//! Linux output stream via SDL2 (Qint / upstream `tsclientlib`
//! `ts_to_audio` pattern).
//!
//! ## Why SDL2 and not cpal on Linux
//!
//! cpal's Linux backend opens raw ALSA's `default` PCM. On most
//! modern distributions (Arch with `pipewire-alsa`, Fedora 38+,
//! Debian/Ubuntu with PipeWire) that virtual device still goes
//! through ALSA's `dmix` + `plug` layers when bypassing the
//! PulseAudio/PipeWire client. The `plug` layer's default
//! resampler is **nearest-neighbour**, which causes pronounced
//! aliasing on the 48 kHz → 44.1 kHz step that the
//! `tsclientlib::AudioHandler` output requires. Users perceive
//! this as constant crackling and popping. cpal also picks a
//! small default period size (≈256 frames), which leaves no
//! headroom for kernel scheduler jitter and causes additional
//! xruns.
//!
//! SDL2 on the same systems opens the audio device through the
//! SDL audio driver — which prefers PulseAudio when available
//! and falls back to ALSA otherwise. On a PipeWire box the SDL
//! PulseAudio driver lands inside PipeWire's PulseAudio
//! compatibility layer, whose resampler is high quality. SDL
//! also defaults to a buffer ≈ samples-requested, so we get
//! exactly one Opus frame (20 ms) per callback.
//!
//! This file replaces the cpal output path on Linux only. The
//! capture path stays on cpal until we have a reason to swap it
//! (the user reports outbound is currently fine). Windows continues
//! to use cpal's WASAPI backend; Apple platforms use direct
//! VoiceProcessingIO AudioUnits for the voice path.
//!
//! ## Threading & lifecycle
//!
//! `sdl2::AudioDevice<CB>` is `!Send + !Sync` — the SDL audio
//! lock is associated with the calling thread. We open the device
//! on the same thread that calls `Self::start_with_gate` (the
//! tokio worker that runs `chanora_core::ChanoraSession::start_audio`)
//! and never move it. The outer `AudioEngine` already carries an
//! `unsafe impl Send` to bypass cpal's identical constraint; we
//! reuse that and stash the SDL device behind a `Mutex<Option<…>>`
//! the same way cpal does.
//!
//! Dropping the `SdlOutput` closes the SDL device cleanly and
//! drops the playback callback — that releases the
//! `Arc<Mutex<AudioHandler>>` clone the callback held.

use std::sync::atomic::{AtomicBool, AtomicU32, Ordering};
use std::sync::{Arc, Mutex};

use sdl2::audio::{AudioCallback, AudioDevice, AudioSpecDesired};
use sdl2::AudioSubsystem;
use tracing::{info, warn};
use tsclientlib::audio::AudioHandler;

use crate::engine::SessionAudioId;
use crate::AudioError;

/// 20 ms at 48 kHz mono == one Opus frame's worth of samples.
/// Stereo doubles the byte count but the frame-count stays the
/// same. Aligning the callback size to the Opus frame keeps the
/// jitter-buffer / playback handshake tight (no fractional frame
/// reads inside fill_buffer).
const FRAME_SAMPLES: u16 = 960;

/// SDL output device wrapper. Holds the live `AudioDevice` so its
/// callback keeps firing for the engine's lifetime, plus a
/// reference to the same `AudioHandler` the inbound forwarder
/// pushes into.
pub struct SdlOutput {
    // Drop order: device first (stops the callback), then any
    // remaining references release naturally. We keep `_subsystem`
    // alive because dropping the AudioSubsystem before the device
    // would invalidate SDL's internal state.
    device: AudioDevice<TsPlaybackCallback>,
    _subsystem: AudioSubsystem,
}

impl SdlOutput {
    /// Open the default SDL playback device at the AudioHandler's
    /// native format (48 kHz stereo f32) and start the device
    /// playing immediately. The callback drains the AudioHandler
    /// directly — no user-side resampling.
    ///
    /// `output_gain` is read on every callback to apply the
    /// master-volume slider; `output_muted` zeroes the output (but
    /// still drains AudioHandler so its jitter buffer doesn't grow
    /// unbounded while muted). These two atomics share the same
    /// definitions the cpal path uses, so the same FFI surface
    /// (`set_output_gain` / `set_output_muted`) drives both.
    pub fn start(
        handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
        output_gain: Arc<AtomicU32>,
        output_muted: Arc<AtomicBool>,
    ) -> Result<Self, AudioError> {
        let sdl = sdl2::init().map_err(|e| AudioError::Backend(format!("sdl init: {e}")))?;
        let subsystem = sdl
            .audio()
            .map_err(|e| AudioError::Backend(format!("sdl audio subsystem: {e}")))?;

        info!(
            target: "chanora_audio",
            driver = subsystem.current_audio_driver(),
            "sdl audio subsystem initialised"
        );

        let desired = AudioSpecDesired {
            freq: Some(48_000),
            channels: Some(2),
            samples: Some(FRAME_SAMPLES),
        };

        let device = AudioDevice::open_playback(&subsystem, None, &desired, |spec| {
            info!(
                target: "chanora_audio",
                freq = spec.freq,
                channels = spec.channels,
                samples = spec.samples,
                "sdl playback spec accepted"
            );
            TsPlaybackCallback {
                handler,
                output_gain,
                output_muted,
            }
        })
        .map_err(|e| AudioError::Backend(format!("sdl open_playback: {e}")))?;

        // Begin pumping audio frames out. SDL's device starts paused;
        // resume() flips it into the playing state. The callback
        // will fire repeatedly at ~50 Hz (every 20 ms) thereafter.
        device.resume();

        Ok(Self {
            device,
            _subsystem: subsystem,
        })
    }
}

impl Drop for SdlOutput {
    fn drop(&mut self) {
        // AudioDevice::drop closes the device which stops the
        // callback. We log so the chanora.log timeline matches
        // engine shutdown.
        warn!(target: "chanora_audio", "sdl playback device closing");
    }
}

/// Playback callback invoked by SDL's audio thread. The shape
/// mirrors the upstream `tsclientlib` example's `SdlCallback`:
/// zero the output buffer (so silent regions emit silence rather
/// than stale memory), then ask the `AudioHandler` to fill in
/// whatever decoded frames it has buffered.
struct TsPlaybackCallback {
    handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
    output_gain: Arc<AtomicU32>,
    output_muted: Arc<AtomicBool>,
}

impl AudioCallback for TsPlaybackCallback {
    type Channel = f32;

    fn callback(&mut self, buffer: &mut [f32]) {
        // The buffer is interleaved stereo at 48 kHz from SDL.
        // Length is FRAME_SAMPLES * 2 (= 1920 f32) per the spec
        // we requested.
        for sample in buffer.iter_mut() {
            *sample = 0.0;
        }
        // Lock window kept as tight as the upstream example —
        // fill_buffer does the actual Opus decode + jitter logic.
        // Contention with the inbound forwarder is the same as
        // in the cpal path, but the upstream design has shipped
        // this way for years.
        {
            let mut data = self.handler.lock().unwrap();
            let _removed_ids = data.fill_buffer(buffer);
            // `_removed_ids` is the list of clients whose stream the
            // handler just finished draining. We could publish that
            // upward as a "stopped speaking" hint, but the existing
            // BridgeEvent::VoiceState already covers that case via
            // the bridge layer; ignoring matches upstream behaviour.
        }

        // Apply local mute + master gain after fill so the jitter
        // buffer still drains while muted (matching the cpal path's
        // contract). `output_gain` is encoded as f32 bits inside an
        // AtomicU32 — same encoding the cpal path uses.
        if self.output_muted.load(Ordering::Relaxed) {
            for sample in buffer.iter_mut() {
                *sample = 0.0;
            }
            return;
        }
        let gain = f32::from_bits(self.output_gain.load(Ordering::Relaxed));
        if (gain - 1.0).abs() > f32::EPSILON {
            for sample in buffer.iter_mut() {
                *sample *= gain;
            }
        }
    }
}