docs: condense implementation comments

core/include/voicecat.h

@@ -132,7 +132,7 @@ typedef enum vc_event_type {
  * vc_confirm_server_identity. Pins the TLS leaf certificate's own SHA-256 fingerprint
  * (verifiable directly from the handshake), NOT the declared Ed25519
  * server_identity_fingerprint from ServerHello — the TLS cert and the server's Ed25519
- * identity key are generated independently with no cryptographic binding between them today
+ * identity key are generated independently with no cryptographic binding between them
  * (docs/security.md §1.1), so pinning the self-declared value would be circular. The Ed25519
  * fingerprint is still available for human-readable display via
  * vc_get_server_identity_display(), it just isn't the value this gate accepts/rejects on. */

core/src/audio/audio_engine.cpp

@@ -632,9 +632,8 @@ void AudioEngine::on_playback(int16_t* out, ma_uint32 frames) {
         // the engine-wide playback channel count 
         // dec_channels/frame_samples are bitstream properties (fixed at decoder init); `frames`
         // below is the *hardware* playback callback's period, an independent value miniaudio
-        // picks on its own — opus_decode's max_samples must be frame_samples, never `frames`
-        // (see RemoteStream::ring in audio_engine.h for what went wrong when it was). The ring
-        // decouples the two: top it up by decoding whole Opus frames, then drain exactly
+        // picks on its own — opus_decode's max_samples must be frame_samples, never `frames`.
+        // The ring decouples the two: top it up by decoding whole Opus frames, then drain exactly
         // `frames` samples-per-channel from it below (silence-padding on underrun = PLC).
         const int dec_channels   = std::max(1, stream.decoder.channels());
         const int frame_samples  = stream.decoder.frame_samples();

core/src/audio/audio_engine.h

@@ -1,7 +1,4 @@
-/*
- * audio/audio_engine.h: capture/playback + DSP + jitter buffer + mixer.
- *
- */
+/* Capture, playback, jitter buffering, and mixing. */
 #ifndef VOICECAT_AUDIO_AUDIO_ENGINE_H
 #define VOICECAT_AUDIO_AUDIO_ENGINE_H
 
@@ -394,13 +391,8 @@ class AudioEngine {
     std::atomic<bool>  running_{false};
     std::atomic<float> output_volume_{1.0f};
 
-    // Capture-side frame accumulators: miniaudio fires the capture (and loopback) callback at
-    // whatever period the hardware/driver chooses — commonly 480 samples (10 ms) on WASAPI
-    // shared mode, while the Opus encoder requires exactly frame_samples_ per call (960 for
-    // 20 ms @ 48 kHz). Accumulate incoming PCM until a full frame is ready, then call
-    // capture_cb_. This mirrors the RemoteStream::ring fix on the playback side. Both
-    // accumulators are pre-allocated once in start(); never resized from the RT callback
-    // thread (satisfies architecture.md §3 — no allocation on RT threads).
+    // Device callback periods are independent of codec frame size. These preallocated
+    // accumulators emit complete frames without allocating on an RT thread.
     struct CaptureAccum {
         std::vector<int16_t> buf;    // pre-sized to frame_samples_ in start()
         int                  count = 0;
@@ -428,15 +420,10 @@ class AudioEngine {
         float            gain  = 1.0f;
         bool             mute  = false;
         uint32_t         playout_ts = 0;
-        // playout_ts free-runs (advances every callback via PLC), so it must be seeded from, and
-        // periodically re-synced to, the actual stream timeline — otherwise it drifts past the
-        // jitter buffer's drop window across VAD/PTT gaps and late joins and every frame is
-        // dropped/never-due (silent playback). false until the first frame seeds it (on_playback).
+        // Re-seeded from the stream timeline after late joins and transmission gaps.
         bool             playout_started = false;
 
-        // Set by push_recv_frame when a kFlagMarker (talkspurt-start) frame arrives; consumed by
-        // on_playback to force an immediate playout-clock reseed at the new talkspurt, so the
-        // bounded-depth target is re-established cleanly across silence gaps. See on_playback.
+        // A talkspurt marker forces playout-clock reseeding.
         bool             pending_marker = false;
 
         // Diagnostic: times the decode/playback ring underran (produced silence because the
@@ -444,12 +431,7 @@ class AudioEngine {
         // "frames arriving but silent / latency starved" signal. Polled via stream_underruns().
         std::atomic<uint64_t> underruns{0};
 
-        // PLC cap (defense-in-depth): consecutive samples produced by packet-loss
-        // concealment since the last real decoded frame. Reset to 0 on every real frame.
-        // When it exceeds kPlcCapSamples (audio_engine.cpp), on_playback stops calling
-        // opus_decode(nullptr,0,...) and emits silence instead — bounding the comfort-noise
-        // hiss to ~2 s so a stale stream can never hiss forever even if remove_stream is
-        // never called. See on_playback's decode loop.
+        // Bounds consecutive PLC output so a stale stream eventually becomes silent.
         int64_t          plc_samples_since_real = 0;
 
         // Listener-chosen, local-only noise reduction (docs/voice.md §10). Lazily
@@ -483,17 +465,8 @@ class AudioEngine {
         std::atomic<int64_t> last_voice_ms{0};
         bool                  talking = false;
 
-        // Decode/playback decoupling ring
-        // opus_decode() must be called with max_samples == the encoder's fixed frame size
-        // (decoder.frame_samples(), e.g. 960 @ 20ms/48kHz) — that's a property of the bitstream,
-        // not a choice. miniaudio's playback callback period is a *separate*, independently
-        // chosen value (often smaller, e.g. ~480 @ low-latency WASAPI defaults) and must never
-        // be passed to opus_decode as max_samples (doing so made decode fail basically every
-        // callback — silent playback bug, fixed by this ring). on_playback() tops this ring up
-        // by decoding whole Opus frames (decoder's channel count) and drains exactly the
-        // hardware-requested sample count from it each callback, padding with silence (PLC) on
-        // underrun. Sized once in init_ring() (called off the audio thread); never resized from
-        // on_playback (real-time rule).
+        // Decoding uses the bitstream frame size, while playback drains the device callback
+        // size. This preallocated ring decouples those clocks and is never resized on the RT path.
         std::vector<int16_t> ring;            // capacity = (frame_samples * 8) frames * ring_channels
         size_t                ring_channels = 1;
         size_t                ring_head     = 0;  // next frame (sample-per-channel) to read

core/src/core/client.cpp

@@ -950,9 +950,7 @@ int64_t client_now_ms() {
         .count();
 }
 
-// Builds an OpusParams from a wire AudioConfig, applying the same field-by-field mapping on
-// both the send (local-stream encoder) and receive (remote-stream decoder) paths — fixes a
-// gap where mode/dtx/complexity/application were silently dropped.
+// Maps the complete wire AudioConfig for both local encoders and remote decoders.
 voicecat::codec::OpusParams opus_params_from_audio_config(const voicecat::v1::AudioConfig& a) {
     voicecat::codec::OpusParams p;
     // Opus always runs at 48 kHz internally: the whole AudioEngine clock is

core/src/net/transport.cpp

@@ -333,9 +333,8 @@ void TcpServerConn::wait_closed() {
 
 
 namespace {
-// Try IPv6 dual-stack first (one socket handles both ::1 and 127.0.0.1 — fixes the common
-// Windows case where `localhost` resolves to ::1 before 127.0.0.1). Falls back to IPv4-only
-// if the OS has IPv6 disabled or the dual-stack bind fails for any reason.
+// Prefer IPv6 dual-stack so one listener accepts both IPv6 and IPv4 localhost addresses.
+// Fall back to IPv4 when dual-stack binding is unavailable.
 asio::ip::tcp::acceptor make_acceptor(asio::io_context& io, uint16_t port) {
     asio::ip::tcp::acceptor acc(io);
     std::error_code ec;