mirror of https://github.com/jetkvm/kvm.git
[WIP] Fix: performance issues
This commit is contained in:
Alex P
parent
c51bdc50b5
commit
767311ec04
3
cloud.go
3
cloud.go
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@ -39,7 +39,8 @@ const (
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// should be lower than the websocket response timeout set in cloud-api
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CloudOidcRequestTimeout = 10 * time.Second
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// WebsocketPingInterval is the interval at which the websocket client sends ping messages to the cloud
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WebsocketPingInterval = 15 * time.Second
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// Increased to 30 seconds for constrained environments to reduce overhead
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WebsocketPingInterval = 30 * time.Second
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)
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var (
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@ -14,8 +14,10 @@ import (
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#include <opus.h>
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#include <stdlib.h>
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#include <string.h>
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#include <errno.h>
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#include <unistd.h>
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// C state for ALSA/Opus
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// C state for ALSA/Opus with safety flags
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static snd_pcm_t *pcm_handle = NULL;
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static snd_pcm_t *pcm_playback_handle = NULL;
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static OpusEncoder *encoder = NULL;
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@ -27,46 +29,193 @@ static int channels = 2;
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static int frame_size = 960; // 20ms for 48kHz
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static int max_packet_size = 1500;
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// Initialize ALSA and Opus encoder
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int jetkvm_audio_init() {
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// State tracking to prevent race conditions during rapid start/stop
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static volatile int capture_initializing = 0;
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static volatile int capture_initialized = 0;
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static volatile int playback_initializing = 0;
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static volatile int playback_initialized = 0;
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// Safe ALSA device opening with retry logic
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static int safe_alsa_open(snd_pcm_t **handle, const char *device, snd_pcm_stream_t stream) {
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int attempts = 3;
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int err;
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snd_pcm_hw_params_t *params;
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if (pcm_handle) return 0;
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if (snd_pcm_open(&pcm_handle, "hw:1,0", SND_PCM_STREAM_CAPTURE, 0) < 0)
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return -1;
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snd_pcm_hw_params_malloc(¶ms);
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snd_pcm_hw_params_any(pcm_handle, params);
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snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
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snd_pcm_hw_params_set_format(pcm_handle, params, SND_PCM_FORMAT_S16_LE);
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snd_pcm_hw_params_set_channels(pcm_handle, params, channels);
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snd_pcm_hw_params_set_rate(pcm_handle, params, sample_rate, 0);
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snd_pcm_hw_params_set_period_size(pcm_handle, params, frame_size, 0);
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snd_pcm_hw_params(pcm_handle, params);
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snd_pcm_hw_params_free(params);
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snd_pcm_prepare(pcm_handle);
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encoder = opus_encoder_create(sample_rate, channels, OPUS_APPLICATION_AUDIO, &err);
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if (!encoder) return -2;
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opus_encoder_ctl(encoder, OPUS_SET_BITRATE(opus_bitrate));
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opus_encoder_ctl(encoder, OPUS_SET_COMPLEXITY(opus_complexity));
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while (attempts-- > 0) {
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err = snd_pcm_open(handle, device, stream, SND_PCM_NONBLOCK);
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if (err >= 0) {
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// Switch to blocking mode after successful open
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snd_pcm_nonblock(*handle, 0);
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return 0;
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}
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// Read and encode one frame, returns encoded size or <0 on error
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if (err == -EBUSY && attempts > 0) {
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// Device busy, wait and retry
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usleep(50000); // 50ms
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continue;
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}
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break;
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}
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return err;
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}
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// Optimized ALSA configuration with stack allocation and performance tuning
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static int configure_alsa_device(snd_pcm_t *handle, const char *device_name) {
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snd_pcm_hw_params_t *params;
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snd_pcm_sw_params_t *sw_params;
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int err;
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if (!handle) return -1;
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// Use stack allocation for better performance
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snd_pcm_hw_params_alloca(¶ms);
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snd_pcm_sw_params_alloca(&sw_params);
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// Hardware parameters
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err = snd_pcm_hw_params_any(handle, params);
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if (err < 0) return err;
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err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0) return err;
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err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
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if (err < 0) return err;
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err = snd_pcm_hw_params_set_channels(handle, params, channels);
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if (err < 0) return err;
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// Set exact rate for better performance
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err = snd_pcm_hw_params_set_rate(handle, params, sample_rate, 0);
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if (err < 0) {
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// Fallback to near rate if exact fails
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unsigned int rate = sample_rate;
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err = snd_pcm_hw_params_set_rate_near(handle, params, &rate, 0);
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if (err < 0) return err;
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}
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// Optimize buffer sizes for low latency
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snd_pcm_uframes_t period_size = frame_size;
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err = snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, 0);
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if (err < 0) return err;
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// Set buffer size to 4 periods for good latency/stability balance
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snd_pcm_uframes_t buffer_size = period_size * 4;
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err = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size);
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if (err < 0) return err;
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err = snd_pcm_hw_params(handle, params);
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if (err < 0) return err;
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// Software parameters for optimal performance
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err = snd_pcm_sw_params_current(handle, sw_params);
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if (err < 0) return err;
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// Start playback/capture when buffer is period_size frames
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err = snd_pcm_sw_params_set_start_threshold(handle, sw_params, period_size);
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if (err < 0) return err;
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// Allow transfers when at least period_size frames are available
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err = snd_pcm_sw_params_set_avail_min(handle, sw_params, period_size);
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if (err < 0) return err;
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err = snd_pcm_sw_params(handle, sw_params);
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if (err < 0) return err;
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return snd_pcm_prepare(handle);
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}
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// Initialize ALSA and Opus encoder with improved safety
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int jetkvm_audio_init() {
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int err;
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// Prevent concurrent initialization
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if (__sync_bool_compare_and_swap(&capture_initializing, 0, 1) == 0) {
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return -EBUSY; // Already initializing
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}
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// Check if already initialized
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if (capture_initialized) {
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capture_initializing = 0;
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return 0;
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}
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// Clean up any existing resources first
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if (encoder) {
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opus_encoder_destroy(encoder);
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encoder = NULL;
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}
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if (pcm_handle) {
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snd_pcm_close(pcm_handle);
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pcm_handle = NULL;
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}
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// Try to open ALSA capture device
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err = safe_alsa_open(&pcm_handle, "hw:1,0", SND_PCM_STREAM_CAPTURE);
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if (err < 0) {
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capture_initializing = 0;
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return -1;
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}
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// Configure the device
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err = configure_alsa_device(pcm_handle, "capture");
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if (err < 0) {
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snd_pcm_close(pcm_handle);
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pcm_handle = NULL;
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capture_initializing = 0;
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return -1;
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}
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// Initialize Opus encoder
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int opus_err = 0;
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encoder = opus_encoder_create(sample_rate, channels, OPUS_APPLICATION_AUDIO, &opus_err);
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if (!encoder || opus_err != OPUS_OK) {
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if (pcm_handle) { snd_pcm_close(pcm_handle); pcm_handle = NULL; }
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capture_initializing = 0;
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return -2;
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}
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opus_encoder_ctl(encoder, OPUS_SET_BITRATE(opus_bitrate));
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opus_encoder_ctl(encoder, OPUS_SET_COMPLEXITY(opus_complexity));
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capture_initialized = 1;
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capture_initializing = 0;
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return 0;
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}
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// Read and encode one frame with enhanced error handling
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int jetkvm_audio_read_encode(void *opus_buf) {
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short pcm_buffer[1920]; // max 2ch*960
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unsigned char *out = (unsigned char*)opus_buf;
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int err = 0;
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// Safety checks
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if (!capture_initialized || !pcm_handle || !encoder || !opus_buf) {
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return -1;
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}
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int pcm_rc = snd_pcm_readi(pcm_handle, pcm_buffer, frame_size);
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// Handle ALSA errors with recovery
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// Handle ALSA errors with enhanced recovery
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if (pcm_rc < 0) {
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if (pcm_rc == -EPIPE) {
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// Buffer underrun - try to recover
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snd_pcm_prepare(pcm_handle);
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err = snd_pcm_prepare(pcm_handle);
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if (err < 0) return -1;
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pcm_rc = snd_pcm_readi(pcm_handle, pcm_buffer, frame_size);
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if (pcm_rc < 0) return -1;
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} else if (pcm_rc == -EAGAIN) {
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// No data available - return 0 to indicate no frame
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return 0;
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} else if (pcm_rc == -ESTRPIPE) {
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// Device suspended, try to resume
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while ((err = snd_pcm_resume(pcm_handle)) == -EAGAIN) {
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usleep(1000); // 1ms
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}
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if (err < 0) {
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err = snd_pcm_prepare(pcm_handle);
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if (err < 0) return -1;
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}
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return 0; // Skip this frame
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} else {
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// Other error - return error code
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return -1;
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@ -82,54 +231,105 @@ int jetkvm_audio_read_encode(void *opus_buf) {
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return nb_bytes;
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}
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// Initialize ALSA playback for microphone input (browser -> USB gadget)
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// Initialize ALSA playback with improved safety
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int jetkvm_audio_playback_init() {
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int err;
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snd_pcm_hw_params_t *params;
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if (pcm_playback_handle) return 0;
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// Try to open the USB gadget audio device for playback
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// This should correspond to the capture endpoint of the USB gadget
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if (snd_pcm_open(&pcm_playback_handle, "hw:1,0", SND_PCM_STREAM_PLAYBACK, 0) < 0) {
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// Fallback to default device if hw:1,0 doesn't work for playback
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if (snd_pcm_open(&pcm_playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0) < 0)
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return -1;
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// Prevent concurrent initialization
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if (__sync_bool_compare_and_swap(&playback_initializing, 0, 1) == 0) {
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return -EBUSY; // Already initializing
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}
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snd_pcm_hw_params_malloc(¶ms);
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snd_pcm_hw_params_any(pcm_playback_handle, params);
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snd_pcm_hw_params_set_access(pcm_playback_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
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snd_pcm_hw_params_set_format(pcm_playback_handle, params, SND_PCM_FORMAT_S16_LE);
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snd_pcm_hw_params_set_channels(pcm_playback_handle, params, channels);
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snd_pcm_hw_params_set_rate(pcm_playback_handle, params, sample_rate, 0);
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snd_pcm_hw_params_set_period_size(pcm_playback_handle, params, frame_size, 0);
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snd_pcm_hw_params(pcm_playback_handle, params);
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snd_pcm_hw_params_free(params);
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snd_pcm_prepare(pcm_playback_handle);
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// Initialize Opus decoder
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decoder = opus_decoder_create(sample_rate, channels, &err);
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if (!decoder) return -2;
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// Check if already initialized
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if (playback_initialized) {
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playback_initializing = 0;
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return 0;
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}
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// Decode Opus and write PCM to playback device
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// Clean up any existing resources first
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if (decoder) {
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opus_decoder_destroy(decoder);
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decoder = NULL;
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}
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if (pcm_playback_handle) {
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snd_pcm_close(pcm_playback_handle);
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pcm_playback_handle = NULL;
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}
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// Try to open the USB gadget audio device for playback
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err = safe_alsa_open(&pcm_playback_handle, "hw:1,0", SND_PCM_STREAM_PLAYBACK);
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if (err < 0) {
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// Fallback to default device
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err = safe_alsa_open(&pcm_playback_handle, "default", SND_PCM_STREAM_PLAYBACK);
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if (err < 0) {
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playback_initializing = 0;
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return -1;
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}
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}
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// Configure the device
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err = configure_alsa_device(pcm_playback_handle, "playback");
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if (err < 0) {
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snd_pcm_close(pcm_playback_handle);
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pcm_playback_handle = NULL;
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playback_initializing = 0;
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return -1;
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}
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// Initialize Opus decoder
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int opus_err = 0;
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decoder = opus_decoder_create(sample_rate, channels, &opus_err);
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if (!decoder || opus_err != OPUS_OK) {
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snd_pcm_close(pcm_playback_handle);
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pcm_playback_handle = NULL;
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playback_initializing = 0;
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return -2;
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}
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playback_initialized = 1;
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playback_initializing = 0;
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return 0;
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}
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// Decode Opus and write PCM with enhanced error handling
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int jetkvm_audio_decode_write(void *opus_buf, int opus_size) {
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short pcm_buffer[1920]; // max 2ch*960
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unsigned char *in = (unsigned char*)opus_buf;
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int err = 0;
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// Safety checks
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if (!playback_initialized || !pcm_playback_handle || !decoder || !opus_buf || opus_size <= 0) {
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return -1;
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}
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// Additional bounds checking
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if (opus_size > max_packet_size) {
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return -1;
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}
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// Decode Opus to PCM
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int pcm_frames = opus_decode(decoder, in, opus_size, pcm_buffer, frame_size, 0);
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if (pcm_frames < 0) return -1;
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// Write PCM to playback device
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// Write PCM to playback device with enhanced recovery
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int pcm_rc = snd_pcm_writei(pcm_playback_handle, pcm_buffer, pcm_frames);
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if (pcm_rc < 0) {
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// Try to recover from underrun
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if (pcm_rc == -EPIPE) {
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snd_pcm_prepare(pcm_playback_handle);
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// Buffer underrun - try to recover
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err = snd_pcm_prepare(pcm_playback_handle);
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if (err < 0) return -2;
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pcm_rc = snd_pcm_writei(pcm_playback_handle, pcm_buffer, pcm_frames);
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} else if (pcm_rc == -ESTRPIPE) {
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// Device suspended, try to resume
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while ((err = snd_pcm_resume(pcm_playback_handle)) == -EAGAIN) {
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usleep(1000); // 1ms
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}
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if (err < 0) {
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err = snd_pcm_prepare(pcm_playback_handle);
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if (err < 0) return -2;
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}
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return 0; // Skip this frame
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}
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if (pcm_rc < 0) return -2;
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}
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@ -137,14 +337,49 @@ int jetkvm_audio_decode_write(void *opus_buf, int opus_size) {
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return pcm_frames;
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}
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// Safe playback cleanup with double-close protection
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void jetkvm_audio_playback_close() {
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if (decoder) { opus_decoder_destroy(decoder); decoder = NULL; }
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if (pcm_playback_handle) { snd_pcm_close(pcm_playback_handle); pcm_playback_handle = NULL; }
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// Wait for any ongoing operations to complete
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while (playback_initializing) {
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usleep(1000); // 1ms
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}
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// Atomic check and set to prevent double cleanup
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if (__sync_bool_compare_and_swap(&playback_initialized, 1, 0) == 0) {
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return; // Already cleaned up
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}
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if (decoder) {
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opus_decoder_destroy(decoder);
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decoder = NULL;
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}
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if (pcm_playback_handle) {
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snd_pcm_drain(pcm_playback_handle);
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snd_pcm_close(pcm_playback_handle);
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pcm_playback_handle = NULL;
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}
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}
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// Safe capture cleanup
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void jetkvm_audio_close() {
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if (encoder) { opus_encoder_destroy(encoder); encoder = NULL; }
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if (pcm_handle) { snd_pcm_close(pcm_handle); pcm_handle = NULL; }
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// Wait for any ongoing operations to complete
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while (capture_initializing) {
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usleep(1000); // 1ms
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}
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capture_initialized = 0;
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if (encoder) {
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opus_encoder_destroy(encoder);
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encoder = NULL;
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}
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if (pcm_handle) {
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snd_pcm_drop(pcm_handle); // Drop pending samples
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snd_pcm_close(pcm_handle);
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pcm_handle = NULL;
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}
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// Also clean up playback
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jetkvm_audio_playback_close();
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}
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*/
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@ -197,7 +432,31 @@ func cgoAudioDecodeWrite(buf []byte) (int, error) {
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if len(buf) == 0 {
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return 0, errors.New("empty buffer")
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}
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n := C.jetkvm_audio_decode_write(unsafe.Pointer(&buf[0]), C.int(len(buf)))
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// Additional safety check to prevent segfault
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if buf == nil {
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return 0, errors.New("nil buffer")
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}
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// Validate buffer size to prevent potential overruns
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if len(buf) > 4096 { // Maximum reasonable Opus frame size
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return 0, errors.New("buffer too large")
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}
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// Ensure buffer is not deallocated by keeping a reference
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bufPtr := unsafe.Pointer(&buf[0])
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if bufPtr == nil {
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return 0, errors.New("invalid buffer pointer")
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}
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||||
// Add recovery mechanism for C function crashes
|
||||
defer func() {
|
||||
if r := recover(); r != nil {
|
||||
// Log the panic but don't crash the entire program
|
||||
// This should not happen with proper validation, but provides safety
|
||||
}
|
||||
}()
|
||||
|
||||
n := C.jetkvm_audio_decode_write(bufPtr, C.int(len(buf)))
|
||||
if n < 0 {
|
||||
return 0, errors.New("audio decode/write error")
|
||||
}
|
||||
|
|
@ -205,26 +464,11 @@ func cgoAudioDecodeWrite(buf []byte) (int, error) {
|
|||
}
|
||||
|
||||
// Wrapper functions for non-blocking audio manager
|
||||
func CGOAudioInit() error {
|
||||
return cgoAudioInit()
|
||||
}
|
||||
|
||||
func CGOAudioClose() {
|
||||
cgoAudioClose()
|
||||
}
|
||||
|
||||
func CGOAudioReadEncode(buf []byte) (int, error) {
|
||||
return cgoAudioReadEncode(buf)
|
||||
}
|
||||
|
||||
func CGOAudioPlaybackInit() error {
|
||||
return cgoAudioPlaybackInit()
|
||||
}
|
||||
|
||||
func CGOAudioPlaybackClose() {
|
||||
cgoAudioPlaybackClose()
|
||||
}
|
||||
|
||||
func CGOAudioDecodeWrite(buf []byte) (int, error) {
|
||||
return cgoAudioDecodeWrite(buf)
|
||||
}
|
||||
var (
|
||||
CGOAudioInit = cgoAudioInit
|
||||
CGOAudioClose = cgoAudioClose
|
||||
CGOAudioReadEncode = cgoAudioReadEncode
|
||||
CGOAudioPlaybackInit = cgoAudioPlaybackInit
|
||||
CGOAudioPlaybackClose = cgoAudioPlaybackClose
|
||||
CGOAudioDecodeWrite = cgoAudioDecodeWrite
|
||||
)
|
||||
|
|
|
|||
|
|
@ -30,28 +30,13 @@ func cgoAudioDecodeWrite(buf []byte) (int, error) {
|
|||
return 0, errors.New("audio not available in lint mode")
|
||||
}
|
||||
|
||||
// Uppercase wrapper functions (called by nonblocking_audio.go)
|
||||
// Uppercase aliases for external API compatibility
|
||||
|
||||
func CGOAudioInit() error {
|
||||
return cgoAudioInit()
|
||||
}
|
||||
|
||||
func CGOAudioClose() {
|
||||
cgoAudioClose()
|
||||
}
|
||||
|
||||
func CGOAudioReadEncode(buf []byte) (int, error) {
|
||||
return cgoAudioReadEncode(buf)
|
||||
}
|
||||
|
||||
func CGOAudioPlaybackInit() error {
|
||||
return cgoAudioPlaybackInit()
|
||||
}
|
||||
|
||||
func CGOAudioPlaybackClose() {
|
||||
cgoAudioPlaybackClose()
|
||||
}
|
||||
|
||||
func CGOAudioDecodeWrite(buf []byte) (int, error) {
|
||||
return cgoAudioDecodeWrite(buf)
|
||||
}
|
||||
var (
|
||||
CGOAudioInit = cgoAudioInit
|
||||
CGOAudioClose = cgoAudioClose
|
||||
CGOAudioReadEncode = cgoAudioReadEncode
|
||||
CGOAudioPlaybackInit = cgoAudioPlaybackInit
|
||||
CGOAudioPlaybackClose = cgoAudioPlaybackClose
|
||||
CGOAudioDecodeWrite = cgoAudioDecodeWrite
|
||||
)
|
||||
|
|
|
|||
|
|
@ -2,6 +2,7 @@ package audio
|
|||
|
||||
import (
|
||||
"context"
|
||||
"strings"
|
||||
"sync"
|
||||
"time"
|
||||
|
||||
|
|
@ -111,6 +112,14 @@ func (aeb *AudioEventBroadcaster) Subscribe(connectionID string, conn *websocket
|
|||
aeb.mutex.Lock()
|
||||
defer aeb.mutex.Unlock()
|
||||
|
||||
// Check if there's already a subscription for this connectionID
|
||||
if _, exists := aeb.subscribers[connectionID]; exists {
|
||||
aeb.logger.Debug().Str("connectionID", connectionID).Msg("duplicate audio events subscription detected; replacing existing entry")
|
||||
// Do NOT close the existing WebSocket connection here because it's shared
|
||||
// with the signaling channel. Just replace the subscriber map entry.
|
||||
delete(aeb.subscribers, connectionID)
|
||||
}
|
||||
|
||||
aeb.subscribers[connectionID] = &AudioEventSubscriber{
|
||||
conn: conn,
|
||||
ctx: ctx,
|
||||
|
|
@ -233,16 +242,37 @@ func (aeb *AudioEventBroadcaster) sendCurrentMetrics(subscriber *AudioEventSubsc
|
|||
|
||||
// startMetricsBroadcasting starts a goroutine that periodically broadcasts metrics
|
||||
func (aeb *AudioEventBroadcaster) startMetricsBroadcasting() {
|
||||
ticker := time.NewTicker(2 * time.Second) // Same interval as current polling
|
||||
// Use 5-second interval instead of 2 seconds for constrained environments
|
||||
ticker := time.NewTicker(5 * time.Second)
|
||||
defer ticker.Stop()
|
||||
|
||||
for range ticker.C {
|
||||
aeb.mutex.RLock()
|
||||
subscriberCount := len(aeb.subscribers)
|
||||
|
||||
// Early exit if no subscribers to save CPU
|
||||
if subscriberCount == 0 {
|
||||
aeb.mutex.RUnlock()
|
||||
continue
|
||||
}
|
||||
|
||||
// Create a copy for safe iteration
|
||||
subscribersCopy := make([]*AudioEventSubscriber, 0, subscriberCount)
|
||||
for _, sub := range aeb.subscribers {
|
||||
subscribersCopy = append(subscribersCopy, sub)
|
||||
}
|
||||
aeb.mutex.RUnlock()
|
||||
|
||||
// Only broadcast if there are subscribers
|
||||
if subscriberCount == 0 {
|
||||
// Pre-check for cancelled contexts to avoid unnecessary work
|
||||
activeSubscribers := 0
|
||||
for _, sub := range subscribersCopy {
|
||||
if sub.ctx.Err() == nil {
|
||||
activeSubscribers++
|
||||
}
|
||||
}
|
||||
|
||||
// Skip metrics gathering if no active subscribers
|
||||
if activeSubscribers == 0 {
|
||||
continue
|
||||
}
|
||||
|
||||
|
|
@ -286,29 +316,54 @@ func (aeb *AudioEventBroadcaster) startMetricsBroadcasting() {
|
|||
// broadcast sends an event to all subscribers
|
||||
func (aeb *AudioEventBroadcaster) broadcast(event AudioEvent) {
|
||||
aeb.mutex.RLock()
|
||||
defer aeb.mutex.RUnlock()
|
||||
|
||||
for connectionID, subscriber := range aeb.subscribers {
|
||||
go func(id string, sub *AudioEventSubscriber) {
|
||||
if !aeb.sendToSubscriber(sub, event) {
|
||||
// Remove failed subscriber
|
||||
aeb.mutex.Lock()
|
||||
delete(aeb.subscribers, id)
|
||||
aeb.mutex.Unlock()
|
||||
aeb.logger.Warn().Str("connectionID", id).Msg("removed failed audio events subscriber")
|
||||
// Create a copy of subscribers to avoid holding the lock during sending
|
||||
subscribersCopy := make(map[string]*AudioEventSubscriber)
|
||||
for id, sub := range aeb.subscribers {
|
||||
subscribersCopy[id] = sub
|
||||
}
|
||||
}(connectionID, subscriber)
|
||||
aeb.mutex.RUnlock()
|
||||
|
||||
// Track failed subscribers to remove them after sending
|
||||
var failedSubscribers []string
|
||||
|
||||
// Send to all subscribers without holding the lock
|
||||
for connectionID, subscriber := range subscribersCopy {
|
||||
if !aeb.sendToSubscriber(subscriber, event) {
|
||||
failedSubscribers = append(failedSubscribers, connectionID)
|
||||
}
|
||||
}
|
||||
|
||||
// Remove failed subscribers if any
|
||||
if len(failedSubscribers) > 0 {
|
||||
aeb.mutex.Lock()
|
||||
for _, connectionID := range failedSubscribers {
|
||||
delete(aeb.subscribers, connectionID)
|
||||
aeb.logger.Warn().Str("connectionID", connectionID).Msg("removed failed audio events subscriber")
|
||||
}
|
||||
aeb.mutex.Unlock()
|
||||
}
|
||||
}
|
||||
|
||||
// sendToSubscriber sends an event to a specific subscriber
|
||||
func (aeb *AudioEventBroadcaster) sendToSubscriber(subscriber *AudioEventSubscriber, event AudioEvent) bool {
|
||||
ctx, cancel := context.WithTimeout(subscriber.ctx, 5*time.Second)
|
||||
// Check if subscriber context is already cancelled
|
||||
if subscriber.ctx.Err() != nil {
|
||||
return false
|
||||
}
|
||||
|
||||
ctx, cancel := context.WithTimeout(subscriber.ctx, 2*time.Second)
|
||||
defer cancel()
|
||||
|
||||
err := wsjson.Write(ctx, subscriber.conn, event)
|
||||
if err != nil {
|
||||
// Don't log network errors for closed connections as warnings, they're expected
|
||||
if strings.Contains(err.Error(), "use of closed network connection") ||
|
||||
strings.Contains(err.Error(), "connection reset by peer") ||
|
||||
strings.Contains(err.Error(), "context canceled") {
|
||||
subscriber.logger.Debug().Err(err).Msg("websocket connection closed during audio event send")
|
||||
} else {
|
||||
subscriber.logger.Warn().Err(err).Msg("failed to send audio event to subscriber")
|
||||
}
|
||||
return false
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -60,6 +60,11 @@ func StopNonBlockingAudioInput() {
|
|||
|
||||
if globalNonBlockingManager != nil && globalNonBlockingManager.IsInputRunning() {
|
||||
globalNonBlockingManager.StopAudioInput()
|
||||
|
||||
// If both input and output are stopped, recreate manager to ensure clean state
|
||||
if !globalNonBlockingManager.IsRunning() {
|
||||
globalNonBlockingManager = nil
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -2,6 +2,7 @@ package audio
|
|||
|
||||
import (
|
||||
"context"
|
||||
"errors"
|
||||
"runtime"
|
||||
"sync"
|
||||
"sync/atomic"
|
||||
|
|
@ -273,7 +274,9 @@ func (nam *NonBlockingAudioManager) inputWorkerThread() {
|
|||
defer runtime.UnlockOSThread()
|
||||
|
||||
defer nam.wg.Done()
|
||||
defer atomic.StoreInt32(&nam.inputWorkerRunning, 0)
|
||||
// Cleanup CGO resources properly to avoid double-close scenarios
|
||||
// The outputWorkerThread's CGOAudioClose() will handle all cleanup
|
||||
atomic.StoreInt32(&nam.inputWorkerRunning, 0)
|
||||
|
||||
atomic.StoreInt32(&nam.inputWorkerRunning, 1)
|
||||
nam.logger.Debug().Msg("input worker thread started")
|
||||
|
|
@ -283,32 +286,102 @@ func (nam *NonBlockingAudioManager) inputWorkerThread() {
|
|||
nam.logger.Error().Err(err).Msg("failed to initialize audio playback in worker thread")
|
||||
return
|
||||
}
|
||||
defer CGOAudioPlaybackClose()
|
||||
|
||||
// Ensure CGO cleanup happens even if we exit unexpectedly
|
||||
cgoInitialized := true
|
||||
defer func() {
|
||||
if cgoInitialized {
|
||||
nam.logger.Debug().Msg("cleaning up CGO audio playback")
|
||||
// Add extra safety: ensure no more CGO calls can happen
|
||||
atomic.StoreInt32(&nam.inputWorkerRunning, 0)
|
||||
// Note: Don't call CGOAudioPlaybackClose() here to avoid double-close
|
||||
// The outputWorkerThread's CGOAudioClose() will handle all cleanup
|
||||
}
|
||||
}()
|
||||
|
||||
for {
|
||||
// If coordinator has stopped, exit worker loop
|
||||
if atomic.LoadInt32(&nam.inputRunning) == 0 {
|
||||
return
|
||||
}
|
||||
select {
|
||||
case <-nam.ctx.Done():
|
||||
nam.logger.Debug().Msg("input worker thread stopping")
|
||||
nam.logger.Debug().Msg("input worker thread stopping due to context cancellation")
|
||||
return
|
||||
|
||||
case workItem := <-nam.inputWorkChan:
|
||||
switch workItem.workType {
|
||||
case audioWorkDecodeWrite:
|
||||
// Perform blocking audio decode/write operation
|
||||
n, err := CGOAudioDecodeWrite(workItem.data)
|
||||
result := audioResult{
|
||||
success: err == nil,
|
||||
length: n,
|
||||
err: err,
|
||||
// Check if we're still supposed to be running before processing
|
||||
if atomic.LoadInt32(&nam.inputWorkerRunning) == 0 || atomic.LoadInt32(&nam.inputRunning) == 0 {
|
||||
nam.logger.Debug().Msg("input worker stopping, ignoring decode work")
|
||||
// Do not send to resultChan; coordinator may have exited
|
||||
return
|
||||
}
|
||||
|
||||
// Send result back (non-blocking)
|
||||
// Validate input data before CGO call
|
||||
if workItem.data == nil || len(workItem.data) == 0 {
|
||||
result := audioResult{
|
||||
success: false,
|
||||
err: errors.New("invalid audio data"),
|
||||
}
|
||||
|
||||
// Check if coordinator is still running before sending result
|
||||
if atomic.LoadInt32(&nam.inputRunning) == 1 {
|
||||
select {
|
||||
case workItem.resultChan <- result:
|
||||
case <-nam.ctx.Done():
|
||||
return
|
||||
default:
|
||||
// Drop result if coordinator is not ready
|
||||
case <-time.After(10 * time.Millisecond):
|
||||
// Timeout - coordinator may have stopped, drop result
|
||||
atomic.AddInt64(&nam.stats.InputFramesDropped, 1)
|
||||
}
|
||||
} else {
|
||||
// Coordinator has stopped, drop result
|
||||
atomic.AddInt64(&nam.stats.InputFramesDropped, 1)
|
||||
}
|
||||
continue
|
||||
}
|
||||
|
||||
// Perform blocking CGO operation with panic recovery
|
||||
var result audioResult
|
||||
func() {
|
||||
defer func() {
|
||||
if r := recover(); r != nil {
|
||||
nam.logger.Error().Interface("panic", r).Msg("CGO decode write panic recovered")
|
||||
result = audioResult{
|
||||
success: false,
|
||||
err: errors.New("CGO decode write panic"),
|
||||
}
|
||||
}
|
||||
}()
|
||||
|
||||
// Double-check we're still running before CGO call
|
||||
if atomic.LoadInt32(&nam.inputWorkerRunning) == 0 {
|
||||
result = audioResult{success: false, err: errors.New("worker shutting down")}
|
||||
return
|
||||
}
|
||||
|
||||
n, err := CGOAudioDecodeWrite(workItem.data)
|
||||
result = audioResult{
|
||||
success: err == nil,
|
||||
length: n,
|
||||
err: err,
|
||||
}
|
||||
}()
|
||||
|
||||
// Send result back (non-blocking) - check if coordinator is still running
|
||||
if atomic.LoadInt32(&nam.inputRunning) == 1 {
|
||||
select {
|
||||
case workItem.resultChan <- result:
|
||||
case <-nam.ctx.Done():
|
||||
return
|
||||
case <-time.After(10 * time.Millisecond):
|
||||
// Timeout - coordinator may have stopped, drop result
|
||||
atomic.AddInt64(&nam.stats.InputFramesDropped, 1)
|
||||
}
|
||||
} else {
|
||||
// Coordinator has stopped, drop result
|
||||
atomic.AddInt64(&nam.stats.InputFramesDropped, 1)
|
||||
}
|
||||
|
||||
|
|
@ -328,6 +401,7 @@ func (nam *NonBlockingAudioManager) inputCoordinatorThread() {
|
|||
nam.logger.Debug().Msg("input coordinator thread started")
|
||||
|
||||
resultChan := make(chan audioResult, 1)
|
||||
// Do not close resultChan to avoid races with worker sends during shutdown
|
||||
|
||||
for atomic.LoadInt32(&nam.inputRunning) == 1 {
|
||||
select {
|
||||
|
|
@ -350,7 +424,7 @@ func (nam *NonBlockingAudioManager) inputCoordinatorThread() {
|
|||
|
||||
select {
|
||||
case nam.inputWorkChan <- workItem:
|
||||
// Wait for result with timeout
|
||||
// Wait for result with timeout and context cancellation
|
||||
select {
|
||||
case result := <-resultChan:
|
||||
if result.success {
|
||||
|
|
@ -362,10 +436,18 @@ func (nam *NonBlockingAudioManager) inputCoordinatorThread() {
|
|||
nam.logger.Warn().Err(result.err).Msg("audio input worker error")
|
||||
}
|
||||
}
|
||||
case <-nam.ctx.Done():
|
||||
nam.logger.Debug().Msg("input coordinator stopping during result wait")
|
||||
return
|
||||
case <-time.After(50 * time.Millisecond):
|
||||
// Timeout waiting for result
|
||||
atomic.AddInt64(&nam.stats.InputFramesDropped, 1)
|
||||
nam.logger.Warn().Msg("timeout waiting for input worker result")
|
||||
// Drain any pending result to prevent worker blocking
|
||||
select {
|
||||
case <-resultChan:
|
||||
default:
|
||||
}
|
||||
}
|
||||
default:
|
||||
// Worker is busy, drop this frame
|
||||
|
|
@ -379,13 +461,7 @@ func (nam *NonBlockingAudioManager) inputCoordinatorThread() {
|
|||
}
|
||||
}
|
||||
|
||||
// Signal worker to close
|
||||
select {
|
||||
case nam.inputWorkChan <- audioWorkItem{workType: audioWorkClose}:
|
||||
case <-time.After(100 * time.Millisecond):
|
||||
nam.logger.Warn().Msg("timeout signaling input worker to close")
|
||||
}
|
||||
|
||||
// Avoid sending close signals or touching channels here; inputRunning=0 will stop worker via checks
|
||||
nam.logger.Info().Msg("input coordinator thread stopped")
|
||||
}
|
||||
|
||||
|
|
@ -413,11 +489,37 @@ func (nam *NonBlockingAudioManager) StopAudioInput() {
|
|||
// Stop only the input coordinator
|
||||
atomic.StoreInt32(&nam.inputRunning, 0)
|
||||
|
||||
// Allow coordinator thread to process the stop signal and update state
|
||||
// This prevents race conditions in state queries immediately after stopping
|
||||
time.Sleep(50 * time.Millisecond)
|
||||
// Drain the receive channel to prevent blocking senders
|
||||
go func() {
|
||||
for {
|
||||
select {
|
||||
case <-nam.inputReceiveChan:
|
||||
// Drain any remaining frames
|
||||
case <-time.After(100 * time.Millisecond):
|
||||
return
|
||||
}
|
||||
}
|
||||
}()
|
||||
|
||||
nam.logger.Info().Msg("audio input stopped")
|
||||
// Wait for the worker to actually stop to prevent race conditions
|
||||
timeout := time.After(2 * time.Second)
|
||||
ticker := time.NewTicker(10 * time.Millisecond)
|
||||
defer ticker.Stop()
|
||||
|
||||
for {
|
||||
select {
|
||||
case <-timeout:
|
||||
nam.logger.Warn().Msg("timeout waiting for input worker to stop")
|
||||
return
|
||||
case <-ticker.C:
|
||||
if atomic.LoadInt32(&nam.inputWorkerRunning) == 0 {
|
||||
nam.logger.Info().Msg("audio input stopped successfully")
|
||||
// Close ALSA playback resources now that input worker has stopped
|
||||
CGOAudioPlaybackClose()
|
||||
return
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// GetStats returns current statistics
|
||||
|
|
|
|||
|
|
@ -150,7 +150,7 @@ export default function Actionbar({
|
|||
"flex origin-top flex-col transition duration-300 ease-out data-closed:translate-y-8 data-closed:opacity-0",
|
||||
)}
|
||||
>
|
||||
{({ open }) => {
|
||||
{({ open }: { open: boolean }) => {
|
||||
checkIfStateChanged(open);
|
||||
return (
|
||||
<div className="mx-auto w-full max-w-xl">
|
||||
|
|
@ -192,7 +192,7 @@ export default function Actionbar({
|
|||
"flex origin-top flex-col transition duration-300 ease-out data-closed:translate-y-8 data-closed:opacity-0",
|
||||
)}
|
||||
>
|
||||
{({ open }) => {
|
||||
{({ open }: { open: boolean }) => {
|
||||
checkIfStateChanged(open);
|
||||
return (
|
||||
<div className="mx-auto w-full max-w-xl">
|
||||
|
|
@ -244,7 +244,7 @@ export default function Actionbar({
|
|||
"flex origin-top flex-col transition duration-300 ease-out data-closed:translate-y-8 data-closed:opacity-0",
|
||||
)}
|
||||
>
|
||||
{({ open }) => {
|
||||
{({ open }: { open: boolean }) => {
|
||||
checkIfStateChanged(open);
|
||||
return (
|
||||
<div className="mx-auto w-full max-w-xl">
|
||||
|
|
@ -287,7 +287,7 @@ export default function Actionbar({
|
|||
"flex origin-top flex-col transition duration-300 ease-out data-closed:translate-y-8 data-closed:opacity-0",
|
||||
)}
|
||||
>
|
||||
{({ open }) => {
|
||||
{({ open }: { open: boolean }) => {
|
||||
checkIfStateChanged(open);
|
||||
return <ExtensionPopover />;
|
||||
}}
|
||||
|
|
@ -369,11 +369,11 @@ export default function Actionbar({
|
|||
"flex origin-top flex-col transition duration-300 ease-out data-closed:translate-y-8 data-closed:opacity-0",
|
||||
)}
|
||||
>
|
||||
{({ open }) => {
|
||||
{({ open }: { open: boolean }) => {
|
||||
checkIfStateChanged(open);
|
||||
return (
|
||||
<div className="mx-auto">
|
||||
<AudioControlPopover microphone={microphone} />
|
||||
<AudioControlPopover microphone={microphone} open={open} />
|
||||
</div>
|
||||
);
|
||||
}}
|
||||
|
|
|
|||
|
|
@ -67,7 +67,12 @@ export default function AudioMetricsDashboard() {
|
|||
|
||||
// Microphone state for audio level monitoring
|
||||
const { isMicrophoneActive, isMicrophoneMuted, microphoneStream } = useMicrophone();
|
||||
const { audioLevel, isAnalyzing } = useAudioLevel(microphoneStream);
|
||||
const { audioLevel, isAnalyzing } = useAudioLevel(
|
||||
isMicrophoneActive ? microphoneStream : null,
|
||||
{
|
||||
enabled: isMicrophoneActive,
|
||||
updateInterval: 120,
|
||||
});
|
||||
|
||||
useEffect(() => {
|
||||
// Load initial configuration (only once)
|
||||
|
|
|
|||
|
|
@ -70,14 +70,18 @@ const qualityLabels = {
|
|||
|
||||
interface AudioControlPopoverProps {
|
||||
microphone: MicrophoneHookReturn;
|
||||
open?: boolean; // whether the popover is open (controls analysis)
|
||||
}
|
||||
|
||||
export default function AudioControlPopover({ microphone }: AudioControlPopoverProps) {
|
||||
export default function AudioControlPopover({ microphone, open }: AudioControlPopoverProps) {
|
||||
const [currentConfig, setCurrentConfig] = useState<AudioConfig | null>(null);
|
||||
const [currentMicrophoneConfig, setCurrentMicrophoneConfig] = useState<AudioConfig | null>(null);
|
||||
const [showAdvanced, setShowAdvanced] = useState(false);
|
||||
const [isLoading, setIsLoading] = useState(false);
|
||||
|
||||
// Add cache flags to prevent unnecessary API calls
|
||||
const [configsLoaded, setConfigsLoaded] = useState(false);
|
||||
|
||||
// Add cooldown to prevent rapid clicking
|
||||
const [lastClickTime, setLastClickTime] = useState(0);
|
||||
const CLICK_COOLDOWN = 500; // 500ms cooldown between clicks
|
||||
|
|
@ -117,8 +121,12 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
|
|||
const micMetrics = wsConnected && microphoneMetrics !== null ? microphoneMetrics : fallbackMicMetrics;
|
||||
const isConnected = wsConnected ? wsConnected : fallbackConnected;
|
||||
|
||||
// Audio level monitoring
|
||||
const { audioLevel, isAnalyzing } = useAudioLevel(microphoneStream);
|
||||
// Audio level monitoring - enable only when popover is open and microphone is active to save resources
|
||||
const analysisEnabled = (open ?? true) && isMicrophoneActive;
|
||||
const { audioLevel, isAnalyzing } = useAudioLevel(analysisEnabled ? microphoneStream : null, {
|
||||
enabled: analysisEnabled,
|
||||
updateInterval: 120, // 8-10 fps to reduce CPU without losing UX quality
|
||||
});
|
||||
|
||||
// Audio devices
|
||||
const {
|
||||
|
|
@ -135,46 +143,61 @@ export default function AudioControlPopover({ microphone }: AudioControlPopoverP
|
|||
|
||||
const { toggleSidebarView } = useUiStore();
|
||||
|
||||
// Load initial configurations once (these don't change frequently)
|
||||
// Load initial configurations once - cache to prevent repeated calls
|
||||
useEffect(() => {
|
||||
if (!configsLoaded) {
|
||||
loadAudioConfigurations();
|
||||
}, []);
|
||||
}
|
||||
}, [configsLoaded]);
|
||||
|
||||
// Load initial audio state and set up fallback polling when WebSocket is not connected
|
||||
// Optimize fallback polling - only run when WebSocket is not connected
|
||||
useEffect(() => {
|
||||
if (!wsConnected) {
|
||||
if (!wsConnected && !configsLoaded) {
|
||||
// Load state once if configs aren't loaded yet
|
||||
loadAudioState();
|
||||
// Only load metrics as fallback when WebSocket is disconnected
|
||||
}
|
||||
|
||||
if (!wsConnected) {
|
||||
loadAudioMetrics();
|
||||
loadMicrophoneMetrics();
|
||||
|
||||
// Set up metrics refresh interval for fallback only
|
||||
// Reduced frequency for fallback polling (every 3 seconds instead of 2)
|
||||
const metricsInterval = setInterval(() => {
|
||||
if (!wsConnected) { // Double-check to prevent unnecessary requests
|
||||
loadAudioMetrics();
|
||||
loadMicrophoneMetrics();
|
||||
}, 2000);
|
||||
}
|
||||
}, 3000);
|
||||
return () => clearInterval(metricsInterval);
|
||||
}
|
||||
|
||||
// Always sync microphone state
|
||||
// Always sync microphone state, but debounce it
|
||||
const syncTimeout = setTimeout(() => {
|
||||
syncMicrophoneState();
|
||||
}, [wsConnected, syncMicrophoneState]);
|
||||
}, 500);
|
||||
|
||||
return () => clearTimeout(syncTimeout);
|
||||
}, [wsConnected, syncMicrophoneState, configsLoaded]);
|
||||
|
||||
const loadAudioConfigurations = async () => {
|
||||
try {
|
||||
// Load quality config
|
||||
const qualityResp = await api.GET("/audio/quality");
|
||||
// Parallel loading for better performance
|
||||
const [qualityResp, micQualityResp] = await Promise.all([
|
||||
api.GET("/audio/quality"),
|
||||
api.GET("/microphone/quality")
|
||||
]);
|
||||
|
||||
if (qualityResp.ok) {
|
||||
const qualityData = await qualityResp.json();
|
||||
setCurrentConfig(qualityData.current);
|
||||
}
|
||||
|
||||
// Load microphone quality config
|
||||
const micQualityResp = await api.GET("/microphone/quality");
|
||||
if (micQualityResp.ok) {
|
||||
const micQualityData = await micQualityResp.json();
|
||||
setCurrentMicrophoneConfig(micQualityData.current);
|
||||
}
|
||||
|
||||
setConfigsLoaded(true);
|
||||
} catch (error) {
|
||||
console.error("Failed to load audio configurations:", error);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -61,16 +61,23 @@ export interface UseAudioEventsReturn {
|
|||
unsubscribe: () => void;
|
||||
}
|
||||
|
||||
// Global subscription management to prevent multiple subscriptions per WebSocket connection
|
||||
let globalSubscriptionState = {
|
||||
isSubscribed: false,
|
||||
subscriberCount: 0,
|
||||
connectionId: null as string | null
|
||||
};
|
||||
|
||||
export function useAudioEvents(): UseAudioEventsReturn {
|
||||
// State for audio data
|
||||
const [audioMuted, setAudioMuted] = useState<boolean | null>(null);
|
||||
const [audioMetrics, setAudioMetrics] = useState<AudioMetricsData | null>(null);
|
||||
const [microphoneState, setMicrophoneState] = useState<MicrophoneStateData | null>(null);
|
||||
const [microphoneMetrics, setMicrophoneMetrics] = useState<MicrophoneMetricsData | null>(null);
|
||||
const [microphoneMetrics, setMicrophoneMetricsData] = useState<MicrophoneMetricsData | null>(null);
|
||||
|
||||
// Subscription state
|
||||
const [isSubscribed, setIsSubscribed] = useState(false);
|
||||
const subscriptionSent = useRef(false);
|
||||
// Local subscription state
|
||||
const [isLocallySubscribed, setIsLocallySubscribed] = useState(false);
|
||||
const subscriptionTimeoutRef = useRef<number | null>(null);
|
||||
|
||||
// Get WebSocket URL
|
||||
const getWebSocketUrl = () => {
|
||||
|
|
@ -79,7 +86,7 @@ export function useAudioEvents(): UseAudioEventsReturn {
|
|||
return `${protocol}//${host}/webrtc/signaling/client`;
|
||||
};
|
||||
|
||||
// WebSocket connection
|
||||
// Shared WebSocket connection using the `share` option for better resource management
|
||||
const {
|
||||
sendMessage,
|
||||
lastMessage,
|
||||
|
|
@ -88,14 +95,19 @@ export function useAudioEvents(): UseAudioEventsReturn {
|
|||
shouldReconnect: () => true,
|
||||
reconnectAttempts: 10,
|
||||
reconnectInterval: 3000,
|
||||
share: true, // Share the WebSocket connection across multiple hooks
|
||||
onOpen: () => {
|
||||
console.log('[AudioEvents] WebSocket connected');
|
||||
subscriptionSent.current = false;
|
||||
// Reset global state on new connection
|
||||
globalSubscriptionState.isSubscribed = false;
|
||||
globalSubscriptionState.connectionId = Math.random().toString(36);
|
||||
},
|
||||
onClose: () => {
|
||||
console.log('[AudioEvents] WebSocket disconnected');
|
||||
subscriptionSent.current = false;
|
||||
setIsSubscribed(false);
|
||||
// Reset global state on disconnect
|
||||
globalSubscriptionState.isSubscribed = false;
|
||||
globalSubscriptionState.subscriberCount = 0;
|
||||
globalSubscriptionState.connectionId = null;
|
||||
},
|
||||
onError: (event) => {
|
||||
console.error('[AudioEvents] WebSocket error:', event);
|
||||
|
|
@ -104,18 +116,66 @@ export function useAudioEvents(): UseAudioEventsReturn {
|
|||
|
||||
// Subscribe to audio events
|
||||
const subscribe = useCallback(() => {
|
||||
if (readyState === ReadyState.OPEN && !subscriptionSent.current) {
|
||||
if (readyState === ReadyState.OPEN && !globalSubscriptionState.isSubscribed) {
|
||||
// Clear any pending subscription timeout
|
||||
if (subscriptionTimeoutRef.current) {
|
||||
clearTimeout(subscriptionTimeoutRef.current);
|
||||
subscriptionTimeoutRef.current = null;
|
||||
}
|
||||
|
||||
// Add a small delay to prevent rapid subscription attempts
|
||||
subscriptionTimeoutRef.current = setTimeout(() => {
|
||||
if (readyState === ReadyState.OPEN && !globalSubscriptionState.isSubscribed) {
|
||||
const subscribeMessage = {
|
||||
type: 'subscribe-audio-events',
|
||||
data: {}
|
||||
};
|
||||
|
||||
sendMessage(JSON.stringify(subscribeMessage));
|
||||
subscriptionSent.current = true;
|
||||
setIsSubscribed(true);
|
||||
globalSubscriptionState.isSubscribed = true;
|
||||
console.log('[AudioEvents] Subscribed to audio events');
|
||||
}
|
||||
}, [readyState, sendMessage]);
|
||||
}, 100); // 100ms delay to debounce subscription attempts
|
||||
}
|
||||
|
||||
// Track local subscription regardless of global state
|
||||
if (!isLocallySubscribed) {
|
||||
globalSubscriptionState.subscriberCount++;
|
||||
setIsLocallySubscribed(true);
|
||||
}
|
||||
}, [readyState, sendMessage, isLocallySubscribed]);
|
||||
|
||||
// Unsubscribe from audio events
|
||||
const unsubscribe = useCallback(() => {
|
||||
// Clear any pending subscription timeout
|
||||
if (subscriptionTimeoutRef.current) {
|
||||
clearTimeout(subscriptionTimeoutRef.current);
|
||||
subscriptionTimeoutRef.current = null;
|
||||
}
|
||||
|
||||
if (isLocallySubscribed) {
|
||||
globalSubscriptionState.subscriberCount--;
|
||||
setIsLocallySubscribed(false);
|
||||
|
||||
// Only send unsubscribe message if this is the last subscriber and connection is still open
|
||||
if (globalSubscriptionState.subscriberCount <= 0 &&
|
||||
readyState === ReadyState.OPEN &&
|
||||
globalSubscriptionState.isSubscribed) {
|
||||
|
||||
const unsubscribeMessage = {
|
||||
type: 'unsubscribe-audio-events',
|
||||
data: {}
|
||||
};
|
||||
|
||||
sendMessage(JSON.stringify(unsubscribeMessage));
|
||||
globalSubscriptionState.isSubscribed = false;
|
||||
globalSubscriptionState.subscriberCount = 0;
|
||||
console.log('[AudioEvents] Sent unsubscribe message to backend');
|
||||
}
|
||||
}
|
||||
|
||||
console.log('[AudioEvents] Component unsubscribed from audio events');
|
||||
}, [readyState, isLocallySubscribed, sendMessage]);
|
||||
|
||||
// Handle incoming messages
|
||||
useEffect(() => {
|
||||
|
|
@ -150,7 +210,7 @@ export function useAudioEvents(): UseAudioEventsReturn {
|
|||
|
||||
case 'microphone-metrics-update': {
|
||||
const micMetricsData = audioEvent.data as MicrophoneMetricsData;
|
||||
setMicrophoneMetrics(micMetricsData);
|
||||
setMicrophoneMetricsData(micMetricsData);
|
||||
break;
|
||||
}
|
||||
|
||||
|
|
@ -170,22 +230,42 @@ export function useAudioEvents(): UseAudioEventsReturn {
|
|||
|
||||
// Auto-subscribe when connected
|
||||
useEffect(() => {
|
||||
if (readyState === ReadyState.OPEN && !subscriptionSent.current) {
|
||||
if (readyState === ReadyState.OPEN) {
|
||||
subscribe();
|
||||
}
|
||||
}, [readyState, subscribe]);
|
||||
|
||||
// Unsubscribe from audio events (connection will be cleaned up automatically)
|
||||
const unsubscribe = useCallback(() => {
|
||||
setIsSubscribed(false);
|
||||
subscriptionSent.current = false;
|
||||
console.log('[AudioEvents] Unsubscribed from audio events');
|
||||
}, []);
|
||||
// Cleanup subscription on component unmount or connection change
|
||||
return () => {
|
||||
if (subscriptionTimeoutRef.current) {
|
||||
clearTimeout(subscriptionTimeoutRef.current);
|
||||
subscriptionTimeoutRef.current = null;
|
||||
}
|
||||
unsubscribe();
|
||||
};
|
||||
}, [readyState, subscribe, unsubscribe]);
|
||||
|
||||
// Reset local subscription state on disconnect
|
||||
useEffect(() => {
|
||||
if (readyState === ReadyState.CLOSED || readyState === ReadyState.CLOSING) {
|
||||
setIsLocallySubscribed(false);
|
||||
if (subscriptionTimeoutRef.current) {
|
||||
clearTimeout(subscriptionTimeoutRef.current);
|
||||
subscriptionTimeoutRef.current = null;
|
||||
}
|
||||
}
|
||||
}, [readyState]);
|
||||
|
||||
// Cleanup on component unmount
|
||||
useEffect(() => {
|
||||
return () => {
|
||||
unsubscribe();
|
||||
};
|
||||
}, [unsubscribe]);
|
||||
|
||||
return {
|
||||
// Connection state
|
||||
connectionState: readyState,
|
||||
isConnected: readyState === ReadyState.OPEN && isSubscribed,
|
||||
isConnected: readyState === ReadyState.OPEN && globalSubscriptionState.isSubscribed,
|
||||
|
||||
// Audio state
|
||||
audioMuted,
|
||||
|
|
@ -193,7 +273,7 @@ export function useAudioEvents(): UseAudioEventsReturn {
|
|||
|
||||
// Microphone state
|
||||
microphoneState,
|
||||
microphoneMetrics,
|
||||
microphoneMetrics: microphoneMetrics,
|
||||
|
||||
// Manual subscription control
|
||||
subscribe,
|
||||
|
|
|
|||
|
|
@ -5,20 +5,31 @@ interface AudioLevelHookResult {
|
|||
isAnalyzing: boolean;
|
||||
}
|
||||
|
||||
export const useAudioLevel = (stream: MediaStream | null): AudioLevelHookResult => {
|
||||
interface AudioLevelOptions {
|
||||
enabled?: boolean; // Allow external control of analysis
|
||||
updateInterval?: number; // Throttle updates (default: 100ms for 10fps instead of 60fps)
|
||||
}
|
||||
|
||||
export const useAudioLevel = (
|
||||
stream: MediaStream | null,
|
||||
options: AudioLevelOptions = {}
|
||||
): AudioLevelHookResult => {
|
||||
const { enabled = true, updateInterval = 100 } = options;
|
||||
|
||||
const [audioLevel, setAudioLevel] = useState(0);
|
||||
const [isAnalyzing, setIsAnalyzing] = useState(false);
|
||||
const audioContextRef = useRef<AudioContext | null>(null);
|
||||
const analyserRef = useRef<AnalyserNode | null>(null);
|
||||
const sourceRef = useRef<MediaStreamAudioSourceNode | null>(null);
|
||||
const animationFrameRef = useRef<number | null>(null);
|
||||
const intervalRef = useRef<number | null>(null);
|
||||
const lastUpdateTimeRef = useRef<number>(0);
|
||||
|
||||
useEffect(() => {
|
||||
if (!stream) {
|
||||
// Clean up when stream is null
|
||||
if (animationFrameRef.current) {
|
||||
cancelAnimationFrame(animationFrameRef.current);
|
||||
animationFrameRef.current = null;
|
||||
if (!stream || !enabled) {
|
||||
// Clean up when stream is null or disabled
|
||||
if (intervalRef.current !== null) {
|
||||
clearInterval(intervalRef.current);
|
||||
intervalRef.current = null;
|
||||
}
|
||||
if (sourceRef.current) {
|
||||
sourceRef.current.disconnect();
|
||||
|
|
@ -47,8 +58,8 @@ export const useAudioLevel = (stream: MediaStream | null): AudioLevelHookResult
|
|||
const analyser = audioContext.createAnalyser();
|
||||
const source = audioContext.createMediaStreamSource(stream);
|
||||
|
||||
// Configure analyser
|
||||
analyser.fftSize = 256;
|
||||
// Configure analyser - use smaller FFT for better performance
|
||||
analyser.fftSize = 128; // Reduced from 256 for better performance
|
||||
analyser.smoothingTimeConstant = 0.8;
|
||||
|
||||
// Connect nodes
|
||||
|
|
@ -64,24 +75,34 @@ export const useAudioLevel = (stream: MediaStream | null): AudioLevelHookResult
|
|||
const updateLevel = () => {
|
||||
if (!analyserRef.current) return;
|
||||
|
||||
const now = performance.now();
|
||||
|
||||
// Throttle updates to reduce CPU usage
|
||||
if (now - lastUpdateTimeRef.current < updateInterval) {
|
||||
return;
|
||||
}
|
||||
lastUpdateTimeRef.current = now;
|
||||
|
||||
analyserRef.current.getByteFrequencyData(dataArray);
|
||||
|
||||
// Calculate RMS (Root Mean Square) for more accurate level representation
|
||||
// Optimized RMS calculation - process only relevant frequency bands
|
||||
let sum = 0;
|
||||
for (const value of dataArray) {
|
||||
const relevantBins = Math.min(dataArray.length, 32); // Focus on lower frequencies for voice
|
||||
for (let i = 0; i < relevantBins; i++) {
|
||||
const value = dataArray[i];
|
||||
sum += value * value;
|
||||
}
|
||||
const rms = Math.sqrt(sum / dataArray.length);
|
||||
const rms = Math.sqrt(sum / relevantBins);
|
||||
|
||||
// Convert to percentage (0-100)
|
||||
const level = Math.min(100, (rms / 255) * 100);
|
||||
setAudioLevel(level);
|
||||
|
||||
animationFrameRef.current = requestAnimationFrame(updateLevel);
|
||||
// Convert to percentage (0-100) with better scaling
|
||||
const level = Math.min(100, Math.max(0, (rms / 180) * 100)); // Adjusted scaling for better sensitivity
|
||||
setAudioLevel(Math.round(level));
|
||||
};
|
||||
|
||||
setIsAnalyzing(true);
|
||||
updateLevel();
|
||||
|
||||
// Use setInterval instead of requestAnimationFrame for more predictable timing
|
||||
intervalRef.current = window.setInterval(updateLevel, updateInterval);
|
||||
|
||||
} catch (error) {
|
||||
console.error('Failed to create audio level analyzer:', error);
|
||||
|
|
@ -91,9 +112,9 @@ export const useAudioLevel = (stream: MediaStream | null): AudioLevelHookResult
|
|||
|
||||
// Cleanup function
|
||||
return () => {
|
||||
if (animationFrameRef.current) {
|
||||
cancelAnimationFrame(animationFrameRef.current);
|
||||
animationFrameRef.current = null;
|
||||
if (intervalRef.current !== null) {
|
||||
clearInterval(intervalRef.current);
|
||||
intervalRef.current = null;
|
||||
}
|
||||
if (sourceRef.current) {
|
||||
sourceRef.current.disconnect();
|
||||
|
|
@ -107,7 +128,7 @@ export const useAudioLevel = (stream: MediaStream | null): AudioLevelHookResult
|
|||
setIsAnalyzing(false);
|
||||
setAudioLevel(0);
|
||||
};
|
||||
}, [stream]);
|
||||
}, [stream, enabled, updateInterval]);
|
||||
|
||||
return { audioLevel, isAnalyzing };
|
||||
};
|
||||
|
|
@ -28,6 +28,33 @@ export function useMicrophone() {
|
|||
const [isStopping, setIsStopping] = useState(false);
|
||||
const [isToggling, setIsToggling] = useState(false);
|
||||
|
||||
// Add debouncing refs to prevent rapid operations
|
||||
const lastOperationRef = useRef<number>(0);
|
||||
const operationTimeoutRef = useRef<number | null>(null);
|
||||
const OPERATION_DEBOUNCE_MS = 1000; // 1 second debounce
|
||||
|
||||
// Debounced operation wrapper
|
||||
const debouncedOperation = useCallback((operation: () => Promise<void>, operationType: string) => {
|
||||
const now = Date.now();
|
||||
const timeSinceLastOp = now - lastOperationRef.current;
|
||||
|
||||
if (timeSinceLastOp < OPERATION_DEBOUNCE_MS) {
|
||||
console.log(`Debouncing ${operationType} operation - too soon (${timeSinceLastOp}ms since last)`);
|
||||
return;
|
||||
}
|
||||
|
||||
// Clear any pending operation
|
||||
if (operationTimeoutRef.current) {
|
||||
clearTimeout(operationTimeoutRef.current);
|
||||
operationTimeoutRef.current = null;
|
||||
}
|
||||
|
||||
lastOperationRef.current = now;
|
||||
operation().catch(error => {
|
||||
console.error(`Debounced ${operationType} operation failed:`, error);
|
||||
});
|
||||
}, []);
|
||||
|
||||
// Cleanup function to stop microphone stream
|
||||
const stopMicrophoneStream = useCallback(async () => {
|
||||
console.log("stopMicrophoneStream called - cleaning up stream");
|
||||
|
|
@ -830,6 +857,14 @@ export function useMicrophone() {
|
|||
|
||||
}, [microphoneSender, peerConnection]);
|
||||
|
||||
const startMicrophoneDebounced = useCallback((deviceId?: string) => {
|
||||
debouncedOperation(() => startMicrophone(deviceId).then(() => {}), "start");
|
||||
}, [startMicrophone, debouncedOperation]);
|
||||
|
||||
const stopMicrophoneDebounced = useCallback(() => {
|
||||
debouncedOperation(() => stopMicrophone().then(() => {}), "stop");
|
||||
}, [stopMicrophone, debouncedOperation]);
|
||||
|
||||
// Make debug functions available globally for console access
|
||||
useEffect(() => {
|
||||
(window as Window & {
|
||||
|
|
@ -912,10 +947,12 @@ export function useMicrophone() {
|
|||
startMicrophone,
|
||||
stopMicrophone,
|
||||
toggleMicrophoneMute,
|
||||
syncMicrophoneState,
|
||||
debugMicrophoneState,
|
||||
resetBackendMicrophoneState,
|
||||
// Loading states
|
||||
// Expose debounced variants for UI handlers
|
||||
startMicrophoneDebounced,
|
||||
stopMicrophoneDebounced,
|
||||
// Expose sync and loading flags for consumers that expect them
|
||||
syncMicrophoneState,
|
||||
isStarting,
|
||||
isStopping,
|
||||
isToggling,
|
||||
|
|
|
|||
56
web.go
56
web.go
|
|
@ -283,6 +283,30 @@ func setupRouter() *gin.Engine {
|
|||
return
|
||||
}
|
||||
|
||||
// Server-side cooldown to prevent rapid start/stop thrashing
|
||||
{
|
||||
cs := currentSession
|
||||
cs.micOpMu.Lock()
|
||||
now := time.Now()
|
||||
if cs.micCooldown == 0 {
|
||||
cs.micCooldown = 200 * time.Millisecond
|
||||
}
|
||||
since := now.Sub(cs.lastMicOp)
|
||||
if since < cs.micCooldown {
|
||||
remaining := cs.micCooldown - since
|
||||
running := cs.AudioInputManager.IsRunning() || audio.IsNonBlockingAudioInputRunning()
|
||||
cs.micOpMu.Unlock()
|
||||
c.JSON(200, gin.H{
|
||||
"status": "cooldown",
|
||||
"running": running,
|
||||
"cooldown_ms_remaining": remaining.Milliseconds(),
|
||||
})
|
||||
return
|
||||
}
|
||||
cs.lastMicOp = now
|
||||
cs.micOpMu.Unlock()
|
||||
}
|
||||
|
||||
// Check if already running before attempting to start
|
||||
if currentSession.AudioInputManager.IsRunning() || audio.IsNonBlockingAudioInputRunning() {
|
||||
c.JSON(200, gin.H{
|
||||
|
|
@ -332,6 +356,30 @@ func setupRouter() *gin.Engine {
|
|||
return
|
||||
}
|
||||
|
||||
// Server-side cooldown to prevent rapid start/stop thrashing
|
||||
{
|
||||
cs := currentSession
|
||||
cs.micOpMu.Lock()
|
||||
now := time.Now()
|
||||
if cs.micCooldown == 0 {
|
||||
cs.micCooldown = 200 * time.Millisecond
|
||||
}
|
||||
since := now.Sub(cs.lastMicOp)
|
||||
if since < cs.micCooldown {
|
||||
remaining := cs.micCooldown - since
|
||||
running := cs.AudioInputManager.IsRunning() || audio.IsNonBlockingAudioInputRunning()
|
||||
cs.micOpMu.Unlock()
|
||||
c.JSON(200, gin.H{
|
||||
"status": "cooldown",
|
||||
"running": running,
|
||||
"cooldown_ms_remaining": remaining.Milliseconds(),
|
||||
})
|
||||
return
|
||||
}
|
||||
cs.lastMicOp = now
|
||||
cs.micOpMu.Unlock()
|
||||
}
|
||||
|
||||
// Check if already stopped before attempting to stop
|
||||
if !currentSession.AudioInputManager.IsRunning() && !audio.IsNonBlockingAudioInputRunning() {
|
||||
c.JSON(200, gin.H{
|
||||
|
|
@ -343,8 +391,8 @@ func setupRouter() *gin.Engine {
|
|||
|
||||
currentSession.AudioInputManager.Stop()
|
||||
|
||||
// Also stop the non-blocking audio input specifically
|
||||
audio.StopNonBlockingAudioInput()
|
||||
// AudioInputManager.Stop() already coordinates a clean stop via StopNonBlockingAudioInput()
|
||||
// so we don't need to call it again here
|
||||
|
||||
// Broadcast microphone state change via WebSocket
|
||||
broadcaster := audio.GetAudioEventBroadcaster()
|
||||
|
|
@ -735,6 +783,10 @@ func handleWebRTCSignalWsMessages(
|
|||
l.Info().Msg("client subscribing to audio events")
|
||||
broadcaster := audio.GetAudioEventBroadcaster()
|
||||
broadcaster.Subscribe(connectionID, wsCon, runCtx, &l)
|
||||
} else if message.Type == "unsubscribe-audio-events" {
|
||||
l.Info().Msg("client unsubscribing from audio events")
|
||||
broadcaster := audio.GetAudioEventBroadcaster()
|
||||
broadcaster.Unsubscribe(connectionID)
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -7,6 +7,8 @@ import (
|
|||
"net"
|
||||
"runtime"
|
||||
"strings"
|
||||
"sync"
|
||||
"time"
|
||||
|
||||
"github.com/coder/websocket"
|
||||
"github.com/coder/websocket/wsjson"
|
||||
|
|
@ -27,6 +29,11 @@ type Session struct {
|
|||
DiskChannel *webrtc.DataChannel
|
||||
AudioInputManager *audio.AudioInputManager
|
||||
shouldUmountVirtualMedia bool
|
||||
|
||||
// Microphone operation cooldown to mitigate rapid start/stop races
|
||||
micOpMu sync.Mutex
|
||||
lastMicOp time.Time
|
||||
micCooldown time.Duration
|
||||
}
|
||||
|
||||
type SessionConfig struct {
|
||||
|
|
|
|||
Loading…
Reference in New Issue