diff --git a/audio.go b/audio.go
index b3bb0f2f..1f6c4163 100644
--- a/audio.go
+++ b/audio.go
@@ -5,6 +5,7 @@ import (
 	"io"
 	"sync"
 	"sync/atomic"
+	"time"
 
 	"github.com/jetkvm/kvm/internal/audio"
 	"github.com/jetkvm/kvm/internal/logging"
@@ -249,8 +250,7 @@ func setPendingInputTrack(track *webrtc.TrackRemote) {
 	go handleInputTrackForSession(track)
 }
 
-// SetAudioOutputEnabled enables or disables audio output capture.
-// When enabling, blocks up to 5 seconds waiting for audio to start.
+// SetAudioOutputEnabled blocks up to 5 seconds when enabling.
 // Returns error if audio fails to start within timeout.
 func SetAudioOutputEnabled(enabled bool) error {
 	if audioOutputEnabled.Swap(enabled) == enabled {
@@ -282,8 +282,7 @@ func SetAudioOutputEnabled(enabled bool) error {
 	return nil
 }
 
-// SetAudioInputEnabled enables or disables audio input playback.
-// When enabling, blocks up to 5 seconds waiting for audio to start.
+// SetAudioInputEnabled blocks up to 5 seconds when enabling.
 // Returns error if audio fails to start within timeout.
 func SetAudioInputEnabled(enabled bool) error {
 	if audioInputEnabled.Swap(enabled) == enabled {
diff --git a/internal/audio/c/audio.c b/internal/audio/c/audio.c
index f736054b..33ef4fb8 100644
--- a/internal/audio/c/audio.c
+++ b/internal/audio/c/audio.c
@@ -54,7 +54,7 @@ static snd_pcm_t *pcm_playback_handle = NULL; // INPUT: Client microphone → de
 
 static const char *alsa_capture_device = NULL;
 static const char *alsa_playback_device = NULL;
-static bool capture_channels_swapped = false;  // True if hardware reports R,L instead of L,R
+static bool capture_channels_swapped = false;
 
 static OpusEncoder *encoder = NULL;
 static OpusDecoder *decoder = NULL;
@@ -104,11 +104,9 @@ static uint32_t max_backoff_us_global = 500000;
 static atomic_int capture_stop_requested = 0;
 static atomic_int playback_stop_requested = 0;
 
-// Mutexes to protect concurrent access to ALSA handles and codecs throughout their lifecycle
-// These prevent race conditions when jetkvm_audio_*_close() is called while
-// jetkvm_audio_read_encode() or jetkvm_audio_decode_write() are executing.
-// The mutexes protect initialization, cleanup, ALSA I/O, codec operations, and handle validation
-// to ensure handles remain valid from acquisition through release.
+// Mutexes protect handle lifecycle and codec operations, NOT the ALSA I/O itself.
+// The mutex is temporarily released during snd_pcm_readi/writei to prevent blocking.
+// Race conditions are detected via handle pointer comparison after reacquiring the lock.
 static pthread_mutex_t capture_mutex = PTHREAD_MUTEX_INITIALIZER;
 static pthread_mutex_t playback_mutex = PTHREAD_MUTEX_INITIALIZER;
 
@@ -187,9 +185,6 @@ static void init_alsa_devices_from_env(void) {
 
 // SIMD-OPTIMIZED BUFFER OPERATIONS (ARM NEON)
 
-/**
- * Clear audio buffer using NEON (16 samples/iteration with 2x unrolling)
- */
 static inline void simd_clear_samples_s16(short * __restrict__ buffer, uint32_t samples) {
     const int16x8_t zero = vdupq_n_s16(0);
     uint32_t i = 0;
@@ -293,11 +288,6 @@ static unsigned int get_hdmi_audio_sample_rate(void) {
 	return detected_rate;
 }
 
-/**
- * Open ALSA device with exponential backoff retry
- * @return 0 on success, negative error code on failure
- */
-// High-precision sleep using nanosleep
 static inline void precise_sleep_us(uint32_t microseconds) {
 	struct timespec ts = {
 		.tv_sec = microseconds / 1000000,
@@ -806,11 +796,6 @@ int jetkvm_audio_capture_init() {
 	return 0;
 }
 
-/**
- * Read HDMI audio, resample with SpeexDSP, encode to Opus (OUTPUT path hot function)
- * @param opus_buf Output buffer for encoded Opus packet
- * @return >0 = Opus packet size in bytes, -1 = error
- */
 __attribute__((hot)) int jetkvm_audio_read_encode(void * __restrict__ opus_buf) {
 	// Two buffers: hardware buffer + resampled buffer (at 48kHz)
 	static short CACHE_ALIGN pcm_hw_buffer[MAX_HARDWARE_FRAME_SIZE * 2];    // Max hardware rate * stereo
@@ -1007,13 +992,6 @@ int jetkvm_audio_playback_init() {
 	return 0;
 }
 
-/**
- * Decode Opus, write to device speakers (INPUT path hot function)
- * Processing pipeline: Opus decode (with FEC) → ALSA playback with error recovery
- * @param opus_buf Encoded Opus packet from client
- * @param opus_size Size of Opus packet in bytes
- * @return >0 = PCM frames written, 0 = frame skipped, -1/-2 = error
- */
 __attribute__((hot)) int jetkvm_audio_decode_write(void * __restrict__ opus_buf, int32_t opus_size) {
 	static short CACHE_ALIGN pcm_buffer[960 * 2];  // Cache-aligned
 	unsigned char * __restrict__ in = (unsigned char*)opus_buf;
diff --git a/internal/audio/cgo_source.go b/internal/audio/cgo_source.go
index 60a7ac7f..0fb130c3 100644
--- a/internal/audio/cgo_source.go
+++ b/internal/audio/cgo_source.go
@@ -83,8 +83,6 @@ func (c *CgoSource) Connect() error {
 func (c *CgoSource) connectOutput() error {
 	os.Setenv("ALSA_CAPTURE_DEVICE", c.alsaDevice)
 
-	// Opus uses fixed 48kHz sample rate (RFC 7587)
-	// SpeexDSP handles any hardware rate conversion
 	const sampleRate = 48000
 	const frameSize = uint16(sampleRate * 20 / 1000) // 20ms frames
 
diff --git a/internal/audio/relay.go b/internal/audio/relay.go
index 92555441..8b42de62 100644
--- a/internal/audio/relay.go
+++ b/internal/audio/relay.go
@@ -77,7 +77,6 @@ func (r *OutputRelay) relayLoop() {
 	consecutiveWriteFailures := 0
 
 	for r.running.Load() {
-		// Connect if not connected
 		if !(*r.source).IsConnected() {
 			if err := (*r.source).Connect(); err != nil {
 				if consecutiveFailures++; consecutiveFailures >= maxRetries {
@@ -93,7 +92,6 @@ func (r *OutputRelay) relayLoop() {
 			retryDelay = 1 * time.Second
 		}
 
-		// Read message from source
 		msgType, payload, err := (*r.source).ReadMessage()
 		if err != nil {
 			if !r.running.Load() {
@@ -110,11 +108,9 @@ func (r *OutputRelay) relayLoop() {
 			continue
 		}
 
-		// Reset retry state on successful read
 		consecutiveFailures = 0
 		retryDelay = 1 * time.Second
 
-		// Write audio sample to WebRTC
 		if msgType == ipcMsgTypeOpus && len(payload) > 0 {
 			r.sample.Data = payload
 			if err := r.audioTrack.WriteSample(r.sample); err != nil {
@@ -129,7 +125,6 @@ func (r *OutputRelay) relayLoop() {
 						Msg("Failed to write sample to WebRTC")
 				}
 
-				// If too many consecutive write failures, reconnect source
 				if consecutiveWriteFailures >= maxConsecutiveWriteFailures {
 					r.logger.Error().
 						Int("failures", consecutiveWriteFailures).
@@ -140,7 +135,7 @@ func (r *OutputRelay) relayLoop() {
 				}
 			} else {
 				r.framesRelayed.Add(1)
-				consecutiveWriteFailures = 0 // Reset on successful write
+				consecutiveWriteFailures = 0
 			}
 		}
 	}