Enable ALSA rate resampling for configurable audio sample rates

Changes the audio subsystem from hw: (direct hardware access) to plughw:
(plugin layer with rate conversion) to enable configurable sample rates.

Changes:
- Update ALSA build to include plug,rate,linear,copy plugins
- Change device names from hw: to plughw: in C and Go code
- Remove 48kHz hardcoding for HDMI audio output
- Keep USB at 48kHz since hardware is fixed at that rate
- Update all comments to reflect plughw usage

Technical details:
- hw: devices bypass all ALSA plugins and require exact hardware rate match
- plughw: devices enable the ALSA plugin layer for automatic rate conversion
- Hardware still receives at native rate (48kHz), resampling happens in userspace
- HDMI can now use 8k/12k/16k/24k/48kHz, USB remains at 48kHz
- NEON-optimized resampling provides good performance on Cortex-A7

Requires rebuilding ALSA library with updated plugin configuration.

.devcontainer/install_audio_deps.sh

@@ -48,7 +48,7 @@ if [ ! -f .built ]; then
   # Use minimal ALSA configuration to avoid FD_SETSIZE issues in devcontainer
   CFLAGS="$OPTIM_CFLAGS" ./configure --host $BUILDKIT_FLAVOR \
     --enable-static=yes --enable-shared=no \
-    --with-pcm-plugins=rate,linear \
+    --with-pcm-plugins=plug,rate,linear,copy \
     --disable-seq --disable-rawmidi --disable-ucm \
     --disable-python --disable-old-symbols \
     --disable-topology --disable-hwdep --disable-mixer \

audio.go

@@ -30,9 +30,9 @@ var (
 
 func getAlsaDevice(source string) string {
 	if source == "hdmi" {
-		return "hw:0,0"
+		return "plughw:0,0"
 	}
-	return "hw:1,0"
+	return "plughw:1,0"
 }
 
 func initAudio() {

internal/audio/c/audio.c

@@ -3,16 +3,16 @@
  *
  * Bidirectional audio processing optimized for ARM NEON SIMD:
  * - OUTPUT PATH: TC358743 HDMI or USB Gadget audio → Client speakers
- *   Pipeline: ALSA hw:0,0 or hw:1,0 capture → Opus encode (192kbps, FEC enabled)
+ *   Pipeline: ALSA plughw:0,0 or plughw:1,0 capture → Opus encode (192kbps, FEC enabled)
  *
  * - INPUT PATH: Client microphone → Device speakers
- *   Pipeline: Opus decode (with FEC) → ALSA hw:1,0 playback
+ *   Pipeline: Opus decode (with FEC) → ALSA plughw:1,0 playback
  *
  * Key features:
  * - ARM NEON SIMD optimization for all audio operations
  * - Opus in-band FEC for packet loss resilience
  * - S16_LE stereo, 20ms frames (sample rate configurable: 8k/12k/16k/24k/48kHz)
- * - ALSA rate plugin resamples hardware output to match requested Opus-compatible rate
+ * - ALSA plughw layer provides automatic rate conversion from hardware to Opus rate
  */
 
 #include <alsa/asoundlib.h>
@@ -140,19 +140,19 @@ void update_audio_decoder_constants(uint32_t sr, uint8_t ch, uint16_t fs, uint16
  * Must be called before jetkvm_audio_capture_init or jetkvm_audio_playback_init
  *
  * Device mapping (set via ALSA_CAPTURE_DEVICE/ALSA_PLAYBACK_DEVICE):
- *   hw:0,0 = TC358743 HDMI audio input (for OUTPUT path capture)
- *   hw:1,0 = USB Audio Gadget (for OUTPUT path capture or INPUT path playback)
+ *   plughw:0,0 = TC358743 HDMI audio with rate conversion (for OUTPUT path capture)
+ *   plughw:1,0 = USB Audio Gadget with rate conversion (for OUTPUT path capture or INPUT path playback)
  */
 static void init_alsa_devices_from_env(void) {
     // Always read from environment to support device switching
     alsa_capture_device = getenv("ALSA_CAPTURE_DEVICE");
     if (alsa_capture_device == NULL || alsa_capture_device[0] == '\0') {
-        alsa_capture_device = "hw:1,0"; // Default: USB gadget audio for capture
+        alsa_capture_device = "plughw:1,0"; // Default: USB gadget audio for capture with rate conversion
     }
 
     alsa_playback_device = getenv("ALSA_PLAYBACK_DEVICE");
     if (alsa_playback_device == NULL || alsa_playback_device[0] == '\0') {
-        alsa_playback_device = "hw:1,0"; // Default: USB gadget audio for playback
+        alsa_playback_device = "plughw:1,0"; // Default: USB gadget audio for playback with rate conversion
     }
 }
 
@@ -430,7 +430,7 @@ static int configure_alsa_device(snd_pcm_t *handle, const char *device_name, uin
 
 /**
  * Initialize OUTPUT path (HDMI or USB Gadget audio capture → Opus encoder)
- * Opens ALSA capture device from ALSA_CAPTURE_DEVICE env (default: hw:1,0, set to hw:0,0 for TC358743 HDMI)
+ * Opens ALSA capture device from ALSA_CAPTURE_DEVICE env (default: plughw:1,0, set to plughw:0,0 for HDMI)
  * and creates Opus encoder with optimized settings
  * @return 0 on success, -EBUSY if initializing, -1/-2/-3 on errors
  */
@@ -606,7 +606,7 @@ retry_read:
 
 /**
  * Initialize INPUT path (Opus decoder → device speakers)
- * Opens ALSA playback device from ALSA_PLAYBACK_DEVICE env (default: hw:1,0)
+ * Opens ALSA playback device from ALSA_PLAYBACK_DEVICE env (default: plughw:1,0)
  * and creates Opus decoder. Returns immediately on device open failure (no fallback).
  * @return 0 on success, -EBUSY if initializing, -1/-2 on errors
  */

internal/audio/cgo_source.go

@@ -83,10 +83,13 @@ func (c *CgoSource) Connect() error {
 func (c *CgoSource) connectOutput() error {
 	os.Setenv("ALSA_CAPTURE_DEVICE", c.alsaDevice)
 
-	// USB Audio Gadget (hw:1,0) only supports 48kHz
-	// For HDMI (hw:0,0), use configured sample rate
+	// Using plughw: enables ALSA rate conversion plugin
+	// USB Gadget hardware is fixed at 48kHz (configfs hardcoded), so keep it at 48kHz
+	// HDMI can use configured rate - plughw resamples from hardware rate to Opus rate
 	sampleRate := c.config.SampleRate
-	if c.alsaDevice == "hw:1,0" {
+	if c.alsaDevice == "plughw:1,0" {
+		sampleRate = 48000
+	} else if sampleRate == 0 {
 		sampleRate = 48000
 	}
 	frameSize := uint16(sampleRate * 20 / 1000)