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Some of these principles apply to all real-time programming, while others are specific to getting stable real-time audio behavior on systems that are not specifically designed or configured for real-time operation i.
These principles are not platform-specific. Real-time waits for nothing Digital audio works by playing a constant stream of audio samples numbers to the digital to analog converter DAC of your sound card or audio interface.
The Thesis classic responsive are played out at a constant rate known as the sampling rate. Every second at the same rate. Not faster, not slower.
For example it might process buffers of samples at a rate of The lower levels of the system then feed the individual samples from each buffer to the DAC at Hz.
In the above example your callback would have to compute each and every buffer in less than 5. No matter how your code is invoked your software has to provide those samples within 5.
Each and every buffer. Real-time does not wait for latecomers. To many users today 5ms is considered a large buffer size.
Applications where low latency is especially important are 1 interactive audio systems such as musical instruments or DJ tools where the UI needs to be responsive to the performer, and 2 real-time audio effects, where the system needs to process analog input say from a guitar and output the processed signal without noticeable delay.
For live audio effects processing many users would prefer latency to be much lower than this. I assume that you want to write low-latency audio software for one or more of these platforms.
Your code has to deliver each and every buffer of audio in a time shorter than one buffer period.
All sources of audio glitches within your code boil down to doing something that takes longer than the buffer period. If not, the internet is full of resources to help you write faster code. Whatever the cause, the result is the same: Therefore, we can state the cardinal rule of real-time audio programming simply as follows: Many are mentioned in the quotes at the start of this post.
I explore them in more detail below. As you might guess, some are more obvious than others.
Doing anything that makes your audio code wait for something else in the system would be blocking. This could be acquiring a mutex, waiting for a resource such as a semaphore, waiting for some other thread or process to do something, waiting for data to be read from disk, waiting for a network socket.
I discuss some of these specific types of blocking in more detail below. Keep in mind that not only do you want to avoid directly writing code that blocks, it is critical that you avoid calling 3rd-party or operating system code that could block internally.
No calls to any system or 3rd-party code that could block. Even so, you might still have a problem: For example, in many applications, an algorithm that runs super-fast If you do something like that in your audio callback, you may get a glitch. For this reason, you should always consider the worst-case execution time of your code.
A simple example would be zeroing a delay line to reset it say using a for-loop to zero every element, or perhaps by calling memset. Another thing to keep in mind here is that many operating system and library functions are implemented using average-case optimised algorithms.
Similar requirements may arise if you need to control audio via a network socket or MIDI. This is a common response.
I remember having it too.
Of course modern operating systems do provide locks mutexes to protect against concurrent access to shared state. You should not use them within an audio callback though.
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