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# Objective Stumbled on one of these and went digging for more ## Solution ```diff - word word + word ```
152 lines
7.2 KiB
Markdown
152 lines
7.2 KiB
Markdown
# Contributing to `bevy_audio`
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This document highlights documents some general explanation and guidelines for
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contributing code to this crate. It assumes knowledge of programming, but not
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necessarily of audio programming specifically. It lays out rules to follow, on
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top of the general programming and contribution guidelines of Bevy, that are of
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particular interest for performance reasons.
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This section applies to the equivalent in abstraction level to working with
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nodes in the render graph, and not manipulating entities with meshes and
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materials.
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Note that these guidelines are general to any audio programming application, and
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not just Bevy.
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## Fundamentals of working with audio
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### A brief introduction to digital audio signals
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Audio signals, when working within a computer, are digital streams of audio
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samples (historically with different types, but nowadays the values are 32-bit
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floats), taken at regular intervals of each other.
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How often this sampling is done is determined by the **sample rate** parameter.
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This parameter is available to the users in OS settings, as well as some
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applications.
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The sample rate directly determines the spectrum of audio frequencies that will
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be representable by the system. That limit sits at half the sample rate, meaning
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that any sound with frequencies higher than that will introduce artifacts.
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If you want to learn more, read about the **Nyquist sampling theorem** and
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**Frequency aliasing**.
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### How the computer interfaces with the sound card
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When requesting for audio input or output, the OS creates a special
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high-priority thread whose task it is to take in the input audio stream, and/or
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produce the output stream. The audio driver passes an audio buffer that you read
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from (for input) or write to (for output). The size of that buffer is also a
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parameter that is configured when opening an audio stream with the sound card,
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and is sometimes reflected in application settings.
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Typical values for buffer size and sample rate are 512 samples at a sample rate
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of 48 kHz. This means that for every 512 samples of audio the driver is going to
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send to the sound card the output callback function is run in this high-priority
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audio thread. Every second, as dictated by the sample rate, the sound card
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needs 48 000 samples of audio data. This means that we can expect the callback
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function to be run every `512/(48000 Hz)` or 10.666... ms.
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This figure is also the latency of the audio engine, that is, how much time it
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takes between a user interaction and hearing the effects out the speakers.
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Therefore, there is a "tug of war" between decreasing the buffer size for
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latency reasons, and increasing it for performance reasons. The threshold for
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instantaneity in audio is around 15 ms, which is why 512 is a good value for
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interactive applications.
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### Real-time programming
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The parts of the code running in the audio thread have exactly
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`buffer_size/samplerate` seconds to complete, beyond which the audio driver
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outputs silence (or worse, the previous buffer output, or garbage data), which
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the user perceives as a glitch and severely deteriorates the quality of the
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audio output of the engine. It is therefore critical to work with code that is
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guaranteed to finish in that time.
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One step to achieving this is making sure that all machines across the spectrum
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of supported CPUs can reliably perform the computations needed for the game in
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that amount of time, and play around with the buffer size to find the best
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compromise between latency and performance. Another is to conditionally enable
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certain effects for more powerful CPUs, when that is possible.
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But the main step is to write code to run in the audio thread following
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real-time programming guidelines. Real-time programming is a set of constraints
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on code and structures that guarantees the code completes at some point, ie. it
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cannot be stuck in an infinite loop nor can it trigger a deadlock situation.
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Practically, the main components of real-time programming are about using
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wait-free and lock-free structures. Examples of things that are *not* correct in
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real-time programming are:
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- Allocating anything on the heap (that is, no direct or indirect creation of a
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`Vec`, `Box`, or any standard collection, as they are not designed with
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real-time programming in mind)
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- Locking a mutex - Generally, any kind of system call gives the OS the
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opportunity to pause the thread, which is an unbounded operation as we don't
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know how long the thread is going to be paused for
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- Waiting by looping until some condition is met (also called a spinloop or a
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spinlock)
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Writing wait-free and lock-free structures is a hard task, and difficult to get
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correct; however many structures already exists, and can be directly used. There
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are crates for most replacements of standard collections.
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### Where in the code should real-time programming principles be applied?
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Any code that is directly or indirectly called by audio threads, needs to be
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real-time safe.
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For the Bevy engine, that is:
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- In the callback of `cpal::Stream::build_input_stream` and
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`cpal::Stream::build_output_stream`, and all functions called from them
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- In implementations of the [`Source`] trait, and all functions called from it
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Code that is run in Bevy systems do not need to be real-time safe, as they are
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not run in the audio thread, but in the main game loop thread.
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## Communication with the audio thread
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To be able to do anything useful with audio, the thread has to be able to
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communicate with the rest of the system, ie. update parameters, send/receive
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audio data, etc., and all of that needs to be done within the constraints of
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real-time programming, of course.
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### Audio parameters
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In most cases, audio parameters can be represented by an atomic floating point
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value, where the game loop updates the parameter, and it gets picked up when
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processing the next buffer. The downside to this approach is that the audio only
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changes once per audio callback, and results in a noticeable "stair-step "
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motion of the parameter. The latter can be mitigated by "smoothing" the change
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over time, using a tween or linear/exponential smoothing.
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Precise timing for non-interactive events (ie. on the beat) need to be setup
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using a clock backed by the audio driver -- that is, counting the number of
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samples processed, and deriving the time elapsed by diving by the sample rate to
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get the number of seconds elapsed. The precise sample at which the parameter
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needs to be changed can then be computed.
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Both interactive and precise events are hard to do, and need very low latency
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(ie. 64 or 128 samples for ~2 ms of latency). It is fundamentally impossible to
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react to user event the very moment it is registered.
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### Audio data
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Audio data is generally transferred between threads with circular buffers, as
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they are simple to implement, fast enough for 99% of use-cases, and are both
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wait-free and lock-free. The only difficulty in using circular buffers is how
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big they should be; however even going for 1 s of audio costs ~50 kB of memory,
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which is small enough to not be noticeable even with potentially 100s of those
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buffers.
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## Additional resources for audio programming
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More in-depth article about audio programming:
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<http://www.rossbencina.com/code/real-time-audio-programming-101-time-waits-for-nothing>
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Awesome Audio DSP: <https://github.com/BillyDM/awesome-audio-dsp>
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