diff --git a/core/src/audio/audio_engine.h b/core/src/audio/audio_engine.h
index 41bf5b1..9fe2051 100644
--- a/core/src/audio/audio_engine.h
+++ b/core/src/audio/audio_engine.h
@@ -478,12 +478,12 @@ class AudioEngine {
         uint32_t user_id   = 0;
         uint32_t stream_id = 0;
 
-        // Talk-indicator edge detection (docs/voice.md §7) — updated by push_recv_frame
+        // Talk-indicator edge detection  updated by push_recv_frame
         // (already off the real-time audio thread), polled by poll_talk_transitions().
         std::atomic<int64_t> last_voice_ms{0};
         bool                  talking = false;
 
-        // ── Decode/playback decoupling ring ─────────────────────────────────────
+        // 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
diff --git a/core/src/codec/opus_codec.cpp b/core/src/codec/opus_codec.cpp
index cbf5b80..c5fd673 100644
--- a/core/src/codec/opus_codec.cpp
+++ b/core/src/codec/opus_codec.cpp
@@ -3,7 +3,7 @@
 namespace voicecat::codec {
 
 // Map an intended channel/capture sample rate (Hz) to the Opus max-bandwidth constant. The
-// codec always runs at 48 kHz internally (docs/voice.md §3); this caps the bandwidth the
+// codec always runs at 48 kHz internally. this caps the bandwidth the
 // encoder will select so a channel can request narrowband/wideband audio for low-bitrate rooms.
 // 0 (unset) and >= 48000 map to FULLBAND, which is the encoder default (i.e. a no-op cap).
 static opus_int32 opus_max_bandwidth_for(uint32_t rate_hz) {
@@ -15,7 +15,7 @@ static opus_int32 opus_max_bandwidth_for(uint32_t rate_hz) {
     return OPUS_BANDWIDTH_FULLBAND;                            // ~20 kHz
 }
 
-// ── OpusEncoder ──────────────────────────────────────────────────────────────
+// OpusEncoder
 
 bool OpusEncoder::init(const OpusParams& p) {
     destroy();
@@ -65,7 +65,7 @@ void OpusEncoder::destroy() {
     if (enc_) { opus_encoder_destroy(enc_); enc_ = nullptr; }
 }
 
-// ── OpusDecoder ──────────────────────────────────────────────────────────────
+// OpusDecoder
 
 bool OpusDecoder::init(const OpusParams& p) {
     destroy();
diff --git a/core/src/codec/opus_codec.h b/core/src/codec/opus_codec.h
index 9f297c9..ac274c8 100644
--- a/core/src/codec/opus_codec.h
+++ b/core/src/codec/opus_codec.h
@@ -1,7 +1,7 @@
 /*
- * codec/opus_codec.h — Opus encode/decode (libopus 1.6).
+ * codec/opus_codec.h: Opus encode/decode (libopus 1.6).
  *
- * Design: docs/voice.md §3–4. Per-channel AudioConfig (mono/stereo, bitrate, frame size,
+ * Per-channel AudioConfig (mono/stereo, bitrate, frame size,
  * FEC, DTX, complexity). The server relays Opus payloads unmodified (no transcode).
  */
 #ifndef VOICECAT_CODEC_OPUS_CODEC_H
@@ -25,9 +25,8 @@ enum class OpusApplication {
 struct OpusParams {
     uint32_t        sample_rate          = 48000;
     // Intended channel/capture sample rate (Hz), used ONLY to cap the encoder's audio bandwidth
-    // (narrowband/wideband/…) via OPUS_SET_MAX_BANDWIDTH. The codec always runs at 48 kHz
-    // internally (docs/voice.md §3); this lets a low-bitrate channel constrain encoded bandwidth
-    // without changing the PCM clock. 0 = unset → full band. Decoder ignores it.
+    // (narrowband/wideband/…) via OPUS_SET_MAX_BANDWIDTH. this lets a low-bitrate channel constrain encoded bandwidth
+    // without changing the PCM clock. 0 = unset  full band. Decoder ignores it.
     uint32_t        max_bandwidth_hz     = 0;
     uint32_t        bitrate_bps          = 24000;
     uint32_t        frame_ms             = 20;
@@ -89,7 +88,7 @@ class OpusDecoder {
     bool init(const OpusParams& p);
 
     // Decode one Opus packet into out_pcm (frame_samples * channels int16 samples).
-    // opus_data=nullptr, len=0 → PLC (free, always enabled by libopus).
+    // opus_data=nullptr, len=0  PLC (free, always enabled by libopus).
     // fec=true, next valid packet in opus_data → FEC recovery from previous loss.
     // Returns number of samples decoded (= frame_samples), or -1 on error.
     int decode(const uint8_t* opus_data, int len, int16_t* out_pcm, int max_samples,