use cpal::traits::{DeviceTrait, HostTrait, StreamTrait}; use prometeu_drivers::{AudioCommand, Channel, MAX_CHANNELS, OUTPUT_SAMPLE_RATE}; use ringbuf::traits::{Consumer, Producer, Split}; use ringbuf::HeapRb; use std::sync::Arc; use std::time::Duration; use prometeu_hal::LoopMode; pub struct HostAudio { pub producer: Option>>>, pub perf_consumer: Option>>>, _stream: Option, } impl HostAudio { pub fn new() -> Self { Self { producer: None, perf_consumer: None, _stream: None, } } pub fn init(&mut self) { let host = cpal::default_host(); let device = host .default_output_device() .expect("no output device available"); let config = cpal::StreamConfig { channels: 2, sample_rate: cpal::SampleRate(OUTPUT_SAMPLE_RATE), buffer_size: cpal::BufferSize::Default, }; let rb = HeapRb::::new(1024); let (prod, mut cons) = rb.split(); self.producer = Some(prod); let mut mixer = AudioMixer::new(); // To pass performance data from the audio thread to the main thread let audio_perf_rb = HeapRb::::new(64); let (mut perf_prod, perf_cons) = audio_perf_rb.split(); let stream = device .build_output_stream( &config, move |data: &mut [f32], _: &cpal::OutputCallbackInfo| { // Consumes commands from ringbuffer while let Some(cmd) = cons.try_pop() { mixer.process_command(cmd); } // Mixes audio mixer.fill_buffer(data); // Sends processing time in microseconds let _ = perf_prod.try_push(mixer.last_processing_time.as_micros() as u64); }, |err| eprintln!("audio stream error: {}", err), None, ) .expect("failed to build audio stream"); stream.play().expect("failed to play audio stream"); self._stream = Some(stream); self.perf_consumer = Some(perf_cons); } pub fn send_commands(&mut self, commands: &mut Vec) { if let Some(producer) = &mut self.producer { for cmd in commands.drain(..) { if let Err(_) = producer.try_push(cmd) { eprintln!("[HostAudio] Command ringbuffer full, dropping command."); } } } } pub fn update_stats(&mut self, stats: &mut crate::stats::HostStats) { if let Some(cons) = &mut self.perf_consumer { while let Some(us) = cons.try_pop() { stats.record_audio_perf(us); } } } } pub struct AudioMixer { voices: [Channel; MAX_CHANNELS], pub last_processing_time: Duration, paused: bool, } impl AudioMixer { pub fn new() -> Self { Self { voices: Default::default(), last_processing_time: Duration::ZERO, paused: false, } } pub fn process_command(&mut self, cmd: AudioCommand) { match cmd { AudioCommand::Play { sample, voice_id, volume, pan, pitch, priority, loop_mode, } => { if voice_id < MAX_CHANNELS { println!("[AudioMixer] Playing voice {}: vol={}, pitch={}, loop={:?}", voice_id, volume, pitch, loop_mode); self.voices[voice_id] = Channel { sample: Some(sample), active: true, pos: 0.0, pitch, volume, pan, loop_mode, priority, }; } } AudioCommand::Stop { voice_id } => { if voice_id < MAX_CHANNELS { self.voices[voice_id].active = false; self.voices[voice_id].sample = None; } } AudioCommand::SetVolume { voice_id, volume } => { if voice_id < MAX_CHANNELS { self.voices[voice_id].volume = volume; } } AudioCommand::SetPan { voice_id, pan } => { if voice_id < MAX_CHANNELS { self.voices[voice_id].pan = pan; } } AudioCommand::SetPitch { voice_id, pitch } => { if voice_id < MAX_CHANNELS { self.voices[voice_id].pitch = pitch; } } AudioCommand::MasterPause => { println!("[AudioMixer] Master Pause"); self.paused = true; } AudioCommand::MasterResume => { println!("[AudioMixer] Master Resume"); self.paused = false; } } } pub fn fill_buffer(&mut self, buffer: &mut [f32]) { let start = std::time::Instant::now(); // Zeroes the buffer (stereo) for sample in buffer.iter_mut() { *sample = 0.0; } if self.paused { self.last_processing_time = start.elapsed(); return; } for voice in self.voices.iter_mut() { let sample_data = match &voice.sample { Some(s) => s, None => continue, }; let pitch_ratio = sample_data.sample_rate as f64 / OUTPUT_SAMPLE_RATE as f64; let step = voice.pitch * pitch_ratio; let vol_f = voice.volume as f32 / 255.0; let pan_f = voice.pan as f32 / 255.0; let vol_l = vol_f * (1.0 - pan_f).sqrt(); let vol_r = vol_f * pan_f.sqrt(); for frame in buffer.chunks_exact_mut(2) { let pos_int = voice.pos as usize; let pos_fract = voice.pos - pos_int as f64; if pos_int >= sample_data.data.len() { voice.active = false; voice.sample = None; break; } // Linear Interpolation let s1 = sample_data.data[pos_int] as f32 / 32768.0; let s2 = if pos_int + 1 < sample_data.data.len() { sample_data.data[pos_int + 1] as f32 / 32768.0 } else if voice.loop_mode == LoopMode::On { let loop_start = sample_data.loop_start.unwrap_or(0) as usize; sample_data.data[loop_start] as f32 / 32768.0 } else { 0.0 }; let sample_val = s1 + (s2 - s1) * pos_fract as f32; frame[0] += sample_val * vol_l; frame[1] += sample_val * vol_r; voice.pos += step; let end_pos = sample_data.loop_end.map(|e| e as f64).unwrap_or(sample_data.data.len() as f64); if voice.pos >= end_pos { if voice.loop_mode == LoopMode::On { let loop_start = sample_data.loop_start.unwrap_or(0) as f64; voice.pos = loop_start + (voice.pos - end_pos); } else { voice.active = false; voice.sample = None; break; } } } } // Final clamp to avoid clipping (optional if using f32, but good for fidelity) for sample in buffer.iter_mut() { *sample = sample.clamp(-1.0, 1.0); } self.last_processing_time = start.elapsed(); } }