feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback

Closes the three items PROGRESS.md's M3 section explicitly carried forward as
out of scope:

- Device enumeration (vc_list_devices) + input device selection
  (vc_set_input_device), backed by AudioEngine::enumerate_devices() via
  miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded
  ma_device_id strings.
- VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk).
  webrtc-audio-processing (the originally-planned APM) has no working
  Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard
  abseil-cpp dependency), so VAD is a new lightweight, dependency-free
  energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor
  interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it.
- True stereo playback: AudioEngine's mixer and output device now carry
  stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded
  stereo streams to mono before mixing.
- Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via
  miniaudio's loopback device type), replacing test-only injection as the
  production capture path.

Also: vccli gains --list-devices, --input-device, --input-mode, and
--share-screen-audio flags, plus a stdin command loop (ptt on/off, mode
vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all
four items (ABI-level + a white-box AudioEngine stereo-mix check).

Docs updated to match: voice.md, roadmap.md (decision-log entry superseding
the original webrtc-audio-processing choice), tech-stack.md, README.md,
architecture.md, CLAUDE.md, PROGRESS.md.

Still explicitly out of scope, documented not silently dropped: real
webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS
SCREEN_AUDIO capture, process-specific loopback, and a pre-existing
RT-thread rule violation in the capture path that predates this work.

Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev
and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive
standalone runs; manually verified live (vccli --list-devices against real
hardware, vccli --voice --input-mode vad streaming without incident).

Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>

core/src/audio/apm_processor.h

@@ -24,8 +24,24 @@ class ApmProcessor {
     // Returns false → caller should skip encode/send (silence gate).
     virtual bool process_capture(int16_t* pcm, int samples, int sample_rate) = 0;
 
-    // Factory: returns a real APM if VOICECAT_HAS_APM is defined, else a passthrough.
+    // Factory: returns a real APM if VOICECAT_HAS_APM is defined, else a passthrough. Used for
+    // recv-side per-stream noise reduction (docs/voice.md §10) — gating doesn't apply there, so
+    // this stays a passthrough until a real APM/NS backend exists (still inert; see
+    // PROGRESS.md). Do not use this for the send-side VAD gate — see create_vad() below.
     static std::unique_ptr<ApmProcessor> create();
+
+    // Factory for the send-side input gate (docs/voice.md §11): a lightweight, dependency-free
+    // energy/RMS VAD with configurable threshold + hang-time. webrtc-audio-processing (the
+    // originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson build,
+    // unfinished MinGW support, hard abseil-cpp dependency — see PROGRESS.md), so this is the
+    // real v1 implementation behind the same ApmProcessor interface, not a passthrough. No AEC
+    // — process_render() is a no-op here; that's a real limitation versus the originally-planned
+    // APM, not just a deferred VAD.
+    //   rms_threshold: normalized 0.0-1.0 RMS-of-int16-range; default ~0.025.
+    //   hang_time_ms:  how long the gate stays open after the last loud frame; default 300 ms
+    //                  (matches AudioEngine's kTalkHangoverMs so "talking" and "gate open" agree).
+    static std::unique_ptr<ApmProcessor> create_vad(float rms_threshold = 0.025f,
+                                                     int64_t hang_time_ms = 300);
 };
 
 }  // namespace voicecat::audio