WIP Rustification of the example

ctru-rs/examples/audio_filters.rs

@@ -1,8 +1,10 @@
 #![feature(allocator_api)]
 
-use ctru::prelude::*;
-use ctru::services::ndsp::{Ndsp, OutputMode, InterpolationType};
 use ctru::linear::LinearAllocator;
+use ctru::prelude::*;
+use ctru::services::ndsp::{
+    AudioFormat, InterpolationType, Ndsp, OutputMode, WaveBuffer, WaveInfo,
+};
 
 const SAMPLERATE: u32 = 22050;
 const SAMPLESPERBUF: u32 = SAMPLERATE / 30; // 735
@@ -13,18 +15,14 @@ fn array_size(array: &[u8]) -> usize {
 } // (sizeof(array)/sizeof(array[0]))
 
 // audioBuffer is stereo PCM16
-void fill_buffer(void* audioBuffer, size_t offset, size_t size, int frequency) {
-	u32* dest = (u32*) audioBuffer;
-
-	for (int i = 0; i < size; i++) {
+fn fill_buffer(audioData: &mut Box<[u8], LinearAlloc>, frequency: i32) {
+    for i in 0..size {
         // This is a simple sine wave, with a frequency of `frequency` Hz, and an amplitude 30% of maximum.
-		s16 sample = 0.3 * 0x7FFF * sin(frequency * (2 * M_PI) * (offset + i) / SAMPLERATE);
+        let sample: i16 = 0.3 * 0x7FFF * sin(frequency * (2 * std::f32::PI) * i / SAMPLERATE);
 
         // Stereo samples are interleaved: left and right channels.
-		dest[i] = (sample << 16) | (sample & 0xffff);
+        audioData[i] = (sample << 16) | (sample & 0xffff);
     }
-
-	DSP_FlushDataCache(audioBuffer, size);
 }
 
 fn main() {
@@ -36,7 +34,15 @@ fn main() {
 
     println!("libctru filtered streamed audio\n");
 
-	let audioBuffer = Box::new_in([0u32; (SAMPLESPERBUF * BYTESPERSAMPLE * 2)], LinearAllocator);
+    let audioBuffer = Box::new_in(
+        [0u32; (SAMPLESPERBUF * BYTESPERSAMPLE * 2)],
+        LinearAllocator,
+    );
+    fill_buffer(audioBuffer, notefreq[note]);
+
+    let audioBuffer1 =
+        WaveBuffer::new(audioBuffer, AudioFormat::PCM16Stereo).expect("Couldn't sync DSP cache");
+    let audioBuffer2 = audioBuffer1.clone();
 
     let fillBlock = false;
 
@@ -60,10 +66,7 @@ fn main() {
     // Note Frequencies
 
     let notefreq = [
-		220,
-		440, 880, 1760, 3520, 7040,
-		14080,
-		7040, 3520, 1760, 880, 440
+        220, 440, 880, 1760, 3520, 7040, 14080, 7040, 3520, 1760, 880, 440,
     ];
 
     let note: i32 = 4;
@@ -76,7 +79,7 @@ fn main() {
         "High-Pass",
         "Band-Pass",
         "Notch",
-		"Peaking"
+        "Peaking",
     ];
 
     let filter = 0;
@@ -84,97 +87,84 @@ fn main() {
     // We set up two wave buffers and alternate between the two,
     // effectively streaming an infinitely long sine wave.
 
-	ndspWaveBuf waveBuf[2];
-	memset(waveBuf,0,sizeof(waveBuf));
-	waveBuf[0].data_vaddr = &audioBuffer[0];
-	waveBuf[0].nsamples = SAMPLESPERBUF;
-	waveBuf[1].data_vaddr = &audioBuffer[SAMPLESPERBUF];
-	waveBuf[1].nsamples = SAMPLESPERBUF;
+    let mut buf1 = WaveInfo::new(&mut audioBuffer1, false);
+    let mut buf2 = WaveInfo::new(&mut audioBuffer2, false);
 
-	let stream_offset = 0;
-
-	fill_buffer(audioBuffer,stream_offset, SAMPLESPERBUF * 2, notefreq[note]);
-
-	stream_offset += SAMPLESPERBUF;
-
-	channel_zero.add_wave_buffer(&waveBuf[0]);
-	channel_zero.add_wave_buffer(&waveBuf[1]);
+    unsafe {
+        channel_zero.add_wave_buffer(buf1);
+        channel_zero.add_wave_buffer(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[7;1Hfilter = {}         ", filter_names[filter]);
 
-	while(aptMainLoop()) {
-
-		gfxSwapBuffers();
-		gfxFlushBuffers();
-		gspWaitForVBlank();
-
-		hidScanInput();
-		u32 kDown = hidKeysDown();
+    while apt.main_loop() {
+        hid.scan_input();
+        let keys_down = hid.keys_down();
 
-		if (kDown & KEY_START)
-			break; // break in order to return to hbmenu
+        if keys_down.contains(KeyPad::KEY_START) {
+            break;
+        } // break in order to return to hbmenu
 
-		if (kDown & KEY_DOWN) {
-			note--;
+        if keys_down.contains(KeyPad::KEY_DOWN) {
+            note -= 1;
             if (note < 0) {
-				note = ARRAY_SIZE(notefreq) - 1;
+                note = notefreq.len() - 1;
             }
             println!("\x1b[6;1Hnote = {} Hz        ", notefreq[note]);
-		} else if (kDown & KEY_UP) {
-			note++;
-			if (note >= ARRAY_SIZE(notefreq)) {
+        } else if keys_down.contains(KeyPad::KEY_UP) {
+            note += 1;
+            if (note >= notefreq.len()) {
                 note = 0;
             }
             println!("\x1b[6;1Hnote = {} Hz        ", notefreq[note]);
         }
 
-		bool update_params = false;
-		if (kDown & KEY_LEFT) {
-			filter--;
+        let update_params = false;
+        if keys_down.contains(KeyPad::KEY_LEFT) {
+            filter -= 1;
             if (filter < 0) {
-				filter = ARRAY_SIZE(filter_names) - 1;
+                filter = filter_names.len() - 1;
             }
             update_params = true;
-		} else if (kDown & KEY_RIGHT) {
-			filter++;
-			if (filter >= ARRAY_SIZE(filter_names)) {
+        } else if keys_down.contains(KeyPad::KEY_LEFT) {
+            filter += 1;
+            if (filter >= filter_names.len()) {
                 filter = 0;
             }
             update_params = true;
         }
 
-		if (update_params) {
+        if update_params {
             println!("\x1b[7;1Hfilter = {}         ", filter_names[filter]);
-			switch (filter) {
-			default:
-				ndspChnIirBiquadSetEnable(0, false);
-				break;
-			case 1:
-				ndspChnIirBiquadSetParamsLowPassFilter(0, 1760.f, 0.707f);
-				break;
-			case 2:
-				ndspChnIirBiquadSetParamsHighPassFilter(0, 1760.f, 0.707f);
-				break;
-			case 3:
-				ndspChnIirBiquadSetParamsBandPassFilter(0, 1760.f, 0.707f);
-				break;
-			case 4:
-				ndspChnIirBiquadSetParamsNotchFilter(0, 1760.f, 0.707f);
-				break;
-			case 5:
-				ndspChnIirBiquadSetParamsPeakingEqualizer(0, 1760.f, 0.707f, 3.0f);
-				break;
+            match filter {
+                1 => ndspChnIirBiquadSetParamsLowPassFilter(0, 1760., 0.707),
+                2 => ndspChnIirBiquadSetParamsHighPassFilter(0, 1760., 0.707),
+                3 => ndspChnIirBiquadSetParamsBandPassFilter(0, 1760., 0.707),
+                4 => ndspChnIirBiquadSetParamsNotchFilter(0, 1760., 0.707),
+                5 => ndspChnIirBiquadSetParamsPeakingEqualizer(0, 1760., 0.707, 3.0),
+                _ => ndspChnIirBiquadSetEnable(0, false),
             }
         }
 
-		if (waveBuf[fillBlock].status == NDSP_WBUF_DONE) {
-			fill_buffer(waveBuf[fillBlock].data_pcm16, stream_offset, waveBuf[fillBlock].nsamples, notefreq[note]);
-			ndspChnWaveBufAdd(0, &waveBuf[fillBlock]);
-			stream_offset += waveBuf[fillBlock].nsamples;
+        if waveBuf[fillBlock].status == NDSP_WBUF_DONE {
+            if fillBlock {
+                fill_buffer(buf1.data_pcm16, notefreq[note]);
+                channel_zero.add_wave_buffer(buf1);
+            } else {
+                fill_buffer(waveBuf[fillBlock].data_pcm16, notefreq[note]);
+                channel_zero.add_wave_buffer(buf2);
+            }
             fillBlock = !fillBlock;
         }
+
+        // Flush and swap framebuffers
+        gfx.flush_buffers();
+        gfx.swap_buffers();
+
+        //Wait for VBlank
+        gfx.wait_for_vblank();
     }
 }

ctru-rs/src/services/ndsp.rs

@@ -32,17 +32,19 @@ pub enum AudioFormat {
 }
 
 /// 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,
-    length: usize,
+    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.
 }
 
-pub struct WaveInfo {
+/// 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: WaveBuffer,
+    buffer: &'b mut WaveBuffer,
     // Holding the data with the raw format is necessary since `libctru` will access it.
     raw_data: ctru_sys::ndspWaveBuf,
 }
@@ -178,12 +180,22 @@ impl AudioFormat {
 }
 
 impl WaveBuffer {
-    pub fn new(data: Box<[u8], LinearAllocator>, audio_format: AudioFormat) -> Self {
-        WaveBuffer {
+    pub fn new(data: Box<[u8], LinearAllocator>, audio_format: AudioFormat) -> crate::Result<Self> {
+        let nsamples = data.len() / format.size();
+
+        unsafe {
+            ResultCode(ctru_sys::DSP_FlushDataCache(data.as_ptr(), data.len()))?;
+        }
+
+        Ok(WaveBuffer {
             data,
             audio_format,
-            length: data.len() / format.size(),
+            nsamples,
+        })
     }
+
+    pub fn get_mut_data(&mut self) -> &mut Box<[u8], LinearAllocator> {
+        &mut self.data
     }
 
     pub fn format(&self) -> AudioFormat {
@@ -195,8 +207,8 @@ impl WaveBuffer {
     }
 }
 
-impl WaveInfo {
-    pub fn new(buffer: WaveBuffer, looping: bool) -> Self {
+impl<'b> WaveInfo<'b> {
+    pub fn new(buffer: &'b mut WaveBuffer, looping: bool) -> Self {
         let raw_data = ctru_sys::ndspWaveBuf {
             __bindgen_anon_1: buffer.data.as_ptr(), // Buffer data virtual address
             nsamples: buffer.length,
@@ -211,6 +223,10 @@ impl WaveInfo {
 
         Self { buffer, raw_data }
     }
+
+    pub fn get_mut_wavebuffer(&mut self) -> &'b mut WaveBuffer {
+        &mut self.buffer
+    }
 }
 
 impl Drop for Ndsp {