feat: Android VoIP client — Phase 1 (audio quality, network adaptation, crate skeleton)

- New wzp-android crate with Oboe C++ backend, lock-free SPSC ring buffers,
  engine orchestrator, codec pipeline, and Android Gradle project structure
- AEC (NLMS adaptive filter), AGC (two-stage with fast attack/slow release),
  windowed-sinc FIR resampler replacing linear interpolation (wzp-codec)
- Opus encoder tuning: complexity 7 default, set_expected_loss support
- Mobile jitter buffer: asymmetric EMA (fast up/slow down), handoff spike
  detection with 2s cooldown, configurable safety margin
- Network-aware quality control: cellular-specific thresholds, faster
  downgrade on cellular, proactive tier drop on WiFi→cellular handoff,
  FEC ratio boost during network transitions
- Handoff detection in PathMonitor via RTT jitter spike analysis

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>

crates/wzp-android/src/stats.rs

@@ -0,0 +1,42 @@
+//! Call statistics for the Android engine.
+
+/// State of the call.
+#[derive(Clone, Debug, Default, serde::Serialize, PartialEq, Eq)]
+pub enum CallState {
+    /// Engine is idle, no active call.
+    #[default]
+    Idle,
+    /// Establishing connection to the relay.
+    Connecting,
+    /// Call is active with audio flowing.
+    Active,
+    /// Temporarily lost connection, attempting to recover.
+    Reconnecting,
+    /// Call has ended.
+    Closed,
+}
+
+/// Aggregated call statistics, serializable for JNI bridge.
+#[derive(Clone, Debug, Default, serde::Serialize)]
+pub struct CallStats {
+    /// Current call state.
+    pub state: CallState,
+    /// Call duration in seconds.
+    pub duration_secs: f64,
+    /// Current quality tier (0=GOOD, 1=DEGRADED, 2=CATASTROPHIC).
+    pub quality_tier: u8,
+    /// Observed packet loss percentage.
+    pub loss_pct: f32,
+    /// Smoothed round-trip time in milliseconds.
+    pub rtt_ms: u32,
+    /// Jitter in milliseconds.
+    pub jitter_ms: u32,
+    /// Current jitter buffer depth in packets.
+    pub jitter_buffer_depth: usize,
+    /// Total frames encoded since call start.
+    pub frames_encoded: u64,
+    /// Total frames decoded since call start.
+    pub frames_decoded: u64,
+    /// Number of playout underruns (buffer empty when audio needed).
+    pub underruns: u64,
+}