From 9837e41f204e12c123e35a84374eda08ac1a99db Mon Sep 17 00:00:00 2001 From: Andrea Ciliberti Date: Sun, 1 Jan 2023 19:57:40 +0100 Subject: [PATCH] Separated module and new anti-drop measures --- ctru-rs/examples/audio_filters.rs | 62 ++++---- ctru-rs/src/error.rs | 4 +- ctru-rs/src/services/ndsp.rs | 253 ------------------------------ 3 files changed, 34 insertions(+), 285 deletions(-) delete mode 100644 ctru-rs/src/services/ndsp.rs diff --git a/ctru-rs/examples/audio_filters.rs b/ctru-rs/examples/audio_filters.rs index e2a1daa..f99957b 100644 --- a/ctru-rs/examples/audio_filters.rs +++ b/ctru-rs/examples/audio_filters.rs @@ -1,5 +1,7 @@ #![feature(allocator_api)] +use std::f32::consts::PI; + use ctru::linear::LinearAllocator; use ctru::prelude::*; use ctru::services::ndsp::{ @@ -10,15 +12,18 @@ const SAMPLERATE: u32 = 22050; const SAMPLESPERBUF: u32 = SAMPLERATE / 30; // 735 const BYTESPERSAMPLE: u32 = 4; +// Note Frequencies +const NOTEFREQ: [u32; 7] = [220, 440, 880, 1760, 3520, 7040, 14080]; + fn array_size(array: &[u8]) -> usize { array.len() } // (sizeof(array)/sizeof(array[0])) // audioBuffer is stereo PCM16 -fn fill_buffer(audioData: &mut Box<[u8], LinearAlloc>, frequency: i32) { +fn fill_buffer(audioData: &mut Box<[u8], LinearAlloc>, frequency: u32) { for i in 0..size { // This is a simple sine wave, with a frequency of `frequency` Hz, and an amplitude 30% of maximum. - let sample: i16 = 0.3 * 0x7FFF * sin(frequency * (2 * std::f32::PI) * i / SAMPLERATE); + let sample: i16 = 0.3 * 0x7FFF * (frequency * (2f32 * PI) * i / SAMPLERATE).sin(); // Stereo samples are interleaved: left and right channels. audioData[i] = (sample << 16) | (sample & 0xffff); @@ -35,10 +40,10 @@ fn main() { println!("libctru filtered streamed audio\n"); let audioBuffer = Box::new_in( - [0u32; (SAMPLESPERBUF * BYTESPERSAMPLE * 2)], + [0u32, (SAMPLESPERBUF * BYTESPERSAMPLE * 2)], LinearAllocator, ); - fill_buffer(audioBuffer, notefreq[note]); + fill_buffer(audioBuffer, NOTEFREQ[note]); let audioBuffer1 = WaveBuffer::new(audioBuffer, AudioFormat::PCM16Stereo).expect("Couldn't sync DSP cache"); @@ -53,23 +58,17 @@ fn main() { let channel_zero = ndsp.channel(0).unwrap(); channel_zero.set_interpolation(InterpolationType::Linear); - channel_zero.set_sample_rate(SAMPLERATE); - channel_zero.set_format(NDSP_FORMAT_STEREO_PCM16); + channel_zero.set_sample_rate(SAMPLERATE as f32); + channel_zero.set_format(AudioFormat::PCM16Stereo); // Output at 100% on the first pair of left and right channels. let mix = [0f32; 12]; mix[0] = 1.0; mix[1] = 1.0; - channel_zero.set_mix(mix); - - // Note Frequencies + channel_zero.set_mix(&mix); - let notefreq = [ - 220, 440, 880, 1760, 3520, 7040, 14080, 7040, 3520, 1760, 880, 440, - ]; - - let note: i32 = 4; + let note: usize = 4; // Filters @@ -91,13 +90,13 @@ fn main() { let mut buf2 = WaveInfo::new(&mut audioBuffer2, false); unsafe { - channel_zero.add_wave_buffer(buf1); - channel_zero.add_wave_buffer(buf2); + channel_zero.queue_wave(&mut buf1); + channel_zero.queue_wave(&mut buf2); }; println!("Press up/down to change tone frequency\n"); println!("Press left/right to change filter\n"); - println!("\x1b[6;1Hnote = {} Hz ", notefreq[note]); + println!("\x1b[6;1Hnote = {} Hz ", NOTEFREQ[note]); println!("\x1b[7;1Hfilter = {} ", filter_names[filter]); while apt.main_loop() { @@ -109,29 +108,32 @@ fn main() { } // break in order to return to hbmenu if keys_down.contains(KeyPad::KEY_DOWN) { - note -= 1; - if (note < 0) { - note = notefreq.len() - 1; + note = note.saturating_sub(1); + if note < 0 { + note = NOTEFREQ.len() - 1; } - println!("\x1b[6;1Hnote = {} Hz ", notefreq[note]); + println!("\x1b[6;1Hnote = {} Hz ", NOTEFREQ[note]); } else if keys_down.contains(KeyPad::KEY_UP) { note += 1; - if (note >= notefreq.len()) { + if note >= NOTEFREQ.len() { note = 0; } - println!("\x1b[6;1Hnote = {} Hz ", notefreq[note]); + println!("\x1b[6;1Hnote = {} Hz ", NOTEFREQ[note]); } + // Check for upper limit + note = std::cmp::max(note, NOTEFREQ.len() - 1); + let update_params = false; if keys_down.contains(KeyPad::KEY_LEFT) { - filter -= 1; - if (filter < 0) { + filter = filter.saturating_sub(1); + if filter < 0 { filter = filter_names.len() - 1; } update_params = true; } else if keys_down.contains(KeyPad::KEY_LEFT) { filter += 1; - if (filter >= filter_names.len()) { + if filter >= filter_names.len() { filter = 0; } update_params = true; @@ -151,11 +153,11 @@ fn main() { if waveBuf[fillBlock].status == NDSP_WBUF_DONE { if fillBlock { - fill_buffer(buf1.data_pcm16, notefreq[note]); - channel_zero.add_wave_buffer(buf1); + fill_buffer(buf1.get_mut_wavebuffer().get_mut_data(), NOTEFREQ[note]); + channel_zero.queue_wave(&mut buf1); } else { - fill_buffer(waveBuf[fillBlock].data_pcm16, notefreq[note]); - channel_zero.add_wave_buffer(buf2); + fill_buffer(buf2.get_mut_wavebuffer().get_mut_data(), NOTEFREQ[note]); + channel_zero.queue_wave(&mut buf2); } fillBlock = !fillBlock; } diff --git a/ctru-rs/src/error.rs b/ctru-rs/src/error.rs index 1d58c45..ccaadaa 100644 --- a/ctru-rs/src/error.rs +++ b/ctru-rs/src/error.rs @@ -114,8 +114,8 @@ impl fmt::Display for Error { Self::ServiceAlreadyActive => write!(f, "Service already active"), Self::OutputAlreadyRedirected => { write!(f, "output streams are already redirected to 3dslink") - }, - Self::InvalidChannel(id) => write!(f, "Audio Channel with id {id} doesn't exist") + } + Self::InvalidChannel(id) => write!(f, "Audio Channel with id {id} doesn't exist"), } } } diff --git a/ctru-rs/src/services/ndsp.rs b/ctru-rs/src/services/ndsp.rs deleted file mode 100644 index bd00cfb..0000000 --- a/ctru-rs/src/services/ndsp.rs +++ /dev/null @@ -1,253 +0,0 @@ -use crate::error::ResultCode; -use crate::linear::LinearAllocator; - -#[derive(Copy, Clone, Debug)] -#[repr(u32)] -pub enum OutputMode { - Mono = ctru_sys::NDSP_OUTPUT_MONO, - Stereo = ctru_sys::NDSP_OUTPUT_STEREO, - Surround = ctru_sys::NDSP_OUTPUT_SURROUND, -} - -#[derive(Copy, Clone, Debug)] -#[repr(u32)] -pub enum InterpolationType { - Polyphase = ctru_sys::NDSP_INTERP_POLYPHASE, - Linear = ctru_sys::NDSP_INTERP_LINEAR, - None = ctru_sys::NDSP_INTERP_NONE, -} - -#[derive(Copy, Clone, Debug)] -#[repr(u32)] -pub enum AudioFormat { - PCM8Mono = ctru_sys::NDSP_FORMAT_MONO_PCM8, - PCM16Mono = ctru_sys::NDSP_FORMAT_MONO_PCM16, - ADPCMMono = ctru_sys::NDSP_FORMAT_MONO_ADPCM, - PCM8Stereo = ctru_sys::NDSP_FORMAT_STEREO_PCM8, - PCM16Stereo = ctru_sys::NDSP_FORMAT_STEREO_PCM16, - FrontBypass = ctru_sys::NDSP_FRONT_BYPASS, - SurroundPreprocessed = ctru_sys::NDSP_3D_SURROUND_PREPROCESSED, -} - -/// Base struct to represent audio wave data. This requires audio format information. -#[derive(Debug, Clone)] -pub struct WaveBuffer { - /// Buffer data. This data must be allocated on the LINEAR memory. - data: Box<[u8], LinearAllocator>, - audio_format: AudioFormat, - nsamples: usize, // We don't use the slice's length here because depending on the format it may vary - // adpcm_data: AdpcmData, TODO: Requires research on how this format is handled. -} - -/// Informational struct holding the raw audio data and playaback info. This corresponds to [ctru_sys::ndspWaveBuf] -pub struct WaveInfo<'b> { - /// Data block of the audio wave (plus its format information). - buffer: &'b mut WaveBuffer, - // Holding the data with the raw format is necessary since `libctru` will access it. - raw_data: ctru_sys::ndspWaveBuf, -} - -pub struct Channel { - id: i32, -} - -#[non_exhaustive] -pub struct Ndsp(()); - -impl Ndsp { - pub fn init() -> crate::Result { - ResultCode(unsafe { ctru_sys::ndspInit() })?; - - Ok(Self(())) - } - - /// Return the representation of the specified channel. - /// - /// # Errors - /// - /// An error will be returned if the channel id is not between 0 and 23. - pub fn channel(&self, id: u8) -> crate::Result { - if id > 23 { - return Err(crate::Error::InvalidChannel(id.into())); - } - - Ok(Channel { id: id.into() }) - } - - /// Set the audio output mode. Defaults to `OutputMode::Stereo`. - pub fn set_output_mode(&mut self, mode: OutputMode) { - unsafe { ctru_sys::ndspSetOutputMode(mode as u32) }; - } -} - -// All channel operations are thread-safe thanks to `libctru`'s use of thread locks. -// As such, there is no need to hold channels to ensure correct mutability. -// With this prospective in mind, this struct looks more like a dummy than an actually functional block. -impl Channel { - /// Reset the channel - pub fn reset(&self) { - unsafe { ctru_sys::ndspChnReset(self.id) }; - } - - /// Initialize the channel's parameters - pub fn init_parameters(&self) { - unsafe { ctru_sys::ndspChnInitParams(self.id) }; - } - - /// Returns whether the channel is playing any audio. - pub fn is_playing(&self) -> bool { - unsafe { ctru_sys::ndspChnIsPlaying(self.id) } - } - - /// Returns whether the channel's playback is currently paused. - pub fn is_paused(&self) -> bool { - unsafe { ctru_sys::ndspChnIsPaused(self.id) } - } - - /// Returns the channel's current sample's position. - pub fn get_sample_position(&self) -> u32 { - unsafe { ctru_sys::ndspChnGetSamplePos(self.id) } - } - - /// Returns the channel's current wave sequence's id. - pub fn get_wave_sequence_id(&self) -> u16 { - unsafe { ctru_sys::ndspChnGetWaveBufSeq(self.id) } - } - - /// Pause or un-pause the channel's playback. - pub fn set_paused(&self, state: bool) { - unsafe { ctru_sys::ndspChnSetPaused(self.id, state) }; - } - - /// Set the channel's output format. - /// Change this setting based on the used sample's format. - pub fn set_format(&self, format: AudioFormat) { - unsafe { ctru_sys::ndspChnSetFormat(self.id, format as u16) }; - } - - /// Set the channel's interpolation mode. - pub fn set_interpolation(&self, interp_type: InterpolationType) { - unsafe { ctru_sys::ndspChnSetInterp(self.id, interp_type as u32) }; - } - - /// Set the channel's volume mix. - /// Docs about the buffer usage: https://libctru.devkitpro.org/channel_8h.html#a30eb26f1972cc3ec28370263796c0444 - pub fn set_mix(&self, mix: &[f32; 12]) { - unsafe { ctru_sys::ndspChnSetMix(self.id, mix.as_ptr().cast_mut()) } - } - - /// Set the channel's rate of sampling. - pub fn set_sample_rate(&self, rate: f32) { - unsafe { ctru_sys::ndspChnSetRate(self.id, rate) }; - } - - // TODO: find a way to wrap `ndspChnSetAdpcmCoefs` - - /// Clear the wave buffer queue and stop playback. - pub fn clear_queue(&self) { - unsafe { ctru_sys::ndspChnWaveBufClear(self.id) }; - } - - /// Add a wave buffer to the channel's queue. - /// Note: if there are no other buffers in queue, playback for this buffer will start. - /// - /// # Unsafe - /// - /// This function is unsafe due to how the buffer is handled internally. - /// `libctru` expects the user to manually keep the info data (in this case [WaveInfo]) alive during playback. - /// Furthermore, there are no checks to see if the used memory is still valid. All responsibility of handling this data is left to the user. - - // INTERNAL NOTE: After extensive research to make a Rust checker for these cases, - // I came to the conclusion that leaving the responsibility to the user is (as of now) the only "good" way to handle this. - // Sadly `libctru` lacks the infrastructure to make runtime checks on the queued objects, like callbacks and iterators. - // Also, in most cases the memory is still accessible even after a `free`, so it may not be a problem to the average user. - // This is still a big "hole" in the Rust wrapper. Upstream changes to `libctru` would be my go-to way to solve this issue. - pub unsafe fn queue_wave(&self, mut buffer: WaveInfo) { - unsafe { ctru_sys::ndspChnWaveBufAdd(self.id, &mut buffer.raw_data) }; - } -} - -impl AudioFormat { - /// Returns the amount of bytes needed to store one sample - /// Eg. - /// 8 bit formats return 1 (byte) - /// 16 bit formats return 2 (bytes) - pub fn bytes_size(self) -> u8 { - match self { - AudioFormat::PCM16Mono | AudioFormat::PCM16Stereo => 2, - AudioFormat::SurroundPreprocessed => { - panic!("Can't find size for Sourround Preprocessed audio: format is under research") - } - _ => 1, - } - } -} - -impl WaveBuffer { - pub fn new(data: Box<[u8], LinearAllocator>, audio_format: AudioFormat) -> crate::Result { - let nsamples: usize = data.len() / (audio_format.bytes_size() as usize); - - unsafe { - ResultCode(ctru_sys::DSP_FlushDataCache(data.as_ptr().cast(), data.len().try_into().unwrap()))?; - } - - Ok(WaveBuffer { - data, - audio_format, - nsamples, - }) - } - - pub fn get_mut_data(&mut self) -> &mut Box<[u8], LinearAllocator> { - &mut self.data - } - - pub fn get_format(&self) -> AudioFormat { - self.audio_format - } - - pub fn get_sample_amount(&self) -> usize { - self.nsamples - } -} - -impl<'b> WaveInfo<'b> { - pub fn new(buffer: &'b mut WaveBuffer, looping: bool) -> Self { - let address = ctru_sys::tag_ndspWaveBuf__bindgen_ty_1{ data_vaddr: buffer.data.as_ptr().cast() }; - - let raw_data = ctru_sys::ndspWaveBuf { - __bindgen_anon_1: address, // Buffer data virtual address - nsamples: buffer.nsamples.try_into().unwrap(), - adpcm_data: std::ptr::null_mut(), - offset: 0, - looping, - // The ones after this point aren't supposed to be setup by the user - status: 0, - sequence_id: 0, - next: std::ptr::null_mut(), - }; - - Self { buffer, raw_data } - } - - pub fn get_mut_wavebuffer(&'b mut self) -> &'b mut WaveBuffer { - &mut self.buffer - } -} - -impl Drop for Ndsp { - fn drop(&mut self) { - unsafe { - ctru_sys::ndspExit(); - } - } -} - -impl Drop for WaveBuffer { - fn drop(&mut self) { - unsafe { - // Result can't be used in any way, let's just shrug it off - let _r = ctru_sys::DSP_InvalidateDataCache(self.data.as_ptr().cast(), self.data.len().try_into().unwrap()); - } - } -}