs(kInLev)
;make modulator frequency oscillate 200 to 1000 Hz
kModFreq  poscil    400, 1/2
kModFreq  =         kModFreq+600
aMod      poscil    1, kModFreq ;modulator signal
aRM       =         aIn * aMod ;ring modulation
          outch     1, aRM, 2, aRM ;output to channel 1 and 2
i 1 0 3600

Live Audio is frequently used with live devices like widgets or MIDI. In CsoundQt, you can find several examples in Examples -> Getting Started -> Realtime Interaction.

  1. You may have to run -iadc999 and -odac999 seperately.^
  2. As Victor Lazzarini explains (mail to Joachim Heintz, 19 march 2013), the role of -b and -B varies between the Audio Modules:
    "1. For portaudio, -B is only used to suggest a latency to the backend, whereas -b is used to set the actual buffersize.
    2. For coreaudio, -B is used as the size of the internal circular buffer, and -b is used for the actual IO buffer size.
    3. For jack, -B is used to determine the number of buffers used in conjunction with -b , num = (N + M + 1) / M. -b is the size of each buffer.
    4. For alsa, -B is the size of the buffer size, -b is the period size (a buffer is divided into periods).
    5. For pulse, -b is the actual buffersize passed to the device, -B is not used.
    In other words, -B is not too significant in 1), not used in 5), but has a part to play in 2), 3) and 4), which is functionally similar." ^
  3. It is always preferable to use power-of-two values for ksmps (which is the same as "block size" in PureData or "vector size" in Max). Just with ksmps = 1, 2, 4, 8, 16 ... you will take advantage of the "full duplex" audio, which provides best real time audio. Make sure your ksmps divides your buffer size with no remainder. So, for -b 128, you can use ksmps = 128, 64, 32, 16, 8, 4, 2 or 1.^

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