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kvm/internal/audio/c/audio.c

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/*
* JetKVM Audio Processing Module
*
* Bidirectional audio processing optimized for ARM NEON SIMD:
* - OUTPUT PATH: TC358743 HDMI or USB Gadget audio → Client speakers
* Pipeline: ALSA hw:0,0 or hw:1,0 capture → SpeexDSP resample → Opus encode (192kbps, FEC enabled)
*
* - INPUT PATH: Client microphone → Device speakers
* Pipeline: Opus decode (with FEC) → ALSA hw:1,0 playback
*
* Key features:
* - ARM NEON SIMD optimization for all audio operations
* - SpeexDSP high-quality resampling (SPEEX_RESAMPLER_QUALITY_DESKTOP)
* - Opus in-band FEC for packet loss resilience
* - S16_LE stereo, 20ms frames at 48kHz (hardware rate auto-negotiated)
* - Direct hardware access with userspace resampling (no ALSA plugin layer)
*/
#include <alsa/asoundlib.h>
#include <opus.h>
#include <speex/speex_resampler.h>
#include <stdio.h>
#include <stdlib.h>
#include <stdbool.h>
#include <string.h>
#include <unistd.h>
#include <errno.h>
#include <sched.h>
#include <time.h>
#include <signal.h>
#include <pthread.h>
#include <stdatomic.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <linux/videodev2.h>
// ARM NEON SIMD optimizations (Cortex-A7 accelerates buffer operations, with scalar fallback)
#include <arm_neon.h>
// TC358743 V4L2 control IDs for audio
#ifndef V4L2_CID_USER_TC35874X_BASE
#define V4L2_CID_USER_TC35874X_BASE (V4L2_CID_USER_BASE + 0x10a0)
#endif
#define TC35874X_CID_AUDIO_SAMPLING_RATE (V4L2_CID_USER_TC35874X_BASE + 0)
// RV1106 (Cortex-A7) has 64-byte cache lines
#define CACHE_LINE_SIZE 64
#define SIMD_ALIGN __attribute__((aligned(16)))
#define CACHE_ALIGN __attribute__((aligned(CACHE_LINE_SIZE)))
#define SIMD_PREFETCH(addr, rw, locality) __builtin_prefetch(addr, rw, locality)
static snd_pcm_t *pcm_capture_handle = NULL; // OUTPUT: TC358743 HDMI audio → client
static snd_pcm_t *pcm_playback_handle = NULL; // INPUT: Client microphone → device speakers
static const char *alsa_capture_device = NULL;
static const char *alsa_playback_device = NULL;
static bool capture_channels_swapped = false;
static OpusEncoder *encoder = NULL;
static OpusDecoder *decoder = NULL;
static SpeexResamplerState *capture_resampler = NULL;
// Audio format - RFC 7587 requires Opus RTP clock rate (not sample rate) to be 48kHz
// The Opus codec itself supports multiple sample rates (8/12/16/24/48 kHz), but the
// RTP timestamp clock must always increment at 48kHz for WebRTC compatibility
static const uint32_t opus_sample_rate = 48000; // RFC 7587: Opus RTP timestamp clock rate (not codec sample rate)
static uint32_t hardware_sample_rate = 48000; // Hardware-negotiated rate (can be 44.1k, 48k, 96k, etc.)
static uint8_t capture_channels = 2; // OUTPUT: Audio source (HDMI or USB) → client (stereo by default)
static uint8_t playback_channels = 1; // INPUT: Client mono mic → device (always mono for USB audio gadget)
static const uint16_t opus_frame_size = 960; // 20ms frames at 48kHz (fixed)
static uint16_t hardware_frame_size = 960; // 20ms frames at hardware rate
// Maximum hardware frame size: 192kHz @ 20ms = 3840 samples/channel
// This is the upper bound for hardware buffer allocation (highest sample rate we support)
#define MAX_HARDWARE_FRAME_SIZE 3840
// Audio initialization error codes
#define ERR_ALSA_OPEN_FAILED -1
#define ERR_ALSA_CONFIG_FAILED -2
#define ERR_RESAMPLER_INIT_FAILED -3
#define ERR_CODEC_INIT_FAILED -4
static uint32_t opus_bitrate = 192000;
static uint8_t opus_complexity = 8;
static uint16_t max_packet_size = 1500;
// Opus encoder configuration constants (see opus_defines.h for full enum values)
#define OPUS_VBR 1 // Variable bitrate mode enabled
#define OPUS_VBR_CONSTRAINT 1 // Constrained VBR maintains bitrate ceiling
#define OPUS_SIGNAL_TYPE 3002 // OPUS_SIGNAL_MUSIC (optimized for music/audio content)
#define OPUS_BANDWIDTH 1105 // OPUS_BANDWIDTH_FULLBAND (0-20kHz frequency range)
#define OPUS_LSB_DEPTH 16 // 16-bit PCM sample depth (S16_LE format)
static uint8_t opus_dtx_enabled = 1;
static uint8_t opus_fec_enabled = 1;
static uint8_t opus_packet_loss_perc = 20; // Default packet loss compensation percentage
static uint8_t buffer_period_count = 24;
static uint32_t sleep_microseconds = 1000;
static uint32_t sleep_milliseconds = 1;
static uint8_t max_attempts_global = 5;
static uint32_t max_backoff_us_global = 500000;
static atomic_int capture_stop_requested = 0;
static atomic_int playback_stop_requested = 0;
// Mutexes protect handle lifecycle and codec operations, NOT the ALSA I/O itself.
// The mutex is temporarily released during snd_pcm_readi/writei to prevent blocking.
// Race conditions are detected via handle pointer comparison after reacquiring the lock.
static pthread_mutex_t capture_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t playback_mutex = PTHREAD_MUTEX_INITIALIZER;
int jetkvm_audio_capture_init();
void jetkvm_audio_capture_close();
int jetkvm_audio_read_encode(void *opus_buf);
int jetkvm_audio_playback_init();
void jetkvm_audio_playback_close();
int jetkvm_audio_decode_write(void *opus_buf, int opus_size);
void update_audio_constants(uint32_t bitrate, uint8_t complexity,
uint32_t sr, uint8_t ch, uint16_t fs, uint16_t max_pkt,
uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff,
uint8_t dtx_enabled, uint8_t fec_enabled, uint8_t buf_periods, uint8_t pkt_loss_perc);
void update_audio_decoder_constants(uint32_t sr, uint8_t ch, uint16_t fs, uint16_t max_pkt,
uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff,
uint8_t buf_periods);
void update_audio_constants(uint32_t bitrate, uint8_t complexity,
uint32_t sr, uint8_t ch, uint16_t fs, uint16_t max_pkt,
uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff,
uint8_t dtx_enabled, uint8_t fec_enabled, uint8_t buf_periods, uint8_t pkt_loss_perc) {
opus_bitrate = (bitrate >= 64000 && bitrate <= 256000) ? bitrate : 192000;
opus_complexity = (complexity <= 10) ? complexity : 5;
capture_channels = (ch == 1 || ch == 2) ? ch : 2;
max_packet_size = max_pkt > 0 ? max_pkt : 1500;
sleep_microseconds = sleep_us > 0 ? sleep_us : 1000;
sleep_milliseconds = sleep_microseconds / 1000;
max_attempts_global = max_attempts > 0 ? max_attempts : 5;
max_backoff_us_global = max_backoff > 0 ? max_backoff : 500000;
opus_dtx_enabled = dtx_enabled ? 1 : 0;
opus_fec_enabled = fec_enabled ? 1 : 0;
buffer_period_count = (buf_periods >= 2 && buf_periods <= 24) ? buf_periods : 12;
opus_packet_loss_perc = (pkt_loss_perc <= 100) ? pkt_loss_perc : 20;
// Note: sr and fs parameters ignored - RFC 7587 requires fixed 48kHz RTP clock rate
// Hardware sample rate conversion is handled by SpeexDSP resampler
}
void update_audio_decoder_constants(uint32_t sr, uint8_t ch, uint16_t fs, uint16_t max_pkt,
uint32_t sleep_us, uint8_t max_attempts, uint32_t max_backoff,
uint8_t buf_periods) {
playback_channels = (ch == 1 || ch == 2) ? ch : 2;
max_packet_size = max_pkt > 0 ? max_pkt : 1500;
sleep_microseconds = sleep_us > 0 ? sleep_us : 1000;
sleep_milliseconds = sleep_microseconds / 1000;
max_attempts_global = max_attempts > 0 ? max_attempts : 5;
max_backoff_us_global = max_backoff > 0 ? max_backoff : 500000;
buffer_period_count = (buf_periods >= 2 && buf_periods <= 24) ? buf_periods : 12;
// Note: sr and fs parameters ignored - decoder uses hardware-negotiated rate
// USB Audio Gadget typically negotiates 48kHz, matching Opus RTP clock (RFC 7587)
}
/**
* Initialize ALSA device names from environment variables
* Must be called before jetkvm_audio_capture_init or jetkvm_audio_playback_init
*
* Device mapping (set via ALSA_CAPTURE_DEVICE/ALSA_PLAYBACK_DEVICE):
* hw:0,0 = TC358743 HDMI audio (direct hardware access, SpeexDSP resampling)
* hw:1,0 = USB Audio Gadget (direct hardware access, SpeexDSP resampling)
*/
static void init_alsa_devices_from_env(void) {
alsa_capture_device = getenv("ALSA_CAPTURE_DEVICE");
if (alsa_capture_device == NULL || alsa_capture_device[0] == '\0') {
alsa_capture_device = "hw:1,0";
}
alsa_playback_device = getenv("ALSA_PLAYBACK_DEVICE");
if (alsa_playback_device == NULL || alsa_playback_device[0] == '\0') {
alsa_playback_device = "hw:1,0";
}
}
// SIMD-OPTIMIZED BUFFER OPERATIONS (ARM NEON)
static inline void simd_clear_samples_s16(short * __restrict__ buffer, uint32_t samples) {
const int16x8_t zero = vdupq_n_s16(0);
uint32_t i = 0;
// Process 16 samples at a time (2x unrolled for better pipeline utilization)
uint32_t simd_samples = samples & ~15U;
for (; i < simd_samples; i += 16) {
vst1q_s16(&buffer[i], zero);
vst1q_s16(&buffer[i + 8], zero);
}
// Handle remaining 8 samples
if (i + 8 <= samples) {
vst1q_s16(&buffer[i], zero);
i += 8;
}
// Scalar: remaining samples
for (; i < samples; i++) {
buffer[i] = 0;
}
}
// INITIALIZATION STATE TRACKING
static volatile sig_atomic_t capture_initializing = 0;
static volatile sig_atomic_t capture_initialized = 0;
static volatile sig_atomic_t playback_initializing = 0;
static volatile sig_atomic_t playback_initialized = 0;
// ALSA UTILITY FUNCTIONS
/**
* Query TC358743 HDMI receiver for detected audio sample rate
* Reads the hardware-detected sample rate from V4L2 control
* @return detected sample rate (44100, 48000, etc.) or 0 if detection fails
*/
static unsigned int get_hdmi_audio_sample_rate(void) {
// TC358743 is a V4L2 subdevice at /dev/v4l-subdev2
int fd = open("/dev/v4l-subdev2", O_RDWR);
if (fd < 0) {
// Distinguish between different failure modes for better diagnostics
if (errno == ENOENT) {
fprintf(stdout, "INFO: TC358743 device not found (USB audio mode or device not present)\n");
} else if (errno == EACCES || errno == EPERM) {
fprintf(stderr, "ERROR: Permission denied accessing TC358743 (/dev/v4l-subdev2)\n");
fprintf(stderr, " Check device permissions or run with appropriate privileges\n");
} else {
fprintf(stderr, "WARNING: Could not open /dev/v4l-subdev2: %s (errno=%d)\n", strerror(errno), errno);
fprintf(stderr, " HDMI audio sample rate detection unavailable, will use 48kHz default\n");
}
fflush(stderr);
fflush(stdout);
return 0;
}
// Use extended controls API for custom V4L2 controls
struct v4l2_ext_control ext_ctrl = {0};
ext_ctrl.id = TC35874X_CID_AUDIO_SAMPLING_RATE;
struct v4l2_ext_controls ext_ctrls = {0};
ext_ctrls.ctrl_class = V4L2_CTRL_CLASS_USER;
ext_ctrls.count = 1;
ext_ctrls.controls = &ext_ctrl;
if (ioctl(fd, VIDIOC_G_EXT_CTRLS, &ext_ctrls) == -1) {
// Provide specific error messages based on errno
if (errno == EINVAL) {
fprintf(stderr, "ERROR: TC358743 sample rate control not supported (driver version mismatch?)\n");
fprintf(stderr, " Ensure kernel driver supports audio_sampling_rate control\n");
} else {
fprintf(stderr, "WARNING: TC358743 ioctl failed: %s (errno=%d)\n", strerror(errno), errno);
fprintf(stderr, " Will use 48kHz default sample rate\n");
}
fflush(stderr);
close(fd);
return 0;
}
close(fd);
unsigned int detected_rate = (unsigned int)ext_ctrl.value;
static unsigned int last_logged_rate = 0; // Track last logged rate to suppress duplicate messages
if (detected_rate == 0) {
if (last_logged_rate != 0) {
fprintf(stdout, "INFO: TC358743 reports 0 Hz (no HDMI signal or audio not detected yet)\n");
fprintf(stdout, " Will use 48kHz default and resample if needed when signal detected\n");
fflush(stdout);
last_logged_rate = 0;
}
return 0;
}
// Validate detected rate is reasonable (log warning only on rate changes)
if (detected_rate < 8000 || detected_rate > 192000) {
if (detected_rate != last_logged_rate) {
fprintf(stderr, "WARNING: TC358743 reported unusual sample rate: %u Hz (expected 32k-192k)\n", detected_rate);
fprintf(stderr, " Using detected rate anyway, but audio may not work correctly\n");
fflush(stderr);
last_logged_rate = detected_rate;
}
}
// Log rate changes and update tracking state to suppress duplicate logging
if (detected_rate != last_logged_rate) {
fprintf(stdout, "INFO: TC358743 detected HDMI audio sample rate: %u Hz\n", detected_rate);
fflush(stdout);
last_logged_rate = detected_rate;
}
return detected_rate;
}
static inline void precise_sleep_us(uint32_t microseconds) {
struct timespec ts = {
.tv_sec = microseconds / 1000000,
.tv_nsec = (microseconds % 1000000) * 1000
};
nanosleep(&ts, NULL);
}
static int safe_alsa_open(snd_pcm_t **handle, const char *device, snd_pcm_stream_t stream) {
uint8_t attempt = 0;
int err;
uint32_t backoff_us = sleep_microseconds;
while (attempt < max_attempts_global) {
err = snd_pcm_open(handle, device, stream, SND_PCM_NONBLOCK);
if (err >= 0) {
// Validate that we can switch to blocking mode
err = snd_pcm_nonblock(*handle, 0);
if (err < 0) {
fprintf(stderr, "ERROR: Failed to set blocking mode on %s: %s\n",
device, snd_strerror(err));
fflush(stderr);
snd_pcm_close(*handle);
*handle = NULL;
return err;
}
return 0;
}
attempt++;
// Apply sleep strategy based on error type
if (err == -EPERM || err == -EACCES) {
precise_sleep_us(backoff_us >> 1); // Shorter wait for permission errors
} else {
precise_sleep_us(backoff_us);
// Exponential backoff for retry-worthy errors
if (err == -EBUSY || err == -EAGAIN || err == -ENODEV || err == -ENOENT) {
backoff_us = (backoff_us < 50000) ? (backoff_us << 1) : 50000;
}
}
}
return err;
}
/**
* Swap stereo channels (L<->R) using ARM NEON SIMD
* Processes 4 frames (8 samples) at a time for optimal performance
* @param buffer Interleaved stereo buffer (L,R,L,R,...)
* @param num_frames Number of stereo frames to swap
*/
static inline void swap_stereo_channels(int16_t *buffer, uint16_t num_frames) {
uint16_t i;
// Process in chunks of 4 frames (8 samples, 128 bits)
for (i = 0; i + 3 < num_frames; i += 4) {
int16x8_t vec = vld1q_s16(&buffer[i * 2]);
int16x8_t swapped = vrev32q_s16(vec);
vst1q_s16(&buffer[i * 2], swapped);
}
// Handle remaining frames with scalar code
for (; i < num_frames; i++) {
int16_t temp = buffer[i * 2];
buffer[i * 2] = buffer[i * 2 + 1];
buffer[i * 2 + 1] = temp;
}
}
/**
* Handle ALSA I/O errors with recovery attempts
* @param handle Pointer to PCM handle to use for recovery operations
* @param valid_handle Pointer to the valid handle to check against (for race detection)
* @param stop_flag Pointer to atomic stop flag
* @param pcm_rc Error code from ALSA I/O operation
* @param recovery_attempts Pointer to uint8_t recovery attempt counter
* @param sleep_ms Milliseconds to sleep during recovery
* @param max_attempts Maximum recovery attempts allowed
* @return Return codes:
* 1 = Retry operation (error was recovered)
* 0 = Skip this frame and continue
* -1 = Fatal error, abort operation
*
* IMPORTANT: This function NEVER unlocks the mutex. The caller is always
* responsible for unlocking after checking the return value. This ensures
* consistent mutex ownership semantics.
*/
static int handle_alsa_error(snd_pcm_t *handle, snd_pcm_t **valid_handle,
atomic_int *stop_flag,
int pcm_rc, uint8_t *recovery_attempts,
uint32_t sleep_ms, uint8_t max_attempts) {
int err;
if (pcm_rc == -EPIPE) {
// Buffer underrun/overrun
(*recovery_attempts)++;
if (*recovery_attempts > max_attempts || handle != *valid_handle) {
return -1;
}
err = snd_pcm_prepare(handle);
if (err < 0) {
if (handle != *valid_handle) {
return -1;
}
snd_pcm_drop(handle);
err = snd_pcm_prepare(handle);
if (err < 0 || handle != *valid_handle) {
return -1;
}
}
return 1; // Retry
} else if (pcm_rc == -EAGAIN) {
// Resource temporarily unavailable
if (handle != *valid_handle) {
return -1;
}
snd_pcm_wait(handle, sleep_ms);
return 1; // Retry
} else if (pcm_rc == -ESTRPIPE) {
// Suspended, need to resume
(*recovery_attempts)++;
if (*recovery_attempts > max_attempts || handle != *valid_handle) {
return -1;
}
uint8_t resume_attempts = 0;
while ((err = snd_pcm_resume(handle)) == -EAGAIN && resume_attempts < 10) {
if (*stop_flag || handle != *valid_handle) {
return -1;
}
snd_pcm_wait(handle, sleep_ms);
resume_attempts++;
}
if (err < 0) {
if (handle != *valid_handle) {
return -1;
}
err = snd_pcm_prepare(handle);
if (err < 0 || handle != *valid_handle) {
return -1;
}
}
return 0; // Skip frame after suspend recovery
} else if (pcm_rc == -ENODEV) {
// Device was removed
return -1;
} else if (pcm_rc == -EIO) {
// I/O error
(*recovery_attempts)++;
if (*recovery_attempts <= max_attempts && handle == *valid_handle) {
snd_pcm_drop(handle);
if (handle != *valid_handle) {
return -1;
}
err = snd_pcm_prepare(handle);
if (err >= 0 && handle == *valid_handle) {
return 1; // Retry
}
}
return -1;
} else {
// Other errors
(*recovery_attempts)++;
if (*recovery_attempts <= 1 && pcm_rc == -EINTR) {
return 1; // Retry on first interrupt
} else if (*recovery_attempts <= 1 && pcm_rc == -EBUSY && handle == *valid_handle) {
snd_pcm_wait(handle, 1);
return 1; // Retry on first busy
}
return -1;
}
}
/**
* Configure ALSA device (S16_LE @ hardware-negotiated rate with optimized buffering)
* @param handle ALSA PCM handle
* @param device_name Device name for logging
* @param num_channels Number of channels (1=mono, 2=stereo)
* @param preferred_rate Preferred sample rate (0 = use default 48kHz)
* @param actual_rate_out Pointer to store the actual hardware-negotiated rate
* @param actual_frame_size_out Pointer to store the actual frame size at hardware rate
* @param channels_swapped_out Pointer to store whether channels are swapped (NULL to ignore)
* @return 0 on success, negative error code on failure
*/
static int configure_alsa_device(snd_pcm_t *handle, const char *device_name, uint8_t num_channels,
unsigned int preferred_rate, unsigned int *actual_rate_out, uint16_t *actual_frame_size_out,
bool *channels_swapped_out) {
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *sw_params;
int err;
snd_pcm_hw_params_alloca(&params);
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) {
fprintf(stderr, "ERROR: %s: Failed to set access mode: %s\n", device_name, snd_strerror(err));
fflush(stderr);
return err;
}
err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
if (err < 0) {
fprintf(stderr, "ERROR: %s: Failed to set format S16_LE: %s\n", device_name, snd_strerror(err));
fflush(stderr);
return err;
}
err = snd_pcm_hw_params_set_channels(handle, params, num_channels);
if (err < 0) {
fprintf(stderr, "ERROR: %s: Failed to set %u channels: %s\n", device_name, num_channels, snd_strerror(err));
fflush(stderr);
return err;
}
// Disable ALSA resampling - we handle it with SpeexDSP
err = snd_pcm_hw_params_set_rate_resample(handle, params, 0);
if (err < 0) {
fprintf(stderr, "ERROR: %s: Failed to disable ALSA resampling: %s\n", device_name, snd_strerror(err));
fflush(stderr);
return err;
}
// Use preferred rate if specified, otherwise default to 48kHz
unsigned int requested_rate = (preferred_rate > 0) ? preferred_rate : opus_sample_rate;
err = snd_pcm_hw_params_set_rate_near(handle, params, &requested_rate, 0);
if (err < 0) return err;
// Calculate frame size for this hardware rate (20ms)
uint16_t hw_frame_size = requested_rate / 50;
snd_pcm_uframes_t period_size = hw_frame_size;
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 = period_size * buffer_period_count;
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;
unsigned int negotiated_rate = 0;
err = snd_pcm_hw_params_get_rate(params, &negotiated_rate, 0);
if (err < 0) return err;
fprintf(stdout, "INFO: %s: Hardware negotiated %u Hz (Opus uses %u Hz with SpeexDSP resampling)\n",
device_name, negotiated_rate, opus_sample_rate);
fflush(stdout);
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;
err = snd_pcm_prepare(handle);
if (err < 0) return err;
if (num_channels == 2 && channels_swapped_out) {
snd_pcm_chmap_t *chmap = snd_pcm_get_chmap(handle);
if (chmap != NULL) {
if (chmap->channels != 2) {
fprintf(stderr, "WARN: %s: Expected 2 channels but channel map has %u\n",
device_name, chmap->channels);
fflush(stderr);
} else if (chmap->pos[0] == SND_CHMAP_UNKNOWN || chmap->pos[1] == SND_CHMAP_UNKNOWN) {
fprintf(stderr, "WARN: %s: Channel map positions are unknown, cannot detect swap\n",
device_name);
fflush(stderr);
} else {
bool is_swapped = (chmap->pos[0] == SND_CHMAP_FR && chmap->pos[1] == SND_CHMAP_FL);
if (is_swapped) {
fprintf(stdout, "INFO: %s: Hardware reports swapped channel map (R,L instead of L,R)\n",
device_name);
fflush(stdout);
}
*channels_swapped_out = is_swapped;
}
free(chmap);
} else {
fprintf(stdout, "INFO: %s: Channel map not available, assuming standard L/R order\n",
device_name);
fflush(stdout);
}
}
if (actual_rate_out) *actual_rate_out = negotiated_rate;
if (actual_frame_size_out) {
// Calculate actual frame size based on negotiated rate (20ms frames)
*actual_frame_size_out = negotiated_rate / 50;
}
return 0;
}
// AUDIO OUTPUT PATH FUNCTIONS (TC358743 HDMI Audio → Client Speakers)
/**
* Initialize OUTPUT path (HDMI or USB Gadget audio capture → Opus encoder)
* Opens ALSA capture device from ALSA_CAPTURE_DEVICE env (default: hw:1,0, set to hw:0,0 for HDMI)
* and creates Opus encoder with optimized settings
* @return 0 on success, -EBUSY if initializing, or:
* ERR_ALSA_OPEN_FAILED (-1), ERR_ALSA_CONFIG_FAILED (-2),
* ERR_RESAMPLER_INIT_FAILED (-3), ERR_CODEC_INIT_FAILED (-4)
*/
int jetkvm_audio_capture_init() {
int err;
init_alsa_devices_from_env();
if (__sync_bool_compare_and_swap(&capture_initializing, 0, 1) == 0) {
return -EBUSY;
}
if (capture_initialized) {
capture_initializing = 0;
return 0;
}
if (encoder != NULL || pcm_capture_handle != NULL) {
capture_initialized = 0;
atomic_store(&capture_stop_requested, 1);
__sync_synchronize();
if (pcm_capture_handle) {
snd_pcm_drop(pcm_capture_handle);
}
pthread_mutex_lock(&capture_mutex);
if (encoder) {
opus_encoder_destroy(encoder);
encoder = NULL;
}
if (pcm_capture_handle) {
snd_pcm_close(pcm_capture_handle);
pcm_capture_handle = NULL;
}
pthread_mutex_unlock(&capture_mutex);
atomic_store(&capture_stop_requested, 0);
}
err = safe_alsa_open(&pcm_capture_handle, alsa_capture_device, SND_PCM_STREAM_CAPTURE);
if (err < 0) {
fprintf(stderr, "Failed to open ALSA capture device %s: %s\n",
alsa_capture_device, snd_strerror(err));
fflush(stderr);
atomic_store(&capture_stop_requested, 0);
capture_initializing = 0;
return ERR_ALSA_OPEN_FAILED;
}
// Query TC358743 for detected HDMI audio sample rate
unsigned int preferred_rate = get_hdmi_audio_sample_rate();
if (preferred_rate > 0) {
fprintf(stdout, "INFO: Using TC358743 detected sample rate: %u Hz\n", preferred_rate);
} else {
fprintf(stdout, "INFO: TC358743 sample rate not detected, using default 48kHz\n");
preferred_rate = 0; // Will default to 48kHz
}
fflush(stdout);
unsigned int actual_rate = 0;
uint16_t actual_frame_size = 0;
bool channels_swapped = false;
err = configure_alsa_device(pcm_capture_handle, "capture", capture_channels, preferred_rate, &actual_rate, &actual_frame_size, &channels_swapped);
if (err < 0) {
snd_pcm_t *handle = pcm_capture_handle;
pcm_capture_handle = NULL;
if (handle) {
snd_pcm_close(handle);
}
atomic_store(&capture_stop_requested, 0);
capture_initializing = 0;
return ERR_ALSA_CONFIG_FAILED;
}
capture_channels_swapped = channels_swapped;
hardware_sample_rate = actual_rate;
hardware_frame_size = actual_frame_size;
if (hardware_frame_size > MAX_HARDWARE_FRAME_SIZE) {
fprintf(stderr, "ERROR: capture: Hardware frame size %u exceeds buffer capacity %u\n",
hardware_frame_size, MAX_HARDWARE_FRAME_SIZE);
fflush(stderr);
snd_pcm_t *handle = pcm_capture_handle;
pcm_capture_handle = NULL;
if (handle) {
snd_pcm_close(handle);
}
atomic_store(&capture_stop_requested, 0);
capture_initializing = 0;
return ERR_CODEC_INIT_FAILED;
}
// Clean up any existing resampler before creating new one (prevents memory leak on re-init)
if (capture_resampler) {
speex_resampler_destroy(capture_resampler);
capture_resampler = NULL;
}
// Initialize Speex resampler if hardware rate != 48kHz
if (hardware_sample_rate != opus_sample_rate) {
int speex_err = 0;
capture_resampler = speex_resampler_init(capture_channels, hardware_sample_rate,
opus_sample_rate, SPEEX_RESAMPLER_QUALITY_DESKTOP,
&speex_err);
if (!capture_resampler || speex_err != 0) {
fprintf(stderr, "ERROR: capture: Failed to create SpeexDSP resampler (%u Hz → %u Hz): %d\n",
hardware_sample_rate, opus_sample_rate, speex_err);
fflush(stderr);
snd_pcm_t *handle = pcm_capture_handle;
pcm_capture_handle = NULL;
if (handle) {
snd_pcm_close(handle);
}
atomic_store(&capture_stop_requested, 0);
capture_initializing = 0;
return ERR_RESAMPLER_INIT_FAILED;
}
}
fprintf(stdout, "INFO: capture: Initializing Opus encoder %sat (%u Hz → %u Hz), %u channels, frame size %u\n",
hardware_sample_rate == opus_sample_rate ? "" : "SpeexDSP resampled ",
hardware_sample_rate, opus_sample_rate,
capture_channels, opus_frame_size);
fflush(stdout);
int opus_err = 0;
encoder = opus_encoder_create(opus_sample_rate, capture_channels, OPUS_APPLICATION_AUDIO, &opus_err);
if (!encoder || opus_err != OPUS_OK) {
if (capture_resampler) {
speex_resampler_destroy(capture_resampler);
capture_resampler = NULL;
}
if (pcm_capture_handle) {
snd_pcm_t *handle = pcm_capture_handle;
pcm_capture_handle = NULL;
if (handle) {
snd_pcm_close(handle);
}
}
atomic_store(&capture_stop_requested, 0);
capture_initializing = 0;
return ERR_CODEC_INIT_FAILED;
}
// Critical settings that must succeed for WebRTC compliance
#define OPUS_CTL_CRITICAL(call, desc) do { \
int _err = call; \
if (_err != OPUS_OK) { \
fprintf(stderr, "ERROR: capture: Failed to set " desc ": %s\n", opus_strerror(_err)); \
fflush(stderr); \
opus_encoder_destroy(encoder); \
encoder = NULL; \
if (capture_resampler) { \
speex_resampler_destroy(capture_resampler); \
capture_resampler = NULL; \
} \
snd_pcm_t *handle = pcm_capture_handle; \
pcm_capture_handle = NULL; \
if (handle) { \
snd_pcm_close(handle); \
} \
atomic_store(&capture_stop_requested, 0); \
capture_initializing = 0; \
return ERR_CODEC_INIT_FAILED; \
} \
} while(0)
// Non-critical settings that can fail without breaking functionality
#define OPUS_CTL_WARN(call, desc) do { \
int _err = call; \
if (_err != OPUS_OK) { \
fprintf(stderr, "WARN: capture: Failed to set " desc ": %s (non-critical, continuing)\n", opus_strerror(_err)); \
fflush(stderr); \
} \
} while(0)
// Critical: Bitrate, VBR mode, FEC are required for proper WebRTC operation
OPUS_CTL_CRITICAL(opus_encoder_ctl(encoder, OPUS_SET_BITRATE(opus_bitrate)), "bitrate");
OPUS_CTL_CRITICAL(opus_encoder_ctl(encoder, OPUS_SET_VBR(OPUS_VBR)), "VBR mode");
OPUS_CTL_CRITICAL(opus_encoder_ctl(encoder, OPUS_SET_VBR_CONSTRAINT(OPUS_VBR_CONSTRAINT)), "VBR constraint");
OPUS_CTL_CRITICAL(opus_encoder_ctl(encoder, OPUS_SET_INBAND_FEC(opus_fec_enabled)), "FEC");
// Non-critical: These optimize quality/performance but aren't required
OPUS_CTL_WARN(opus_encoder_ctl(encoder, OPUS_SET_COMPLEXITY(opus_complexity)), "complexity");
OPUS_CTL_WARN(opus_encoder_ctl(encoder, OPUS_SET_SIGNAL(OPUS_SIGNAL_TYPE)), "signal type");
OPUS_CTL_WARN(opus_encoder_ctl(encoder, OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH)), "bandwidth");
OPUS_CTL_WARN(opus_encoder_ctl(encoder, OPUS_SET_DTX(opus_dtx_enabled)), "DTX");
OPUS_CTL_WARN(opus_encoder_ctl(encoder, OPUS_SET_LSB_DEPTH(OPUS_LSB_DEPTH)), "LSB depth");
OPUS_CTL_WARN(opus_encoder_ctl(encoder, OPUS_SET_PACKET_LOSS_PERC(opus_packet_loss_perc)), "packet loss percentage");
#undef OPUS_CTL_CRITICAL
#undef OPUS_CTL_WARN
capture_initialized = 1;
atomic_store(&capture_stop_requested, 0);
capture_initializing = 0;
return 0;
}
__attribute__((hot)) int jetkvm_audio_read_encode(void * __restrict__ opus_buf) {
// Two buffers: hardware buffer + resampled buffer (at 48kHz)
static short CACHE_ALIGN pcm_hw_buffer[MAX_HARDWARE_FRAME_SIZE * 2]; // Max hardware rate * stereo
static short CACHE_ALIGN pcm_opus_buffer[960 * 2]; // 48kHz @ 20ms * 2 channels
static uint16_t sample_rate_check_counter = 0;
unsigned char * __restrict__ out = (unsigned char*)opus_buf;
int32_t pcm_rc, nb_bytes;
uint8_t recovery_attempts = 0;
const uint8_t max_recovery_attempts = 3;
if (__builtin_expect(atomic_load(&capture_stop_requested), 0)) {
return -1;
}
SIMD_PREFETCH(out, 1, 0);
SIMD_PREFETCH(pcm_hw_buffer, 0, 0);
SIMD_PREFETCH(pcm_hw_buffer + 64, 0, 1);
pthread_mutex_lock(&capture_mutex);
if (__builtin_expect(!capture_initialized || !pcm_capture_handle || !encoder || !opus_buf, 0)) {
pthread_mutex_unlock(&capture_mutex);
return -1;
}
retry_read:
if (__builtin_expect(atomic_load(&capture_stop_requested), 0)) {
pthread_mutex_unlock(&capture_mutex);
return -1;
}
snd_pcm_t *handle = pcm_capture_handle;
pthread_mutex_unlock(&capture_mutex);
pcm_rc = snd_pcm_readi(handle, pcm_hw_buffer, hardware_frame_size);
pthread_mutex_lock(&capture_mutex);
if (handle != pcm_capture_handle || atomic_load(&capture_stop_requested)) {
pthread_mutex_unlock(&capture_mutex);
return -1;
}
if (__builtin_expect(pcm_rc < 0, 0)) {
int err_result = handle_alsa_error(handle, &pcm_capture_handle, &capture_stop_requested,
pcm_rc, &recovery_attempts,
sleep_milliseconds, max_recovery_attempts);
if (err_result == 1) {
// Recovery successful, retry (mutex still held)
goto retry_read;
} else {
// Fatal error or skip frame (err_result == -1 or 0)
pthread_mutex_unlock(&capture_mutex);
return (err_result == 0) ? 0 : -1;
}
}
// Periodic sample rate change detection (every 50 frames = ~1 second)
if (__builtin_expect(++sample_rate_check_counter >= 50, 0)) {
sample_rate_check_counter = 0;
unsigned int current_rate = get_hdmi_audio_sample_rate();
if (current_rate != 0 && current_rate != hardware_sample_rate) {
fprintf(stderr, "ERROR: capture: HDMI sample rate changed from %u to %u Hz\n",
hardware_sample_rate, current_rate);
fprintf(stderr, " Triggering reconnection for automatic reconfiguration\n");
fflush(stderr);
pthread_mutex_unlock(&capture_mutex);
return -1;
}
}
if (__builtin_expect(pcm_rc < hardware_frame_size, 0)) {
uint32_t remaining_samples = (hardware_frame_size - pcm_rc) * capture_channels;
simd_clear_samples_s16(&pcm_hw_buffer[pcm_rc * capture_channels], remaining_samples);
}
if (capture_channels_swapped) {
swap_stereo_channels(pcm_hw_buffer, hardware_frame_size);
}
short *pcm_to_encode;
if (capture_resampler) {
spx_uint32_t in_len = hardware_frame_size;
spx_uint32_t out_len = opus_frame_size;
int res_err = speex_resampler_process_interleaved_int(capture_resampler,
pcm_hw_buffer, &in_len,
pcm_opus_buffer, &out_len);
if (res_err != 0 || out_len != opus_frame_size) {
fprintf(stderr, "ERROR: capture: Resampling failed (err=%d, out_len=%u, expected=%u)\n",
res_err, out_len, opus_frame_size);
fflush(stderr);
pthread_mutex_unlock(&capture_mutex);
return -1;
}
pcm_to_encode = pcm_opus_buffer;
} else {
pcm_to_encode = pcm_hw_buffer;
}
OpusEncoder *enc = encoder;
if (!enc) {
pthread_mutex_unlock(&capture_mutex);
return -1;
}
nb_bytes = opus_encode(enc, pcm_to_encode, opus_frame_size, out, max_packet_size);
if (__builtin_expect(nb_bytes < 0, 0)) {
fprintf(stderr, "ERROR: capture: Opus encoding failed: %s\n", opus_strerror(nb_bytes));
fflush(stderr);
}
pthread_mutex_unlock(&capture_mutex);
return nb_bytes;
}
// AUDIO INPUT PATH FUNCTIONS (Client Microphone → Device Speakers)
/**
* Initialize INPUT path (Opus decoder → device speakers)
* Opens ALSA playback device from ALSA_PLAYBACK_DEVICE env (default: hw:1,0)
* and creates Opus decoder. Returns immediately on device open failure (no fallback).
* @return 0 on success, -EBUSY if initializing, or:
* ERR_ALSA_OPEN_FAILED (-1), ERR_ALSA_CONFIG_FAILED (-2), ERR_CODEC_INIT_FAILED (-4)
*/
int jetkvm_audio_playback_init() {
int err;
init_alsa_devices_from_env();
if (__sync_bool_compare_and_swap(&playback_initializing, 0, 1) == 0) {
return -EBUSY;
}
if (playback_initialized) {
playback_initializing = 0;
return 0;
}
if (decoder != NULL || pcm_playback_handle != NULL) {
playback_initialized = 0;
atomic_store(&playback_stop_requested, 1);
__sync_synchronize();
if (pcm_playback_handle) {
snd_pcm_drop(pcm_playback_handle);
}
pthread_mutex_lock(&playback_mutex);
if (decoder) {
opus_decoder_destroy(decoder);
decoder = NULL;
}
if (pcm_playback_handle) {
snd_pcm_close(pcm_playback_handle);
pcm_playback_handle = NULL;
}
pthread_mutex_unlock(&playback_mutex);
atomic_store(&playback_stop_requested, 0);
}
err = safe_alsa_open(&pcm_playback_handle, alsa_playback_device, SND_PCM_STREAM_PLAYBACK);
if (err < 0) {
fprintf(stderr, "Failed to open ALSA playback device %s: %s\n",
alsa_playback_device, snd_strerror(err));
fflush(stderr);
atomic_store(&playback_stop_requested, 0);
playback_initializing = 0;
return ERR_ALSA_OPEN_FAILED;
}
unsigned int actual_rate = 0;
uint16_t actual_frame_size = 0;
err = configure_alsa_device(pcm_playback_handle, "playback", playback_channels, 0, &actual_rate, &actual_frame_size, NULL);
if (err < 0) {
snd_pcm_t *handle = pcm_playback_handle;
pcm_playback_handle = NULL;
if (handle) {
snd_pcm_close(handle);
}
atomic_store(&playback_stop_requested, 0);
playback_initializing = 0;
return ERR_ALSA_CONFIG_FAILED;
}
fprintf(stdout, "INFO: playback: Initializing Opus decoder at %u Hz, %u channels, frame size %u\n",
actual_rate, playback_channels, actual_frame_size);
fflush(stdout);
int opus_err = 0;
decoder = opus_decoder_create(actual_rate, playback_channels, &opus_err);
if (!decoder || opus_err != OPUS_OK) {
snd_pcm_t *handle = pcm_playback_handle;
pcm_playback_handle = NULL;
if (handle) {
snd_pcm_close(handle);
}
atomic_store(&playback_stop_requested, 0);
playback_initializing = 0;
return ERR_CODEC_INIT_FAILED;
}
playback_initialized = 1;
atomic_store(&playback_stop_requested, 0);
playback_initializing = 0;
return 0;
}
__attribute__((hot)) int jetkvm_audio_decode_write(void * __restrict__ opus_buf, int32_t opus_size) {
static short CACHE_ALIGN pcm_buffer[960 * 2]; // Cache-aligned
unsigned char * __restrict__ in = (unsigned char*)opus_buf;
int32_t pcm_frames, pcm_rc;
uint8_t recovery_attempts = 0;
const uint8_t max_recovery_attempts = 3;
// Validate inputs before acquiring mutex to reduce lock contention
if (__builtin_expect(!opus_buf || opus_size <= 0 || opus_size > max_packet_size, 0)) {
return -1;
}
if (__builtin_expect(atomic_load(&playback_stop_requested), 0)) {
return -1;
}
SIMD_PREFETCH(in, 0, 0);
pthread_mutex_lock(&playback_mutex);
if (__builtin_expect(!playback_initialized || !pcm_playback_handle || !decoder, 0)) {
pthread_mutex_unlock(&playback_mutex);
return -1;
}
OpusDecoder *dec = decoder;
if (!dec) {
pthread_mutex_unlock(&playback_mutex);
return -1;
}
pcm_frames = opus_decode(dec, in, opus_size, pcm_buffer, opus_frame_size, 0);
if (__builtin_expect(pcm_frames < 0, 0)) {
// Initial decode failed, try Forward Error Correction from previous packets
fprintf(stderr, "WARN: playback: Opus decode failed (%d), attempting FEC recovery\n", pcm_frames);
fflush(stderr);
pcm_frames = opus_decode(dec, NULL, 0, pcm_buffer, opus_frame_size, 1);
if (pcm_frames < 0) {
fprintf(stderr, "ERROR: playback: FEC recovery also failed (%d), dropping frame\n", pcm_frames);
fflush(stderr);
pthread_mutex_unlock(&playback_mutex);
return -1;
}
if (pcm_frames > 0) {
fprintf(stdout, "INFO: playback: FEC recovered %d frames\n", pcm_frames);
fflush(stdout);
} else {
pthread_mutex_unlock(&playback_mutex);
return 0; // FEC returned no frames, nothing to write
}
}
if (__builtin_expect(pcm_frames <= 0, 0)) {
pthread_mutex_unlock(&playback_mutex);
return 0; // Nothing to write
}
retry_write:
if (__builtin_expect(atomic_load(&playback_stop_requested), 0)) {
pthread_mutex_unlock(&playback_mutex);
return -1;
}
snd_pcm_t *handle = pcm_playback_handle;
pthread_mutex_unlock(&playback_mutex);
pcm_rc = snd_pcm_writei(handle, pcm_buffer, pcm_frames);
pthread_mutex_lock(&playback_mutex);
if (handle != pcm_playback_handle || atomic_load(&playback_stop_requested)) {
pthread_mutex_unlock(&playback_mutex);
return -1;
}
if (__builtin_expect(pcm_rc < 0, 0)) {
int err_result = handle_alsa_error(handle, &pcm_playback_handle, &playback_stop_requested,
pcm_rc, &recovery_attempts,
sleep_milliseconds, max_recovery_attempts);
if (err_result == 1) {
// Recovery successful, retry (mutex still held)
goto retry_write;
} else {
// Fatal error or skip frame (err_result == -1 or 0)
pthread_mutex_unlock(&playback_mutex);
return (err_result == 0) ? 0 : -2;
}
}
pthread_mutex_unlock(&playback_mutex);
return pcm_frames;
}
// CLEANUP FUNCTIONS
/**
* Close audio stream (shared cleanup logic for capture and playback)
* @param stop_requested Pointer to stop flag
* @param initializing Pointer to initializing flag
* @param initialized Pointer to initialized flag
* @param mutex Mutex to protect cleanup
* @param pcm_handle Pointer to PCM handle
* @param codec Pointer to codec (encoder or decoder)
* @param destroy_codec Function to destroy the codec
*/
typedef void (*codec_destroy_fn)(void*);
static void close_audio_stream(atomic_int *stop_requested, volatile int *initializing,
volatile int *initialized, pthread_mutex_t *mutex,
snd_pcm_t **pcm_handle, void **codec,
codec_destroy_fn destroy_codec) {
atomic_store(stop_requested, 1);
while (*initializing) {
sched_yield();
}
if (__sync_bool_compare_and_swap(initialized, 1, 0) == 0) {
atomic_store(stop_requested, 0);
return;
}
struct timespec short_delay = { .tv_sec = 0, .tv_nsec = 5000000 };
nanosleep(&short_delay, NULL);
pthread_mutex_lock(mutex);
snd_pcm_t *handle_to_close = *pcm_handle;
void *codec_to_destroy = *codec;
*pcm_handle = NULL;
*codec = NULL;
// Clean up resampler inside mutex to prevent race with encoding thread
if (mutex == &capture_mutex && capture_resampler) {
SpeexResamplerState *res = capture_resampler;
capture_resampler = NULL;
speex_resampler_destroy(res);
}
pthread_mutex_unlock(mutex);
if (handle_to_close) {
snd_pcm_drop(handle_to_close);
snd_pcm_close(handle_to_close);
}
if (codec_to_destroy) {
destroy_codec(codec_to_destroy);
}
atomic_store(stop_requested, 0);
}
void jetkvm_audio_playback_close() {
close_audio_stream(&playback_stop_requested, &playback_initializing,
&playback_initialized, &playback_mutex,
&pcm_playback_handle, (void**)&decoder,
(codec_destroy_fn)opus_decoder_destroy);
}
void jetkvm_audio_capture_close() {
close_audio_stream(&capture_stop_requested, &capture_initializing,
&capture_initialized, &capture_mutex,
&pcm_capture_handle, (void**)&encoder,
(codec_destroy_fn)opus_encoder_destroy);
}
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