//  server/src/audio.rs
#![forbid(unsafe_code)]

use anyhow::{anyhow, Context};
use chrono::Local;
use futures_util::Stream;
use gstreamer as gst;
use gstreamer_app as gst_app;
use gst::prelude::*;
use gst::ElementFactory;
use gst::MessageView::*;
use tokio_stream::wrappers::ReceiverStream;
use tonic::Status;
use tracing::{debug, error, warn};
use std::time::{Instant, Duration, SystemTime, UNIX_EPOCH};
use std::sync::{Arc, Mutex};

use lesavka_common::lesavka::AudioPacket;

/// “Speaker” stream coming **from** the remote host (UAC2‑gadget playback
/// endpoint) **towards** the client.
pub struct AudioStream {
    _pipeline: gst::Pipeline,
    inner:     ReceiverStream<Result<AudioPacket, Status>>,
}

impl Stream for AudioStream {
    type Item = Result<AudioPacket, Status>;
    fn poll_next(
        mut self: std::pin::Pin<&mut Self>,
        cx: &mut std::task::Context<'_>,
    ) -> std::task::Poll<Option<Self::Item>> {
        Stream::poll_next(std::pin::Pin::new(&mut self.inner), cx)
    }
}

impl Drop for AudioStream {
    fn drop(&mut self) {
        let _ = self._pipeline.set_state(gst::State::Null);
    }
}

/*───────────────────────────────────────────────────────────────────────────*/
/*  ear() - capture from ALSA (“speaker”) and push AAC AUs via gRPC          */
/*───────────────────────────────────────────────────────────────────────────*/

pub async fn ear(alsa_dev: &str, id: u32) -> anyhow::Result<AudioStream> {
    //   NB: one *logical* speaker → id==0.  A 2nd logical stream could be
    //   added later (for multi‑channel) without changing the client.
    gst::init().context("gst init")?;

    /*──────────── pipeline description ────────────
     *
     *    ALSA (UAC2 gadget)       AAC+ADTS            AppSink
     *   ┌───────────┐ raw 48 kHz ┌─────────┐ AU/ADTS ┌──────────┐
     *   │  alsasrc  │────────────► voaacenc │────────► appsink   │
     *   └───────────┘            └─────────┘         └──────────┘
     */
    let desc = build_pipeline_desc(alsa_dev)?;

    let pipeline: gst::Pipeline = gst::parse::launch(&desc)?
        .downcast()
        .expect("pipeline");

    let sink: gst_app::AppSink = pipeline
        .by_name("asink")
        .expect("asink")
        .downcast()
        .expect("appsink");
    
    let tap = Arc::new(Mutex::new(ClipTap::new("🎧 - ear", Duration::from_secs(60))));
    // sink.connect("underrun", false, |_| {
    //     tracing::warn!("⚠️  USB playback underrun – host muted or not reading");
    //     None
    // });

    let (tx, rx) = tokio::sync::mpsc::channel(8192);

    let bus = pipeline.bus().expect("bus");
    std::thread::spawn(move || {
        for msg in bus.iter_timed(gst::ClockTime::NONE) {
            match msg.view() {
                Error(e)   => error!("💥 audio pipeline: {} ({})",
                                      e.error(), e.debug().unwrap_or_default()),
                Warning(w) => warn!("⚠️  audio pipeline: {} ({})",
                                      w.error(), w.debug().unwrap_or_default()),
                StateChanged(s) if s.current() == gst::State::Playing =>
                    debug!("🎶 audio pipeline PLAYING"),
                _ => {}
            }
        }
    });

    /*──────────── callbacks ────────────*/
    sink.set_callbacks(
        gst_app::AppSinkCallbacks::builder()
        .new_sample({
            let tap = tap.clone();
            move |s| {
                let sample = s.pull_sample().map_err(|_| gst::FlowError::Eos)?;
                let buffer = sample.buffer().ok_or(gst::FlowError::Error)?;
                let map    = buffer.map_readable().map_err(|_| gst::FlowError::Error)?;

                // -------- clip‑tap (minute dumps) ------------
                tap.lock().unwrap().feed(map.as_slice());

                static CNT: std::sync::atomic::AtomicU64 =
                    std::sync::atomic::AtomicU64::new(0);
                let n = CNT.fetch_add(1, std::sync::atomic::Ordering::Relaxed);
                if n < 10 || n % 300 == 0 {
                    debug!("🎧 ear #{n}: {} bytes", map.len());
                }

                let pts_us = buffer
                    .pts()
                    .unwrap_or(gst::ClockTime::ZERO)
                    .nseconds() / 1_000;

                // push non‑blocking; drop oldest on overflow
                if tx.try_send(Ok(AudioPacket {
                    id,
                    pts: pts_us,
                    data: map.as_slice().to_vec(),
                })).is_err() {
                    static DROPS: std::sync::atomic::AtomicU64 =
                        std::sync::atomic::AtomicU64::new(0);
                    let d = DROPS.fetch_add(1, std::sync::atomic::Ordering::Relaxed);
                    if d % 300 == 0 {
                        warn!("🎧💔 dropped {d} audio AUs (client too slow)");
                    }
                }
                Ok(gst::FlowSuccess::Ok)
            }
        }).build(),
    );

    pipeline.set_state(gst::State::Playing)
        .context("starting audio pipeline")?;

    Ok(AudioStream {
        _pipeline: pipeline,
        inner:     ReceiverStream::new(rx),
    })
}

/*────────────────────────── build_pipeline_desc ───────────────────────────*/
fn build_pipeline_desc(dev: &str) -> anyhow::Result<String> {
    let reg = gst::Registry::get();

    // first available encoder
    let enc = ["fdkaacenc", "voaacenc", "avenc_aac"]
        .into_iter()
        .find(|&e| {
            reg.find_plugin(e).is_some()
                || reg
                    .find_feature(e, ElementFactory::static_type())
                    .is_some()
        })
        .ok_or_else(|| anyhow!("no AAC encoder plugin available"))?;

    Ok(format!(
        concat!(
            "alsasrc device=\"{dev}\" do-timestamp=true ! ",
            "audio/x-raw,format=S16LE,channels=2,rate=48000 ! ",
            "audioconvert ! audioresample ! {enc} bitrate=192000 ! ",
            "aacparse ! ",
            "capsfilter caps=audio/mpeg,stream-format=adts,channels=2,rate=48000 ! ",
            "tee name=t ",
            "t. ! queue ! appsink name=asink emit-signals=true ",
            "t. ! queue ! appsink name=debugtap emit-signals=true max-buffers=500 drop=true"
        ),
        dev = dev,
        enc = enc
    ))
}

// ────────────────────── minute‑clip helper ───────────────────────────────
pub struct ClipTap {
    buf:    Vec<u8>,
    tag:    &'static str,
    next_dump: Instant,
    period: Duration,
}

impl ClipTap {
    pub fn new(tag: &'static str, period: Duration) -> Self {
        Self {
            buf: Vec::with_capacity(260_000),
            tag,
            next_dump: Instant::now() + period,
            period,
        }
    }
    pub fn feed(&mut self, bytes: &[u8]) {
        self.buf.extend_from_slice(bytes);
        if self.buf.len() > 256_000 {
            self.buf.drain(..self.buf.len() - 256_000);
        }
        if Instant::now() >= self.next_dump {
            self.flush();
            self.next_dump += self.period;
        }
    }
    pub fn flush(&mut self) {
        if self.buf.is_empty() {
            return;
        }
        let ts = chrono::Local::now().format("%Y%m%d-%H%M%S");
        let path = format!("/tmp/{}-{}.aac", self.tag, ts);
        if std::fs::write(&path, &self.buf).is_ok() {
            tracing::debug!("📼 wrote {} clip → {}", self.tag, path);
        }
        self.buf.clear();
    }
}
impl Drop for ClipTap {
    fn drop(&mut self) {
        self.flush()
    }
}

// ────────────────────── microphone sink  ────────────────────────────────
pub struct Voice {
    appsrc: gst_app::AppSrc,
    _pipe:  gst::Pipeline,        // keep pipeline alive
    tap:    ClipTap,
}

impl Voice {
    pub async fn new(alsa_dev: &str) -> anyhow::Result<Self> {
        use gst::prelude::*;

        gst::init().context("gst init")?;

        // pipeline
        let pipeline = gst::Pipeline::new();

        // elements
        let appsrc     = gst::ElementFactory::make("appsrc")
            .build()
            .context("make appsrc")?
            .downcast::<gst_app::AppSrc>()
            .unwrap();

        // dedicated AppSrc helpers exist and avoid the needless `?`
        appsrc.set_format(gst::Format::Time);
        appsrc.set_is_live(true);

        let decodebin  = gst::ElementFactory::make("decodebin")
            .build()
            .context("make decodebin")?;
        let alsa_sink  = gst::ElementFactory::make("alsasink")
            .build()
            .context("make alsasink")?;

        alsa_sink.set_property("device", &alsa_dev);

        pipeline.add_many(&[appsrc.upcast_ref(), &decodebin, &alsa_sink])?;
        appsrc.link(&decodebin)?;

        /*------------ decodebin autolink ----------------*/
        let sink_clone = alsa_sink.clone();          // keep original for later
        decodebin.connect_pad_added(move |_db, pad| {
            let sink_pad = sink_clone.static_pad("sink").unwrap();
            if !sink_pad.is_linked() {
                let _ = pad.link(&sink_pad);
            }
        });

        // underrun ≠ error – just show a warning
        // let _id = alsa_sink.connect("underrun", false, |_| {
        //     tracing::warn!("⚠️  USB playback underrun – host muted/not reading");
        //     None
        // });

        pipeline.set_state(gst::State::Playing)?;

        Ok(Self {
            appsrc,
            _pipe: pipeline,
            tap: ClipTap::new("voice", Duration::from_secs(60)),
        })
    }

    pub fn push(&mut self, pkt: &AudioPacket) {
        use gst::prelude::*;

        self.tap.feed(&pkt.data);

        let mut buf = gst::Buffer::from_slice(pkt.data.clone());
        buf.get_mut()
            .unwrap()
            .set_pts(Some(gst::ClockTime::from_useconds(pkt.pts)));

        if let Err(e) = self.appsrc.push_buffer(buf) {
            tracing::warn!("🎤 AppSrc push failed: {e:?}");
        }
    }
    pub fn finish(&mut self) {
        self.tap.flush();
        let _ = self.appsrc.end_of_stream();
    }
}