kvm/internal/audio/cgo_audio.go

1258 lines
40 KiB
Go

//go:build cgo
package audio
import (
"errors"
"fmt"
"sync"
"sync/atomic"
"time"
"unsafe"
)
/*
#cgo CFLAGS: -I$HOME/.jetkvm/audio-libs/alsa-lib-$ALSA_VERSION/include -I$HOME/.jetkvm/audio-libs/opus-$OPUS_VERSION/include -I$HOME/.jetkvm/audio-libs/opus-$OPUS_VERSION/celt
#cgo LDFLAGS: -L$HOME/.jetkvm/audio-libs/alsa-lib-$ALSA_VERSION/src/.libs -lasound -L$HOME/.jetkvm/audio-libs/opus-$OPUS_VERSION/.libs -lopus -lm -ldl -static
#include <alsa/asoundlib.h>
#include <opus.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <errno.h>
#include <sys/mman.h>
// C state for ALSA/Opus with safety flags
static snd_pcm_t *pcm_handle = NULL;
static snd_pcm_t *pcm_playback_handle = NULL;
static OpusEncoder *encoder = NULL;
static OpusDecoder *decoder = NULL;
// Opus encoder settings - initialized from Go configuration
static int opus_bitrate = 96000; // Will be set from Config.CGOOpusBitrate
static int opus_complexity = 3; // Will be set from Config.CGOOpusComplexity
static int opus_vbr = 1; // Will be set from Config.CGOOpusVBR
static int opus_vbr_constraint = 1; // Will be set from Config.CGOOpusVBRConstraint
static int opus_signal_type = 3; // Will be set from Config.CGOOpusSignalType
static int opus_bandwidth = 1105; // OPUS_BANDWIDTH_WIDEBAND for compatibility (was 1101)
static int opus_dtx = 0; // Will be set from Config.CGOOpusDTX
static int opus_lsb_depth = 16; // LSB depth for improved bit allocation on constrained hardware
static int sample_rate = 48000; // Will be set from Config.CGOSampleRate
static int channels = 2; // Will be set from Config.CGOChannels
static int frame_size = 960; // Will be set from Config.CGOFrameSize
static int max_packet_size = 1500; // Will be set from Config.CGOMaxPacketSize
static int sleep_microseconds = 1000; // Will be set from Config.CGOUsleepMicroseconds
static int max_attempts_global = 5; // Will be set from Config.CGOMaxAttempts
static int max_backoff_us_global = 500000; // Will be set from Config.CGOMaxBackoffMicroseconds
// Hardware optimization flags for constrained environments
static int use_mmap_access = 0; // Disable MMAP for compatibility (was 1)
static int optimized_buffer_size = 0; // Disable optimized buffer sizing for stability (was 1)
// C function declarations (implementations are below)
int jetkvm_audio_init();
void jetkvm_audio_close();
int jetkvm_audio_read_encode(void *opus_buf);
int jetkvm_audio_decode_write(void *opus_buf, int opus_size);
int jetkvm_audio_playback_init();
void jetkvm_audio_playback_close();
// Function to update constants from Go configuration
void update_audio_constants(int bitrate, int complexity, int vbr, int vbr_constraint,
int signal_type, int bandwidth, int dtx, int lsb_depth, int sr, int ch,
int fs, int max_pkt, int sleep_us, int max_attempts, int max_backoff) {
opus_bitrate = bitrate;
opus_complexity = complexity;
opus_vbr = vbr;
opus_vbr_constraint = vbr_constraint;
opus_signal_type = signal_type;
opus_bandwidth = bandwidth;
opus_dtx = dtx;
opus_lsb_depth = lsb_depth;
sample_rate = sr;
channels = ch;
frame_size = fs;
max_packet_size = max_pkt;
sleep_microseconds = sleep_us;
max_attempts_global = max_attempts;
max_backoff_us_global = max_backoff;
}
// State tracking to prevent race conditions during rapid start/stop
static volatile int capture_initializing = 0;
static volatile int capture_initialized = 0;
static volatile int playback_initializing = 0;
static volatile int playback_initialized = 0;
// Function to dynamically update Opus encoder parameters
int update_opus_encoder_params(int bitrate, int complexity, int vbr, int vbr_constraint,
int signal_type, int bandwidth, int dtx) {
// This function works for both audio input and output encoder parameters
// Require either capture (output) or playback (input) initialization
if (!encoder || (!capture_initialized && !playback_initialized)) {
return -1; // Audio encoder not initialized
}
// Update the static variables
opus_bitrate = bitrate;
opus_complexity = complexity;
opus_vbr = vbr;
opus_vbr_constraint = vbr_constraint;
opus_signal_type = signal_type;
opus_bandwidth = bandwidth;
opus_dtx = dtx;
// Apply the new settings to the encoder
int result = 0;
result |= opus_encoder_ctl(encoder, OPUS_SET_BITRATE(opus_bitrate));
result |= opus_encoder_ctl(encoder, OPUS_SET_COMPLEXITY(opus_complexity));
result |= opus_encoder_ctl(encoder, OPUS_SET_VBR(opus_vbr));
result |= opus_encoder_ctl(encoder, OPUS_SET_VBR_CONSTRAINT(opus_vbr_constraint));
result |= opus_encoder_ctl(encoder, OPUS_SET_SIGNAL(opus_signal_type));
result |= opus_encoder_ctl(encoder, OPUS_SET_BANDWIDTH(opus_bandwidth));
result |= opus_encoder_ctl(encoder, OPUS_SET_DTX(opus_dtx));
return result; // 0 on success, non-zero on error
}
// Enhanced ALSA device opening with exponential backoff retry logic
static int safe_alsa_open(snd_pcm_t **handle, const char *device, snd_pcm_stream_t stream) {
int attempt = 0;
int err;
int backoff_us = sleep_microseconds; // Start with base sleep time
while (attempt < max_attempts_global) {
err = snd_pcm_open(handle, device, stream, SND_PCM_NONBLOCK);
if (err >= 0) {
// Switch to blocking mode after successful open
snd_pcm_nonblock(*handle, 0);
return 0;
}
attempt++;
if (attempt >= max_attempts_global) break;
// Enhanced error handling with specific retry strategies
if (err == -EBUSY || err == -EAGAIN) {
// Device busy or temporarily unavailable - retry with backoff
usleep(backoff_us);
backoff_us = (backoff_us * 2 < max_backoff_us_global) ? backoff_us * 2 : max_backoff_us_global;
} else if (err == -ENODEV || err == -ENOENT) {
// Device not found - longer wait as device might be initializing
usleep(backoff_us * 2);
backoff_us = (backoff_us * 2 < max_backoff_us_global) ? backoff_us * 2 : max_backoff_us_global;
} else if (err == -EPERM || err == -EACCES) {
// Permission denied - shorter wait, likely persistent issue
usleep(backoff_us / 2);
} else {
// Other errors - standard backoff
usleep(backoff_us);
backoff_us = (backoff_us * 2 < max_backoff_us_global) ? backoff_us * 2 : max_backoff_us_global;
}
}
return err;
}
// Optimized ALSA configuration with stack allocation and performance tuning
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;
// Use MMAP access for direct hardware memory access if enabled
if (use_mmap_access) {
err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_MMAP_INTERLEAVED);
if (err < 0) {
// Fallback to RW access if MMAP fails
err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
}
} else {
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) {
// Fallback to near rate if exact fails
unsigned int rate = sample_rate;
err = snd_pcm_hw_params_set_rate_near(handle, params, &rate, 0);
if (err < 0) return err;
}
// Optimize buffer sizes for constrained hardware
snd_pcm_uframes_t period_size = frame_size;
if (optimized_buffer_size) {
// Use smaller periods for lower latency on constrained hardware
period_size = frame_size / 2;
if (period_size < 64) period_size = 64; // Minimum safe period size
}
err = snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, 0);
if (err < 0) return err;
// Optimize buffer size based on hardware constraints
snd_pcm_uframes_t buffer_size;
if (optimized_buffer_size) {
// Use 2 periods for ultra-low latency on constrained hardware
buffer_size = period_size * 2;
} else {
// Standard 4 periods for good latency/stability balance
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 (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) {
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
capture_initializing = 0;
return -1;
}
// Initialize Opus encoder with optimized settings
int opus_err = 0;
encoder = opus_encoder_create(sample_rate, channels, OPUS_APPLICATION_AUDIO, &opus_err);
if (!encoder || opus_err != OPUS_OK) {
if (pcm_handle) { snd_pcm_close(pcm_handle); pcm_handle = NULL; }
capture_initializing = 0;
return -2;
}
// Apply optimized Opus encoder settings for constrained hardware
opus_encoder_ctl(encoder, OPUS_SET_BITRATE(opus_bitrate));
opus_encoder_ctl(encoder, OPUS_SET_COMPLEXITY(opus_complexity));
opus_encoder_ctl(encoder, OPUS_SET_VBR(opus_vbr));
opus_encoder_ctl(encoder, OPUS_SET_VBR_CONSTRAINT(opus_vbr_constraint));
opus_encoder_ctl(encoder, OPUS_SET_SIGNAL(opus_signal_type));
opus_encoder_ctl(encoder, OPUS_SET_BANDWIDTH(opus_bandwidth)); // WIDEBAND for compatibility
opus_encoder_ctl(encoder, OPUS_SET_DTX(opus_dtx));
// Set LSB depth for improved bit allocation on constrained hardware (disabled for compatibility)
// opus_encoder_ctl(encoder, OPUS_SET_LSB_DEPTH(opus_lsb_depth));
// Enable packet loss concealment for better resilience
opus_encoder_ctl(encoder, OPUS_SET_PACKET_LOSS_PERC(5));
// Set prediction disabled for lower latency
opus_encoder_ctl(encoder, OPUS_SET_PREDICTION_DISABLED(1));
capture_initialized = 1;
capture_initializing = 0;
return 0;
}
// jetkvm_audio_read_encode captures audio from ALSA, encodes with Opus, and handles errors.
// Implements robust error recovery for buffer underruns and device suspension.
// Returns: >0 (bytes written), -1 (init error), -2 (unrecoverable error)
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;
int recovery_attempts = 0;
const int max_recovery_attempts = 3;
// Safety checks
if (!capture_initialized || !pcm_handle || !encoder || !opus_buf) {
return -1;
}
retry_read:
;
int pcm_rc = snd_pcm_readi(pcm_handle, pcm_buffer, frame_size);
// Handle ALSA errors with robust recovery strategies
if (pcm_rc < 0) {
if (pcm_rc == -EPIPE) {
// Buffer underrun - implement progressive recovery
recovery_attempts++;
if (recovery_attempts > max_recovery_attempts) {
return -1; // Give up after max attempts
}
// Try to recover with prepare
err = snd_pcm_prepare(pcm_handle);
if (err < 0) {
// If prepare fails, try drop and prepare
snd_pcm_drop(pcm_handle);
err = snd_pcm_prepare(pcm_handle);
if (err < 0) return -1;
}
// Wait before retry to allow device to stabilize
usleep(sleep_microseconds * recovery_attempts);
goto retry_read;
} else if (pcm_rc == -EAGAIN) {
// No data available - return 0 to indicate no frame
return 0;
} else if (pcm_rc == -ESTRPIPE) {
// Device suspended, implement robust resume logic
recovery_attempts++;
if (recovery_attempts > max_recovery_attempts) {
return -1;
}
// Try to resume with timeout
int resume_attempts = 0;
while ((err = snd_pcm_resume(pcm_handle)) == -EAGAIN && resume_attempts < 10) {
usleep(sleep_microseconds);
resume_attempts++;
}
if (err < 0) {
// Resume failed, try prepare as fallback
err = snd_pcm_prepare(pcm_handle);
if (err < 0) return -1;
}
// Wait before retry to allow device to stabilize
usleep(sleep_microseconds * recovery_attempts);
return 0; // Skip this frame but don't fail
} else if (pcm_rc == -ENODEV) {
// Device disconnected - critical error
return -1;
} else if (pcm_rc == -EIO) {
// I/O error - try recovery once
recovery_attempts++;
if (recovery_attempts <= max_recovery_attempts) {
snd_pcm_drop(pcm_handle);
err = snd_pcm_prepare(pcm_handle);
if (err >= 0) {
usleep(sleep_microseconds);
goto retry_read;
}
}
return -1;
} else {
// Other errors - limited retry for transient issues
recovery_attempts++;
if (recovery_attempts <= 1 && (pcm_rc == -EINTR || pcm_rc == -EBUSY)) {
usleep(sleep_microseconds / 2);
goto retry_read;
}
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;
}
// 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 (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) {
// Fallback to default device
err = safe_alsa_open(&pcm_playback_handle, "default", SND_PCM_STREAM_PLAYBACK);
if (err < 0) {
playback_initializing = 0;
return -1;
}
}
// Configure the device
err = configure_alsa_device(pcm_playback_handle, "playback");
if (err < 0) {
snd_pcm_close(pcm_playback_handle);
pcm_playback_handle = NULL;
playback_initializing = 0;
return -1;
}
// Initialize Opus decoder
int opus_err = 0;
decoder = opus_decoder_create(sample_rate, channels, &opus_err);
if (!decoder || opus_err != OPUS_OK) {
snd_pcm_close(pcm_playback_handle);
pcm_playback_handle = NULL;
playback_initializing = 0;
return -2;
}
playback_initialized = 1;
playback_initializing = 0;
return 0;
}
// jetkvm_audio_decode_write decodes Opus data and writes PCM to ALSA playback device
// with error recovery and packet loss concealment
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;
int recovery_attempts = 0;
const int max_recovery_attempts = 3;
// 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 with error handling
int pcm_frames = opus_decode(decoder, in, opus_size, pcm_buffer, frame_size, 0);
if (pcm_frames < 0) {
// Try packet loss concealment on decode error
pcm_frames = opus_decode(decoder, NULL, 0, pcm_buffer, frame_size, 0);
if (pcm_frames < 0) return -1;
}
retry_write:
;
// Write PCM to playback device with robust recovery
int pcm_rc = snd_pcm_writei(pcm_playback_handle, pcm_buffer, pcm_frames);
if (pcm_rc < 0) {
if (pcm_rc == -EPIPE) {
// Buffer underrun - implement progressive recovery
recovery_attempts++;
if (recovery_attempts > max_recovery_attempts) {
return -2;
}
// Try to recover with prepare
err = snd_pcm_prepare(pcm_playback_handle);
if (err < 0) {
// If prepare fails, try drop and prepare
snd_pcm_drop(pcm_playback_handle);
err = snd_pcm_prepare(pcm_playback_handle);
if (err < 0) return -2;
}
// Wait before retry to allow device to stabilize
usleep(sleep_microseconds * recovery_attempts);
goto retry_write;
} else if (pcm_rc == -ESTRPIPE) {
// Device suspended, implement robust resume logic
recovery_attempts++;
if (recovery_attempts > max_recovery_attempts) {
return -2;
}
// Try to resume with timeout
int resume_attempts = 0;
while ((err = snd_pcm_resume(pcm_playback_handle)) == -EAGAIN && resume_attempts < 10) {
usleep(sleep_microseconds);
resume_attempts++;
}
if (err < 0) {
// Resume failed, try prepare as fallback
err = snd_pcm_prepare(pcm_playback_handle);
if (err < 0) return -2;
}
// Wait before retry to allow device to stabilize
usleep(sleep_microseconds * recovery_attempts);
return 0; // Skip this frame but don't fail
} else if (pcm_rc == -ENODEV) {
// Device disconnected - critical error
return -2;
} else if (pcm_rc == -EIO) {
// I/O error - try recovery once
recovery_attempts++;
if (recovery_attempts <= max_recovery_attempts) {
snd_pcm_drop(pcm_playback_handle);
err = snd_pcm_prepare(pcm_playback_handle);
if (err >= 0) {
usleep(sleep_microseconds);
goto retry_write;
}
}
return -2;
} else if (pcm_rc == -EAGAIN) {
// Device not ready - brief wait and retry
recovery_attempts++;
if (recovery_attempts <= max_recovery_attempts) {
usleep(sleep_microseconds / 4);
goto retry_write;
}
return -2;
} else {
// Other errors - limited retry for transient issues
recovery_attempts++;
if (recovery_attempts <= 1 && (pcm_rc == -EINTR || pcm_rc == -EBUSY)) {
usleep(sleep_microseconds / 2);
goto retry_write;
}
return -2;
}
}
return pcm_frames;
}
// Safe playback cleanup with double-close protection
void jetkvm_audio_playback_close() {
// Wait for any ongoing operations to complete
while (playback_initializing) {
usleep(sleep_microseconds); // Use centralized constant
}
// 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 (pcm_playback_handle) {
snd_pcm_drain(pcm_playback_handle);
snd_pcm_close(pcm_playback_handle);
pcm_playback_handle = NULL;
}
}
// Safe capture cleanup
void jetkvm_audio_close() {
// Wait for any ongoing operations to complete
while (capture_initializing) {
usleep(sleep_microseconds); // Use centralized constant
}
capture_initialized = 0;
if (encoder) {
opus_encoder_destroy(encoder);
encoder = NULL;
}
if (pcm_handle) {
snd_pcm_drop(pcm_handle); // Drop pending samples
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
}
// Also clean up playback
jetkvm_audio_playback_close();
}
*/
import "C"
// Optimized Go wrappers with reduced overhead
var (
// Base error types for wrapping with context
errAudioInitFailed = errors.New("failed to init ALSA/Opus")
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")
errInvalidBufferPtr = errors.New("invalid buffer pointer")
)
// Error creation functions with enhanced context
func newBufferTooSmallError(actual, required int) error {
baseErr := fmt.Errorf("buffer too small: got %d bytes, need at least %d bytes", actual, required)
return WrapWithMetadata(baseErr, "cgo_audio", "buffer_validation", map[string]interface{}{
"actual_size": actual,
"required_size": required,
"error_type": "buffer_undersize",
})
}
func newBufferTooLargeError(actual, max int) error {
baseErr := fmt.Errorf("buffer too large: got %d bytes, maximum allowed %d bytes", actual, max)
return WrapWithMetadata(baseErr, "cgo_audio", "buffer_validation", map[string]interface{}{
"actual_size": actual,
"max_size": max,
"error_type": "buffer_oversize",
})
}
func newAudioInitError(cErrorCode int) error {
baseErr := fmt.Errorf("%w: C error code %d", errAudioInitFailed, cErrorCode)
return WrapWithMetadata(baseErr, "cgo_audio", "initialization", map[string]interface{}{
"c_error_code": cErrorCode,
"error_type": "init_failure",
"severity": "critical",
})
}
func newAudioPlaybackInitError(cErrorCode int) error {
baseErr := fmt.Errorf("%w: C error code %d", errAudioPlaybackInit, cErrorCode)
return WrapWithMetadata(baseErr, "cgo_audio", "playback_init", map[string]interface{}{
"c_error_code": cErrorCode,
"error_type": "playback_init_failure",
"severity": "high",
})
}
func newAudioReadEncodeError(cErrorCode int) error {
baseErr := fmt.Errorf("%w: C error code %d", errAudioReadEncode, cErrorCode)
return WrapWithMetadata(baseErr, "cgo_audio", "read_encode", map[string]interface{}{
"c_error_code": cErrorCode,
"error_type": "read_encode_failure",
"severity": "medium",
})
}
func newAudioDecodeWriteError(cErrorCode int) error {
baseErr := fmt.Errorf("%w: C error code %d", errAudioDecodeWrite, cErrorCode)
return WrapWithMetadata(baseErr, "cgo_audio", "decode_write", map[string]interface{}{
"c_error_code": cErrorCode,
"error_type": "decode_write_failure",
"severity": "medium",
})
}
func cgoAudioInit() error {
// Get cached config and ensure it's updated
cache := GetCachedConfig()
cache.Update()
// Update C constants from cached config (atomic access, no locks)
C.update_audio_constants(
C.int(cache.opusBitrate.Load()),
C.int(cache.opusComplexity.Load()),
C.int(cache.opusVBR.Load()),
C.int(cache.opusVBRConstraint.Load()),
C.int(cache.opusSignalType.Load()),
C.int(cache.opusBandwidth.Load()),
C.int(cache.opusDTX.Load()),
C.int(16), // LSB depth for improved bit allocation
C.int(cache.sampleRate.Load()),
C.int(cache.channels.Load()),
C.int(cache.frameSize.Load()),
C.int(cache.maxPacketSize.Load()),
C.int(Config.CGOUsleepMicroseconds),
C.int(Config.CGOMaxAttempts),
C.int(Config.CGOMaxBackoffMicroseconds),
)
result := C.jetkvm_audio_init()
if result != 0 {
return newAudioInitError(int(result))
}
return nil
}
func cgoAudioClose() {
C.jetkvm_audio_close()
}
// AudioConfigCache provides a comprehensive caching system for audio configuration
type AudioConfigCache struct {
// Atomic int64 fields MUST be first for ARM32 alignment (8-byte alignment required)
minFrameDuration atomic.Int64 // Store as nanoseconds
maxFrameDuration atomic.Int64 // Store as nanoseconds
maxLatency atomic.Int64 // Store as nanoseconds
minMetricsUpdateInterval atomic.Int64 // Store as nanoseconds
maxMetricsUpdateInterval atomic.Int64 // Store as nanoseconds
restartWindow atomic.Int64 // Store as nanoseconds
restartDelay atomic.Int64 // Store as nanoseconds
maxRestartDelay atomic.Int64 // Store as nanoseconds
// Atomic int32 fields for lock-free access to frequently used values
minReadEncodeBuffer atomic.Int32
maxDecodeWriteBuffer atomic.Int32
maxPacketSize atomic.Int32
maxPCMBufferSize atomic.Int32
opusBitrate atomic.Int32
opusComplexity atomic.Int32
opusVBR atomic.Int32
opusVBRConstraint atomic.Int32
opusSignalType atomic.Int32
opusBandwidth atomic.Int32
opusDTX atomic.Int32
sampleRate atomic.Int32
channels atomic.Int32
frameSize atomic.Int32
// Additional cached values for validation functions
maxAudioFrameSize atomic.Int32
maxChannels atomic.Int32
minOpusBitrate atomic.Int32
maxOpusBitrate atomic.Int32
// Socket and buffer configuration values
socketMaxBuffer atomic.Int32
socketMinBuffer atomic.Int32
inputProcessingTimeoutMS atomic.Int32
maxRestartAttempts atomic.Int32
// Batch processing related values
BatchProcessingTimeout time.Duration
BatchProcessorFramesPerBatch int
BatchProcessorTimeout time.Duration
BatchProcessingDelay time.Duration
MinBatchSizeForThreadPinning int
BatchProcessorMaxQueueSize int
BatchProcessorAdaptiveThreshold float64
BatchProcessorThreadPinningThreshold int
// Mutex for updating the cache
mutex sync.RWMutex
lastUpdate time.Time
cacheExpiry time.Duration
initialized atomic.Bool
// Pre-allocated errors to avoid allocations in hot path
bufferTooSmallReadEncode error
bufferTooLargeDecodeWrite error
}
// Global audio config cache instance
var globalAudioConfigCache = &AudioConfigCache{
cacheExpiry: 30 * time.Second, // Increased from 10s to 30s to further reduce cache updates
}
// GetCachedConfig returns the global audio config cache instance
func GetCachedConfig() *AudioConfigCache {
return globalAudioConfigCache
}
// Update refreshes the cached config values if needed
func (c *AudioConfigCache) Update() {
// Fast path: if cache is initialized and not expired, return immediately
if c.initialized.Load() {
c.mutex.RLock()
cacheExpired := time.Since(c.lastUpdate) > c.cacheExpiry
c.mutex.RUnlock()
if !cacheExpired {
return
}
}
// Slow path: update cache
c.mutex.Lock()
defer c.mutex.Unlock()
// Double-check after acquiring lock
if !c.initialized.Load() || time.Since(c.lastUpdate) > c.cacheExpiry {
// Update atomic values for lock-free access - CGO values
c.minReadEncodeBuffer.Store(int32(Config.MinReadEncodeBuffer))
c.maxDecodeWriteBuffer.Store(int32(Config.MaxDecodeWriteBuffer))
c.maxPacketSize.Store(int32(Config.CGOMaxPacketSize))
c.maxPCMBufferSize.Store(int32(Config.MaxPCMBufferSize))
c.opusBitrate.Store(int32(Config.CGOOpusBitrate))
c.opusComplexity.Store(int32(Config.CGOOpusComplexity))
c.opusVBR.Store(int32(Config.CGOOpusVBR))
c.opusVBRConstraint.Store(int32(Config.CGOOpusVBRConstraint))
c.opusSignalType.Store(int32(Config.CGOOpusSignalType))
c.opusBandwidth.Store(int32(Config.CGOOpusBandwidth))
c.opusDTX.Store(int32(Config.CGOOpusDTX))
c.sampleRate.Store(int32(Config.CGOSampleRate))
c.channels.Store(int32(Config.CGOChannels))
c.frameSize.Store(int32(Config.CGOFrameSize))
// Update additional validation values
c.maxAudioFrameSize.Store(int32(Config.MaxAudioFrameSize))
c.maxChannels.Store(int32(Config.MaxChannels))
c.minFrameDuration.Store(int64(Config.MinFrameDuration))
c.maxFrameDuration.Store(int64(Config.MaxFrameDuration))
c.minOpusBitrate.Store(int32(Config.MinOpusBitrate))
c.maxOpusBitrate.Store(int32(Config.MaxOpusBitrate))
// Update batch processing related values
c.BatchProcessingTimeout = 100 * time.Millisecond // Fixed timeout for batch processing
c.BatchProcessorFramesPerBatch = Config.BatchProcessorFramesPerBatch
c.BatchProcessorTimeout = Config.BatchProcessorTimeout
c.BatchProcessingDelay = Config.BatchProcessingDelay
c.MinBatchSizeForThreadPinning = Config.MinBatchSizeForThreadPinning
c.BatchProcessorMaxQueueSize = Config.BatchProcessorMaxQueueSize
c.BatchProcessorAdaptiveThreshold = Config.BatchProcessorAdaptiveThreshold
c.BatchProcessorThreadPinningThreshold = Config.BatchProcessorThreadPinningThreshold
// Pre-allocate common errors
c.bufferTooSmallReadEncode = newBufferTooSmallError(0, Config.MinReadEncodeBuffer)
c.bufferTooLargeDecodeWrite = newBufferTooLargeError(Config.MaxDecodeWriteBuffer+1, Config.MaxDecodeWriteBuffer)
c.lastUpdate = time.Now()
c.initialized.Store(true)
// Update the global validation cache as well
if cachedMaxFrameSize != 0 {
cachedMaxFrameSize = Config.MaxAudioFrameSize
}
}
}
// GetMinReadEncodeBuffer returns the cached MinReadEncodeBuffer value
func (c *AudioConfigCache) GetMinReadEncodeBuffer() int {
return int(c.minReadEncodeBuffer.Load())
}
// GetMaxDecodeWriteBuffer returns the cached MaxDecodeWriteBuffer value
func (c *AudioConfigCache) GetMaxDecodeWriteBuffer() int {
return int(c.maxDecodeWriteBuffer.Load())
}
// GetMaxPacketSize returns the cached MaxPacketSize value
func (c *AudioConfigCache) GetMaxPacketSize() int {
return int(c.maxPacketSize.Load())
}
// GetMaxPCMBufferSize returns the cached MaxPCMBufferSize value
func (c *AudioConfigCache) GetMaxPCMBufferSize() int {
return int(c.maxPCMBufferSize.Load())
}
// GetBufferTooSmallError returns the pre-allocated buffer too small error
func (c *AudioConfigCache) GetBufferTooSmallError() error {
return c.bufferTooSmallReadEncode
}
// GetBufferTooLargeError returns the pre-allocated buffer too large error
func (c *AudioConfigCache) GetBufferTooLargeError() error {
return c.bufferTooLargeDecodeWrite
}
// Removed duplicate config caching system - using AudioConfigCache instead
// updateCacheIfNeeded updates cache only if expired to avoid overhead
func updateCacheIfNeeded(cache *AudioConfigCache) {
if cache.initialized.Load() {
cache.mutex.RLock()
cacheExpired := time.Since(cache.lastUpdate) > cache.cacheExpiry
cache.mutex.RUnlock()
if cacheExpired {
cache.Update()
}
} else {
cache.Update()
}
}
func cgoAudioReadEncode(buf []byte) (int, error) {
// Minimal buffer validation - assume caller provides correct size
if len(buf) == 0 {
return 0, errEmptyBuffer
}
// Direct CGO call - hotpath optimization
n := C.jetkvm_audio_read_encode(unsafe.Pointer(&buf[0]))
// Fast path for success
if n > 0 {
return int(n), nil
}
// Error handling with static errors
if n < 0 {
if n == -1 {
return 0, errAudioInitFailed
}
return 0, errAudioReadEncode
}
return 0, nil
}
// Audio playback functions
func cgoAudioPlaybackInit() error {
// Get cached config and ensure it's updated
cache := GetCachedConfig()
cache.Update()
// No need to update C constants here as they're already set in cgoAudioInit
ret := C.jetkvm_audio_playback_init()
if ret != 0 {
return newAudioPlaybackInitError(int(ret))
}
return nil
}
func cgoAudioPlaybackClose() {
C.jetkvm_audio_playback_close()
}
func cgoAudioDecodeWrite(buf []byte) (int, error) {
// Minimal validation - assume caller provides correct size
if len(buf) == 0 {
return 0, errEmptyBuffer
}
// Direct CGO call - hotpath optimization
n := int(C.jetkvm_audio_decode_write(unsafe.Pointer(&buf[0]), C.int(len(buf))))
// Fast path for success
if n >= 0 {
return n, nil
}
// Error handling with static errors
if n == -1 {
return 0, errAudioInitFailed
}
return 0, errAudioDecodeWrite
}
// updateOpusEncoderParams dynamically updates OPUS encoder parameters
func updateOpusEncoderParams(bitrate, complexity, vbr, vbrConstraint, signalType, bandwidth, dtx int) error {
result := C.update_opus_encoder_params(
C.int(bitrate),
C.int(complexity),
C.int(vbr),
C.int(vbrConstraint),
C.int(signalType),
C.int(bandwidth),
C.int(dtx),
)
if result != 0 {
return fmt.Errorf("failed to update OPUS encoder parameters: C error code %d", result)
}
return nil
}
// Buffer pool for reusing buffers in CGO functions
var (
// Using SizedBufferPool for better memory management
// Track buffer pool usage
cgoBufferPoolGets atomic.Int64
cgoBufferPoolPuts atomic.Int64
// Batch processing statistics - only enabled in debug builds
batchProcessingCount atomic.Int64
batchFrameCount atomic.Int64
batchProcessingTime atomic.Int64
// Batch time tracking removed
)
// GetBufferFromPool gets a buffer from the pool with at least the specified capacity
func GetBufferFromPool(minCapacity int) []byte {
cgoBufferPoolGets.Add(1)
return GetOptimalBuffer(minCapacity)
}
// ReturnBufferToPool returns a buffer to the pool
func ReturnBufferToPool(buf []byte) {
cgoBufferPoolPuts.Add(1)
ReturnOptimalBuffer(buf)
}
// ReadEncodeWithPooledBuffer reads audio data and encodes it using a buffer from the pool
func ReadEncodeWithPooledBuffer() ([]byte, int, error) {
cache := GetCachedConfig()
updateCacheIfNeeded(cache)
bufferSize := cache.GetMinReadEncodeBuffer()
if bufferSize == 0 {
bufferSize = 1500
}
buf := GetBufferFromPool(bufferSize)
n, err := cgoAudioReadEncode(buf)
if err != nil {
ReturnBufferToPool(buf)
return nil, 0, err
}
return buf[:n], n, nil
}
// DecodeWriteWithPooledBuffer decodes and writes audio data using a pooled buffer
func DecodeWriteWithPooledBuffer(data []byte) (int, error) {
if len(data) == 0 {
return 0, errEmptyBuffer
}
cache := GetCachedConfig()
updateCacheIfNeeded(cache)
maxPacketSize := cache.GetMaxPacketSize()
if len(data) > maxPacketSize {
return 0, newBufferTooLargeError(len(data), maxPacketSize)
}
pcmBuffer := GetBufferFromPool(cache.GetMaxPCMBufferSize())
defer ReturnBufferToPool(pcmBuffer)
return CGOAudioDecodeWrite(data, pcmBuffer)
}
// BatchReadEncode reads and encodes multiple audio frames in a single batch
// with optimized zero-copy frame management and batch reference counting
func BatchReadEncode(batchSize int) ([][]byte, error) {
// Simple batch processing without complex overhead
frames := make([][]byte, 0, batchSize)
frameSize := 4096 // Fixed frame size for performance
for i := 0; i < batchSize; i++ {
buf := make([]byte, frameSize)
n, err := cgoAudioReadEncode(buf)
if err != nil {
if i > 0 {
return frames, nil // Return partial batch
}
return nil, err
}
if n > 0 {
frames = append(frames, buf[:n])
}
}
return frames, nil
}
// BatchDecodeWrite decodes and writes multiple audio frames in a single batch
// This reduces CGO call overhead by processing multiple frames at once
// with optimized zero-copy frame management and batch reference counting
func BatchDecodeWrite(frames [][]byte) error {
// Validate input
if len(frames) == 0 {
return nil
}
// Convert to zero-copy frames for optimized processing
zeroCopyFrames := make([]*ZeroCopyAudioFrame, 0, len(frames))
for _, frameData := range frames {
if len(frameData) > 0 {
frame := GetZeroCopyFrame()
frame.SetDataDirect(frameData) // Direct assignment without copy
zeroCopyFrames = append(zeroCopyFrames, frame)
}
}
// Use batch reference counting for efficient management
if len(zeroCopyFrames) > 0 {
// Batch AddRef all frames at once
err := BatchAddRefFrames(zeroCopyFrames)
if err != nil {
return err
}
// Ensure cleanup with batch release
defer func() {
if _, err := BatchReleaseFrames(zeroCopyFrames); err != nil {
// Log release error but don't fail the operation
_ = err
}
}()
}
// Get cached config
cache := GetCachedConfig()
// Only update cache if expired - avoid unnecessary overhead
// Use proper locking to avoid race condition
if cache.initialized.Load() {
cache.mutex.RLock()
cacheExpired := time.Since(cache.lastUpdate) > cache.cacheExpiry
cache.mutex.RUnlock()
if cacheExpired {
cache.Update()
}
} else {
cache.Update()
}
// Track batch processing statistics - only if enabled
var startTime time.Time
// Batch time tracking removed
trackTime := false
if trackTime {
startTime = time.Now()
}
batchProcessingCount.Add(1)
// Get a PCM buffer from the pool for optimized decode-write
pcmBuffer := GetBufferFromPool(cache.GetMaxPCMBufferSize())
defer ReturnBufferToPool(pcmBuffer)
// Process each zero-copy frame with optimized batch processing
frameCount := 0
for _, zcFrame := range zeroCopyFrames {
// Get frame data from zero-copy frame
frameData := zcFrame.Data()[:zcFrame.Length()]
if len(frameData) == 0 {
continue
}
// Process this frame using optimized implementation
_, err := CGOAudioDecodeWrite(frameData, pcmBuffer)
if err != nil {
// Update statistics before returning error
batchFrameCount.Add(int64(frameCount))
if trackTime {
batchProcessingTime.Add(time.Since(startTime).Microseconds())
}
return err
}
frameCount++
}
// Update statistics
batchFrameCount.Add(int64(frameCount))
if trackTime {
batchProcessingTime.Add(time.Since(startTime).Microseconds())
}
return nil
}
// GetBatchProcessingStats returns statistics about batch processing
func GetBatchProcessingStats() (count, frames, avgTimeUs int64) {
count = batchProcessingCount.Load()
frames = batchFrameCount.Load()
totalTime := batchProcessingTime.Load()
// Calculate average time per batch
if count > 0 {
avgTimeUs = totalTime / count
}
return count, frames, avgTimeUs
}
// cgoAudioDecodeWriteWithBuffers decodes opus data and writes to PCM buffer
// This implementation uses separate buffers for opus data and PCM output
func cgoAudioDecodeWriteWithBuffers(opusData []byte, pcmBuffer []byte) (int, error) {
// Validate input
if len(opusData) == 0 {
return 0, errEmptyBuffer
}
if len(pcmBuffer) == 0 {
return 0, errEmptyBuffer
}
// Get cached config
cache := GetCachedConfig()
// Only update cache if expired - avoid unnecessary overhead
// Use proper locking to avoid race condition
if cache.initialized.Load() {
cache.mutex.RLock()
cacheExpired := time.Since(cache.lastUpdate) > cache.cacheExpiry
cache.mutex.RUnlock()
if cacheExpired {
cache.Update()
}
} else {
cache.Update()
}
// Ensure data doesn't exceed max packet size
maxPacketSize := cache.GetMaxPacketSize()
if len(opusData) > maxPacketSize {
return 0, newBufferTooLargeError(len(opusData), maxPacketSize)
}
// Direct CGO call with minimal overhead - unsafe.Pointer(&slice[0]) is never nil for non-empty slices
n := int(C.jetkvm_audio_decode_write(unsafe.Pointer(&opusData[0]), C.int(len(opusData))))
// Fast path for success case
if n >= 0 {
return n, nil
}
// Handle error cases with static error codes to reduce allocations
switch n {
case -1:
return 0, errAudioInitFailed
case -2:
return 0, errAudioDecodeWrite
default:
return 0, newAudioDecodeWriteError(n)
}
}
// Optimized CGO function aliases - use direct function calls to reduce overhead
// These are now direct function aliases instead of variable assignments
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 CGOAudioDecodeWriteLegacy(buf []byte) (int, error) { return cgoAudioDecodeWrite(buf) }
func CGOAudioDecodeWrite(opusData []byte, pcmBuffer []byte) (int, error) {
return cgoAudioDecodeWriteWithBuffers(opusData, pcmBuffer)
}
func CGOUpdateOpusEncoderParams(bitrate, complexity, vbr, vbrConstraint, signalType, bandwidth, dtx int) error {
return updateOpusEncoderParams(bitrate, complexity, vbr, vbrConstraint, signalType, bandwidth, dtx)
}