Add runtime configurable audio parameters with UI controls

- Add config fields for bitrate, complexity, DTX, FEC, buffer periods
- Add RPC methods for get/set audio config and restart
- Add UI settings page with controls for all audio parameters
- Add Apply Settings button to restart audio with new config
- Add config migration for backwards compatibility
- Add translations for all 9 languages
- Clean up redundant comments and optimize log levels

audio.go

@@ -39,7 +39,23 @@ func initAudio() {
 	audioInitialized = true
 }
 
-// startAudio starts audio sources and relays (skips already running ones)
+func getAudioConfig() audio.AudioConfig {
+	ensureConfigLoaded()
+	cfg := audio.DefaultAudioConfig()
+	if config.AudioBitrate >= 64 && config.AudioBitrate <= 256 {
+		cfg.Bitrate = uint16(config.AudioBitrate)
+	}
+	if config.AudioComplexity >= 0 && config.AudioComplexity <= 10 {
+		cfg.Complexity = uint8(config.AudioComplexity)
+	}
+	cfg.DTXEnabled = config.AudioDTXEnabled
+	cfg.FECEnabled = config.AudioFECEnabled
+	if config.AudioBufferPeriods >= 2 && config.AudioBufferPeriods <= 24 {
+		cfg.BufferPeriods = uint8(config.AudioBufferPeriods)
+	}
+	return cfg
+}
+
 func startAudio() error {
 	audioMutex.Lock()
 	defer audioMutex.Unlock()
@@ -49,30 +65,32 @@ func startAudio() error {
 		return nil
 	}
 
-	// Start output audio if not running, enabled, and we have a track
 	if outputSource == nil && audioOutputEnabled.Load() && currentAudioTrack != nil {
 		ensureConfigLoaded()
-		alsaDevice := "hw:1,0" // USB audio (default)
+		alsaDevice := "hw:1,0"
 		if config.AudioOutputSource == "hdmi" {
-			alsaDevice = "hw:0,0" // HDMI audio
+			alsaDevice = "hw:0,0"
 		}
 
-		outputSource = audio.NewCgoOutputSource(alsaDevice)
+		source := audio.NewCgoOutputSource(alsaDevice)
+		source.SetConfig(getAudioConfig())
+		outputSource = source
 		outputRelay = audio.NewOutputRelay(outputSource, currentAudioTrack)
 		if err := outputRelay.Start(); err != nil {
 			audioLogger.Error().Err(err).Msg("Failed to start audio output relay")
 		}
 	}
 
-	// Start input audio if not running, USB audio enabled, and input enabled
 	ensureConfigLoaded()
 	if inputSource.Load() == nil && audioInputEnabled.Load() && config.UsbDevices != nil && config.UsbDevices.Audio {
-		alsaPlaybackDevice := "hw:1,0" // USB speakers
+		alsaPlaybackDevice := "hw:1,0"
 
-		var source audio.AudioSource = audio.NewCgoInputSource(alsaPlaybackDevice)
+		source := audio.NewCgoInputSource(alsaPlaybackDevice)
-		inputSource.Store(&source)
+		source.SetConfig(getAudioConfig())
+		var audioSource audio.AudioSource = source
+		inputSource.Store(&audioSource)
 
-		inputRelay = audio.NewInputRelay(source)
+		inputRelay = audio.NewInputRelay(audioSource)
 		if err := inputRelay.Start(); err != nil {
 			audioLogger.Error().Err(err).Msg("Failed to start input relay")
 		}
@@ -158,11 +176,12 @@ func setAudioTrack(audioTrack *webrtc.TrackLocalStaticSample) {
 	audioMutex.Lock()
 	if currentAudioTrack != nil && audioOutputEnabled.Load() {
 		ensureConfigLoaded()
-		alsaDevice := "hw:1,0" // USB audio (default)
+		alsaDevice := "hw:1,0"
 		if config.AudioOutputSource == "hdmi" {
-			alsaDevice = "hw:0,0" // HDMI audio
+			alsaDevice = "hw:0,0"
 		}
 		newSource := audio.NewCgoOutputSource(alsaDevice)
+		newSource.SetConfig(getAudioConfig())
 		newRelay := audio.NewOutputRelay(newSource, currentAudioTrack)
 		outputSource = newSource
 		outputRelay = newRelay
@@ -187,10 +206,9 @@ func setPendingInputTrack(track *webrtc.TrackRemote) {
 	go handleInputTrackForSession(track)
 }
 
-// SetAudioOutputEnabled enables or disables audio output
 func SetAudioOutputEnabled(enabled bool) error {
 	if audioOutputEnabled.Swap(enabled) == enabled {
-		return nil // Already in desired state
+		return nil
 	}
 
 	if enabled {
@@ -204,10 +222,9 @@ func SetAudioOutputEnabled(enabled bool) error {
 	return nil
 }
 
-// SetAudioInputEnabled enables or disables audio input
 func SetAudioInputEnabled(enabled bool) error {
 	if audioInputEnabled.Swap(enabled) == enabled {
-		return nil // Already in desired state
+		return nil
 	}
 
 	if enabled {
@@ -221,7 +238,6 @@ func SetAudioInputEnabled(enabled bool) error {
 	return nil
 }
 
-// SetAudioOutputSource switches between HDMI and USB audio sources
 func SetAudioOutputSource(source string) error {
 	if source != "hdmi" && source != "usb" {
 		return nil
@@ -237,12 +253,13 @@ func SetAudioOutputSource(source string) error {
 	stopOutputAudio()
 
 	if audioOutputEnabled.Load() && activeConnections.Load() > 0 && currentAudioTrack != nil {
-		alsaDevice := "hw:1,0" // USB
+		alsaDevice := "hw:1,0"
 		if source == "hdmi" {
-			alsaDevice = "hw:0,0" // HDMI
+			alsaDevice = "hw:0,0"
 		}
 
 		newSource := audio.NewCgoOutputSource(alsaDevice)
+		newSource.SetConfig(getAudioConfig())
 		newRelay := audio.NewOutputRelay(newSource, currentAudioTrack)
 
 		audioMutex.Lock()
@@ -258,54 +275,79 @@ func SetAudioOutputSource(source string) error {
 	return SaveConfig()
 }
 
-// handleInputTrackForSession runs for the entire WebRTC session lifetime
+func RestartAudioOutput() {
-// It continuously reads from the track and sends to whatever relay is currently active
+	audioMutex.Lock()
+	hasActiveOutput := outputSource != nil && currentAudioTrack != nil && audioOutputEnabled.Load()
+	audioMutex.Unlock()
+
+	if !hasActiveOutput {
+		return
+	}
+
+	audioLogger.Info().Msg("Restarting audio output")
+
+	stopOutputAudio()
+
+	ensureConfigLoaded()
+	alsaDevice := "hw:1,0"
+	if config.AudioOutputSource == "hdmi" {
+		alsaDevice = "hw:0,0"
+	}
+
+	newSource := audio.NewCgoOutputSource(alsaDevice)
+	newSource.SetConfig(getAudioConfig())
+	newRelay := audio.NewOutputRelay(newSource, currentAudioTrack)
+
+	audioMutex.Lock()
+	outputSource = newSource
+	outputRelay = newRelay
+	audioMutex.Unlock()
+
+	if err := newRelay.Start(); err != nil {
+		audioLogger.Error().Err(err).Msg("Failed to restart audio output")
+	}
+}
+
 func handleInputTrackForSession(track *webrtc.TrackRemote) {
 	myTrackID := track.ID()
 
 	audioLogger.Debug().
 		Str("codec", track.Codec().MimeType).
 		Str("track_id", myTrackID).
-		Msg("starting session-lifetime track handler")
+		Msg("starting input track handler")
 
 	for {
-		// Check if we've been superseded by a new track
 		currentTrackID := currentInputTrack.Load()
 		if currentTrackID != nil && *currentTrackID != myTrackID {
 			audioLogger.Debug().
 				Str("my_track_id", myTrackID).
 				Str("current_track_id", *currentTrackID).
-				Msg("audio track handler exiting - superseded by new track")
+				Msg("input track handler exiting - superseded")
 			return
 		}
 
-		// Read RTP packet (must always read to keep track alive)
 		rtpPacket, _, err := track.ReadRTP()
 		if err != nil {
 			if err == io.EOF {
-				audioLogger.Debug().Str("track_id", myTrackID).Msg("audio track ended")
+				audioLogger.Debug().Str("track_id", myTrackID).Msg("input track ended")
 				return
 			}
 			audioLogger.Warn().Err(err).Str("track_id", myTrackID).Msg("failed to read RTP packet")
 			continue
 		}
 
-		// Extract Opus payload
 		opusData := rtpPacket.Payload
 		if len(opusData) == 0 {
 			continue
 		}
 
-		// Only send if input is enabled
 		if !audioInputEnabled.Load() {
-			continue // Drop frame but keep reading
+			continue
 		}
 
-		// Lock-free source access (hot path optimization)
 		source := inputSource.Load()
-
 		if source == nil {
-			continue // No relay, drop frame but keep reading
+			continue
 		}
 
 		inputSourceMutex.Lock()

config.go

@@ -110,6 +110,11 @@ type Config struct {
 	AudioInputAutoEnable bool                 `json:"audio_input_auto_enable"`
 	AudioOutputEnabled   bool                 `json:"audio_output_enabled"`
 	AudioOutputSource    string               `json:"audio_output_source"` // "hdmi" or "usb"
+	AudioBitrate         int                  `json:"audio_bitrate"`       // kbps (64-256)
+	AudioComplexity      int                  `json:"audio_complexity"`    // 0-10
+	AudioDTXEnabled      bool                 `json:"audio_dtx_enabled"`
+	AudioFECEnabled      bool                 `json:"audio_fec_enabled"`
+	AudioBufferPeriods   int                  `json:"audio_buffer_periods"` // 2-24
 }
 
 func (c *Config) GetDisplayRotation() uint16 {
@@ -186,6 +191,11 @@ func getDefaultConfig() Config {
 		AudioInputAutoEnable: false,
 		AudioOutputEnabled:   true,
 		AudioOutputSource:    "usb",
+		AudioBitrate:         128,
+		AudioComplexity:      5,
+		AudioDTXEnabled:      true,
+		AudioFECEnabled:      true,
+		AudioBufferPeriods:   12,
 	}
 }
 
@@ -256,6 +266,15 @@ func LoadConfig() {
 		loadedConfig.JigglerConfig = getDefaultConfig().JigglerConfig
 	}
 
+	if loadedConfig.AudioBitrate == 0 {
+		defaults := getDefaultConfig()
+		loadedConfig.AudioBitrate = defaults.AudioBitrate
+		loadedConfig.AudioComplexity = defaults.AudioComplexity
+		loadedConfig.AudioDTXEnabled = defaults.AudioDTXEnabled
+		loadedConfig.AudioFECEnabled = defaults.AudioFECEnabled
+		loadedConfig.AudioBufferPeriods = defaults.AudioBufferPeriods
+	}
+
 	// fixup old keyboard layout value
 	if loadedConfig.KeyboardLayout == "en_US" {
 		loadedConfig.KeyboardLayout = "en-US"

internal/audio/c/audio.c

@@ -2,7 +2,6 @@
  * JetKVM Audio Processing Module
  *
  * Bidirectional audio processing optimized for ARM NEON SIMD:
-* TODO: Remove USB Gadget audio once new system image release is made available
  * - OUTPUT PATH: TC358743 HDMI or USB Gadget audio → Client speakers
  *   Pipeline: ALSA hw:0,0 or hw:1,0 capture → Opus encode (128kbps, FEC enabled)
  *
@@ -56,18 +55,20 @@ static uint8_t channels = 2;
 static uint16_t frame_size = 960;  // 20ms frames at 48kHz
 
 static uint32_t opus_bitrate = 128000;
-static uint8_t opus_complexity = 5;  // Higher complexity for better quality
+static uint8_t opus_complexity = 5;
 static uint16_t max_packet_size = 1500;
 
-// Opus encoder constants (hardcoded for production)
+#define OPUS_VBR 1
-#define OPUS_VBR 1                      // VBR enabled
+#define OPUS_VBR_CONSTRAINT 1
-#define OPUS_VBR_CONSTRAINT 1           // Constrained VBR (prevents bitrate starvation at low volumes)
+#define OPUS_SIGNAL_TYPE 3002
-#define OPUS_SIGNAL_TYPE 3002           // OPUS_SIGNAL_MUSIC (better transient handling)
+#define OPUS_BANDWIDTH 1104
-#define OPUS_BANDWIDTH 1104             // OPUS_BANDWIDTH_SUPERWIDEBAND (16kHz)
+#define OPUS_LSB_DEPTH 16
-#define OPUS_DTX 1                      // DTX enabled (bandwidth optimization)
+
-#define OPUS_LSB_DEPTH 16               // 16-bit depth
+static uint8_t opus_dtx_enabled = 1;
+static uint8_t opus_fec_enabled = 1;
+static uint8_t opus_packet_loss_perc = 20;
+static uint8_t buffer_period_count = 12;
 
-// ALSA retry configuration
 static uint32_t sleep_microseconds = 1000;
 static uint32_t sleep_milliseconds = 1;
 static uint8_t max_attempts_global = 5;
@@ -86,43 +87,45 @@ int jetkvm_audio_decode_write(void *opus_buf, int opus_size);
 
 void update_audio_constants(uint32_t bitrate, uint8_t complexity,
                            uint32_t sr, uint8_t ch, uint16_t fs, uint16_t max_pkt,
-                           uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff);
+                           uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff,
+                           uint8_t dtx_enabled, uint8_t fec_enabled, uint8_t buf_periods);
 void update_audio_decoder_constants(uint32_t sr, uint8_t ch, uint16_t fs, uint16_t max_pkt,
-                                    uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff);
+                                    uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff,
+                                    uint8_t buf_periods);
 int update_opus_encoder_params(uint32_t bitrate, uint8_t complexity);
 
 
-/**
- * Sync encoder configuration from Go to C
- */
 void update_audio_constants(uint32_t bitrate, uint8_t complexity,
                            uint32_t sr, uint8_t ch, uint16_t fs, uint16_t max_pkt,
-                           uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff) {
+                           uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff,
-    opus_bitrate = bitrate;
+                           uint8_t dtx_enabled, uint8_t fec_enabled, uint8_t buf_periods) {
-    opus_complexity = complexity;
+    opus_bitrate = (bitrate >= 64000 && bitrate <= 256000) ? bitrate : 128000;
-    sample_rate = sr;
+    opus_complexity = (complexity <= 10) ? complexity : 5;
-    channels = ch;
+    sample_rate = sr > 0 ? sr : 48000;
-    frame_size = fs;
+    channels = (ch == 1 || ch == 2) ? ch : 2;
-    max_packet_size = max_pkt;
+    frame_size = fs > 0 ? fs : 960;
-    sleep_microseconds = sleep_us;
+    max_packet_size = max_pkt > 0 ? max_pkt : 1500;
-    sleep_milliseconds = sleep_us / 1000;  // Precompute for snd_pcm_wait
+    sleep_microseconds = sleep_us > 0 ? sleep_us : 1000;
-    max_attempts_global = max_attempts;
+    sleep_milliseconds = sleep_microseconds / 1000;
-    max_backoff_us_global = max_backoff;
+    max_attempts_global = max_attempts > 0 ? max_attempts : 5;
+    max_backoff_us_global = max_backoff > 0 ? max_backoff : 500000;
+    opus_dtx_enabled = dtx_enabled ? 1 : 0;
+    opus_fec_enabled = fec_enabled ? 1 : 0;
+    buffer_period_count = (buf_periods >= 2 && buf_periods <= 24) ? buf_periods : 12;
 }
 
-/**
- * Sync decoder configuration from Go to C (no encoder-only params)
- */
 void update_audio_decoder_constants(uint32_t sr, uint8_t ch, uint16_t fs, uint16_t max_pkt,
-                                    uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff) {
+                                    uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff,
-    sample_rate = sr;
+                                    uint8_t buf_periods) {
-    channels = ch;
+    sample_rate = sr > 0 ? sr : 48000;
-    frame_size = fs;
+    channels = (ch == 1 || ch == 2) ? ch : 2;
-    max_packet_size = max_pkt;
+    frame_size = fs > 0 ? fs : 960;
-    sleep_microseconds = sleep_us;
+    max_packet_size = max_pkt > 0 ? max_pkt : 1500;
-    sleep_milliseconds = sleep_us / 1000;  // Precompute for snd_pcm_wait
+    sleep_microseconds = sleep_us > 0 ? sleep_us : 1000;
-    max_attempts_global = max_attempts;
+    sleep_milliseconds = sleep_microseconds / 1000;
-    max_backoff_us_global = max_backoff;
+    max_attempts_global = max_attempts > 0 ? max_attempts : 5;
+    max_backoff_us_global = max_backoff > 0 ? max_backoff : 500000;
+    buffer_period_count = (buf_periods >= 2 && buf_periods <= 24) ? buf_periods : 12;
 }
 
 /**
@@ -233,18 +236,17 @@ static int safe_alsa_open(snd_pcm_t **handle, const char *device, snd_pcm_stream
 
 		attempt++;
 
-		// Exponential backoff with bit shift (faster than multiplication)
 		if (err == -EBUSY || err == -EAGAIN) {
 			precise_sleep_us(backoff_us);
-			backoff_us = (backoff_us << 1 < max_backoff_us_global) ? (backoff_us << 1) : max_backoff_us_global;
+			backoff_us = (backoff_us < 50000) ? (backoff_us << 1) : 50000;
 		} else if (err == -ENODEV || err == -ENOENT) {
-			precise_sleep_us(backoff_us << 1);
+			precise_sleep_us(backoff_us);
-			backoff_us = (backoff_us << 1 < max_backoff_us_global) ? (backoff_us << 1) : max_backoff_us_global;
+			backoff_us = (backoff_us < 50000) ? (backoff_us << 1) : 50000;
 		} else if (err == -EPERM || err == -EACCES) {
 			precise_sleep_us(backoff_us >> 1);
 		} else {
 			precise_sleep_us(backoff_us);
-			backoff_us = (backoff_us << 1 < max_backoff_us_global) ? (backoff_us << 1) : max_backoff_us_global;
+			backoff_us = (backoff_us < 50000) ? (backoff_us << 1) : 50000;
 		}
 	}
 	return err;
@@ -285,13 +287,13 @@ static int configure_alsa_device(snd_pcm_t *handle, const char *device_name) {
 		if (err < 0) return err;
 	}
 
-	snd_pcm_uframes_t period_size = frame_size;  // Optimized: use full frame as period
+	snd_pcm_uframes_t period_size = frame_size;
 	if (period_size < 64) period_size = 64;
 
 	err = snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, 0);
 	if (err < 0) return err;
 
-	snd_pcm_uframes_t buffer_size = period_size * 12;  // 12 periods = 240ms buffer for better jitter tolerance
+	snd_pcm_uframes_t buffer_size = period_size * buffer_period_count;
 	err = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size);
 	if (err < 0) return err;
 
@@ -378,11 +380,11 @@ int jetkvm_audio_capture_init() {
 	opus_encoder_ctl(encoder, OPUS_SET_VBR_CONSTRAINT(OPUS_VBR_CONSTRAINT));
 	opus_encoder_ctl(encoder, OPUS_SET_SIGNAL(OPUS_SIGNAL_TYPE));
 	opus_encoder_ctl(encoder, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH));
-	opus_encoder_ctl(encoder, OPUS_SET_DTX(OPUS_DTX));
+	opus_encoder_ctl(encoder, OPUS_SET_DTX(opus_dtx_enabled));
 	opus_encoder_ctl(encoder, OPUS_SET_LSB_DEPTH(OPUS_LSB_DEPTH));
 
-	opus_encoder_ctl(encoder, OPUS_SET_INBAND_FEC(1));
+	opus_encoder_ctl(encoder, OPUS_SET_INBAND_FEC(opus_fec_enabled));
-	opus_encoder_ctl(encoder, OPUS_SET_PACKET_LOSS_PERC(20));
+	opus_encoder_ctl(encoder, OPUS_SET_PACKET_LOSS_PERC(opus_packet_loss_perc));
 
 	capture_initialized = 1;
 	capture_initializing = 0;

internal/audio/cgo_source.go

@@ -21,21 +21,20 @@ import (
 )
 
 const (
-	ipcMaxFrameSize = 1024 // Max Opus frame size: 128kbps @ 20ms = ~600 bytes
+	ipcMaxFrameSize = 1024
 )
 
-// CgoSource implements AudioSource via direct CGO calls to C audio functions (in-process)
 type CgoSource struct {
-	direction   string // "output" or "input"
+	direction   string
 	alsaDevice  string
 	initialized bool
 	connected   bool
 	mu          sync.Mutex
 	logger      zerolog.Logger
-	opusBuf     []byte // Reusable buffer for Opus packets
+	opusBuf     []byte
+	config      AudioConfig
 }
 
-// NewCgoOutputSource creates a new CGO audio source for output (HDMI/USB → browser)
 func NewCgoOutputSource(alsaDevice string) *CgoSource {
 	logger := logging.GetDefaultLogger().With().Str("component", "audio-output-cgo").Logger()
 
@@ -44,10 +43,10 @@ func NewCgoOutputSource(alsaDevice string) *CgoSource {
 		alsaDevice: alsaDevice,
 		logger:     logger,
 		opusBuf:    make([]byte, ipcMaxFrameSize),
+		config:     DefaultAudioConfig(),
 	}
 }
 
-// NewCgoInputSource creates a new CGO audio source for input (browser → USB speakers)
 func NewCgoInputSource(alsaDevice string) *CgoSource {
 	logger := logging.GetDefaultLogger().With().Str("component", "audio-input-cgo").Logger()
 
@@ -56,10 +55,16 @@ func NewCgoInputSource(alsaDevice string) *CgoSource {
 		alsaDevice: alsaDevice,
 		logger:     logger,
 		opusBuf:    make([]byte, ipcMaxFrameSize),
+		config:     DefaultAudioConfig(),
 	}
 }
 
-// Connect initializes the C audio subsystem
+func (c *CgoSource) SetConfig(cfg AudioConfig) {
+	c.mu.Lock()
+	defer c.mu.Unlock()
+	c.config = cfg
+}
+
 func (c *CgoSource) Connect() error {
 	c.mu.Lock()
 	defer c.mu.Unlock()
@@ -68,46 +73,61 @@ func (c *CgoSource) Connect() error {
 		return nil
 	}
 
-	// Set ALSA device via environment for C code to read via init_alsa_devices_from_env()
 	if c.direction == "output" {
-		// Set capture device for output path via environment variable
 		os.Setenv("ALSA_CAPTURE_DEVICE", c.alsaDevice)
 
-		// Initialize constants
+		dtx := C.uchar(0)
+		if c.config.DTXEnabled {
+			dtx = C.uchar(1)
+		}
+		fec := C.uchar(0)
+		if c.config.FECEnabled {
+			fec = C.uchar(1)
+		}
+
+		c.logger.Debug().
+			Uint16("bitrate_kbps", c.config.Bitrate).
+			Uint8("complexity", c.config.Complexity).
+			Bool("dtx", c.config.DTXEnabled).
+			Bool("fec", c.config.FECEnabled).
+			Uint8("buffer_periods", c.config.BufferPeriods).
+			Str("alsa_device", c.alsaDevice).
+			Msg("Initializing audio capture")
+
 		C.update_audio_constants(
-			C.uint(128000), // bitrate
+			C.uint(uint32(c.config.Bitrate)*1000),
-			C.uchar(5),     // complexity
+			C.uchar(c.config.Complexity),
-			C.uint(48000),  // sample_rate
+			C.uint(48000),
-			C.uchar(2),     // channels
+			C.uchar(2),
-			C.ushort(960),  // frame_size
+			C.ushort(960),
-			C.ushort(1500), // max_packet_size
+			C.ushort(1500),
-			C.uint(1000),   // sleep_us
+			C.uint(1000),
-			C.uchar(5),     // max_attempts
+			C.uchar(5),
-			C.uint(500000), // max_backoff_us
+			C.uint(500000),
+			dtx,
+			fec,
+			C.uchar(c.config.BufferPeriods),
 		)
 
-		// Initialize capture (HDMI/USB → browser)
 		rc := C.jetkvm_audio_capture_init()
 		if rc != 0 {
 			c.logger.Error().Int("rc", int(rc)).Msg("Failed to initialize audio capture")
 			return fmt.Errorf("jetkvm_audio_capture_init failed: %d", rc)
 		}
 	} else {
-		// Set playback device for input path via environment variable
 		os.Setenv("ALSA_PLAYBACK_DEVICE", c.alsaDevice)
 
-		// Initialize decoder constants
 		C.update_audio_decoder_constants(
-			C.uint(48000),  // sample_rate
+			C.uint(48000),
-			C.uchar(2),     // channels
+			C.uchar(2),
-			C.ushort(960),  // frame_size
+			C.ushort(960),
-			C.ushort(1500), // max_packet_size
+			C.ushort(1500),
-			C.uint(1000),   // sleep_us
+			C.uint(1000),
-			C.uchar(5),     // max_attempts
+			C.uchar(5),
-			C.uint(500000), // max_backoff_us
+			C.uint(500000),
+			C.uchar(c.config.BufferPeriods),
 		)
 
-		// Initialize playback (browser → USB speakers)
 		rc := C.jetkvm_audio_playback_init()
 		if rc != 0 {
 			c.logger.Error().Int("rc", int(rc)).Msg("Failed to initialize audio playback")
@@ -120,7 +140,6 @@ func (c *CgoSource) Connect() error {
 	return nil
 }
 
-// Disconnect closes the C audio subsystem
 func (c *CgoSource) Disconnect() {
 	c.mu.Lock()
 	defer c.mu.Unlock()
@@ -138,17 +157,12 @@ func (c *CgoSource) Disconnect() {
 	c.connected = false
 }
 
-// IsConnected returns true if currently connected
 func (c *CgoSource) IsConnected() bool {
 	c.mu.Lock()
 	defer c.mu.Unlock()
 	return c.connected
 }
 
-// ReadMessage reads the next audio frame from C audio subsystem
-// For output path: reads HDMI/USB audio and encodes to Opus
-// For input path: not used (input uses WriteMessage instead)
-// Returns message type (0 = Opus), payload data, and error
 func (c *CgoSource) ReadMessage() (uint8, []byte, error) {
 	c.mu.Lock()
 	defer c.mu.Unlock()
@@ -161,25 +175,18 @@ func (c *CgoSource) ReadMessage() (uint8, []byte, error) {
 		return 0, nil, fmt.Errorf("ReadMessage only supported for output direction")
 	}
 
-	// Call C function to read HDMI/USB audio and encode to Opus
-	// Returns Opus packet size (>0) or error (<0)
 	opusSize := C.jetkvm_audio_read_encode(unsafe.Pointer(&c.opusBuf[0]))
-
 	if opusSize < 0 {
 		return 0, nil, fmt.Errorf("jetkvm_audio_read_encode failed: %d", opusSize)
 	}
 
 	if opusSize == 0 {
-		// No data available (silence/DTX)
 		return 0, nil, nil
 	}
 
-	// Return slice of opusBuf - caller must use immediately
 	return ipcMsgTypeOpus, c.opusBuf[:opusSize], nil
 }
 
-// WriteMessage writes an Opus packet to the C audio subsystem for playback
-// Only used for input path (browser → USB speakers)
 func (c *CgoSource) WriteMessage(msgType uint8, payload []byte) error {
 	c.mu.Lock()
 	defer c.mu.Unlock()
@@ -193,7 +200,6 @@ func (c *CgoSource) WriteMessage(msgType uint8, payload []byte) error {
 	}
 
 	if msgType != ipcMsgTypeOpus {
-		// Ignore non-Opus messages
 		return nil
 	}
 
@@ -201,9 +207,7 @@ func (c *CgoSource) WriteMessage(msgType uint8, payload []byte) error {
 		return nil
 	}
 
-	// Call C function to decode Opus and write to USB speakers
 	rc := C.jetkvm_audio_decode_write(unsafe.Pointer(&payload[0]), C.int(len(payload)))
-
 	if rc < 0 {
 		return fmt.Errorf("jetkvm_audio_decode_write failed: %d", rc)
 	}

internal/audio/cgo_source_stub.go

@@ -33,3 +33,7 @@ func (c *CgoSource) ReadMessage() (uint8, []byte, error) {
 func (c *CgoSource) WriteMessage(msgType uint8, payload []byte) error {
 	panic("audio CGO source not supported on this platform")
 }
+
+func (c *CgoSource) SetConfig(cfg AudioConfig) {
+	panic("audio CGO source not supported on this platform")
+}

internal/audio/relay.go

@@ -12,7 +12,6 @@ import (
 	"github.com/rs/zerolog"
 )
 
-// OutputRelay forwards audio from AudioSource (CGO) to WebRTC (browser)
 type OutputRelay struct {
 	source     AudioSource
 	audioTrack *webrtc.TrackLocalStaticSample
@@ -23,12 +22,10 @@ type OutputRelay struct {
 	sample     media.Sample
 	stopped    chan struct{}
 
-	// Stats (Uint32: overflows after 2.7 years @ 50fps, faster atomics on 32-bit ARM)
 	framesRelayed atomic.Uint32
 	framesDropped atomic.Uint32
 }
 
-// NewOutputRelay creates a relay for output audio (device → browser)
 func NewOutputRelay(source AudioSource, audioTrack *webrtc.TrackLocalStaticSample) *OutputRelay {
 	ctx, cancel := context.WithCancel(context.Background())
 	logger := logging.GetDefaultLogger().With().Str("component", "audio-output-relay").Logger()
@@ -46,7 +43,6 @@ func NewOutputRelay(source AudioSource, audioTrack *webrtc.TrackLocalStaticSampl
 	}
 }
 
-// Start begins relaying audio frames
 func (r *OutputRelay) Start() error {
 	if r.running.Swap(true) {
 		return fmt.Errorf("output relay already running")
@@ -57,7 +53,6 @@ func (r *OutputRelay) Start() error {
 	return nil
 }
 
-// Stop stops the relay and waits for goroutine to exit
 func (r *OutputRelay) Stop() {
 	if !r.running.Swap(false) {
 		return
@@ -72,14 +67,12 @@ func (r *OutputRelay) Stop() {
 		Msg("output relay stopped")
 }
 
-// relayLoop continuously reads from audio source and writes to WebRTC
 func (r *OutputRelay) relayLoop() {
 	defer close(r.stopped)
 
 	const reconnectDelay = 1 * time.Second
 
 	for r.running.Load() {
-		// Ensure connected
 		if !r.source.IsConnected() {
 			if err := r.source.Connect(); err != nil {
 				r.logger.Debug().Err(err).Msg("failed to connect, will retry")
@@ -88,10 +81,8 @@ func (r *OutputRelay) relayLoop() {
 			}
 		}
 
-		// Read message from audio source
 		msgType, payload, err := r.source.ReadMessage()
 		if err != nil {
-			// Connection error - reconnect
 			if r.running.Load() {
 				r.logger.Warn().Err(err).Msg("read error, reconnecting")
 				r.source.Disconnect()
@@ -100,11 +91,8 @@ func (r *OutputRelay) relayLoop() {
 			continue
 		}
 
-		// Handle message
 		if msgType == ipcMsgTypeOpus && len(payload) > 0 {
-			// Reuse sample struct (zero-allocation hot path)
 			r.sample.Data = payload
-
 			if err := r.audioTrack.WriteSample(r.sample); err != nil {
 				r.framesDropped.Add(1)
 				r.logger.Warn().Err(err).Msg("failed to write sample to WebRTC")
@@ -115,7 +103,6 @@ func (r *OutputRelay) relayLoop() {
 	}
 }
 
-// InputRelay forwards audio from WebRTC (browser microphone) to AudioSource (USB audio)
 type InputRelay struct {
 	source  AudioSource
 	ctx     context.Context
@@ -124,7 +111,6 @@ type InputRelay struct {
 	running atomic.Bool
 }
 
-// NewInputRelay creates a relay for input audio (browser → device)
 func NewInputRelay(source AudioSource) *InputRelay {
 	ctx, cancel := context.WithCancel(context.Background())
 	logger := logging.GetDefaultLogger().With().Str("component", "audio-input-relay").Logger()
@@ -137,7 +123,6 @@ func NewInputRelay(source AudioSource) *InputRelay {
 	}
 }
 
-// Start begins relaying audio frames
 func (r *InputRelay) Start() error {
 	if r.running.Swap(true) {
 		return fmt.Errorf("input relay already running")
@@ -147,7 +132,6 @@ func (r *InputRelay) Start() error {
 	return nil
 }
 
-// Stop stops the relay
 func (r *InputRelay) Stop() {
 	if !r.running.Swap(false) {
 		return

internal/audio/source.go

@@ -1,28 +1,32 @@
 package audio
 
-// IPC message types
 const (
-	ipcMsgTypeOpus = 0 // Message type for Opus audio data
+	ipcMsgTypeOpus = 0
 )
 
-// AudioSource provides audio frames via CGO (in-process) C audio functions
+type AudioConfig struct {
-type AudioSource interface {
+	Bitrate       uint16
-	// ReadMessage reads the next audio message
+	Complexity    uint8
-	// Returns message type, payload data, and error
+	BufferPeriods uint8
-	// Blocks until data is available or error occurs
+	DTXEnabled    bool
-	// Used for output path (device → browser)
+	FECEnabled    bool
-	ReadMessage() (msgType uint8, payload []byte, err error)
+}
-
+
-	// WriteMessage writes an audio message
+func DefaultAudioConfig() AudioConfig {
-	// Used for input path (browser → device)
+	return AudioConfig{
-	WriteMessage(msgType uint8, payload []byte) error
+		Bitrate:       128,
-
+		Complexity:    5,
-	// IsConnected returns true if the source is connected and ready
+		BufferPeriods: 12,
-	IsConnected() bool
+		DTXEnabled:    true,
-
+		FECEnabled:    true,
-	// Connect initializes the C audio subsystem
+	}
-	Connect() error
+}
-
+
-	// Disconnect closes the connection and releases resources
+type AudioSource interface {
-	Disconnect()
+	ReadMessage() (msgType uint8, payload []byte, err error)
+	WriteMessage(msgType uint8, payload []byte) error
+	IsConnected() bool
+	Connect() error
+	Disconnect()
+	SetConfig(cfg AudioConfig)
 }

jsonrpc.go

@@ -1000,6 +1000,60 @@ func rpcSetAudioOutputSource(source string) error {
 	return SetAudioOutputSource(source)
 }
 
+type AudioConfigResponse struct {
+	Bitrate       int  `json:"bitrate"`
+	Complexity    int  `json:"complexity"`
+	DTXEnabled    bool `json:"dtx_enabled"`
+	FECEnabled    bool `json:"fec_enabled"`
+	BufferPeriods int  `json:"buffer_periods"`
+}
+
+func rpcGetAudioConfig() (AudioConfigResponse, error) {
+	ensureConfigLoaded()
+	bitrate := config.AudioBitrate
+	if bitrate < 64 || bitrate > 256 {
+		bitrate = 128
+	}
+	bufferPeriods := config.AudioBufferPeriods
+	if bufferPeriods < 2 || bufferPeriods > 24 {
+		bufferPeriods = 12
+	}
+	return AudioConfigResponse{
+		Bitrate:       bitrate,
+		Complexity:    config.AudioComplexity,
+		DTXEnabled:    config.AudioDTXEnabled,
+		FECEnabled:    config.AudioFECEnabled,
+		BufferPeriods: bufferPeriods,
+	}, nil
+}
+
+func rpcSetAudioConfig(bitrate int, complexity int, dtxEnabled bool, fecEnabled bool, bufferPeriods int) error {
+	ensureConfigLoaded()
+
+	if bitrate < 64 || bitrate > 256 {
+		return fmt.Errorf("bitrate must be between 64 and 256 kbps")
+	}
+	if complexity < 0 || complexity > 10 {
+		return fmt.Errorf("complexity must be between 0 and 10")
+	}
+	if bufferPeriods < 2 || bufferPeriods > 24 {
+		return fmt.Errorf("buffer periods must be between 2 and 24")
+	}
+
+	config.AudioBitrate = bitrate
+	config.AudioComplexity = complexity
+	config.AudioDTXEnabled = dtxEnabled
+	config.AudioFECEnabled = fecEnabled
+	config.AudioBufferPeriods = bufferPeriods
+
+	return SaveConfig()
+}
+
+func rpcRestartAudioOutput() error {
+	RestartAudioOutput()
+	return nil
+}
+
 func rpcGetAudioInputAutoEnable() (bool, error) {
 	ensureConfigLoaded()
 	return config.AudioInputAutoEnable, nil
@@ -1340,6 +1394,9 @@ var rpcHandlers = map[string]RPCHandler{
 	"getAudioOutputSource":    {Func: rpcGetAudioOutputSource},
 	"setAudioOutputSource":    {Func: rpcSetAudioOutputSource, Params: []string{"source"}},
 	"refreshHdmiConnection":   {Func: rpcRefreshHdmiConnection},
+	"getAudioConfig":          {Func: rpcGetAudioConfig},
+	"setAudioConfig":          {Func: rpcSetAudioConfig, Params: []string{"bitrate", "complexity", "dtxEnabled", "fecEnabled", "bufferPeriods"}},
+	"restartAudioOutput":      {Func: rpcRestartAudioOutput},
 	"getAudioInputAutoEnable": {Func: rpcGetAudioInputAutoEnable},
 	"setAudioInputAutoEnable": {Func: rpcSetAudioInputAutoEnable, Params: []string{"enabled"}},
 	"setCloudUrl":             {Func: rpcSetCloudUrl, Params: []string{"apiUrl", "appUrl"}},

ui/localization/messages/da.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "Aktiver mikrofon automatisk",
     "audio_settings_auto_enable_microphone_description": "Aktiver automatisk browsermikrofon ved tilslutning (ellers skal du aktivere det manuelt ved hver session)",
+    "audio_settings_bitrate_title": "Opus Bitrate",
+    "audio_settings_bitrate_description": "Lydkodningsbitrate (højere = bedre kvalitet, mere båndbredde)",
+    "audio_settings_complexity_title": "Opus Kompleksitet",
+    "audio_settings_complexity_description": "Encoder-kompleksitet (0-10, højere = bedre kvalitet, mere CPU)",
+    "audio_settings_dtx_title": "DTX (Diskontinuerlig Transmission)",
+    "audio_settings_dtx_description": "Spar båndbredde under stilhed",
+    "audio_settings_fec_title": "FEC (Fremadrettet Fejlkorrektion)",
+    "audio_settings_fec_description": "Forbedre lydkvaliteten på tabende forbindelser",
+    "audio_settings_buffer_title": "Bufferperioder",
+    "audio_settings_buffer_description": "ALSA bufferstørrelse (højere = mere stabil, mere latens)",
+    "audio_settings_config_updated": "Lydkonfiguration opdateret",
+    "audio_settings_apply_button": "Anvend indstillinger",
+    "audio_settings_applied": "Lydindstillinger anvendt",
     "action_bar_extension": "Udvidelse",
     "action_bar_fullscreen": "Fuldskærm",
     "action_bar_settings": "Indstillinger",

ui/localization/messages/de.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "Mikrofon automatisch aktivieren",
     "audio_settings_auto_enable_microphone_description": "Browser-Mikrofon beim Verbinden automatisch aktivieren (andernfalls müssen Sie es in jeder Sitzung manuell aktivieren)",
+    "audio_settings_bitrate_title": "Opus Bitrate",
+    "audio_settings_bitrate_description": "Audio-Codierungsbitrate (höher = bessere Qualität, mehr Bandbreite)",
+    "audio_settings_complexity_title": "Opus Komplexität",
+    "audio_settings_complexity_description": "Encoder-Komplexität (0-10, höher = bessere Qualität, mehr CPU)",
+    "audio_settings_dtx_title": "DTX (Discontinuous Transmission)",
+    "audio_settings_dtx_description": "Bandbreite während Stille sparen",
+    "audio_settings_fec_title": "FEC (Forward Error Correction)",
+    "audio_settings_fec_description": "Audioqualität bei verlustbehafteten Verbindungen verbessern",
+    "audio_settings_buffer_title": "Pufferperioden",
+    "audio_settings_buffer_description": "ALSA-Puffergröße (höher = stabiler, mehr Latenz)",
+    "audio_settings_config_updated": "Audiokonfiguration aktualisiert",
+    "audio_settings_apply_button": "Einstellungen anwenden",
+    "audio_settings_applied": "Audioeinstellungen angewendet",
     "action_bar_extension": "Erweiterung",
     "action_bar_fullscreen": "Vollbild",
     "action_bar_settings": "Einstellungen",

ui/localization/messages/en.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "Auto-enable Microphone",
     "audio_settings_auto_enable_microphone_description": "Automatically enable browser microphone when connecting (otherwise you must manually enable each session)",
+    "audio_settings_bitrate_title": "Opus Bitrate",
+    "audio_settings_bitrate_description": "Audio encoding bitrate (higher = better quality, more bandwidth)",
+    "audio_settings_complexity_title": "Opus Complexity",
+    "audio_settings_complexity_description": "Encoder complexity (0-10, higher = better quality, more CPU)",
+    "audio_settings_dtx_title": "DTX (Discontinuous Transmission)",
+    "audio_settings_dtx_description": "Save bandwidth during silence",
+    "audio_settings_fec_title": "FEC (Forward Error Correction)",
+    "audio_settings_fec_description": "Improve audio quality on lossy connections",
+    "audio_settings_buffer_title": "Buffer Periods",
+    "audio_settings_buffer_description": "ALSA buffer size (higher = more stable, more latency)",
+    "audio_settings_config_updated": "Audio configuration updated",
+    "audio_settings_apply_button": "Apply Settings",
+    "audio_settings_applied": "Audio settings applied",
     "action_bar_extension": "Extension",
     "action_bar_fullscreen": "Fullscreen",
     "action_bar_settings": "Settings",

ui/localization/messages/es.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "Habilitar micrófono automáticamente",
     "audio_settings_auto_enable_microphone_description": "Habilitar automáticamente el micrófono del navegador al conectar (de lo contrario, debe habilitarlo manualmente en cada sesión)",
+    "audio_settings_bitrate_title": "Bitrate Opus",
+    "audio_settings_bitrate_description": "Tasa de bits de codificación de audio (mayor = mejor calidad, más ancho de banda)",
+    "audio_settings_complexity_title": "Complejidad Opus",
+    "audio_settings_complexity_description": "Complejidad del codificador (0-10, mayor = mejor calidad, más CPU)",
+    "audio_settings_dtx_title": "DTX (Transmisión Discontinua)",
+    "audio_settings_dtx_description": "Ahorrar ancho de banda durante el silencio",
+    "audio_settings_fec_title": "FEC (Corrección de Errores)",
+    "audio_settings_fec_description": "Mejorar la calidad de audio en conexiones con pérdida",
+    "audio_settings_buffer_title": "Períodos de Buffer",
+    "audio_settings_buffer_description": "Tamaño del buffer ALSA (mayor = más estable, más latencia)",
+    "audio_settings_config_updated": "Configuración de audio actualizada",
+    "audio_settings_apply_button": "Aplicar configuración",
+    "audio_settings_applied": "Configuración de audio aplicada",
     "action_bar_extension": "Extensión",
     "action_bar_fullscreen": "Pantalla completa",
     "action_bar_settings": "Ajustes",

ui/localization/messages/fr.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "Activer automatiquement le microphone",
     "audio_settings_auto_enable_microphone_description": "Activer automatiquement le microphone du navigateur lors de la connexion (sinon vous devez l'activer manuellement à chaque session)",
+    "audio_settings_bitrate_title": "Débit Opus",
+    "audio_settings_bitrate_description": "Débit d'encodage audio (plus élevé = meilleure qualité, plus de bande passante)",
+    "audio_settings_complexity_title": "Complexité Opus",
+    "audio_settings_complexity_description": "Complexité de l'encodeur (0-10, plus élevé = meilleure qualité, plus de CPU)",
+    "audio_settings_dtx_title": "DTX (Transmission Discontinue)",
+    "audio_settings_dtx_description": "Économiser la bande passante pendant le silence",
+    "audio_settings_fec_title": "FEC (Correction d'Erreur)",
+    "audio_settings_fec_description": "Améliorer la qualité audio sur les connexions avec perte",
+    "audio_settings_buffer_title": "Périodes de Tampon",
+    "audio_settings_buffer_description": "Taille du tampon ALSA (plus élevé = plus stable, plus de latence)",
+    "audio_settings_config_updated": "Configuration audio mise à jour",
+    "audio_settings_apply_button": "Appliquer les paramètres",
+    "audio_settings_applied": "Paramètres audio appliqués",
     "action_bar_extension": "Extension",
     "action_bar_fullscreen": "Plein écran",
     "action_bar_settings": "Paramètres",

ui/localization/messages/it.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "Abilita automaticamente il microfono",
     "audio_settings_auto_enable_microphone_description": "Abilita automaticamente il microfono del browser durante la connessione (altrimenti devi abilitarlo manualmente ad ogni sessione)",
+    "audio_settings_bitrate_title": "Bitrate Opus",
+    "audio_settings_bitrate_description": "Bitrate di codifica audio (più alto = migliore qualità, più banda)",
+    "audio_settings_complexity_title": "Complessità Opus",
+    "audio_settings_complexity_description": "Complessità dell'encoder (0-10, più alto = migliore qualità, più CPU)",
+    "audio_settings_dtx_title": "DTX (Trasmissione Discontinua)",
+    "audio_settings_dtx_description": "Risparmia banda durante il silenzio",
+    "audio_settings_fec_title": "FEC (Correzione Errori)",
+    "audio_settings_fec_description": "Migliora la qualità audio su connessioni con perdita",
+    "audio_settings_buffer_title": "Periodi Buffer",
+    "audio_settings_buffer_description": "Dimensione buffer ALSA (più alto = più stabile, più latenza)",
+    "audio_settings_config_updated": "Configurazione audio aggiornata",
+    "audio_settings_apply_button": "Applica impostazioni",
+    "audio_settings_applied": "Impostazioni audio applicate",
     "action_bar_extension": "Estensione",
     "action_bar_fullscreen": "A schermo intero",
     "action_bar_settings": "Impostazioni",

ui/localization/messages/nb.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "Aktiver mikrofon automatisk",
     "audio_settings_auto_enable_microphone_description": "Aktiver automatisk nettlesermikrofon ved tilkobling (ellers må du aktivere det manuelt hver økt)",
+    "audio_settings_bitrate_title": "Opus Bitrate",
+    "audio_settings_bitrate_description": "Lydkodingsbitrate (høyere = bedre kvalitet, mer båndbredde)",
+    "audio_settings_complexity_title": "Opus Kompleksitet",
+    "audio_settings_complexity_description": "Encoder-kompleksitet (0-10, høyere = bedre kvalitet, mer CPU)",
+    "audio_settings_dtx_title": "DTX (Diskontinuerlig Overføring)",
+    "audio_settings_dtx_description": "Spar båndbredde under stillhet",
+    "audio_settings_fec_title": "FEC (Fremadrettet Feilkorreksjon)",
+    "audio_settings_fec_description": "Forbedre lydkvaliteten på tapende tilkoblinger",
+    "audio_settings_buffer_title": "Bufferperioder",
+    "audio_settings_buffer_description": "ALSA bufferstørrelse (høyere = mer stabil, mer latens)",
+    "audio_settings_config_updated": "Lydkonfigurasjon oppdatert",
+    "audio_settings_apply_button": "Bruk innstillinger",
+    "audio_settings_applied": "Lydinnstillinger brukt",
     "action_bar_extension": "Forlengelse",
     "action_bar_fullscreen": "Fullskjerm",
     "action_bar_settings": "Innstillinger",

ui/localization/messages/sv.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "Aktivera mikrofon automatiskt",
     "audio_settings_auto_enable_microphone_description": "Aktivera automatiskt webbläsarmikrofon vid anslutning (annars måste du aktivera den manuellt varje session)",
+    "audio_settings_bitrate_title": "Opus Bitrate",
+    "audio_settings_bitrate_description": "Ljudkodningsbitrate (högre = bättre kvalitet, mer bandbredd)",
+    "audio_settings_complexity_title": "Opus Komplexitet",
+    "audio_settings_complexity_description": "Encoder-komplexitet (0-10, högre = bättre kvalitet, mer CPU)",
+    "audio_settings_dtx_title": "DTX (Diskontinuerlig Överföring)",
+    "audio_settings_dtx_description": "Spara bandbredd under tystnad",
+    "audio_settings_fec_title": "FEC (Framåtriktad Felkorrigering)",
+    "audio_settings_fec_description": "Förbättra ljudkvaliteten på förlustdrabbade anslutningar",
+    "audio_settings_buffer_title": "Bufferperioder",
+    "audio_settings_buffer_description": "ALSA bufferstorlek (högre = mer stabil, mer latens)",
+    "audio_settings_config_updated": "Ljudkonfiguration uppdaterad",
+    "audio_settings_apply_button": "Tillämpa inställningar",
+    "audio_settings_applied": "Ljudinställningar tillämpade",
     "action_bar_extension": "Förlängning",
     "action_bar_fullscreen": "Helskärm",
     "action_bar_settings": "Inställningar",

ui/localization/messages/zh.json

@@ -86,6 +86,19 @@
     "audio_settings_usb_label": "USB",
     "audio_settings_auto_enable_microphone_title": "自动启用麦克风",
     "audio_settings_auto_enable_microphone_description": "连接时自动启用浏览器麦克风（否则您必须在每次会话中手动启用）",
+    "audio_settings_bitrate_title": "Opus 比特率",
+    "audio_settings_bitrate_description": "音频编码比特率（越高 = 质量越好，带宽越大）",
+    "audio_settings_complexity_title": "Opus 复杂度",
+    "audio_settings_complexity_description": "编码器复杂度（0-10，越高 = 质量越好，CPU 使用越多）",
+    "audio_settings_dtx_title": "DTX（不连续传输）",
+    "audio_settings_dtx_description": "在静音时节省带宽",
+    "audio_settings_fec_title": "FEC（前向纠错）",
+    "audio_settings_fec_description": "改善有损连接上的音频质量",
+    "audio_settings_buffer_title": "缓冲周期",
+    "audio_settings_buffer_description": "ALSA 缓冲大小（越高 = 越稳定，延迟越高）",
+    "audio_settings_config_updated": "音频配置已更新",
+    "audio_settings_apply_button": "应用设置",
+    "audio_settings_applied": "音频设置已应用",
     "action_bar_extension": "扩展",
     "action_bar_fullscreen": "全屏",
     "action_bar_settings": "设置",

ui/src/hooks/stores.ts

@@ -392,6 +392,18 @@ export interface SettingsState {
   audioInputAutoEnable: boolean;
   setAudioInputAutoEnable: (enabled: boolean) => void;
 
+  // Audio codec settings
+  audioBitrate: number;
+  setAudioBitrate: (value: number) => void;
+  audioComplexity: number;
+  setAudioComplexity: (value: number) => void;
+  audioDTXEnabled: boolean;
+  setAudioDTXEnabled: (enabled: boolean) => void;
+  audioFECEnabled: boolean;
+  setAudioFECEnabled: (enabled: boolean) => void;
+  audioBufferPeriods: number;
+  setAudioBufferPeriods: (value: number) => void;
+
   resetMicrophoneState: () => void;
 }
 
@@ -450,6 +462,17 @@ export const useSettingsStore = create(
       audioInputAutoEnable: false,
       setAudioInputAutoEnable: (enabled: boolean) => set({ audioInputAutoEnable: enabled }),
 
+      audioBitrate: 128,
+      setAudioBitrate: (value: number) => set({ audioBitrate: value }),
+      audioComplexity: 5,
+      setAudioComplexity: (value: number) => set({ audioComplexity: value }),
+      audioDTXEnabled: true,
+      setAudioDTXEnabled: (enabled: boolean) => set({ audioDTXEnabled: enabled }),
+      audioFECEnabled: true,
+      setAudioFECEnabled: (enabled: boolean) => set({ audioFECEnabled: enabled }),
+      audioBufferPeriods: 12,
+      setAudioBufferPeriods: (value: number) => set({ audioBufferPeriods: value }),
+
       resetMicrophoneState: () => set({ microphoneEnabled: false }),
     }),
     {

ui/src/routes/devices.$id.settings.audio.tsx

@@ -10,9 +10,34 @@ import { m } from "@localizations/messages.js";
 
 import notifications from "../notifications";
 
+interface AudioConfigResult {
+  bitrate: number;
+  complexity: number;
+  dtx_enabled: boolean;
+  fec_enabled: boolean;
+  buffer_periods: number;
+}
+
 export default function SettingsAudioRoute() {
   const { send } = useJsonRpc();
-  const { setAudioOutputEnabled, setAudioInputAutoEnable, setAudioOutputSource, audioOutputEnabled, audioInputAutoEnable, audioOutputSource } = useSettingsStore();
+  const {
+    setAudioOutputEnabled,
+    setAudioInputAutoEnable,
+    setAudioOutputSource,
+    audioOutputEnabled,
+    audioInputAutoEnable,
+    audioOutputSource,
+    audioBitrate,
+    setAudioBitrate,
+    audioComplexity,
+    setAudioComplexity,
+    audioDTXEnabled,
+    setAudioDTXEnabled,
+    audioFECEnabled,
+    setAudioFECEnabled,
+    audioBufferPeriods,
+    setAudioBufferPeriods,
+  } = useSettingsStore();
 
   useEffect(() => {
     send("getAudioOutputEnabled", {}, (resp: JsonRpcResponse) => {
@@ -29,7 +54,17 @@ export default function SettingsAudioRoute() {
       if ("error" in resp) return;
       setAudioOutputSource(resp.result as string);
     });
-  }, [send, setAudioOutputEnabled, setAudioInputAutoEnable, setAudioOutputSource]);
+
+    send("getAudioConfig", {}, (resp: JsonRpcResponse) => {
+      if ("error" in resp) return;
+      const config = resp.result as AudioConfigResult;
+      setAudioBitrate(config.bitrate);
+      setAudioComplexity(config.complexity);
+      setAudioDTXEnabled(config.dtx_enabled);
+      setAudioFECEnabled(config.fec_enabled);
+      setAudioBufferPeriods(config.buffer_periods);
+    });
+  }, [send, setAudioOutputEnabled, setAudioInputAutoEnable, setAudioOutputSource, setAudioBitrate, setAudioComplexity, setAudioDTXEnabled, setAudioFECEnabled, setAudioBufferPeriods]);
 
   const handleAudioOutputEnabledChange = (enabled: boolean) => {
     send("setAudioOutputEnabled", { enabled }, (resp: JsonRpcResponse) => {
@@ -71,6 +106,41 @@ export default function SettingsAudioRoute() {
     });
   };
 
+  const handleAudioConfigChange = (
+    bitrate: number,
+    complexity: number,
+    dtxEnabled: boolean,
+    fecEnabled: boolean,
+    bufferPeriods: number
+  ) => {
+    send(
+      "setAudioConfig",
+      { bitrate, complexity, dtxEnabled, fecEnabled, bufferPeriods },
+      (resp: JsonRpcResponse) => {
+        if ("error" in resp) {
+          notifications.error(String(resp.error.data || m.unknown_error()));
+          return;
+        }
+        setAudioBitrate(bitrate);
+        setAudioComplexity(complexity);
+        setAudioDTXEnabled(dtxEnabled);
+        setAudioFECEnabled(fecEnabled);
+        setAudioBufferPeriods(bufferPeriods);
+        notifications.success(m.audio_settings_config_updated());
+      }
+    );
+  };
+
+  const handleRestartAudio = () => {
+    send("restartAudioOutput", {}, (resp: JsonRpcResponse) => {
+      if ("error" in resp) {
+        notifications.error(String(resp.error.data || m.unknown_error()));
+        return;
+      }
+      notifications.success(m.audio_settings_applied());
+    });
+  };
+
   return (
     <div className="space-y-4">
       <SettingsPageHeader
@@ -112,6 +182,130 @@ export default function SettingsAudioRoute() {
             onChange={(e) => handleAudioInputAutoEnableChange(e.target.checked)}
           />
         </SettingsItem>
+
+        <SettingsItem
+          title={m.audio_settings_bitrate_title()}
+          description={m.audio_settings_bitrate_description()}
+        >
+          <SelectMenuBasic
+            size="SM"
+            value={String(audioBitrate)}
+            options={[
+              { value: "64", label: "64 kbps" },
+              { value: "96", label: "96 kbps" },
+              { value: "128", label: "128 kbps" },
+              { value: "160", label: "160 kbps" },
+              { value: "192", label: "192 kbps" },
+              { value: "256", label: "256 kbps" },
+            ]}
+            onChange={(e) =>
+              handleAudioConfigChange(
+                parseInt(e.target.value),
+                audioComplexity,
+                audioDTXEnabled,
+                audioFECEnabled,
+                audioBufferPeriods
+              )
+            }
+          />
+        </SettingsItem>
+
+        <SettingsItem
+          title={m.audio_settings_complexity_title()}
+          description={m.audio_settings_complexity_description()}
+        >
+          <SelectMenuBasic
+            size="SM"
+            value={String(audioComplexity)}
+            options={[
+              { value: "0", label: "0 (fastest)" },
+              { value: "2", label: "2" },
+              { value: "5", label: "5 (balanced)" },
+              { value: "8", label: "8" },
+              { value: "10", label: "10 (best)" },
+            ]}
+            onChange={(e) =>
+              handleAudioConfigChange(
+                audioBitrate,
+                parseInt(e.target.value),
+                audioDTXEnabled,
+                audioFECEnabled,
+                audioBufferPeriods
+              )
+            }
+          />
+        </SettingsItem>
+
+        <SettingsItem
+          title={m.audio_settings_dtx_title()}
+          description={m.audio_settings_dtx_description()}
+        >
+          <Checkbox
+            checked={audioDTXEnabled}
+            onChange={(e) =>
+              handleAudioConfigChange(
+                audioBitrate,
+                audioComplexity,
+                e.target.checked,
+                audioFECEnabled,
+                audioBufferPeriods
+              )
+            }
+          />
+        </SettingsItem>
+
+        <SettingsItem
+          title={m.audio_settings_fec_title()}
+          description={m.audio_settings_fec_description()}
+        >
+          <Checkbox
+            checked={audioFECEnabled}
+            onChange={(e) =>
+              handleAudioConfigChange(
+                audioBitrate,
+                audioComplexity,
+                audioDTXEnabled,
+                e.target.checked,
+                audioBufferPeriods
+              )
+            }
+          />
+        </SettingsItem>
+
+        <SettingsItem
+          title={m.audio_settings_buffer_title()}
+          description={m.audio_settings_buffer_description()}
+        >
+          <SelectMenuBasic
+            size="SM"
+            value={String(audioBufferPeriods)}
+            options={[
+              { value: "4", label: "4 (80ms)" },
+              { value: "8", label: "8 (160ms)" },
+              { value: "12", label: "12 (240ms)" },
+              { value: "16", label: "16 (320ms)" },
+              { value: "24", label: "24 (480ms)" },
+            ]}
+            onChange={(e) =>
+              handleAudioConfigChange(
+                audioBitrate,
+                audioComplexity,
+                audioDTXEnabled,
+                audioFECEnabled,
+                parseInt(e.target.value)
+              )
+            }
+          />
+        </SettingsItem>
+
+        <div className="pt-4">
+          <button
+            onClick={handleRestartAudio}
+            className="inline-flex items-center px-4 py-2 border border-transparent text-sm font-medium rounded-md shadow-sm text-white bg-blue-600 hover:bg-blue-700 focus:outline-none focus:ring-2 focus:ring-offset-2 focus:ring-blue-500"
+          >
+            {m.audio_settings_apply_button()}
+          </button>
+        </div>
       </div>
     </div>
   );