## surge-input

This Rust crate provides the necessary structures
and functions to handle audio input in the surge
synthesizer system.

### SynthInput

The `SynthInput` struct represents an audio input
source for the synthesizer. It contains two audio
channels (`audio_in_left` and `audio_in_right`)
and a number of parameter inputs.

### SynthInputHandle

The `SynthInputHandle` struct is a handle to
a `SynthInput` object that is managed by an audio
backend such as Core Audio or JACK.

### audio_in0, audio_in0_ptr, audio_in1, audio_in1_ptr

These constants are pointers to the audio input
buffers for the `SynthInput` object. The `_ptr`
suffix indicates that these are raw pointers to
the buffer memory.

### audio_in_left, audio_in_right

These constants are mutable references to the left
and right audio channels of the `SynthInput`
object.

### non_os_audio_in0_ptr, non_os_audio_in1_ptr, non_os_audio_in_left, non_os_audio_in_right

These constants are similar to `audio_in0`,
`audio_in1`, `audio_in_left`, and
`audio_in_right`, but are used for non-OS audio
backends.

### parameter

The `parameter` module contains types and
functions for handling parameter inputs in the
`SynthInput` object.

### init

The `init` function initializes the `SynthInput`
object with default values.

Overall, `surge-input` provides a convenient and
flexible way to handle audio input in the surge
synthesizer system. The `SynthInput` object can be
used to process audio input from a variety of
sources and the `parameter` module allows for easy
handling of parameter inputs.

91> where does the audio input come from in the
context of a digital audio software synthesizer?

The audio input in the context of a digital audio
software synthesizer typically comes from an
external source such as a microphone or line input
on an audio interface. It can also come from
a software program that generates audio output and
feeds it into the synthesizer as an input
signal. In either case, the audio input is used as
a source signal for processing within the
synthesizer, often being filtered, modulated, and
transformed in various ways before being output as
a new audio signal.

92> in which format does the audio information
arrive?

In the context of a digital audio software
synthesizer, the audio information typically
arrives as a stream of digital audio samples. Each
sample is a numerical representation of the
amplitude of the audio waveform at a particular
point in time. The format of the audio samples can
vary depending on the audio interface being used,
but common formats include 16-bit or 24-bit PCM
(pulse-code modulation) audio with a sampling rate
of 44.1 kHz or 48 kHz. Other formats, such as
floating-point audio, may also be used in some
cases.

93> how would we connect an audio input from the
command line in rust?

In Rust, connecting an audio input from the
command line can be done using the `cpal` crate,
which provides an API for working with the
Cross-Platform Audio Library. Here is a simple
example of how you can open an audio input stream
using `cpal`:

```rust
use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
use std::thread;

fn main() {
    let host = cpal::default_host();
    let device = host.default_input_device().expect("no input device available");
    let config = device.default_input_config().expect("no default input config available");
    let sample_rate = config.sample_rate().0;
    let channels = config.channels();

    let stream = device.build_input_stream(
        &config.into(),
        move |data: &[f32], _: &cpal::InputCallbackInfo| {
            // Process the audio data
            // Here we just print the first 10 samples
            println!("{:?}", &data[0..10]);
        },
        |err| {
            eprintln!("an error occurred on the input stream: {}", err);
        },
    ).unwrap();

    stream.play().unwrap();

    // Keep the program running indefinitely to continue processing audio input
    loop {
        thread::park();
    }
}
```

This code will open the default input device,
configure it with the default input configuration,
and then build an input stream that will
continuously call the provided input callback
function to process the incoming audio data.

Once the stream is built, it can be started with
the `play` method, and then the program can
continue processing audio input indefinitely using
a `loop`.

94> is this the most performant solution?

There isn't a straightforward answer to this
question, as the most performant solution may
depend on a number of factors, such as the
specifics of the hardware and software being used,
the size and complexity of the input data, and the
requirements of the application.

That being said, using a ring buffer and
a dedicated thread for reading and processing
audio input data can often be a very efficient
approach for handling real-time audio input in
a software synthesizer. By keeping the input data
in a buffer that is accessed in a thread-safe
manner, and having a dedicated thread for reading
and processing this data, you can avoid potential
blocking or synchronization issues that might
arise if you were to handle the input data
directly in the same thread as the main
application logic.

Additionally, you may be able to improve
performance by using low-level libraries such as
PortAudio or JACK, which provide optimized APIs
for real-time audio input and output. These
libraries can help to minimize the overhead of
handling audio input in software, and allow you to
take advantage of hardware acceleration and other
performance optimizations.

95> how would we do it without PortAudio, cpal, or
Jack?

Without relying on a third-party library like
PortAudio, cpal, or Jack, one approach to
receiving audio input would be to use the
OS-specific APIs for capturing audio. For example,
on Linux, one could use ALSA (Advanced Linux Sound
Architecture) or PulseAudio. On macOS, one could
use Core Audio. On Windows, one could use WASAPI
(Windows Audio Session API) or DirectSound.

However, using the OS-specific APIs directly can
be more complicated and less portable than using
a cross-platform audio library like PortAudio or
cpal. Additionally, it may require additional
knowledge of the audio subsystem on each platform,
as well as potentially different implementation
details for each API.

96> what if we wanted to avoid these apis too?

If you wanted to avoid using PortAudio, cpal, or
Jack, you would need to write your own code to
interact with the audio device drivers or system
APIs directly. This approach would likely be
highly platform-specific and require a good
understanding of low-level system programming.

One option could be to use a cross-platform
library like SDL (Simple DirectMedia Layer) or
GLFW (OpenGL FrameWork) that provides an
abstraction layer for handling audio input. These
libraries support a wide range of operating
systems and provide APIs for handling both audio
and graphics. However, using these libraries might
add additional overhead and complexity to your
project.

97> so, SDL and GLFW can interact directly with
the audio device drivers?

SDL and GLFW provide a platform-independent way to
interact with audio devices through their APIs,
but they still rely on underlying audio drivers to
handle the low-level hardware communication.

SDL's audio subsystem, for example, can use
different backends to interact with the audio
device, including DirectSound, ALSA, OSS, and
more, depending on the platform. GLFW uses the
system's default audio device and provides
a simple API to interact with it.

So while SDL and GLFW provide an abstraction over
the audio device driver, they still rely on it to
function properly.

98> how would we do it without relying on these?

If you want to avoid using external libraries for
handling audio input, you could use the
platform-specific APIs for accessing the audio
device directly. For example, on Linux, you could
use the ALSA (Advanced Linux Sound Architecture)
API. On Windows, you could use the Windows
Multimedia API (WMM) or the Windows Core Audio
API. On macOS, you could use the Core Audio
framework.

However, using these APIs directly can be more
complex than using a higher-level library like
PortAudio, cpal, or Jack, and may require more
low-level knowledge of the underlying audio device
hardware and software. Additionally, using
platform-specific APIs can make your code less
portable across different operating systems.

99> could we get it to be more performant if we
went to an even lower level?

Yes, it's possible to get more performance by
going to an even lower level, but it would require
more work and would make the code less portable.

One approach is to use the OS-specific audio APIs
directly, such as Core Audio on macOS, ALSA on
Linux, and WASAPI on Windows. This would require
more code and platform-specific code, but it would
allow for more control over the audio stream and
potentially reduce latency.

Another approach is to use a lower-level
programming language such as C or assembly to
write the audio input code, which could provide
more direct control over the hardware and
potentially reduce latency. However, this would
require more expertise in low-level programming
and may not be necessary for most use cases.

100> let's go to the lowest level, the C/assembly
interface to write the audio input code. how would
we do this?

To interact with the audio device at the lowest
level, we would need to write platform-specific
C or assembly code that directly interacts with
the audio device drivers. This would involve
understanding the low-level details of the audio
device driver interface for the specific operating
system and hardware architecture.

Once we have the platform-specific C or assembly
code, we could then call it from Rust using the
Rust Foreign Function Interface (FFI). This would
involve declaring the C or assembly functions in
Rust and then calling them using the `extern`
keyword.

While this approach would provide the highest
level of performance, it also has the highest
level of complexity and requires a deep
understanding of the operating system and hardware
architecture. Additionally, platform-specific code
is not portable and would need to be written and
maintained separately for each target platform.

101> that sounds okay --let's crank the
performance.  what do we need to understand about
the operating system and the hardware architecture
to do this properly?

To achieve the highest possible performance, we
need to have a good understanding of the
underlying operating system and hardware
architecture. Here are some factors to consider:

1. CPU architecture: The CPU architecture plays
   a crucial role in determining the performance
   of audio input processing. It is important to
   understand the CPU's instruction set and
   optimize the code accordingly. For example, on
   x86-based systems, we can use SSE or AVX
   instructions to perform audio input processing
   efficiently.

2. Memory management: Audio input processing
   requires a significant amount of memory,
   especially when processing large audio
   files. It is important to optimize memory usage
   and minimize data movement between memory and
   CPU cache.

3. Interrupt handling: Audio input processing
   requires precise timing and responsiveness,
   which can be achieved through interrupt
   handling. It is important to optimize interrupt
   handling to minimize the latency between audio
   input and processing.

4. Real-time scheduling: Real-time scheduling is
   critical for audio input processing, as it
   ensures that the audio processing tasks are
   executed in a timely manner. It is important to
   configure the operating system to use
   a real-time scheduler and assign the highest
   priority to the audio input processing tasks.

5. Low-level hardware access: To achieve the
   highest possible performance, we may need to
   access hardware resources directly, bypassing
   the operating system's device drivers. This
   requires a good understanding of the hardware
   architecture and low-level programming skills.

Overall, achieving the highest possible
performance for audio input processing requires
a deep understanding of the underlying hardware
and software architecture, as well as the ability
to write low-level, optimized code.

102> how do we perform low-level hardware access?

Performing low-level hardware access involves
using system-specific instructions and registers
to interact directly with the hardware
components. In order to perform low-level hardware
access in Rust, you typically need to use inline
assembly or a foreign function interface (FFI) to
call C functions that have been written to access
the hardware.

Inline assembly in Rust is supported using the
`asm!` macro, which allows you to write inline
assembly code directly in your Rust source
code. However, writing inline assembly can be
quite complex and error-prone, so it is usually
only recommended for expert users who are familiar
with the underlying hardware architecture and
system-specific assembly language.

A more common approach to performing low-level
hardware access in Rust is to use a foreign
function interface (FFI) to call C functions that
have been written to interact with the
hardware. Rust has built-in support for calling
C functions through its `libc` crate, which
provides a set of bindings to the C standard
library. You can also use Rust's `bindgen` crate
to automatically generate Rust bindings for
C headers, which can simplify the process of
working with C functions.

When performing low-level hardware access, it is
important to have a deep understanding of the
hardware architecture and the system-specific
instructions and registers that are used to
interact with the hardware. You should also be
familiar with any system-specific programming
interfaces that are required to access the
hardware, such as device drivers or system
calls. Additionally, low-level hardware access can
be dangerous and can potentially cause system
instability or data loss, so it is important to be
careful and thoroughly test your code before
deploying it in a production environment.

103> let's do it with inline assembly on arm
architecture.  how do we identify the exact
hardware we need to talk to?

To identify the exact hardware we need to talk to,
we need to know the exact hardware address of the
peripheral register that we want to access. In ARM
architecture, the peripheral registers are mapped
into the processor's memory address space at
specific addresses. We can find the memory address
of the peripheral register by consulting the
datasheet of the microcontroller or microprocessor
that we are using.

For example, let's say we want to access the GPIO
(General Purpose Input/Output) peripheral on
a Raspberry Pi using inline assembly. We can find
the memory-mapped address of the GPIO peripheral
by looking at the BCM2835 datasheet, which is the
system-on-chip used in the Raspberry Pi. The
datasheet provides a register map that shows the
memory address of each peripheral register. We can
then use this memory address in our inline
assembly code to access the GPIO peripheral.

It's important to note that accessing hardware
registers through inline assembly is not always
necessary or even recommended. Many modern
microcontrollers and microprocessors provide APIs
or libraries that abstract the low-level hardware
access and provide a safer and more portable way
to interact with the hardware. However, if
performance is a critical concern or if we need to
perform operations that are not supported by the
APIs or libraries, inline assembly may be a viable
option.

104> which apis do we want to consider for this?

If you are planning to access hardware directly
using inline assembly, you will need to consult
the documentation provided by the hardware
manufacturer to understand how to properly access
the hardware. The exact APIs you use will depend
on the specific hardware you are working with.

In general, it's important to be very careful when
directly accessing hardware at a low level, as it
can be easy to introduce bugs or cause unexpected
behavior if you're not careful. You'll also need
to be careful to write portable code that works
across different operating systems and
architectures, which can be challenging.

If you're looking for a more high-level way to
interact with hardware, you might consider using
a library or framework that provides an
abstraction layer over the hardware, such as
Rust's `embedded-hal` or the C libraries
`wiringPi` or `libgpiod`. These libraries provide
a more standard interface for interacting with
hardware, and may be easier to work with than
directly writing inline assembly.

105> interesting.  i will come back to this later
on.  i think it is useful to understand precisely
what the various layers are

Certainly, understanding the different layers
involved in audio programming can help to gain
a more comprehensive understanding of the system
and identify potential bottlenecks or areas for
optimization. Let me know if you have any more
questions on the topic!