lesavka/server/src/audio.rs

//  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, MessageView};
use gst::MessageView::*;
use lesavka_common::lesavka::AudioPacket;
use tokio_stream::wrappers::ReceiverStream;
use tonic::Status;
use tracing::{debug, error, warn};
use std::time::{Duration, SystemTime, UNIX_EPOCH};

/// “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");
    if let Some(tap) = pipeline
        .by_name("debugtap")
        .and_then(|e| e.downcast::<gst_app::AppSink>().ok())
    {
        clip_tap(tap);
    }

    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(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)?;

                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
    ))
}

/*────────────────────────────── clip_tap() ────────────────────────────────*/
/// Called once per pipeline; spawns a thread that writes a 1 s AAC file
/// at the start of every wall‑clock minute **while log‑level == TRACE**.
fn clip_tap(tap: gst_app::AppSink) {
    use gst::prelude::*;

    std::thread::spawn(move || {
        use std::fs::File;
        use std::io::Write;

        let mut collecting = Vec::with_capacity(200_000); // ~1 s
        let mut next_min_boundary = next_minute();

        loop {
            match tap.pull_sample() {
                Ok(s) => {
                    let buf = s.buffer().unwrap();
                    let map = buf.map_readable().unwrap();
                    collecting.extend_from_slice(map.as_slice());

                    // once per minute boundary & trace‑level
                    if tracing::enabled!(tracing::Level::TRACE) &&
                       SystemTime::now() >= next_min_boundary {
                        if !collecting.is_empty() {
                            let ts = chrono::Local::now()
                                .format("%Y%m%d-%H%M%S")
                                .to_string();
                            let path = format!("/tmp/ear-{ts}.aac");
                            if let Ok(mut f) = File::create(&path) {
                                let _ = f.write_all(&collecting);
                                tracing::debug!("📼 wrote 1 s clip → {}", path);
                            }
                        }
                        collecting.clear();
                        next_min_boundary = next_minute();
                    }

                    if collecting.len() > 192_000 {   // keep at most ~1 s
                        collecting.truncate(192_000);
                    }
                }
                Err(_) => break, // EOS
            }
        }
    });

    fn next_minute() -> SystemTime {
        let now = SystemTime::now()
            .duration_since(UNIX_EPOCH)
            .unwrap();
        let secs = now.as_secs();
        let next = (secs / 60 + 1) * 60;
        UNIX_EPOCH + Duration::from_secs(next)
    }
}