mirror of https://github.com/jetkvm/kvm.git
1098 lines
37 KiB
C
1098 lines
37 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; // OUTPUT: TC358743 HDMI audio → client
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static snd_pcm_t *pcm_playback_handle = NULL; // INPUT: Client microphone → device speakers
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// Opus codec instances
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static OpusEncoder *encoder = NULL;
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static OpusDecoder *decoder = NULL;
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// Audio format (S16_LE @ 48kHz stereo)
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static int sample_rate = 48000;
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static int channels = 2;
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static int frame_size = 960; // 20ms frames at 48kHz
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// Opus encoder settings (optimized for minimal CPU ~0.5% on RV1106)
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static int opus_bitrate = 96000; // 96 kbps
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static int opus_complexity = 1; // Complexity 1 (minimal CPU)
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static int opus_vbr = 1; // Variable bitrate enabled
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static int opus_vbr_constraint = 1; // Constrained VBR for predictable bandwidth
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static int opus_signal_type = -1000; // OPUS_AUTO (-1000)
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static int opus_bandwidth = 1103; // OPUS_BANDWIDTH_WIDEBAND (1103)
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static int opus_dtx = 0; // DTX disabled
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static int opus_lsb_depth = 16; // 16-bit depth matches S16_LE
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// Network configuration
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static int max_packet_size = 1500;
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// ALSA retry configuration
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static int sleep_microseconds = 1000;
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static int max_attempts_global = 5;
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static int max_backoff_us_global = 500000;
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// Buffer optimization (1 = use 2-period ultra-low latency, 0 = use 4-period balanced)
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static const int optimized_buffer_size = 1;
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// ============================================================================
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// FUNCTION DECLARATIONS
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// ============================================================================
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int jetkvm_audio_capture_init();
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void jetkvm_audio_capture_close();
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int jetkvm_audio_read_encode(void *opus_buf);
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int jetkvm_audio_playback_init();
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void jetkvm_audio_playback_close();
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int jetkvm_audio_decode_write(void *opus_buf, int opus_size);
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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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* Sync configuration from Go to C
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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/disable trace logging (zero overhead when disabled)
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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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* Clear audio buffer using NEON (8 samples/iteration)
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* @param buffer Audio buffer to clear
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* @param samples Number of samples to zero out
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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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int simd_samples = samples & ~7;
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const int16x8_t zero = vdupq_n_s16(0);
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// SIMD path: zero 8 samples per iteration
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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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// Scalar path: handle remaining samples
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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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* Interleave L/R channels using NEON (8 frames/iteration)
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* Converts separate left/right buffers to interleaved stereo (LRLRLR...)
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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 buffer
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* @param frames Number of stereo 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;
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// SIMD path: interleave 8 frames (16 samples) per iteration
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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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int16x8x2_t interleaved = vzipq_s16(left_vec, right_vec);
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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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// Scalar path: 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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* Apply gain using NEON Q15 fixed-point math (8 samples/iteration)
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* Uses vqrdmulhq_s16 for single-instruction saturating rounded multiply-high
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* @param samples Audio buffer to scale in-place
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* @param count Number of samples to process
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* @param volume Gain multiplier (e.g., 2.5 for 2.5x gain)
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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 float gain to Q14 fixed-point for vqrdmulhq_s16
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// vqrdmulhq_s16 extracts bits [30:15], so multiply by 16384 (2^14) instead of 32768 (2^15)
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int16_t vol_fixed = (int16_t)(volume * 16384.0f);
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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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// SIMD path: process 8 samples per iteration
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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 result = vqrdmulhq_s16(samples_vec, vol_vec);
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vst1q_s16(&samples[i], result);
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}
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// Scalar path: 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) >> 14);
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}
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}
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/**
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* Byte-swap 16-bit samples using NEON (8 samples/iteration)
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* Converts between little-endian and big-endian formats
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* @param samples Audio buffer to byte-swap in-place
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* @param count Number of samples to process
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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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// SIMD path: swap 8 samples per iteration
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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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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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// Scalar path: 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 S16 to float using NEON (4 samples/iteration)
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* Converts 16-bit signed integers to normalized float [-1.0, 1.0]
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* @param input S16 audio samples
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* @param output Float output buffer
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* @param count Number of samples to convert
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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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int simd_count = count & ~3;
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float32x4_t scale_vec = vdupq_n_f32(scale);
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// SIMD path: convert 4 samples per iteration
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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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// Scalar path: 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 float to S16 using NEON (4 samples/iteration)
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* Converts normalized float [-1.0, 1.0] to 16-bit signed integers with saturation
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* @param input Float audio samples
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* @param output S16 output buffer
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* @param count Number of samples to convert
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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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int simd_count = count & ~3;
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float32x4_t scale_vec = vdupq_n_f32(scale);
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// SIMD path: convert 4 samples per iteration with saturation
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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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// Scalar path: handle remaining samples with clamping
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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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* Mono → stereo (duplicate samples) using NEON (4 frames/iteration)
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* Duplicates mono samples to both L and R channels
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* @param mono Mono input buffer
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* @param stereo Stereo output buffer
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* @param frames Number of frames to process
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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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int simd_frames = frames & ~3;
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// SIMD path: duplicate 4 frames (8 samples) per iteration
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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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// Scalar path: 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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* Stereo → mono (average L+R) using NEON (4 frames/iteration)
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* Downmixes stereo to mono by averaging left and right channels
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* @param stereo Interleaved stereo input buffer
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* @param mono Mono output buffer
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* @param frames Number of frames to process
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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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int simd_frames = frames & ~3;
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// SIMD path: average 4 stereo frames per iteration
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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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// Scalar path: 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 L/R balance using NEON (4 frames/iteration)
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* Adjusts stereo balance: negative = more left, positive = more right
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* @param stereo Interleaved stereo buffer to modify in-place
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* @param frames Number of stereo frames to process
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* @param balance Balance factor [-1.0 = full left, 0.0 = center, 1.0 = full right]
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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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int simd_frames = frames & ~3;
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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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// SIMD path: apply balance to 4 stereo frames per iteration
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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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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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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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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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// Scalar path: 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 → L/R channels using NEON (4 frames/iteration)
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* Separates interleaved stereo (LRLRLR...) into separate L and R buffers
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* @param interleaved Interleaved stereo input buffer
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* @param left Left channel output buffer
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* @param right Right channel output buffer
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* @param frames Number of stereo frames to process
|
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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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int simd_frames = frames & ~3;
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// SIMD path: deinterleave 4 frames (8 samples) per iteration
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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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// Scalar path: 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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/**
|
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* Find max absolute sample value for silence detection using NEON (8 samples/iteration)
|
||
* Used to detect silence (threshold < 50 = ~0.15% max volume) and audio discontinuities
|
||
* @param samples Audio buffer to analyze
|
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* @param count Number of samples to process
|
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* @return 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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int simd_count = count & ~7;
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int16x8_t max_vec = vdupq_n_s16(0);
|
||
|
||
// SIMD path: find max of 8 samples per iteration
|
||
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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// Horizontal reduction: extract single max value from vector
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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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||
// Scalar path: handle remaining samples
|
||
for (int i = simd_count; i < count; i++) {
|
||
short abs_sample = samples[i] < 0 ? -samples[i] : samples[i];
|
||
if (abs_sample > max_sample) {
|
||
max_sample = abs_sample;
|
||
}
|
||
}
|
||
|
||
return max_sample;
|
||
}
|
||
|
||
// ============================================================================
|
||
// INITIALIZATION STATE TRACKING
|
||
// ============================================================================
|
||
|
||
static volatile int capture_initializing = 0;
|
||
static volatile int capture_initialized = 0;
|
||
static volatile int playback_initializing = 0;
|
||
static volatile int playback_initialized = 0;
|
||
|
||
/**
|
||
* Update Opus encoder settings at runtime
|
||
* @return 0 on success, -1 if 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;
|
||
}
|
||
|
||
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;
|
||
|
||
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
|
||
// ============================================================================
|
||
|
||
/**
|
||
* Open ALSA device with exponential backoff retry
|
||
* @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;
|
||
|
||
while (attempt < max_attempts_global) {
|
||
err = snd_pcm_open(handle, device, stream, SND_PCM_NONBLOCK);
|
||
if (err >= 0) {
|
||
snd_pcm_nonblock(*handle, 0);
|
||
return 0;
|
||
}
|
||
|
||
attempt++;
|
||
|
||
if (err == -EBUSY || err == -EAGAIN) {
|
||
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) {
|
||
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) {
|
||
usleep(backoff_us / 2);
|
||
} else {
|
||
usleep(backoff_us);
|
||
backoff_us = (backoff_us * 2 < max_backoff_us_global) ? backoff_us * 2 : max_backoff_us_global;
|
||
}
|
||
}
|
||
return err;
|
||
}
|
||
|
||
/**
|
||
* Configure ALSA device (S16_LE @ 48kHz stereo with optimized buffering)
|
||
* @param handle ALSA PCM handle
|
||
* @param device_name Unused (for debugging only)
|
||
* @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;
|
||
|
||
snd_pcm_hw_params_alloca(¶ms);
|
||
snd_pcm_sw_params_alloca(&sw_params);
|
||
|
||
err = snd_pcm_hw_params_any(handle, params);
|
||
if (err < 0) return err;
|
||
|
||
err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
|
||
if (err < 0) return err;
|
||
|
||
err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
|
||
if (err < 0) return err;
|
||
|
||
err = snd_pcm_hw_params_set_channels(handle, params, channels);
|
||
if (err < 0) return err;
|
||
|
||
err = snd_pcm_hw_params_set_rate(handle, params, sample_rate, 0);
|
||
if (err < 0) {
|
||
unsigned int rate = sample_rate;
|
||
err = snd_pcm_hw_params_set_rate_near(handle, params, &rate, 0);
|
||
if (err < 0) return err;
|
||
}
|
||
|
||
snd_pcm_uframes_t period_size = optimized_buffer_size ? frame_size : frame_size / 2;
|
||
if (period_size < 64) period_size = 64;
|
||
|
||
err = snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, 0);
|
||
if (err < 0) return err;
|
||
|
||
snd_pcm_uframes_t buffer_size = optimized_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;
|
||
|
||
err = snd_pcm_sw_params_current(handle, sw_params);
|
||
if (err < 0) return err;
|
||
|
||
err = snd_pcm_sw_params_set_start_threshold(handle, sw_params, period_size);
|
||
if (err < 0) return err;
|
||
|
||
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 OUTPUT path (TC358743 HDMI capture → Opus encoder)
|
||
* Opens hw:0,0 (TC358743) and creates Opus encoder with optimized settings
|
||
* @return 0 on success, -EBUSY if initializing, -1/-2/-3 on errors
|
||
*/
|
||
int jetkvm_audio_capture_init() {
|
||
int err;
|
||
|
||
simd_init_once();
|
||
|
||
if (__sync_bool_compare_and_swap(&capture_initializing, 0, 1) == 0) {
|
||
return -EBUSY;
|
||
}
|
||
|
||
if (capture_initialized) {
|
||
capture_initializing = 0;
|
||
return 0;
|
||
}
|
||
|
||
if (encoder) {
|
||
opus_encoder_destroy(encoder);
|
||
encoder = NULL;
|
||
}
|
||
if (pcm_capture_handle) {
|
||
snd_pcm_close(pcm_capture_handle);
|
||
pcm_capture_handle = NULL;
|
||
}
|
||
|
||
err = safe_alsa_open(&pcm_capture_handle, "hw:0,0", SND_PCM_STREAM_CAPTURE);
|
||
if (err < 0) {
|
||
capture_initializing = 0;
|
||
return -1;
|
||
}
|
||
|
||
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;
|
||
}
|
||
|
||
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;
|
||
}
|
||
|
||
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));
|
||
opus_encoder_ctl(encoder, OPUS_SET_DTX(opus_dtx));
|
||
opus_encoder_ctl(encoder, OPUS_SET_LSB_DEPTH(opus_lsb_depth));
|
||
|
||
// Enable in-band FEC for packet loss resilience (adds ~2-5% bitrate)
|
||
opus_encoder_ctl(encoder, OPUS_SET_INBAND_FEC(1));
|
||
opus_encoder_ctl(encoder, OPUS_SET_PACKET_LOSS_PERC(10));
|
||
|
||
capture_initialized = 1;
|
||
capture_initializing = 0;
|
||
return 0;
|
||
}
|
||
|
||
/**
|
||
* Read HDMI audio, encode to Opus (OUTPUT path hot function)
|
||
* Processing pipeline: ALSA capture → silence detection → discontinuity detection → 2.5x gain → Opus encode
|
||
* @param opus_buf Output buffer for encoded Opus packet
|
||
* @return >0 = Opus packet size in bytes, 0 = silence/no data, -1 = error
|
||
*/
|
||
__attribute__((hot)) int jetkvm_audio_read_encode(void * __restrict__ opus_buf) {
|
||
// Static buffers persist across calls for better cache locality
|
||
static short SIMD_ALIGN pcm_buffer[1920]; // 960 frames × 2 channels
|
||
static short prev_max_sample = 0; // Previous frame peak for discontinuity detection
|
||
|
||
// Local variables
|
||
unsigned char * __restrict__ out = (unsigned char*)opus_buf;
|
||
int pcm_rc;
|
||
int err = 0;
|
||
int recovery_attempts = 0;
|
||
const int max_recovery_attempts = 3;
|
||
int total_samples;
|
||
short max_sample;
|
||
int nb_bytes;
|
||
|
||
// Prefetch output buffer for write
|
||
SIMD_PREFETCH(out, 1, 3);
|
||
SIMD_PREFETCH(pcm_buffer, 0, 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:
|
||
// Read 960 frames (20ms) from ALSA capture device
|
||
pcm_rc = snd_pcm_readi(pcm_capture_handle, pcm_buffer, frame_size);
|
||
|
||
if (__builtin_expect(pcm_rc < 0, 0)) {
|
||
if (pcm_rc == -EPIPE) {
|
||
recovery_attempts++;
|
||
if (recovery_attempts > max_recovery_attempts) {
|
||
return -1;
|
||
}
|
||
err = snd_pcm_prepare(pcm_capture_handle);
|
||
if (err < 0) {
|
||
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) {
|
||
return 0;
|
||
} else if (pcm_rc == -ESTRPIPE) {
|
||
recovery_attempts++;
|
||
if (recovery_attempts > max_recovery_attempts) {
|
||
return -1;
|
||
}
|
||
int resume_attempts = 0;
|
||
while ((err = snd_pcm_resume(pcm_capture_handle)) == -EAGAIN && resume_attempts < 10) {
|
||
usleep(sleep_microseconds);
|
||
resume_attempts++;
|
||
}
|
||
if (err < 0) {
|
||
err = snd_pcm_prepare(pcm_capture_handle);
|
||
if (err < 0) return -1;
|
||
}
|
||
return 0;
|
||
} else if (pcm_rc == -ENODEV) {
|
||
return -1;
|
||
} else if (pcm_rc == -EIO) {
|
||
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 {
|
||
recovery_attempts++;
|
||
if (recovery_attempts <= 1 && pcm_rc == -EINTR) {
|
||
goto retry_read;
|
||
} else if (recovery_attempts <= 1 && pcm_rc == -EBUSY) {
|
||
usleep(sleep_microseconds / 2);
|
||
goto retry_read;
|
||
}
|
||
return -1;
|
||
}
|
||
}
|
||
|
||
// Zero-pad if we got a short read
|
||
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: skip frames below ~0.15% of maximum volume
|
||
total_samples = frame_size * channels;
|
||
max_sample = simd_find_max_abs_s16(pcm_buffer, total_samples);
|
||
|
||
if (max_sample < 50) {
|
||
prev_max_sample = 0; // Reset discontinuity tracker on silence
|
||
if (trace_logging_enabled) {
|
||
printf("[AUDIO_OUTPUT] jetkvm_audio_read_encode: Silence detected (max=%d), skipping frame\n", max_sample);
|
||
}
|
||
return 0;
|
||
}
|
||
|
||
// Discontinuity detection: reset encoder on abrupt level changes (video seeks)
|
||
// Prevents crackling when audio stream jumps due to video seeking
|
||
if (prev_max_sample > 0) {
|
||
int level_ratio = (max_sample > prev_max_sample * 5) || (prev_max_sample > max_sample * 5);
|
||
if (level_ratio) {
|
||
if (trace_logging_enabled) {
|
||
printf("[AUDIO_OUTPUT] Discontinuity detected (%d→%d), resetting encoder\n", prev_max_sample, max_sample);
|
||
}
|
||
opus_encoder_ctl(encoder, OPUS_RESET_STATE);
|
||
}
|
||
}
|
||
prev_max_sample = max_sample;
|
||
|
||
// Apply 2.5x gain boost to prevent quantization noise at low volumes
|
||
// HDMI audio typically transmitted at -6 to -12dB; boost prevents Opus noise floor artifacts
|
||
simd_scale_volume_s16(pcm_buffer, frame_size * channels, 2.5f);
|
||
|
||
// Encode PCM to Opus (20ms frame → ~200 bytes at 96kbps)
|
||
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 INPUT path (Opus decoder → device speakers)
|
||
* Opens hw:1,0 (USB gadget) or "default" and creates Opus decoder
|
||
* @return 0 on success, -EBUSY if initializing, -1/-2 on errors
|
||
*/
|
||
int jetkvm_audio_playback_init() {
|
||
int err;
|
||
|
||
simd_init_once();
|
||
|
||
if (__sync_bool_compare_and_swap(&playback_initializing, 0, 1) == 0) {
|
||
return -EBUSY;
|
||
}
|
||
|
||
if (playback_initialized) {
|
||
playback_initializing = 0;
|
||
return 0;
|
||
}
|
||
|
||
if (decoder) {
|
||
opus_decoder_destroy(decoder);
|
||
decoder = NULL;
|
||
}
|
||
if (pcm_playback_handle) {
|
||
snd_pcm_close(pcm_playback_handle);
|
||
pcm_playback_handle = NULL;
|
||
}
|
||
|
||
err = safe_alsa_open(&pcm_playback_handle, "hw:1,0", SND_PCM_STREAM_PLAYBACK);
|
||
if (err < 0) {
|
||
err = safe_alsa_open(&pcm_playback_handle, "default", SND_PCM_STREAM_PLAYBACK);
|
||
if (err < 0) {
|
||
playback_initializing = 0;
|
||
return -1;
|
||
}
|
||
}
|
||
|
||
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;
|
||
}
|
||
|
||
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, write to device speakers (INPUT path hot function)
|
||
* Processing pipeline: Opus decode (with FEC) → ALSA playback with error recovery
|
||
* @param opus_buf Encoded Opus packet from client
|
||
* @param opus_size Size of Opus packet in bytes
|
||
* @return >0 = PCM frames written, 0 = frame skipped, -1/-2 = error
|
||
*/
|
||
__attribute__((hot)) int jetkvm_audio_decode_write(void * __restrict__ opus_buf, int opus_size) {
|
||
// Static buffer persists across calls for better cache locality
|
||
static short SIMD_ALIGN pcm_buffer[1920]; // 960 frames × 2 channels
|
||
|
||
// Local variables
|
||
unsigned char * __restrict__ in = (unsigned char*)opus_buf;
|
||
int pcm_frames;
|
||
int pcm_rc;
|
||
int err = 0;
|
||
int recovery_attempts = 0;
|
||
const int max_recovery_attempts = 3;
|
||
|
||
// Prefetch input buffer for read
|
||
SIMD_PREFETCH(in, 0, 3);
|
||
|
||
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;
|
||
}
|
||
|
||
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 packet to PCM (FEC automatically applied if embedded in packet)
|
||
// decode_fec=0 means normal decode (FEC data is used automatically when present)
|
||
pcm_frames = opus_decode(decoder, in, opus_size, pcm_buffer, frame_size, 0);
|
||
|
||
if (__builtin_expect(pcm_frames < 0, 0)) {
|
||
// Decode failed - attempt packet loss concealment using FEC from previous packet
|
||
if (trace_logging_enabled) {
|
||
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Opus decode failed with error %d, attempting packet loss concealment\n", pcm_frames);
|
||
}
|
||
|
||
// decode_fec=1 means use FEC data from the NEXT packet to reconstruct THIS lost packet
|
||
pcm_frames = opus_decode(decoder, NULL, 0, pcm_buffer, frame_size, 1);
|
||
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 decoded PCM to ALSA playback device
|
||
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) {
|
||
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);
|
||
}
|
||
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));
|
||
}
|
||
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) {
|
||
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);
|
||
}
|
||
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));
|
||
}
|
||
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;
|
||
} else if (pcm_rc == -ENODEV) {
|
||
if (trace_logging_enabled) {
|
||
printf("[AUDIO_INPUT] jetkvm_audio_decode_write: Device disconnected (ENODEV) - critical error\n");
|
||
}
|
||
return -2;
|
||
} else if (pcm_rc == -EIO) {
|
||
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) {
|
||
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);
|
||
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 {
|
||
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 device\n", pcm_frames);
|
||
}
|
||
return pcm_frames;
|
||
}
|
||
|
||
// ============================================================================
|
||
// CLEANUP FUNCTIONS
|
||
// ============================================================================
|
||
|
||
/**
|
||
* Close INPUT path (thread-safe with drain)
|
||
*/
|
||
void jetkvm_audio_playback_close() {
|
||
while (playback_initializing) {
|
||
usleep(sleep_microseconds);
|
||
}
|
||
|
||
if (__sync_bool_compare_and_swap(&playback_initialized, 1, 0) == 0) {
|
||
return;
|
||
}
|
||
|
||
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;
|
||
}
|
||
}
|
||
|
||
/**
|
||
* Close OUTPUT path (thread-safe with drain)
|
||
*/
|
||
void jetkvm_audio_capture_close() {
|
||
while (capture_initializing) {
|
||
usleep(sleep_microseconds);
|
||
}
|
||
|
||
if (__sync_bool_compare_and_swap(&capture_initialized, 1, 0) == 0) {
|
||
return;
|
||
}
|
||
|
||
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;
|
||
}
|
||
}
|