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// This file is part of the 64k demo project.
// It implements the production audio backend using miniaudio.
// Moved from audio.cc to enable backend abstraction for testing.
#include "miniaudio_backend.h"
#include "audio.h"
#include "ring_buffer.h"
#include "util/debug.h"
#include <stdio.h>
#include <stdlib.h> // for abort()
// Static callback for miniaudio (C API requirement)
void MiniaudioBackend::audio_callback(ma_device* pDevice, void* pOutput,
const void* pInput,
ma_uint32 frameCount) {
(void)pInput;
#if defined(DEBUG_LOG_AUDIO)
// Validate callback parameters
static ma_uint32 last_frameCount = 0;
static int callback_reentry = 0;
static double last_time = 0.0;
static int timing_initialized = 0;
static uint64_t total_frames_requested = 0;
static uint64_t callback_number = 0;
callback_number++;
total_frames_requested += frameCount;
// Track timing
struct timespec ts;
clock_gettime(CLOCK_MONOTONIC, &ts);
double now = ts.tv_sec + ts.tv_nsec / 1000000000.0;
if (timing_initialized) {
double delta = (now - last_time) * 1000.0; // ms
double expected = ((double)frameCount / pDevice->sampleRate) * 1000.0;
double jitter = delta - expected;
// Enhanced logging: Log first 20 callbacks in detail, then periodic summary
if (callback_number <= 20 || callback_number % 50 == 0) {
const double elapsed_by_frames = (double)total_frames_requested / pDevice->sampleRate * 1000.0;
const double elapsed_by_time = now * 1000.0; // Convert to ms
DEBUG_AUDIO("[CB#%llu] frameCount=%u, Delta=%.2fms, Expected=%.2fms, Jitter=%.2fms, "
"TotalFrames=%llu (%.1fms), TotalTime=%.1fms, Drift=%.2fms\n",
callback_number, frameCount, delta, expected, jitter,
total_frames_requested, elapsed_by_frames, elapsed_by_time,
elapsed_by_time - elapsed_by_frames);
}
// Detect large timing anomalies (>5ms off from expected)
if (fabs(jitter) > 5.0) {
DEBUG_AUDIO("[TIMING ANOMALY] CB#%llu Delta=%.2fms, Expected=%.2fms, Jitter=%.2fms\n",
callback_number, delta, expected, jitter);
}
}
last_time = now;
timing_initialized = 1;
// Check for re-entrant calls
if (callback_reentry > 0) {
DEBUG_AUDIO("FATAL: Callback re-entered! depth=%d\n", callback_reentry);
abort();
}
callback_reentry++;
// Check if frameCount changed unexpectedly
if (last_frameCount != 0 && frameCount != last_frameCount) {
DEBUG_AUDIO("WARNING: frameCount changed! was=%u, now=%u\n",
last_frameCount, frameCount);
}
last_frameCount = frameCount;
// Validate device state
if (!pDevice || pDevice->sampleRate == 0) {
DEBUG_AUDIO("FATAL: Invalid device in callback!\n");
abort();
}
// Check actual sample rate matches our expectation
if (pDevice->sampleRate != 32000) {
static int rate_warning = 0;
if (rate_warning++ == 0) {
DEBUG_AUDIO("WARNING: Device sample rate is %u, not 32000! Resampling may occur.\n",
pDevice->sampleRate);
}
}
#endif /* defined(DEBUG_LOG_AUDIO) */
float* fOutput = (float*)pOutput;
// BOUNDS CHECK: Sanity check on frameCount
if (frameCount > 8192 || frameCount == 0) {
fprintf(stderr, "AUDIO CALLBACK ERROR: frameCount=%u (unreasonable!)\n", frameCount);
abort();
}
// Read from ring buffer instead of calling synth directly
AudioRingBuffer* ring_buffer = audio_get_ring_buffer();
if (ring_buffer != nullptr) {
const int samples_to_read = (int)frameCount * 2; // Stereo
#if defined(DEBUG_LOG_RING_BUFFER)
// Track buffer level and detect drops
static int min_available = 99999;
const int available = ring_buffer->available_read();
if (available < min_available) {
min_available = available;
DEBUG_RING_BUFFER("[BUFFER] CB#%llu NEW MIN: available=%d (%.1fms)\n",
callback_number, available, (float)available / (32000.0f * 2.0f) * 1000.0f);
}
// Log buffer state for first 20 callbacks and periodically
if (callback_number <= 20 || callback_number % 50 == 0) {
DEBUG_RING_BUFFER("[BUFFER] CB#%llu requested=%d, available=%d (%.1fms), min=%d\n",
callback_number, samples_to_read, available,
(float)available / (32000.0f * 2.0f) * 1000.0f, min_available);
}
// CRITICAL: Verify we have enough samples
if (available < samples_to_read) {
DEBUG_RING_BUFFER("[BUFFER UNDERRUN] CB#%llu requested=%d, available=%d, SHORT=%d\n",
callback_number, samples_to_read, available, samples_to_read - available);
}
#endif /* defined(DEBUG_LOG_RING_BUFFER) */
const int actually_read = ring_buffer->read(fOutput, samples_to_read);
#if defined(DEBUG_LOG_RING_BUFFER)
if (actually_read < samples_to_read) {
DEBUG_RING_BUFFER("[PARTIAL READ] CB#%llu requested=%d, got=%d, padded=%d with silence\n",
callback_number, samples_to_read, actually_read, samples_to_read - actually_read);
}
#endif /* defined(DEBUG_LOG_RING_BUFFER) */
}
#if defined(DEBUG_LOG_AUDIO)
// Clear reentry flag
callback_reentry--;
#endif /* defined(DEBUG_LOG_AUDIO) */
}
MiniaudioBackend::MiniaudioBackend() : initialized_(false) {
}
MiniaudioBackend::~MiniaudioBackend() {
if (initialized_) {
shutdown();
}
}
void MiniaudioBackend::init() {
if (initialized_) {
return;
}
ma_device_config config = ma_device_config_init(ma_device_type_playback);
config.playback.format = ma_format_f32;
config.playback.channels = 2;
config.sampleRate = 32000;
// Core Audio Backend-Specific Configuration
// Problem: Core Audio uses 10ms periods optimized for 44.1kHz, causing
// uneven callback timing (10ms/10ms/20ms) when resampling to 32kHz
//
// Solution 1: Force OS-level sample rate to 32kHz to avoid resampling
config.coreaudio.allowNominalSampleRateChange = MA_TRUE;
// Solution 2: Use conservative performance profile for larger buffers
config.performanceProfile = ma_performance_profile_conservative;
// Let Core Audio choose the period size based on conservative profile
config.periodSizeInFrames = 0; // 0 = let backend decide
config.periods = 0; // 0 = let backend decide based on performance profile
config.dataCallback = MiniaudioBackend::audio_callback;
config.pUserData = this;
if (ma_device_init(NULL, &config, &device_) != MA_SUCCESS) {
printf("Failed to open playback device.\n");
return;
}
#if defined(DEBUG_LOG_AUDIO)
// Log actual device configuration (to stderr for visibility)
DEBUG_AUDIO("\n=== MINIAUDIO DEVICE CONFIGURATION ===\n");
DEBUG_AUDIO(" Sample rate: %u (requested: 32000)\n", device_.sampleRate);
DEBUG_AUDIO(" Channels: %u (requested: 2)\n", device_.playback.channels);
DEBUG_AUDIO(" Format: %d (requested: %d, f32=%d)\n",
device_.playback.format, config.playback.format, ma_format_f32);
DEBUG_AUDIO(" Period size: %u frames (%.1fms at %uHz)\n",
device_.playback.internalPeriodSizeInFrames,
(float)device_.playback.internalPeriodSizeInFrames / device_.sampleRate * 1000.0f,
device_.sampleRate);
DEBUG_AUDIO(" Periods: %u (buffer multiplier)\n", device_.playback.internalPeriods);
DEBUG_AUDIO(" Backend: %s\n", ma_get_backend_name(device_.pContext->backend));
DEBUG_AUDIO(" Total buffer size: %u frames (%.2fms) [period * periods]\n",
device_.playback.internalPeriodSizeInFrames * device_.playback.internalPeriods,
(float)(device_.playback.internalPeriodSizeInFrames * device_.playback.internalPeriods) / device_.sampleRate * 1000.0f);
// Calculate expected callback interval
if (device_.playback.internalPeriodSizeInFrames > 0) {
const float expected_callback_ms = (float)device_.playback.internalPeriodSizeInFrames / device_.sampleRate * 1000.0f;
DEBUG_AUDIO(" Expected callback interval: %.2fms (based on period size)\n", expected_callback_ms);
DEBUG_AUDIO(" WARNING: If actual callback interval differs, audio corruption may occur!\n");
}
DEBUG_AUDIO("======================================\n\n");
fflush(stderr);
#endif /* defined(DEBUG_LOG_AUDIO) */
initialized_ = true;
}
void MiniaudioBackend::start() {
if (!initialized_) {
printf("Cannot start: backend not initialized.\n");
return;
}
if (ma_device_start(&device_) != MA_SUCCESS) {
printf("Failed to start playback device.\n");
ma_device_uninit(&device_);
initialized_ = false;
return;
}
}
void MiniaudioBackend::shutdown() {
if (!initialized_) {
return;
}
ma_device_stop(&device_);
ma_device_uninit(&device_);
initialized_ = false;
}
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