Fix: linter errors

This commit is contained in:
Alex P 2025-08-22 22:28:15 +00:00
parent e360348829
commit 6ecb829334
3 changed files with 102 additions and 102 deletions

View File

@ -14,7 +14,7 @@ const (
// Input RPC Direct Handlers
// This module provides optimized direct handlers for high-frequency input events,
// bypassing the reflection-based RPC system for improved performance.
//
//
// Performance benefits:
// - Eliminates reflection overhead (~2-3ms per call)
// - Reduces memory allocations
@ -214,4 +214,4 @@ func isInputMethod(method string) bool {
default:
return false
}
}
}

View File

@ -39,7 +39,7 @@ static volatile int playback_initialized = 0;
static int safe_alsa_open(snd_pcm_t **handle, const char *device, snd_pcm_stream_t stream) {
int attempts = 3;
int err;
while (attempts-- > 0) {
err = snd_pcm_open(handle, device, stream, SND_PCM_NONBLOCK);
if (err >= 0) {
@ -47,7 +47,7 @@ static int safe_alsa_open(snd_pcm_t **handle, const char *device, snd_pcm_stream
snd_pcm_nonblock(*handle, 0);
return 0;
}
if (err == -EBUSY && attempts > 0) {
// Device busy, wait and retry
usleep(50000); // 50ms
@ -63,26 +63,26 @@ static int configure_alsa_device(snd_pcm_t *handle, const char *device_name) {
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *sw_params;
int err;
if (!handle) return -1;
// Use stack allocation for better performance
snd_pcm_hw_params_alloca(&params);
snd_pcm_sw_params_alloca(&sw_params);
// Hardware parameters
err = snd_pcm_hw_params_any(handle, params);
if (err < 0) return err;
err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) return err;
err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
if (err < 0) return err;
err = snd_pcm_hw_params_set_channels(handle, params, channels);
if (err < 0) return err;
// Set exact rate for better performance
err = snd_pcm_hw_params_set_rate(handle, params, sample_rate, 0);
if (err < 0) {
@ -91,70 +91,70 @@ static int configure_alsa_device(snd_pcm_t *handle, const char *device_name) {
err = snd_pcm_hw_params_set_rate_near(handle, params, &rate, 0);
if (err < 0) return err;
}
// Optimize buffer sizes for low latency
snd_pcm_uframes_t period_size = frame_size;
err = snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, 0);
if (err < 0) return err;
// Set buffer size to 4 periods for good latency/stability balance
snd_pcm_uframes_t buffer_size = period_size * 4;
err = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size);
if (err < 0) return err;
err = snd_pcm_hw_params(handle, params);
if (err < 0) return err;
// Software parameters for optimal performance
err = snd_pcm_sw_params_current(handle, sw_params);
if (err < 0) return err;
// Start playback/capture when buffer is period_size frames
err = snd_pcm_sw_params_set_start_threshold(handle, sw_params, period_size);
if (err < 0) return err;
// Allow transfers when at least period_size frames are available
err = snd_pcm_sw_params_set_avail_min(handle, sw_params, period_size);
if (err < 0) return err;
err = snd_pcm_sw_params(handle, sw_params);
if (err < 0) return err;
return snd_pcm_prepare(handle);
}
// Initialize ALSA and Opus encoder with improved safety
int jetkvm_audio_init() {
int err;
// Prevent concurrent initialization
if (__sync_bool_compare_and_swap(&capture_initializing, 0, 1) == 0) {
return -EBUSY; // Already initializing
}
// Check if already initialized
if (capture_initialized) {
capture_initializing = 0;
return 0;
}
// Clean up any existing resources first
if (encoder) {
opus_encoder_destroy(encoder);
encoder = NULL;
if (encoder) {
opus_encoder_destroy(encoder);
encoder = NULL;
}
if (pcm_handle) {
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
if (pcm_handle) {
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
}
// Try to open ALSA capture device
err = safe_alsa_open(&pcm_handle, "hw:1,0", SND_PCM_STREAM_CAPTURE);
if (err < 0) {
capture_initializing = 0;
return -1;
}
// Configure the device
err = configure_alsa_device(pcm_handle, "capture");
if (err < 0) {
@ -163,7 +163,7 @@ int jetkvm_audio_init() {
capture_initializing = 0;
return -1;
}
// Initialize Opus encoder
int opus_err = 0;
encoder = opus_encoder_create(sample_rate, channels, OPUS_APPLICATION_AUDIO, &opus_err);
@ -172,10 +172,10 @@ int jetkvm_audio_init() {
capture_initializing = 0;
return -2;
}
opus_encoder_ctl(encoder, OPUS_SET_BITRATE(opus_bitrate));
opus_encoder_ctl(encoder, OPUS_SET_COMPLEXITY(opus_complexity));
capture_initialized = 1;
capture_initializing = 0;
return 0;
@ -186,21 +186,21 @@ int jetkvm_audio_read_encode(void *opus_buf) {
short pcm_buffer[1920]; // max 2ch*960
unsigned char *out = (unsigned char*)opus_buf;
int err = 0;
// Safety checks
if (!capture_initialized || !pcm_handle || !encoder || !opus_buf) {
return -1;
}
int pcm_rc = snd_pcm_readi(pcm_handle, pcm_buffer, frame_size);
// Handle ALSA errors with enhanced recovery
if (pcm_rc < 0) {
if (pcm_rc == -EPIPE) {
// Buffer underrun - try to recover
err = snd_pcm_prepare(pcm_handle);
if (err < 0) return -1;
pcm_rc = snd_pcm_readi(pcm_handle, pcm_buffer, frame_size);
if (pcm_rc < 0) return -1;
} else if (pcm_rc == -EAGAIN) {
@ -221,12 +221,12 @@ int jetkvm_audio_read_encode(void *opus_buf) {
return -1;
}
}
// If we got fewer frames than expected, pad with silence
if (pcm_rc < frame_size) {
memset(&pcm_buffer[pcm_rc * channels], 0, (frame_size - pcm_rc) * channels * sizeof(short));
}
int nb_bytes = opus_encode(encoder, pcm_buffer, frame_size, out, max_packet_size);
return nb_bytes;
}
@ -234,28 +234,28 @@ int jetkvm_audio_read_encode(void *opus_buf) {
// Initialize ALSA playback with improved safety
int jetkvm_audio_playback_init() {
int err;
// Prevent concurrent initialization
if (__sync_bool_compare_and_swap(&playback_initializing, 0, 1) == 0) {
return -EBUSY; // Already initializing
}
// Check if already initialized
if (playback_initialized) {
playback_initializing = 0;
return 0;
}
// Clean up any existing resources first
if (decoder) {
opus_decoder_destroy(decoder);
decoder = NULL;
if (decoder) {
opus_decoder_destroy(decoder);
decoder = NULL;
}
if (pcm_playback_handle) {
snd_pcm_close(pcm_playback_handle);
pcm_playback_handle = NULL;
if (pcm_playback_handle) {
snd_pcm_close(pcm_playback_handle);
pcm_playback_handle = NULL;
}
// Try to open the USB gadget audio device for playback
err = safe_alsa_open(&pcm_playback_handle, "hw:1,0", SND_PCM_STREAM_PLAYBACK);
if (err < 0) {
@ -266,7 +266,7 @@ int jetkvm_audio_playback_init() {
return -1;
}
}
// Configure the device
err = configure_alsa_device(pcm_playback_handle, "playback");
if (err < 0) {
@ -275,7 +275,7 @@ int jetkvm_audio_playback_init() {
playback_initializing = 0;
return -1;
}
// Initialize Opus decoder
int opus_err = 0;
decoder = opus_decoder_create(sample_rate, channels, &opus_err);
@ -285,7 +285,7 @@ int jetkvm_audio_playback_init() {
playback_initializing = 0;
return -2;
}
playback_initialized = 1;
playback_initializing = 0;
return 0;
@ -296,21 +296,21 @@ int jetkvm_audio_decode_write(void *opus_buf, int opus_size) {
short pcm_buffer[1920]; // max 2ch*960
unsigned char *in = (unsigned char*)opus_buf;
int err = 0;
// Safety checks
if (!playback_initialized || !pcm_playback_handle || !decoder || !opus_buf || opus_size <= 0) {
return -1;
}
// Additional bounds checking
if (opus_size > max_packet_size) {
return -1;
}
// Decode Opus to PCM
int pcm_frames = opus_decode(decoder, in, opus_size, pcm_buffer, frame_size, 0);
if (pcm_frames < 0) return -1;
// Write PCM to playback device with enhanced recovery
int pcm_rc = snd_pcm_writei(pcm_playback_handle, pcm_buffer, pcm_frames);
if (pcm_rc < 0) {
@ -318,7 +318,7 @@ int jetkvm_audio_decode_write(void *opus_buf, int opus_size) {
// Buffer underrun - try to recover
err = snd_pcm_prepare(pcm_playback_handle);
if (err < 0) return -2;
pcm_rc = snd_pcm_writei(pcm_playback_handle, pcm_buffer, pcm_frames);
} else if (pcm_rc == -ESTRPIPE) {
// Device suspended, try to resume
@ -333,7 +333,7 @@ int jetkvm_audio_decode_write(void *opus_buf, int opus_size) {
}
if (pcm_rc < 0) return -2;
}
return pcm_frames;
}
@ -343,20 +343,20 @@ void jetkvm_audio_playback_close() {
while (playback_initializing) {
usleep(1000); // 1ms
}
// Atomic check and set to prevent double cleanup
if (__sync_bool_compare_and_swap(&playback_initialized, 1, 0) == 0) {
return; // Already cleaned up
}
if (decoder) {
opus_decoder_destroy(decoder);
decoder = NULL;
if (decoder) {
opus_decoder_destroy(decoder);
decoder = NULL;
}
if (pcm_playback_handle) {
if (pcm_playback_handle) {
snd_pcm_drain(pcm_playback_handle);
snd_pcm_close(pcm_playback_handle);
pcm_playback_handle = NULL;
snd_pcm_close(pcm_playback_handle);
pcm_playback_handle = NULL;
}
}
@ -366,19 +366,19 @@ void jetkvm_audio_close() {
while (capture_initializing) {
usleep(1000); // 1ms
}
capture_initialized = 0;
if (encoder) {
opus_encoder_destroy(encoder);
encoder = NULL;
if (encoder) {
opus_encoder_destroy(encoder);
encoder = NULL;
}
if (pcm_handle) {
if (pcm_handle) {
snd_pcm_drop(pcm_handle); // Drop pending samples
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
}
// Also clean up playback
jetkvm_audio_playback_close();
}
@ -387,15 +387,15 @@ import "C"
// Optimized Go wrappers with reduced overhead
var (
errAudioInitFailed = errors.New("failed to init ALSA/Opus")
errBufferTooSmall = errors.New("buffer too small")
errAudioReadEncode = errors.New("audio read/encode error")
errAudioDecodeWrite = errors.New("audio decode/write error")
errAudioPlaybackInit = errors.New("failed to init ALSA playback/Opus decoder")
errEmptyBuffer = errors.New("empty buffer")
errNilBuffer = errors.New("nil buffer")
errBufferTooLarge = errors.New("buffer too large")
errInvalidBufferPtr = errors.New("invalid buffer pointer")
errAudioInitFailed = errors.New("failed to init ALSA/Opus")
errBufferTooSmall = errors.New("buffer too small")
errAudioReadEncode = errors.New("audio read/encode error")
errAudioDecodeWrite = errors.New("audio decode/write error")
errAudioPlaybackInit = errors.New("failed to init ALSA playback/Opus decoder")
errEmptyBuffer = errors.New("empty buffer")
errNilBuffer = errors.New("nil buffer")
errBufferTooLarge = errors.New("buffer too large")
errInvalidBufferPtr = errors.New("invalid buffer pointer")
)
func cgoAudioInit() error {
@ -416,7 +416,7 @@ func cgoAudioReadEncode(buf []byte) (int, error) {
if len(buf) < 1276 {
return 0, errBufferTooSmall
}
n := C.jetkvm_audio_read_encode(unsafe.Pointer(&buf[0]))
if n < 0 {
return 0, errAudioReadEncode
@ -449,18 +449,18 @@ func cgoAudioDecodeWrite(buf []byte) (int, error) {
if buf == nil {
return 0, errors.New("nil buffer")
}
// Validate buffer size to prevent potential overruns
if len(buf) > 4096 { // Maximum reasonable Opus frame size
return 0, errors.New("buffer too large")
}
// Ensure buffer is not deallocated by keeping a reference
bufPtr := unsafe.Pointer(&buf[0])
if bufPtr == nil {
return 0, errors.New("invalid buffer pointer")
}
// Add recovery mechanism for C function crashes
defer func() {
if r := recover(); r != nil {
@ -469,7 +469,7 @@ func cgoAudioDecodeWrite(buf []byte) (int, error) {
_ = r // Explicitly ignore the panic value
}
}()
n := C.jetkvm_audio_decode_write(bufPtr, C.int(len(buf)))
if n < 0 {
return 0, errors.New("audio decode/write error")
@ -479,10 +479,10 @@ func cgoAudioDecodeWrite(buf []byte) (int, error) {
// Wrapper functions for non-blocking audio manager
var (
CGOAudioInit = cgoAudioInit
CGOAudioClose = cgoAudioClose
CGOAudioReadEncode = cgoAudioReadEncode
CGOAudioPlaybackInit = cgoAudioPlaybackInit
CGOAudioPlaybackClose = cgoAudioPlaybackClose
CGOAudioDecodeWrite = cgoAudioDecodeWrite
CGOAudioInit = cgoAudioInit
CGOAudioClose = cgoAudioClose
CGOAudioReadEncode = cgoAudioReadEncode
CGOAudioPlaybackInit = cgoAudioPlaybackInit
CGOAudioPlaybackClose = cgoAudioPlaybackClose
CGOAudioDecodeWrite = cgoAudioDecodeWrite
)

14
main.go
View File

@ -14,10 +14,10 @@ import (
)
var (
appCtx context.Context
isAudioServer bool
appCtx context.Context
isAudioServer bool
audioProcessDone chan struct{}
audioSupervisor *audio.AudioServerSupervisor
audioSupervisor *audio.AudioServerSupervisor
)
func runAudioServer() {
@ -68,7 +68,7 @@ func startAudioSubprocess() error {
// onProcessStart
func(pid int) {
logger.Info().Int("pid", pid).Msg("audio server process started")
// Start audio relay system for main process without a track initially
// The track will be updated when a WebRTC session is created
if err := audio.StartAudioRelay(nil); err != nil {
@ -82,7 +82,7 @@ func startAudioSubprocess() error {
} else {
logger.Info().Int("pid", pid).Msg("audio server process exited gracefully")
}
// Stop audio relay when process exits
audio.StopAudioRelay()
},
@ -100,12 +100,12 @@ func startAudioSubprocess() error {
// Monitor supervisor and handle cleanup
go func() {
defer close(audioProcessDone)
// Wait for supervisor to stop
for audioSupervisor.IsRunning() {
time.Sleep(100 * time.Millisecond)
}
logger.Info().Msg("audio supervisor stopped")
}()