mirror of https://github.com/jetkvm/kvm.git
[WIP] Updates: simplify audio system
This commit is contained in:
Alex P
parent
753c613708
commit
01719e01dd
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@ -14,27 +14,13 @@
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#include <unistd.h>
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#include <errno.h>
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// ARM NEON SIMD support for Cortex-A7
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#ifdef __ARM_NEON
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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_ENABLED 1
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#else
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#define SIMD_ENABLED 0
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#endif
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// Performance optimization flags
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static int trace_logging_enabled = 0; // Enable detailed trace logging
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// SIMD feature detection and optimization macros
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#if SIMD_ENABLED
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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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#else
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#define SIMD_ALIGN
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#define SIMD_PREFETCH(addr, rw, locality)
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#endif
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// SIMD initialization and feature detection
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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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@ -140,10 +126,9 @@ void set_trace_logging(int enabled) {
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}
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// ============================================================================
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// SIMD-OPTIMIZED BUFFER OPERATIONS
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// SIMD-OPTIMIZED BUFFER OPERATIONS (ARM NEON)
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// ============================================================================
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#if SIMD_ENABLED
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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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@ -414,85 +399,38 @@ static inline void simd_deinterleave_stereo_s16(const short *interleaved, short
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}
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}
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#else
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// Fallback implementations for non-SIMD builds
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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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/**
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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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memset(buffer, 0, samples * sizeof(short));
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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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for (int i = 0; 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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// 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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}
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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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// 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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for (int i = 0; i < count; i++) {
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samples[i] = (short)(samples[i] * volume);
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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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static inline void simd_swap_endian_s16(short *samples, int count) {
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for (int i = 0; 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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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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for (int i = 0; i < count; i++) {
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output[i] = (float)input[i] * scale;
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}
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return max_sample;
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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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for (int i = 0; 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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static inline void simd_mono_to_stereo_s16(const short *mono, short *stereo, int frames) {
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for (int i = 0; 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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static inline void simd_stereo_to_mono_s16(const short *stereo, short *mono, int frames) {
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for (int i = 0; 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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static inline void simd_apply_stereo_balance_s16(short *stereo, int frames, float balance) {
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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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for (int i = 0; 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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static inline void simd_deinterleave_stereo_s16(const short *interleaved, short *left,
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short *right, int frames) {
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for (int i = 0; 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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#endif
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// ============================================================================
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// INITIALIZATION STATE TRACKING
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// ============================================================================
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@ -870,29 +808,23 @@ retry_read:
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simd_clear_samples_s16(&pcm_buffer[pcm_rc * channels], remaining_samples);
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}
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// Silence detection: check if all samples are below threshold
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// Threshold: 100 = ~0.3% of max volume (very quiet)
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const short silence_threshold = 100;
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// Silence detection using SIMD-optimized max peak detection
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// Find the maximum absolute sample value in the frame
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int total_samples = frame_size * channels;
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int is_silence = 1;
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for (int i = 0; i < total_samples; i++) {
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short abs_sample = pcm_buffer[i] < 0 ? -pcm_buffer[i] : pcm_buffer[i];
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if (abs_sample > silence_threshold) {
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is_silence = 0;
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break;
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}
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}
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short max_sample = simd_find_max_abs_s16(pcm_buffer, total_samples);
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// If silence detected, return 0 to skip sending this frame
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if (is_silence) {
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// If max peak is below threshold, consider it silence
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// Threshold: 50 = ~0.15% of max volume (very quiet background noise)
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if (max_sample < 50) {
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if (trace_logging_enabled) {
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printf("[AUDIO_OUTPUT] jetkvm_audio_read_encode: Silence detected, skipping frame\n");
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printf("[AUDIO_OUTPUT] jetkvm_audio_read_encode: Silence detected (max=%d), skipping frame\n", max_sample);
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}
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return 0;
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}
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// Apply 4x gain boost to fix quantization noise on transients at normal volumes to prevent crackling issues
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simd_scale_volume_s16(pcm_buffer, frame_size * channels, 4.0f);
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// Apply 5x gain boost to fix quantization noise on transients at normal volumes to prevent crackling issues
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// This allows comfortable listening at low remote volumes (10-40% range)
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simd_scale_volume_s16(pcm_buffer, frame_size * channels, 5.0f);
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int nb_bytes = opus_encode(encoder, pcm_buffer, frame_size, out, max_packet_size);
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