mirror of https://github.com/jetkvm/kvm.git
1193 lines
42 KiB
C
1193 lines
42 KiB
C
/*
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* JetKVM Audio Processing Module
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*
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* This module handles bidirectional audio processing for JetKVM:
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* - Audio INPUT: Client microphone → Device speakers (decode Opus → ALSA playback)
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* - Audio OUTPUT: TC358743 HDMI audio → Client speakers (ALSA capture → encode Opus)
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*/
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#include <alsa/asoundlib.h>
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#include <opus.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <unistd.h>
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#include <errno.h>
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// ARM NEON SIMD support (always available on JetKVM's ARM Cortex-A7)
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#include <arm_neon.h>
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#define SIMD_ALIGN __attribute__((aligned(16)))
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#define SIMD_PREFETCH(addr, rw, locality) __builtin_prefetch(addr, rw, locality)
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static int trace_logging_enabled = 0;
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static int simd_initialized = 0;
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static void simd_init_once(void) {
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if (simd_initialized) return;
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simd_initialized = 1;
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}
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// ============================================================================
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// GLOBAL STATE VARIABLES
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// ============================================================================
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// ALSA device handles
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static snd_pcm_t *pcm_capture_handle = NULL; // TC358743 HDMI audio capture (OUTPUT path)
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static snd_pcm_t *pcm_playback_handle = NULL; // Device speakers (INPUT path)
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// Opus codec instances
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static OpusEncoder *encoder = NULL; // For OUTPUT path (TC358743 HDMI → client)
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static OpusDecoder *decoder = NULL; // For INPUT path (client → device speakers)
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// Audio format configuration
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static int sample_rate = 48000; // Sample rate in Hz
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static int channels = 2; // Number of audio channels (stereo)
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static int frame_size = 960; // Frames per Opus packet
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// Opus encoder configuration - Optimized for S16_LE @ 48kHz with MINIMAL CPU usage
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static int opus_bitrate = 96000; // Bitrate: 96 kbps (optimal for stereo @ 48kHz)
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static int opus_complexity = 1; // Complexity: 1 (minimal CPU, ~0.5% on RV1106)
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static int opus_vbr = 1; // VBR: enabled for efficient encoding
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static int opus_vbr_constraint = 1; // Constrained VBR: predictable bitrate
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static int opus_signal_type = -1000; // Signal: OPUS_AUTO (automatic voice/music detection)
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static int opus_bandwidth = 1103; // Bandwidth: WIDEBAND (1103 = native 48kHz, no resampling)
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static int opus_dtx = 0; // DTX: disabled (continuous audio stream)
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static int opus_lsb_depth = 16; // LSB depth: 16-bit matches S16_LE input
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// Network and buffer configuration
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static int max_packet_size = 1500; // Maximum Opus packet size
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// Error handling and retry configuration
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static int sleep_microseconds = 1000; // Base sleep time for retries
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static int max_attempts_global = 5; // Maximum retry attempts
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static int max_backoff_us_global = 500000; // Maximum backoff time
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// Performance optimization flags
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static const int optimized_buffer_size = 1; // Use optimized buffer sizing
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// ============================================================================
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// FUNCTION DECLARATIONS
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// ============================================================================
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// Audio OUTPUT path functions (TC358743 HDMI audio → client speakers)
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int jetkvm_audio_capture_init(); // Initialize TC358743 capture and Opus encoder
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void jetkvm_audio_capture_close(); // Cleanup capture resources
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int jetkvm_audio_read_encode(void *opus_buf); // Read PCM from TC358743, encode to Opus
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// Audio INPUT path functions (client microphone → device speakers)
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int jetkvm_audio_playback_init(); // Initialize playback device and Opus decoder
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void jetkvm_audio_playback_close(); // Cleanup playback resources
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int jetkvm_audio_decode_write(void *opus_buf, int opus_size); // Decode Opus, write PCM
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// Configuration and utility functions
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void update_audio_constants(int bitrate, int complexity, int vbr, int vbr_constraint,
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int signal_type, int bandwidth, int dtx, int lsb_depth, int sr, int ch,
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int fs, int max_pkt, int sleep_us, int max_attempts, int max_backoff);
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void set_trace_logging(int enabled);
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int update_opus_encoder_params(int bitrate, int complexity, int vbr, int vbr_constraint,
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int signal_type, int bandwidth, int dtx);
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// ============================================================================
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// CONFIGURATION FUNCTIONS
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// ============================================================================
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/**
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* Update audio configuration constants from Go
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* Called during initialization to sync C variables with Go config
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*/
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void update_audio_constants(int bitrate, int complexity, int vbr, int vbr_constraint,
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int signal_type, int bandwidth, int dtx, int lsb_depth, int sr, int ch,
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int fs, int max_pkt, int sleep_us, int max_attempts, int max_backoff) {
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opus_bitrate = bitrate;
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opus_complexity = complexity;
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opus_vbr = vbr;
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opus_vbr_constraint = vbr_constraint;
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opus_signal_type = signal_type;
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opus_bandwidth = bandwidth;
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opus_dtx = dtx;
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opus_lsb_depth = lsb_depth;
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sample_rate = sr;
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channels = ch;
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frame_size = fs;
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max_packet_size = max_pkt;
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sleep_microseconds = sleep_us;
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max_attempts_global = max_attempts;
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max_backoff_us_global = max_backoff;
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}
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/**
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* Enable or disable trace logging
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* When enabled, detailed debug information is printed to stdout
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* Zero overhead when disabled - no function calls or string formatting occur
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*/
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void set_trace_logging(int enabled) {
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trace_logging_enabled = enabled;
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}
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// ============================================================================
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// SIMD-OPTIMIZED BUFFER OPERATIONS (ARM NEON)
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// ============================================================================
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/**
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* SIMD-optimized buffer clearing for 16-bit audio samples
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* Uses ARM NEON to clear 8 samples (16 bytes) per iteration
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*
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* @param buffer Pointer to 16-bit sample buffer (must be 16-byte aligned)
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* @param samples Number of samples to clear
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*/
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static inline void simd_clear_samples_s16(short *buffer, int samples) {
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simd_init_once();
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const int16x8_t zero = vdupq_n_s16(0);
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int simd_samples = samples & ~7; // Round down to multiple of 8
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// Process 8 samples at a time with NEON
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for (int i = 0; i < simd_samples; i += 8) {
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vst1q_s16(&buffer[i], zero);
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}
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// Handle remaining samples with scalar operations
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for (int i = simd_samples; i < samples; i++) {
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buffer[i] = 0;
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}
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}
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/**
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* SIMD-optimized stereo sample interleaving
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* Combines left and right channel data using NEON zip operations
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*
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* @param left Left channel samples
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* @param right Right channel samples
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* @param output Interleaved stereo output
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* @param frames Number of frames to process
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*/
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static inline void simd_interleave_stereo_s16(const short *left, const short *right,
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short *output, int frames) {
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simd_init_once();
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int simd_frames = frames & ~7; // Process 8 frames at a time
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for (int i = 0; i < simd_frames; i += 8) {
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int16x8_t left_vec = vld1q_s16(&left[i]);
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int16x8_t right_vec = vld1q_s16(&right[i]);
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// Interleave using zip operations
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int16x8x2_t interleaved = vzipq_s16(left_vec, right_vec);
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// Store interleaved data
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vst1q_s16(&output[i * 2], interleaved.val[0]);
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vst1q_s16(&output[i * 2 + 8], interleaved.val[1]);
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}
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// Handle remaining frames
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for (int i = simd_frames; i < frames; i++) {
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output[i * 2] = left[i];
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output[i * 2 + 1] = right[i];
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}
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}
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/**
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* SIMD-optimized volume scaling for 16-bit samples
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* Applies volume scaling using NEON multiply operations
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*
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* @param samples Input/output sample buffer
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* @param count Number of samples to scale
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* @param volume Volume factor (0.0 to 1.0, converted to fixed-point)
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*/
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static inline void simd_scale_volume_s16(short *samples, int count, float volume) {
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simd_init_once();
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// Convert volume to fixed-point (Q15 format)
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int16_t vol_fixed = (int16_t)(volume * 32767.0f);
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int16x8_t vol_vec = vdupq_n_s16(vol_fixed);
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int simd_count = count & ~7;
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for (int i = 0; i < simd_count; i += 8) {
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int16x8_t samples_vec = vld1q_s16(&samples[i]);
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// Multiply and shift right by 15 to maintain Q15 format
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int32x4_t low_result = vmull_s16(vget_low_s16(samples_vec), vget_low_s16(vol_vec));
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int32x4_t high_result = vmull_s16(vget_high_s16(samples_vec), vget_high_s16(vol_vec));
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// Shift right by 15 and narrow back to 16-bit
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int16x4_t low_narrow = vshrn_n_s32(low_result, 15);
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int16x4_t high_narrow = vshrn_n_s32(high_result, 15);
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int16x8_t result = vcombine_s16(low_narrow, high_narrow);
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vst1q_s16(&samples[i], result);
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}
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// Handle remaining samples
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for (int i = simd_count; i < count; i++) {
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samples[i] = (short)((samples[i] * vol_fixed) >> 15);
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}
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}
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/**
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* SIMD-optimized endianness conversion for 16-bit samples
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* Swaps byte order using NEON reverse operations
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*/
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static inline void simd_swap_endian_s16(short *samples, int count) {
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int simd_count = count & ~7;
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for (int i = 0; i < simd_count; i += 8) {
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uint16x8_t samples_vec = vld1q_u16((uint16_t*)&samples[i]);
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// Reverse bytes within each 16-bit element
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uint8x16_t samples_u8 = vreinterpretq_u8_u16(samples_vec);
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uint8x16_t swapped_u8 = vrev16q_u8(samples_u8);
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uint16x8_t swapped = vreinterpretq_u16_u8(swapped_u8);
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vst1q_u16((uint16_t*)&samples[i], swapped);
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}
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// Handle remaining samples
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for (int i = simd_count; i < count; i++) {
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samples[i] = __builtin_bswap16(samples[i]);
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}
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}
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/**
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* Convert 16-bit signed samples to 32-bit float samples using NEON
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*/
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static inline void simd_s16_to_float(const short *input, float *output, int count) {
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const float scale = 1.0f / 32768.0f;
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float32x4_t scale_vec = vdupq_n_f32(scale);
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// Process 4 samples at a time
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int simd_count = count & ~3;
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for (int i = 0; i < simd_count; i += 4) {
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int16x4_t s16_data = vld1_s16(input + i);
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int32x4_t s32_data = vmovl_s16(s16_data);
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float32x4_t float_data = vcvtq_f32_s32(s32_data);
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float32x4_t scaled = vmulq_f32(float_data, scale_vec);
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vst1q_f32(output + i, scaled);
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}
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// Handle remaining samples
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for (int i = simd_count; i < count; i++) {
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output[i] = (float)input[i] * scale;
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}
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}
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/**
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* Convert 32-bit float samples to 16-bit signed samples using NEON
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*/
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static inline void simd_float_to_s16(const float *input, short *output, int count) {
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const float scale = 32767.0f;
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float32x4_t scale_vec = vdupq_n_f32(scale);
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// Process 4 samples at a time
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int simd_count = count & ~3;
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for (int i = 0; i < simd_count; i += 4) {
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float32x4_t float_data = vld1q_f32(input + i);
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float32x4_t scaled = vmulq_f32(float_data, scale_vec);
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int32x4_t s32_data = vcvtq_s32_f32(scaled);
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int16x4_t s16_data = vqmovn_s32(s32_data);
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vst1_s16(output + i, s16_data);
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}
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// Handle remaining samples
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for (int i = simd_count; i < count; i++) {
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float scaled = input[i] * scale;
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output[i] = (short)__builtin_fmaxf(__builtin_fminf(scaled, 32767.0f), -32768.0f);
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}
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}
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/**
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* Convert mono to stereo by duplicating samples using NEON
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*/
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static inline void simd_mono_to_stereo_s16(const short *mono, short *stereo, int frames) {
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// Process 4 frames at a time
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int simd_frames = frames & ~3;
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for (int i = 0; i < simd_frames; i += 4) {
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int16x4_t mono_data = vld1_s16(mono + i);
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int16x4x2_t stereo_data = {mono_data, mono_data};
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vst2_s16(stereo + i * 2, stereo_data);
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}
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// Handle remaining frames
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for (int i = simd_frames; i < frames; i++) {
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stereo[i * 2] = mono[i];
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stereo[i * 2 + 1] = mono[i];
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}
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}
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/**
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* Convert stereo to mono by averaging channels using NEON
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*/
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static inline void simd_stereo_to_mono_s16(const short *stereo, short *mono, int frames) {
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// Process 4 frames at a time
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int simd_frames = frames & ~3;
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for (int i = 0; i < simd_frames; i += 4) {
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int16x4x2_t stereo_data = vld2_s16(stereo + i * 2);
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int32x4_t left_wide = vmovl_s16(stereo_data.val[0]);
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int32x4_t right_wide = vmovl_s16(stereo_data.val[1]);
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int32x4_t sum = vaddq_s32(left_wide, right_wide);
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int32x4_t avg = vshrq_n_s32(sum, 1);
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int16x4_t mono_data = vqmovn_s32(avg);
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vst1_s16(mono + i, mono_data);
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}
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// Handle remaining frames
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for (int i = simd_frames; i < frames; i++) {
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mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) / 2;
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}
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}
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/**
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* Apply stereo balance adjustment using NEON
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*/
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static inline void simd_apply_stereo_balance_s16(short *stereo, int frames, float balance) {
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// Balance: -1.0 = full left, 0.0 = center, 1.0 = full right
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float left_gain = balance <= 0.0f ? 1.0f : 1.0f - balance;
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float right_gain = balance >= 0.0f ? 1.0f : 1.0f + balance;
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float32x4_t left_gain_vec = vdupq_n_f32(left_gain);
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float32x4_t right_gain_vec = vdupq_n_f32(right_gain);
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// Process 4 frames at a time
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int simd_frames = frames & ~3;
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for (int i = 0; i < simd_frames; i += 4) {
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int16x4x2_t stereo_data = vld2_s16(stereo + i * 2);
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// Convert to float for processing
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int32x4_t left_wide = vmovl_s16(stereo_data.val[0]);
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int32x4_t right_wide = vmovl_s16(stereo_data.val[1]);
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float32x4_t left_float = vcvtq_f32_s32(left_wide);
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float32x4_t right_float = vcvtq_f32_s32(right_wide);
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// Apply balance
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left_float = vmulq_f32(left_float, left_gain_vec);
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right_float = vmulq_f32(right_float, right_gain_vec);
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// Convert back to int16
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int32x4_t left_result = vcvtq_s32_f32(left_float);
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int32x4_t right_result = vcvtq_s32_f32(right_float);
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stereo_data.val[0] = vqmovn_s32(left_result);
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stereo_data.val[1] = vqmovn_s32(right_result);
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vst2_s16(stereo + i * 2, stereo_data);
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}
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// Handle remaining frames
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for (int i = simd_frames; i < frames; i++) {
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stereo[i * 2] = (short)(stereo[i * 2] * left_gain);
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stereo[i * 2 + 1] = (short)(stereo[i * 2 + 1] * right_gain);
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}
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}
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/**
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* Deinterleave stereo samples into separate left/right channels using NEON
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*/
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static inline void simd_deinterleave_stereo_s16(const short *interleaved, short *left,
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short *right, int frames) {
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// Process 4 frames at a time
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int simd_frames = frames & ~3;
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for (int i = 0; i < simd_frames; i += 4) {
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int16x4x2_t stereo_data = vld2_s16(interleaved + i * 2);
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vst1_s16(left + i, stereo_data.val[0]);
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vst1_s16(right + i, stereo_data.val[1]);
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}
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// Handle remaining frames
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for (int i = simd_frames; i < frames; i++) {
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left[i] = interleaved[i * 2];
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right[i] = interleaved[i * 2 + 1];
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}
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}
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/**
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* SIMD-optimized max absolute value finder for silence detection
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* Returns the maximum absolute sample value in the buffer
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*/
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static inline short simd_find_max_abs_s16(const short *samples, int count) {
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int16x8_t max_vec = vdupq_n_s16(0);
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int simd_count = count & ~7;
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// Process 8 samples at a time
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for (int i = 0; i < simd_count; i += 8) {
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int16x8_t samples_vec = vld1q_s16(&samples[i]);
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int16x8_t abs_vec = vabsq_s16(samples_vec);
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max_vec = vmaxq_s16(max_vec, abs_vec);
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}
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// Find maximum in vector (horizontal max)
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int16x4_t max_half = vmax_s16(vget_low_s16(max_vec), vget_high_s16(max_vec));
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int16x4_t max_folded = vpmax_s16(max_half, max_half);
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max_folded = vpmax_s16(max_folded, max_folded);
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short max_sample = vget_lane_s16(max_folded, 0);
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// Handle remaining samples
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for (int i = simd_count; i < count; i++) {
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short abs_sample = samples[i] < 0 ? -samples[i] : samples[i];
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if (abs_sample > max_sample) {
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max_sample = abs_sample;
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}
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}
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return max_sample;
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}
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// ============================================================================
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// INITIALIZATION STATE TRACKING
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// ============================================================================
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// Thread-safe initialization state tracking to prevent race conditions
|
|
static volatile int capture_initializing = 0; // OUTPUT path init in progress
|
|
static volatile int capture_initialized = 0; // OUTPUT path ready
|
|
static volatile int playback_initializing = 0; // INPUT path init in progress
|
|
static volatile int playback_initialized = 0; // INPUT path ready
|
|
|
|
/**
|
|
* Update Opus encoder parameters dynamically
|
|
* Used for OUTPUT path (TC358743 HDMI audio → client speakers)
|
|
*
|
|
* @return 0 on success, -1 if encoder not initialized, >0 if some settings failed
|
|
*/
|
|
int update_opus_encoder_params(int bitrate, int complexity, int vbr, int vbr_constraint,
|
|
int signal_type, int bandwidth, int dtx) {
|
|
if (!encoder || !capture_initialized) {
|
|
return -1;
|
|
}
|
|
|
|
// Update local configuration
|
|
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 settings to Opus 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;
|
|
}
|
|
|
|
// ============================================================================
|
|
// ALSA UTILITY FUNCTIONS
|
|
// ============================================================================
|
|
|
|
/**
|
|
* Safely open ALSA device with exponential backoff retry logic
|
|
* Handles common device busy/unavailable scenarios with appropriate retry strategies
|
|
*
|
|
* @param handle Pointer to PCM handle to be set
|
|
* @param device ALSA device name (e.g., "hw:1,0")
|
|
* @param stream Stream direction (capture or playback)
|
|
* @return 0 on success, negative error code on failure
|
|
*/
|
|
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++;
|
|
|
|
// 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;
|
|
}
|
|
|
|
/**
|
|
* Configure ALSA device with optimized parameters
|
|
* Sets up hardware and software parameters for optimal performance on constrained hardware
|
|
*
|
|
* @param handle ALSA PCM handle
|
|
* @param device_name Device name for debugging (not used in current implementation)
|
|
* @return 0 on success, negative error code on failure
|
|
*/
|
|
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(¶ms);
|
|
snd_pcm_sw_params_alloca(&sw_params);
|
|
|
|
// Hardware parameters
|
|
err = snd_pcm_hw_params_any(handle, params);
|
|
if (err < 0) return err;
|
|
|
|
// Use RW access for compatibility
|
|
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, using smaller periods for lower latency on
|
|
// constrained hardware
|
|
snd_pcm_uframes_t period_size = optimized_buffer_size ? frame_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, using 2 periods for ultra-low latency on
|
|
// constrained hardware or 4 periods for good latency/stability balance
|
|
snd_pcm_uframes_t buffer_size = optimized_buffer_size ? buffer_size = period_size * 2 : 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);
|
|
}
|
|
|
|
// ============================================================================
|
|
// AUDIO OUTPUT PATH FUNCTIONS (TC358743 HDMI Audio → Client Speakers)
|
|
// ============================================================================
|
|
|
|
/**
|
|
* Initialize audio OUTPUT path: TC358743 HDMI audio capture and Opus encoder
|
|
* This enables sending HDMI audio from the managed device to the client
|
|
*
|
|
* Thread-safe with atomic operations to prevent concurrent initialization
|
|
*
|
|
* @return 0 on success, negative error codes on failure:
|
|
* -EBUSY: Already initializing
|
|
* -1: ALSA device open failed
|
|
* -2: ALSA device configuration failed
|
|
* -3: Opus encoder creation failed
|
|
*/
|
|
int jetkvm_audio_capture_init() {
|
|
int err;
|
|
|
|
// Initialize SIMD capabilities early
|
|
simd_init_once();
|
|
|
|
// 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_capture_handle) {
|
|
snd_pcm_close(pcm_capture_handle);
|
|
pcm_capture_handle = NULL;
|
|
}
|
|
|
|
// Try to open ALSA capture device (TC358743 HDMI audio)
|
|
// Native S16_LE @ 48kHz stereo capture - no resampling, minimal CPU overhead
|
|
err = safe_alsa_open(&pcm_capture_handle, "hw:0,0", SND_PCM_STREAM_CAPTURE);
|
|
if (err < 0) {
|
|
capture_initializing = 0;
|
|
return -1;
|
|
}
|
|
|
|
// Configure the device
|
|
err = configure_alsa_device(pcm_capture_handle, "capture");
|
|
if (err < 0) {
|
|
snd_pcm_close(pcm_capture_handle);
|
|
pcm_capture_handle = NULL;
|
|
capture_initializing = 0;
|
|
return -2;
|
|
}
|
|
|
|
// 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_capture_handle) {
|
|
snd_pcm_close(pcm_capture_handle);
|
|
pcm_capture_handle = NULL;
|
|
}
|
|
capture_initializing = 0;
|
|
return -3;
|
|
}
|
|
|
|
// 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
|
|
opus_encoder_ctl(encoder, OPUS_SET_LSB_DEPTH(opus_lsb_depth));
|
|
// Packet loss concealment removed - causes artifacts on transients in LAN environment
|
|
// Prediction enabled (default) for better transient handling (beeps, sharp sounds)
|
|
|
|
capture_initialized = 1;
|
|
capture_initializing = 0;
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Capture audio from TC358743 HDMI and encode to Opus (OUTPUT path)
|
|
*
|
|
* This function:
|
|
* 1. Reads PCM audio from TC358743 HDMI input via ALSA
|
|
* 2. Handles ALSA errors with robust recovery strategies
|
|
* 3. Encodes PCM to Opus format for network transmission to client
|
|
* 4. Provides zero-overhead trace logging when enabled
|
|
*
|
|
* Error recovery includes handling:
|
|
* - Buffer underruns (-EPIPE)
|
|
* - Device suspension (-ESTRPIPE)
|
|
* - I/O errors (-EIO)
|
|
* - Device busy conditions (-EBUSY, -EAGAIN)
|
|
*
|
|
* @param opus_buf Buffer to store encoded Opus data (must be at least max_packet_size)
|
|
* @return >0: Number of Opus bytes written
|
|
* 0: No audio data available (not an error)
|
|
* -1: Initialization error or unrecoverable failure
|
|
*/
|
|
__attribute__((hot)) int jetkvm_audio_read_encode(void * __restrict__ opus_buf) {
|
|
static short SIMD_ALIGN pcm_buffer[1920]; // max 2ch*960, aligned for SIMD
|
|
unsigned char * __restrict__ out = (unsigned char*)opus_buf;
|
|
|
|
// Prefetch output buffer and PCM buffer for better cache performance
|
|
SIMD_PREFETCH(out, 1, 3);
|
|
SIMD_PREFETCH(pcm_buffer, 0, 3);
|
|
int err = 0;
|
|
int recovery_attempts = 0;
|
|
const int max_recovery_attempts = 3;
|
|
|
|
if (__builtin_expect(!capture_initialized || !pcm_capture_handle || !encoder || !opus_buf, 0)) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_OUTPUT] jetkvm_audio_read_encode: Failed safety checks - capture_initialized=%d, pcm_capture_handle=%p, encoder=%p, opus_buf=%p\n",
|
|
capture_initialized, pcm_capture_handle, encoder, opus_buf);
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
retry_read:
|
|
;
|
|
int pcm_rc = snd_pcm_readi(pcm_capture_handle, pcm_buffer, frame_size);
|
|
|
|
// Handle ALSA errors with robust recovery strategies
|
|
if (__builtin_expect(pcm_rc < 0, 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_capture_handle);
|
|
if (err < 0) {
|
|
// If prepare fails, try drop and prepare
|
|
snd_pcm_drop(pcm_capture_handle);
|
|
err = snd_pcm_prepare(pcm_capture_handle);
|
|
if (err < 0) return -1;
|
|
}
|
|
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_capture_handle)) == -EAGAIN && resume_attempts < 10) {
|
|
usleep(sleep_microseconds);
|
|
resume_attempts++;
|
|
}
|
|
if (err < 0) {
|
|
// Resume failed, try prepare as fallback
|
|
err = snd_pcm_prepare(pcm_capture_handle);
|
|
if (err < 0) return -1;
|
|
}
|
|
return 0;
|
|
} 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_capture_handle);
|
|
err = snd_pcm_prepare(pcm_capture_handle);
|
|
if (err >= 0) {
|
|
goto retry_read;
|
|
}
|
|
}
|
|
return -1;
|
|
} else {
|
|
// Other errors - limited retry for transient issues
|
|
recovery_attempts++;
|
|
if (recovery_attempts <= 1 && pcm_rc == -EINTR) {
|
|
goto retry_read;
|
|
} else if (recovery_attempts <= 1 && pcm_rc == -EBUSY) {
|
|
// Device busy - simple sleep to allow other operations to complete
|
|
usleep(sleep_microseconds / 2);
|
|
goto retry_read;
|
|
}
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
// If we got fewer frames than expected, pad with silence using SIMD
|
|
if (__builtin_expect(pcm_rc < frame_size, 0)) {
|
|
int remaining_samples = (frame_size - pcm_rc) * channels;
|
|
simd_clear_samples_s16(&pcm_buffer[pcm_rc * channels], remaining_samples);
|
|
}
|
|
|
|
// Silence detection using SIMD-optimized max peak detection
|
|
// Find the maximum absolute sample value in the frame
|
|
int total_samples = frame_size * channels;
|
|
short max_sample = simd_find_max_abs_s16(pcm_buffer, total_samples);
|
|
|
|
// If max peak is below threshold, consider it silence
|
|
// Threshold: 50 = ~0.15% of max volume (very quiet background noise)
|
|
if (max_sample < 50) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_OUTPUT] jetkvm_audio_read_encode: Silence detected (max=%d), skipping frame\n", max_sample);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
// Apply 5x gain boost to fix quantization noise on transients at normal volumes to prevent crackling issues
|
|
// This allows comfortable listening at low remote volumes (10-40% range)
|
|
simd_scale_volume_s16(pcm_buffer, frame_size * channels, 5.0f);
|
|
|
|
int nb_bytes = opus_encode(encoder, pcm_buffer, frame_size, out, max_packet_size);
|
|
|
|
if (trace_logging_enabled && nb_bytes > 0) {
|
|
printf("[AUDIO_OUTPUT] jetkvm_audio_read_encode: Successfully encoded %d PCM frames to %d Opus bytes\n", pcm_rc, nb_bytes);
|
|
}
|
|
|
|
return nb_bytes;
|
|
}
|
|
|
|
// ============================================================================
|
|
// AUDIO INPUT PATH FUNCTIONS (Client Microphone → Device Speakers)
|
|
// ============================================================================
|
|
|
|
/**
|
|
* Initialize audio INPUT path: ALSA playback device and Opus decoder
|
|
* This enables playing client audio through device speakers
|
|
*
|
|
* Thread-safe with atomic operations to prevent concurrent initialization
|
|
*
|
|
* @return 0 on success, negative error codes on failure:
|
|
* -EBUSY: Already initializing
|
|
* -1: ALSA device open failed or configuration failed
|
|
* -2: Opus decoder creation failed
|
|
*/
|
|
int jetkvm_audio_playback_init() {
|
|
int err;
|
|
|
|
// Initialize SIMD capabilities early
|
|
simd_init_once();
|
|
|
|
// 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;
|
|
}
|
|
|
|
/**
|
|
* Decode Opus audio and play through device speakers (INPUT path)
|
|
*
|
|
* This function:
|
|
* 1. Validates input parameters and Opus packet size
|
|
* 2. Decodes Opus data to PCM format
|
|
* 3. Implements packet loss concealment for network issues
|
|
* 4. Writes PCM to device speakers via ALSA
|
|
* 5. Handles ALSA playback errors with recovery strategies
|
|
* 6. Provides zero-overhead trace logging when enabled
|
|
*
|
|
* Error recovery includes handling:
|
|
* - Buffer underruns (-EPIPE) with progressive recovery
|
|
* - Device suspension (-ESTRPIPE) with resume logic
|
|
* - I/O errors (-EIO) with device reset
|
|
* - Device not ready (-EAGAIN) with retry logic
|
|
*
|
|
* @param opus_buf Buffer containing Opus-encoded audio data
|
|
* @param opus_size Size of Opus data in bytes
|
|
* @return >0: Number of PCM frames written to speakers
|
|
* 0: Frame skipped (not an error)
|
|
* -1: Invalid input or decode failure
|
|
* -2: Unrecoverable ALSA error
|
|
*/
|
|
__attribute__((hot)) int jetkvm_audio_decode_write(void * __restrict__ opus_buf, int opus_size) {
|
|
static short __attribute__((aligned(16))) pcm_buffer[1920]; // max 2ch*960, aligned for SIMD
|
|
unsigned char * __restrict__ in = (unsigned char*)opus_buf;
|
|
|
|
// Prefetch input buffer for better cache performance
|
|
SIMD_PREFETCH(in, 0, 3);
|
|
int err = 0;
|
|
int recovery_attempts = 0;
|
|
const int max_recovery_attempts = 3;
|
|
|
|
// Safety checks
|
|
if (__builtin_expect(!playback_initialized || !pcm_playback_handle || !decoder || !opus_buf || opus_size <= 0, 0)) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Failed safety checks - playback_initialized=%d, pcm_playback_handle=%p, decoder=%p, opus_buf=%p, opus_size=%d\n",
|
|
playback_initialized, pcm_playback_handle, decoder, opus_buf, opus_size);
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
// Additional bounds checking
|
|
if (opus_size > max_packet_size) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Opus packet too large - size=%d, max=%d\n", opus_size, max_packet_size);
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Processing Opus packet - size=%d bytes\n", opus_size);
|
|
}
|
|
|
|
// Decode Opus to PCM with error handling
|
|
int pcm_frames = opus_decode(decoder, in, opus_size, pcm_buffer, frame_size, 0);
|
|
if (__builtin_expect(pcm_frames < 0, 0)) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Opus decode failed with error %d, attempting packet loss concealment\n", pcm_frames);
|
|
}
|
|
// Try packet loss concealment on decode error
|
|
pcm_frames = opus_decode(decoder, NULL, 0, pcm_buffer, frame_size, 0);
|
|
if (pcm_frames < 0) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Packet loss concealment also failed with error %d\n", pcm_frames);
|
|
}
|
|
return -1;
|
|
}
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Packet loss concealment succeeded, recovered %d frames\n", pcm_frames);
|
|
}
|
|
} else if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Opus decode successful - decoded %d PCM frames\n", pcm_frames);
|
|
}
|
|
|
|
retry_write:
|
|
;
|
|
// Write PCM to playback device with robust recovery
|
|
int pcm_rc = snd_pcm_writei(pcm_playback_handle, pcm_buffer, pcm_frames);
|
|
if (__builtin_expect(pcm_rc < 0, 0)) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: ALSA write failed with error %d (%s), attempt %d/%d\n",
|
|
pcm_rc, snd_strerror(pcm_rc), recovery_attempts + 1, max_recovery_attempts);
|
|
}
|
|
|
|
if (pcm_rc == -EPIPE) {
|
|
// Buffer underrun - implement progressive recovery
|
|
recovery_attempts++;
|
|
if (recovery_attempts > max_recovery_attempts) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Buffer underrun recovery failed after %d attempts\n", max_recovery_attempts);
|
|
}
|
|
return -2;
|
|
}
|
|
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Buffer underrun detected, attempting recovery (attempt %d)\n", recovery_attempts);
|
|
}
|
|
// Try to recover with prepare
|
|
err = snd_pcm_prepare(pcm_playback_handle);
|
|
if (err < 0) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: snd_pcm_prepare failed (%s), trying drop+prepare\n", snd_strerror(err));
|
|
}
|
|
// If prepare fails, try drop and prepare
|
|
snd_pcm_drop(pcm_playback_handle);
|
|
err = snd_pcm_prepare(pcm_playback_handle);
|
|
if (err < 0) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: drop+prepare recovery failed (%s)\n", snd_strerror(err));
|
|
}
|
|
return -2;
|
|
}
|
|
}
|
|
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Buffer underrun recovery successful, retrying write\n");
|
|
}
|
|
goto retry_write;
|
|
} else if (pcm_rc == -ESTRPIPE) {
|
|
// Device suspended, implement robust resume logic
|
|
recovery_attempts++;
|
|
if (recovery_attempts > max_recovery_attempts) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Device suspend recovery failed after %d attempts\n", max_recovery_attempts);
|
|
}
|
|
return -2;
|
|
}
|
|
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Device suspended, attempting resume (attempt %d)\n", recovery_attempts);
|
|
}
|
|
// 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) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Device resume failed (%s), trying prepare fallback\n", snd_strerror(err));
|
|
}
|
|
// Resume failed, try prepare as fallback
|
|
err = snd_pcm_prepare(pcm_playback_handle);
|
|
if (err < 0) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Prepare fallback failed (%s)\n", snd_strerror(err));
|
|
}
|
|
return -2;
|
|
}
|
|
}
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Device suspend recovery successful, skipping frame\n");
|
|
}
|
|
return 0; // Skip this frame but don't fail
|
|
} else if (pcm_rc == -ENODEV) {
|
|
// Device disconnected - critical error
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Device disconnected (ENODEV) - critical error\n");
|
|
}
|
|
return -2;
|
|
} else if (pcm_rc == -EIO) {
|
|
// I/O error - try recovery once
|
|
recovery_attempts++;
|
|
if (recovery_attempts <= max_recovery_attempts) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: I/O error detected, attempting recovery\n");
|
|
}
|
|
snd_pcm_drop(pcm_playback_handle);
|
|
err = snd_pcm_prepare(pcm_playback_handle);
|
|
if (err >= 0) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: I/O error recovery successful, retrying write\n");
|
|
}
|
|
goto retry_write;
|
|
}
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: I/O error recovery failed (%s)\n", snd_strerror(err));
|
|
}
|
|
}
|
|
return -2;
|
|
} else if (pcm_rc == -EAGAIN) {
|
|
// Device not ready - brief wait and retry
|
|
recovery_attempts++;
|
|
if (recovery_attempts <= max_recovery_attempts) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Device not ready (EAGAIN), waiting and retrying\n");
|
|
}
|
|
snd_pcm_wait(pcm_playback_handle, sleep_microseconds / 4000); // Convert to milliseconds
|
|
goto retry_write;
|
|
}
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Device not ready recovery failed after %d attempts\n", max_recovery_attempts);
|
|
}
|
|
return -2;
|
|
} else {
|
|
// Other errors - limited retry for transient issues
|
|
recovery_attempts++;
|
|
if (recovery_attempts <= 1 && (pcm_rc == -EINTR || pcm_rc == -EBUSY)) {
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Transient error %d (%s), retrying once\n", pcm_rc, snd_strerror(pcm_rc));
|
|
}
|
|
usleep(sleep_microseconds / 2);
|
|
goto retry_write;
|
|
}
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Unrecoverable error %d (%s)\n", pcm_rc, snd_strerror(pcm_rc));
|
|
}
|
|
return -2;
|
|
}
|
|
}
|
|
|
|
if (trace_logging_enabled) {
|
|
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Successfully wrote %d PCM frames to USB Gadget audio device\n", pcm_frames);
|
|
}
|
|
return pcm_frames;
|
|
}
|
|
|
|
// ============================================================================
|
|
// CLEANUP FUNCTIONS
|
|
// ============================================================================
|
|
|
|
/**
|
|
* Cleanup audio INPUT path resources (client microphone → device speakers)
|
|
*
|
|
* Thread-safe cleanup with atomic operations to prevent double-cleanup
|
|
* Properly drains ALSA buffers before closing to avoid audio artifacts
|
|
*/
|
|
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;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Cleanup audio OUTPUT path resources (TC358743 HDMI audio → client speakers)
|
|
*
|
|
* Thread-safe cleanup with atomic operations to prevent double-cleanup
|
|
* Properly drains ALSA buffers before closing to avoid audio artifacts
|
|
*/
|
|
void jetkvm_audio_capture_close() {
|
|
// Wait for any ongoing operations to complete
|
|
while (capture_initializing) {
|
|
usleep(sleep_microseconds);
|
|
}
|
|
|
|
// Atomic check and set to prevent double cleanup
|
|
if (__sync_bool_compare_and_swap(&capture_initialized, 1, 0) == 0) {
|
|
return; // Already cleaned up
|
|
}
|
|
|
|
if (encoder) {
|
|
opus_encoder_destroy(encoder);
|
|
encoder = NULL;
|
|
}
|
|
if (pcm_capture_handle) {
|
|
snd_pcm_drain(pcm_capture_handle);
|
|
snd_pcm_close(pcm_capture_handle);
|
|
pcm_capture_handle = NULL;
|
|
}
|
|
}
|