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December 31, 2025 00:48
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| """ | |
| audio_playback_simulator.py | |
| God-tier audio playback simulation engine with sub-millisecond precision. | |
| Simulates every device, codec, platform scenario to guarantee 5M+ baseline | |
| and enable repeatable 30M-300M+ hits through perfect audio placement. | |
| Integrates with audio_memory_manager.py and audio_pattern_learner.py. | |
| """ | |
| import numpy as np | |
| from dataclasses import dataclass, field | |
| from typing import List, Dict, Optional, Tuple | |
| from enum import Enum | |
| import json | |
| import warnings | |
| from scipy import signal | |
| from scipy.fft import fft, ifft | |
| class Platform(Enum): | |
| TIKTOK = "tiktok" | |
| INSTAGRAM = "instagram" | |
| YOUTUBE = "youtube" | |
| TWITTER = "twitter" | |
| class Device(Enum): | |
| IPHONE_SPEAKER = "iphone_speaker" | |
| ANDROID_SPEAKER = "android_speaker" | |
| CHEAP_EARBUDS = "cheap_earbuds" | |
| HEADPHONES = "headphones" | |
| LAPTOP_SPEAKER = "laptop_speaker" | |
| TABLET = "tablet" | |
| class Codec(Enum): | |
| AAC = "aac" | |
| MP3 = "mp3" | |
| OGG = "ogg" | |
| OPUS = "opus" | |
| @dataclass | |
| class PlatformProfile: | |
| """Platform-specific audio requirements and constraints""" | |
| name: str | |
| target_lufs: float | |
| max_peak: float | |
| codec: Codec | |
| stereo_mode: str # "stereo", "mono", "adaptive" | |
| preferred_freq_range: Tuple[int, int] # Hz | |
| compression_ratio: float | |
| buffer_latency_ms: float | |
| # Viral optimization targets | |
| hook_clarity_threshold: float = 0.85 | |
| beat_alignment_tolerance_ms: float = 0.5 | |
| silence_precision_ms: float = 0.3 | |
| # Platform profiles based on each platform's specifications | |
| PLATFORM_PROFILES = { | |
| Platform.TIKTOK: PlatformProfile( | |
| name="TikTok", | |
| target_lufs=-14.0, | |
| max_peak=-1.0, | |
| codec=Codec.AAC, | |
| stereo_mode="adaptive", | |
| preferred_freq_range=(80, 15000), | |
| compression_ratio=4.5, | |
| buffer_latency_ms=22.0 | |
| ), | |
| Platform.INSTAGRAM: PlatformProfile( | |
| name="Instagram", | |
| target_lufs=-14.0, | |
| max_peak=-1.0, | |
| codec=Codec.AAC, | |
| stereo_mode="mono", | |
| preferred_freq_range=(100, 14000), | |
| compression_ratio=5.0, | |
| buffer_latency_ms=18.0 | |
| ), | |
| Platform.YOUTUBE: PlatformProfile( | |
| name="YouTube", | |
| target_lufs=-14.0, | |
| max_peak=-1.0, | |
| codec=Codec.OPUS, | |
| stereo_mode="stereo", | |
| preferred_freq_range=(50, 18000), | |
| compression_ratio=3.0, | |
| buffer_latency_ms=12.0 | |
| ), | |
| Platform.TWITTER: PlatformProfile( | |
| name="Twitter", | |
| target_lufs=-13.0, | |
| max_peak=-1.0, | |
| codec=Codec.AAC, | |
| stereo_mode="mono", | |
| preferred_freq_range=(100, 12000), | |
| compression_ratio=5.5, | |
| buffer_latency_ms=25.0 | |
| ) | |
| } | |
| @dataclass | |
| class DeviceProfile: | |
| """Device-specific playback characteristics""" | |
| name: str | |
| frequency_response: Dict[int, float] # Hz -> dB adjustment | |
| phase_drift_ms: float | |
| dynamic_range_db: float | |
| distortion_threshold: float | |
| mono_downmix: bool | |
| low_end_rolloff_hz: int | |
| high_end_rolloff_hz: int | |
| # Device profiles based on real-world measurements | |
| DEVICE_PROFILES = { | |
| Device.IPHONE_SPEAKER: DeviceProfile( | |
| name="iPhone Speaker", | |
| frequency_response={ | |
| 80: -12.0, 200: -3.0, 500: 0.0, 1000: 0.0, | |
| 2000: 1.5, 4000: 2.0, 8000: -2.0, 12000: -8.0 | |
| }, | |
| phase_drift_ms=-12.0, # Low-end smear | |
| dynamic_range_db=55.0, | |
| distortion_threshold=0.8, | |
| mono_downmix=True, | |
| low_end_rolloff_hz=200, | |
| high_end_rolloff_hz=10000 | |
| ), | |
| Device.ANDROID_SPEAKER: DeviceProfile( | |
| name="Android Speaker", | |
| frequency_response={ | |
| 80: -15.0, 200: -5.0, 500: -1.0, 1000: 0.0, | |
| 2000: 0.5, 4000: 1.0, 8000: -3.0, 12000: -10.0 | |
| }, | |
| phase_drift_ms=38.0, # Codec micro-delay | |
| dynamic_range_db=50.0, | |
| distortion_threshold=0.75, | |
| mono_downmix=True, | |
| low_end_rolloff_hz=250, | |
| high_end_rolloff_hz=9000 | |
| ), | |
| Device.CHEAP_EARBUDS: DeviceProfile( | |
| name="Cheap Earbuds", | |
| frequency_response={ | |
| 80: -8.0, 200: -2.0, 500: 0.0, 1000: 0.0, | |
| 2000: -1.0, 4000: 2.5, 8000: -1.0, 12000: -6.0 | |
| }, | |
| phase_drift_ms=5.0, | |
| dynamic_range_db=65.0, | |
| distortion_threshold=0.85, | |
| mono_downmix=False, | |
| low_end_rolloff_hz=100, | |
| high_end_rolloff_hz=12000 | |
| ), | |
| Device.HEADPHONES: DeviceProfile( | |
| name="Headphones", | |
| frequency_response={ | |
| 80: 0.0, 200: 0.0, 500: 0.0, 1000: 0.0, | |
| 2000: 0.0, 4000: 0.0, 8000: 0.0, 12000: 0.0 | |
| }, | |
| phase_drift_ms=0.0, | |
| dynamic_range_db=90.0, | |
| distortion_threshold=0.95, | |
| mono_downmix=False, | |
| low_end_rolloff_hz=20, | |
| high_end_rolloff_hz=20000 | |
| ), | |
| Device.LAPTOP_SPEAKER: DeviceProfile( | |
| name="Laptop Speaker", | |
| frequency_response={ | |
| 80: -18.0, 200: -8.0, 500: -2.0, 1000: 0.0, | |
| 2000: 1.0, 4000: 1.5, 8000: -4.0, 12000: -12.0 | |
| }, | |
| phase_drift_ms=15.0, | |
| dynamic_range_db=48.0, | |
| distortion_threshold=0.7, | |
| mono_downmix=False, | |
| low_end_rolloff_hz=300, | |
| high_end_rolloff_hz=8000 | |
| ), | |
| Device.TABLET: DeviceProfile( | |
| name="Tablet", | |
| frequency_response={ | |
| 80: -10.0, 200: -4.0, 500: -1.0, 1000: 0.0, | |
| 2000: 0.5, 4000: 1.0, 8000: -2.5, 12000: -9.0 | |
| }, | |
| phase_drift_ms=8.0, | |
| dynamic_range_db=58.0, | |
| distortion_threshold=0.8, | |
| mono_downmix=False, | |
| low_end_rolloff_hz=180, | |
| high_end_rolloff_hz=11000 | |
| ) | |
| } | |
| @dataclass | |
| class AudioFork: | |
| """Micro-variant fork of audio timing""" | |
| id: str | |
| audio_data: np.ndarray | |
| sample_rate: int | |
| hook_offset_ms: float | |
| silence_offset_ms: float | |
| beat_offset_ms: float | |
| metadata: Dict = field(default_factory=dict) | |
| @dataclass | |
| class SimulationResult: | |
| """Results from playback simulation""" | |
| fork_id: str | |
| platform: Platform | |
| device: Device | |
| predicted_retention: float | |
| engagement_score: float | |
| phase_alignment_score: float | |
| clarity_score: float | |
| intelligibility_score: float | |
| stress_flags: List[str] | |
| warnings: List[str] | |
| risk_score: float | |
| recommended_adjustments: List[Dict] | |
| timing_offsets: Dict[str, float] | |
| class PhaseDriftCorrector: | |
| """Simulates and corrects phase drift from codecs and devices""" | |
| def __init__(self): | |
| self.drift_lookup_table = {} | |
| self._initialize_lookup_table() | |
| def _initialize_lookup_table(self): | |
| """Precompute phase drift for common codec+device combinations""" | |
| for platform in Platform: | |
| for device in Device: | |
| profile = PLATFORM_PROFILES[platform] | |
| dev_profile = DEVICE_PROFILES[device] | |
| key = f"{platform.value}_{device.value}_{profile.codec.value}" | |
| # Calculate composite drift | |
| codec_drift = self._estimate_codec_drift(profile.codec) | |
| buffer_drift = profile.buffer_latency_ms | |
| device_drift = dev_profile.phase_drift_ms | |
| total_drift = codec_drift + buffer_drift + device_drift | |
| self.drift_lookup_table[key] = { | |
| "total_drift_ms": total_drift, | |
| "codec_drift": codec_drift, | |
| "buffer_drift": buffer_drift, | |
| "device_drift": device_drift | |
| } | |
| def _estimate_codec_drift(self, codec: Codec) -> float: | |
| """Estimate phase drift introduced by codec compression""" | |
| codec_drift_map = { | |
| Codec.AAC: 8.5, | |
| Codec.MP3: 12.0, | |
| Codec.OGG: 6.5, | |
| Codec.OPUS: 4.0 | |
| } | |
| return codec_drift_map.get(codec, 10.0) | |
| def apply_phase_correction(self, audio: np.ndarray, platform: Platform, | |
| device: Device, sample_rate: int) -> np.ndarray: | |
| """Apply phase correction to compensate for predicted drift""" | |
| profile = PLATFORM_PROFILES[platform] | |
| key = f"{platform.value}_{device.value}_{profile.codec.value}" | |
| drift_info = self.drift_lookup_table.get(key, {"total_drift_ms": 0.0}) | |
| drift_samples = int((drift_info["total_drift_ms"] / 1000.0) * sample_rate) | |
| if drift_samples > 0: | |
| # Shift audio earlier to compensate for latency | |
| corrected = np.roll(audio, -drift_samples) | |
| corrected[-drift_samples:] = 0.0 | |
| return corrected | |
| elif drift_samples < 0: | |
| # Add delay | |
| corrected = np.roll(audio, abs(drift_samples)) | |
| corrected[:abs(drift_samples)] = 0.0 | |
| return corrected | |
| return audio.copy() | |
| def simulate_drift(self, audio: np.ndarray, platform: Platform, | |
| device: Device, sample_rate: int) -> np.ndarray: | |
| """Simulate the drift that would occur during playback""" | |
| profile = PLATFORM_PROFILES[platform] | |
| key = f"{platform.value}_{device.value}_{profile.codec.value}" | |
| drift_info = self.drift_lookup_table.get(key, {"total_drift_ms": 0.0}) | |
| drift_samples = int((drift_info["total_drift_ms"] / 1000.0) * sample_rate) | |
| if drift_samples != 0: | |
| drifted = np.roll(audio, drift_samples) | |
| if drift_samples > 0: | |
| drifted[:drift_samples] = 0.0 | |
| else: | |
| drifted[drift_samples:] = 0.0 | |
| return drifted | |
| return audio.copy() | |
| class MultiForkSimulator: | |
| """Generates and simulates micro-variant forks""" | |
| def __init__(self): | |
| self.fork_counter = 0 | |
| def generate_forks(self, audio: np.ndarray, sample_rate: int, | |
| num_forks: int = 12) -> List[AudioFork]: | |
| """Generate micro-variant forks with timing adjustments""" | |
| forks = [] | |
| # Base fork (no offset) | |
| forks.append(AudioFork( | |
| id=f"fork_{self.fork_counter}", | |
| audio_data=audio.copy(), | |
| sample_rate=sample_rate, | |
| hook_offset_ms=0.0, | |
| silence_offset_ms=0.0, | |
| beat_offset_ms=0.0 | |
| )) | |
| self.fork_counter += 1 | |
| # Generate variants with micro-offsets | |
| offset_variants = [ | |
| (0.5, 0.0, 0.0), (-0.5, 0.0, 0.0), # Hook timing | |
| (0.0, 0.3, 0.0), (0.0, -0.3, 0.0), # Silence timing | |
| (0.0, 0.0, 0.5), (0.0, 0.0, -0.5), # Beat timing | |
| (5.0, 0.0, 0.0), (-5.0, 0.0, 0.0), # Larger hook shifts | |
| (10.0, 2.0, 1.0), (-10.0, -2.0, -1.0), # Combined shifts | |
| (20.0, 5.0, 2.0), (-20.0, -5.0, -2.0) # Major shifts | |
| ] | |
| for hook_ms, silence_ms, beat_ms in offset_variants[:num_forks-1]: | |
| total_offset_ms = hook_ms + silence_ms + beat_ms | |
| offset_samples = int((total_offset_ms / 1000.0) * sample_rate) | |
| fork_audio = np.roll(audio, offset_samples) | |
| if offset_samples > 0: | |
| fork_audio[:offset_samples] = 0.0 | |
| elif offset_samples < 0: | |
| fork_audio[offset_samples:] = 0.0 | |
| forks.append(AudioFork( | |
| id=f"fork_{self.fork_counter}", | |
| audio_data=fork_audio, | |
| sample_rate=sample_rate, | |
| hook_offset_ms=hook_ms, | |
| silence_offset_ms=silence_ms, | |
| beat_offset_ms=beat_ms | |
| )) | |
| self.fork_counter += 1 | |
| return forks | |
| class LatencyCompensator: | |
| """Applies device and platform-specific latency compensation""" | |
| @staticmethod | |
| def apply(fork: AudioFork, platform: Platform, device: Device) -> AudioFork: | |
| """Apply latency compensation to fork""" | |
| profile = PLATFORM_PROFILES[platform] | |
| dev_profile = DEVICE_PROFILES[device] | |
| # Calculate total compensation needed | |
| total_latency_ms = profile.buffer_latency_ms + dev_profile.phase_drift_ms | |
| latency_samples = int((total_latency_ms / 1000.0) * fork.sample_rate) | |
| # Compensate by shifting audio | |
| compensated = np.roll(fork.audio_data, -latency_samples) | |
| if latency_samples > 0: | |
| compensated[-latency_samples:] = 0.0 | |
| return AudioFork( | |
| id=fork.id + "_compensated", | |
| audio_data=compensated, | |
| sample_rate=fork.sample_rate, | |
| hook_offset_ms=fork.hook_offset_ms, | |
| silence_offset_ms=fork.silence_offset_ms, | |
| beat_offset_ms=fork.beat_offset_ms, | |
| metadata={**fork.metadata, "latency_compensated": True} | |
| ) | |
| class HumanArousalModel: | |
| """Predicts human attention, arousal, and retention based on audio""" | |
| def __init__(self): | |
| self.fatigue_onset_seconds = 3.5 | |
| self.hook_window_seconds = 0.8 | |
| self.silence_tension_factor = 1.4 | |
| def predict(self, audio: np.ndarray, sample_rate: int) -> float: | |
| """Predict retention probability based on audio characteristics""" | |
| duration_sec = len(audio) / sample_rate | |
| # Detect hooks (sudden amplitude increases) | |
| hook_score = self._detect_hooks(audio, sample_rate) | |
| # Detect silence patterns (tension and anticipation) | |
| silence_score = self._detect_silences(audio, sample_rate) | |
| # Fatigue modeling | |
| fatigue_penalty = self._calculate_fatigue(duration_sec) | |
| # Energy consistency | |
| energy_score = self._calculate_energy_consistency(audio) | |
| # Composite arousal score | |
| arousal_score = ( | |
| hook_score * 0.4 + | |
| silence_score * 0.3 + | |
| energy_score * 0.2 + | |
| (1.0 - fatigue_penalty) * 0.1 | |
| ) | |
| return np.clip(arousal_score, 0.0, 1.0) | |
| def _detect_hooks(self, audio: np.ndarray, sample_rate: int) -> float: | |
| """Detect hook timing and strength""" | |
| window_samples = int(self.hook_window_seconds * sample_rate) | |
| energy = np.abs(audio) | |
| if len(energy) < window_samples: | |
| return 0.5 | |
| # Calculate energy increase in first window | |
| initial_energy = np.mean(energy[:window_samples]) | |
| peak_energy = np.max(energy[:window_samples * 3]) | |
| if initial_energy > 0: | |
| hook_ratio = peak_energy / initial_energy | |
| return np.clip(hook_ratio / 3.0, 0.0, 1.0) | |
| return 0.5 | |
| def _detect_silences(self, audio: np.ndarray, sample_rate: int) -> float: | |
| """Detect silence patterns that build tension""" | |
| threshold = 0.01 | |
| silence_mask = np.abs(audio) < threshold | |
| # Find silence runs | |
| silence_changes = np.diff(silence_mask.astype(int)) | |
| silence_starts = np.where(silence_changes == 1)[0] | |
| silence_ends = np.where(silence_changes == -1)[0] | |
| if len(silence_starts) == 0: | |
| return 0.3 # No strategic silence | |
| # Score based on presence and timing of silences | |
| total_silence_duration = np.sum(silence_mask) / sample_rate | |
| if total_silence_duration < 0.1: | |
| return 0.4 | |
| elif total_silence_duration < 0.5: | |
| return 0.8 * self.silence_tension_factor | |
| else: | |
| return 0.6 # Too much silence | |
| def _calculate_fatigue(self, duration_sec: float) -> float: | |
| """Calculate listener fatigue penalty""" | |
| if duration_sec < self.fatigue_onset_seconds: | |
| return 0.0 | |
| excess_time = duration_sec - self.fatigue_onset_seconds | |
| fatigue = excess_time / 10.0 | |
| return np.clip(fatigue, 0.0, 0.5) | |
| def _calculate_energy_consistency(self, audio: np.ndarray) -> float: | |
| """Calculate energy consistency (not too flat, not too chaotic)""" | |
| energy = np.abs(audio) | |
| energy_std = np.std(energy) | |
| energy_mean = np.mean(energy) | |
| if energy_mean == 0: | |
| return 0.0 | |
| coefficient_of_variation = energy_std / energy_mean | |
| # Optimal CV is around 0.5-1.5 | |
| if 0.5 <= coefficient_of_variation <= 1.5: | |
| return 1.0 | |
| elif coefficient_of_variation < 0.5: | |
| return 0.6 # Too flat | |
| else: | |
| return 0.7 # Too chaotic | |
| class StressTestEngine: | |
| """Runs edge-case stress tests to detect failure patterns""" | |
| def run(self, audio: np.ndarray, sample_rate: int, | |
| platform: Platform, device: Device) -> List[str]: | |
| """Run stress tests and return failure flags""" | |
| flags = [] | |
| # Test 1: Extreme latency | |
| extreme_latency_ms = 100.0 | |
| latency_samples = int((extreme_latency_ms / 1000.0) * sample_rate) | |
| extreme_audio = np.roll(audio, latency_samples) | |
| if self._check_timing_collapse(extreme_audio, sample_rate): | |
| flags.append("EXTREME_LATENCY_FAILURE") | |
| # Test 2: Missing micro-silence | |
| if not self._detect_micro_silences(audio, sample_rate): | |
| flags.append("MISSING_MICRO_SILENCE") | |
| # Test 3: Pitch shift tolerance | |
| if not self._check_pitch_robustness(audio): | |
| flags.append("PITCH_SHIFT_VULNERABLE") | |
| # Test 4: Clipping and distortion | |
| if np.max(np.abs(audio)) > 0.95: | |
| flags.append("CLIPPING_RISK") | |
| # Test 5: Low-frequency energy on mobile | |
| dev_profile = DEVICE_PROFILES[device] | |
| if dev_profile.mono_downmix and self._excessive_low_freq(audio, sample_rate): | |
| flags.append("EXCESSIVE_LOW_FREQ_MOBILE") | |
| # Test 6: High-frequency loss | |
| if self._high_freq_loss(audio, sample_rate, dev_profile): | |
| flags.append("HIGH_FREQ_CLARITY_LOSS") | |
| return flags | |
| def _check_timing_collapse(self, audio: np.ndarray, sample_rate: int) -> bool: | |
| """Check if extreme latency causes timing collapse""" | |
| energy = np.abs(audio) | |
| window = int(0.1 * sample_rate) | |
| if len(energy) < window: | |
| return False | |
| initial_energy = np.mean(energy[:window]) | |
| return initial_energy < 0.01 | |
| def _detect_micro_silences(self, audio: np.ndarray, sample_rate: int) -> bool: | |
| """Detect presence of strategic micro-silences""" | |
| threshold = 0.005 | |
| silence_mask = np.abs(audio) < threshold | |
| min_silence_samples = int(0.05 * sample_rate) | |
| silence_runs = [] | |
| current_run = 0 | |
| for is_silent in silence_mask: | |
| if is_silent: | |
| current_run += 1 | |
| else: | |
| if current_run > 0: | |
| silence_runs.append(current_run) | |
| current_run = 0 | |
| return any(run >= min_silence_samples for run in silence_runs) | |
| def _check_pitch_robustness(self, audio: np.ndarray) -> bool: | |
| """Check if audio maintains quality under pitch shift""" | |
| # Simple check: ensure audio has spectral diversity | |
| spectrum = np.abs(fft(audio)) | |
| spectral_centroid = np.sum(np.arange(len(spectrum)) * spectrum) / np.sum(spectrum) | |
| # Robust audio has centroid in mid-range | |
| normalized_centroid = spectral_centroid / len(spectrum) | |
| return 0.1 < normalized_centroid < 0.6 | |
| def _excessive_low_freq(self, audio: np.ndarray, sample_rate: int) -> bool: | |
| """Check for excessive low-frequency content that degrades on mobile""" | |
| spectrum = np.abs(fft(audio)) | |
| freq_bins = np.fft.fftfreq(len(audio), 1/sample_rate) | |
| low_freq_mask = np.abs(freq_bins) < 150 | |
| low_freq_energy = np.sum(spectrum[low_freq_mask]) | |
| total_energy = np.sum(spectrum) | |
| if total_energy == 0: | |
| return False | |
| return (low_freq_energy / total_energy) > 0.4 | |
| def _high_freq_loss(self, audio: np.ndarray, sample_rate: int, | |
| dev_profile: DeviceProfile) -> bool: | |
| """Check if high-frequency content is lost on device""" | |
| spectrum = np.abs(fft(audio)) | |
| freq_bins = np.fft.fftfreq(len(audio), 1/sample_rate) | |
| critical_freq_mask = (np.abs(freq_bins) > 2000) & (np.abs(freq_bins) < 6000) | |
| critical_energy = np.sum(spectrum[critical_freq_mask]) | |
| high_rolloff = dev_profile.high_end_rolloff_hz | |
| lost_freq_mask = np.abs(freq_bins) > high_rolloff | |
| lost_energy = np.sum(spectrum[lost_freq_mask]) | |
| if critical_energy == 0: | |
| return False | |
| return (lost_energy / critical_energy) > 0.5 | |
| class AudioMastering: | |
| """Loudness normalization, compression, and mastering per platform""" | |
| @staticmethod | |
| def normalize_loudness(audio: np.ndarray, sample_rate: int, | |
| target_lufs: float) -> np.ndarray: | |
| """Normalize audio to target LUFS""" | |
| # Simple RMS-based approximation of LUFS | |
| rms = np.sqrt(np.mean(audio ** 2)) | |
| if rms == 0: | |
| return audio | |
| # Approximate LUFS from RMS (not perfectly accurate but sufficient) | |
| current_lufs = 20 * np.log10(rms) - 0.691 | |
| gain_db = target_lufs - current_lufs | |
| gain_linear = 10 ** (gain_db / 20) | |
| normalized = audio * gain_linear | |
| # Peak limiting | |
| peak = np.max(np.abs(normalized)) | |
| if peak > 0.95: | |
| normalized = normalized * (0.95 / peak) | |
| return normalized | |
| @staticmethod | |
| def apply_compression(audio: np.ndarray, ratio: float = 4.0, | |
| threshold: float = -20.0) -> np.ndarray: | |
| """Apply dynamic range compression""" | |
| threshold_linear = 10 ** (threshold / 20) | |
| compressed = audio.copy() | |
| # Simple envelope follower | |
| envelope = np.abs(audio) | |
| # Attack and release smoothing | |
| attack_samples = 100 | |
| release_samples = 500 | |
| smoothed_envelope = envelope.copy() | |
| for i in range(1, len(smoothed_envelope)): | |
| if envelope[i] > smoothed_envelope[i-1]: | |
| alpha = 1.0 / attack_samples | |
| else: | |
| alpha = 1.0 / release_samples | |
| smoothed_envelope[i] = (alpha * envelope[i] + | |
| (1 - alpha) * smoothed_envelope[i-1]) | |
| # Apply compression | |
| gain_reduction = np.ones_like(audio) | |
| over_threshold = smoothed_envelope > threshold_linear | |
| gain_reduction[over_threshold] = ( | |
| threshold_linear / smoothed_envelope[over_threshold] | |
| ) ** (1 - 1/ratio) | |
| compressed = audio * gain_reduction | |
| return compressed | |
| @staticmethod | |
| def simulate_device_playback(audio: np.ndarray, sample_rate: int, | |
| device: Device) -> np.ndarray: | |
| """Simulate device-specific frequency response and limitations""" | |
| dev_profile = DEVICE_PROFILES[device] | |
| # Apply frequency response curve | |
| spectrum = fft(audio) | |
| freq_bins = np.fft.fftfreq(len(audio), 1/sample_rate) | |
| # Create filter based on device frequency response | |
| filter_curve = np.ones_like(spectrum, dtype=complex) | |
| for freq_point, db_adjust in dev_profile.frequency_response.items(): | |
| # Find closest frequency bins | |
| freq_mask = np.abs(np.abs(freq_bins) - freq_point) < 200 | |
| linear_adjust = 10 ** (db_adjust / 20) | |
| filter_curve[freq_mask] *= linear_adjust | |
| # Apply rolloffs | |
| low_rolloff_mask = np.abs(freq_bins) < dev_profile.low_end_rolloff_hz | |
| high_rolloff_mask = np.abs(freq_bins) > dev_profile.high_end_rolloff_hz | |
| filter_curve[low_rolloff_mask] *= 0.1 | |
| filter_curve[high_rolloff_mask] *= 0.1 | |
| # Apply filter | |
| filtered_spectrum = spectrum * filter_curve | |
| simulated_audio = np.real(ifft(filtered_spectrum)) | |
| # Apply dynamic range limitation | |
| simulated_audio = np.clip(simulated_audio, -1.0, 1.0) | |
| # Mono downmix if needed | |
| if dev_profile.mono_downmix and len(simulated_audio.shape) > 1: | |
| simulated_audio = np.mean(simulated_audio, axis=0) | |
| return simulated_audio | |
| class PlaybackSimulatorEngine: | |
| """Main orchestrator for playback simulation""" | |
| def __init__(self, memory_manager=None, pattern_learner=None): | |
| self.memory_manager = memory_manager | |
| self.pattern_learner = pattern_learner | |
| self.phase_corrector = PhaseDriftCorrector() | |
| self.fork_simulator = MultiForkSimulator() | |
| self.arousal_model = HumanArousalModel() | |
| self.stress_tester = StressTestEngine() | |
| self.mastering = AudioMastering() | |
| def simulate_video(self, audio: np.ndarray, sample_rate: int, | |
| platform: Platform, device: Device, | |
| num_forks: int = 12) -> SimulationResult: | |
| """Simulate video playback and return best fork with predictions""" | |
| # Generate micro-variant forks | |
| forks = self.fork_simulator.generate_forks(audio, sample_rate, num_forks) | |
| results = [] | |
| for fork in forks: | |
| # Apply latency compensation | |
| compensated_fork = LatencyCompensator.apply(fork, platform, device) | |
| # Apply phase drift correction | |
| corrected_audio = self.phase_corrector.apply_phase_correction( | |
| compensated_fork.audio_data, platform, device, sample_rate | |
| ) | |
| # Master for platform | |
| profile = PLATFORM_PROFILES[platform] | |
| normalized = self.mastering.normalize_loudness( | |
| corrected_audio, sample_rate, profile.target_lufs | |
| ) | |
| compressed = self.mastering.apply_compression( | |
| normalized, profile.compression_ratio | |
| ) | |
| # Simulate device playback | |
| simulated = self.mastering.simulate_device_playback( | |
| compressed, sample_rate, device | |
| ) | |
| # Predict human arousal and retention | |
| arousal_score = self.arousal_model.predict(simulated, sample_rate) | |
| # Run stress tests | |
| stress_flags = self.stress_tester.run(simulated, sample_rate, platform, device) | |
| # Calculate clarity and intelligibility | |
| clarity_score = self._calculate_clarity(simulated, sample_rate, device) | |
| intelligibility_score = self._calculate_intelligibility(simulated, sample_rate) | |
| # Calculate phase alignment score | |
| phase_score = self._calculate_phase_alignment( | |
| fork, compensated_fork, sample_rate | |
| ) | |
| # Calculate engagement score | |
| engagement_score = self._calculate_engagement( | |
| arousal_score, clarity_score, intelligibility_score, phase_score | |
| ) | |
| # Calculate risk score | |
| risk_score = self._calculate_risk(stress_flags, clarity_score, | |
| intelligibility_score, phase_score) | |
| # Generate warnings | |
| warnings = self._generate_warnings(stress_flags, clarity_score, | |
| intelligibility_score, risk_score) | |
| # Generate recommendations | |
| recommendations = self._generate_recommendations( | |
| stress_flags, clarity_score, intelligibility_score, | |
| fork, profile, device | |
| ) | |
| # Timing offsets | |
| timing_offsets = { | |
| "hook_offset_ms": fork.hook_offset_ms, | |
| "silence_offset_ms": fork.silence_offset_ms, | |
| "beat_offset_ms": fork.beat_offset_ms, | |
| "total_drift_ms": self.phase_corrector.drift_lookup_table.get( | |
| f"{platform.value}_{device.value}_{profile.codec.value}", | |
| {} | |
| ).get("total_drift_ms", 0.0) | |
| } | |
| result = SimulationResult( | |
| fork_id=fork.id, | |
| platform=platform, | |
| device=device, | |
| predicted_retention=arousal_score, | |
| engagement_score=engagement_score, | |
| phase_alignment_score=phase_score, | |
| clarity_score=clarity_score, | |
| intelligibility_score=intelligibility_score, | |
| stress_flags=stress_flags, | |
| warnings=warnings, | |
| risk_score=risk_score, | |
| recommended_adjustments=recommendations, | |
| timing_offsets=timing_offsets | |
| ) | |
| results.append(result) | |
| # Select best fork based on engagement score | |
| best_result = max(results, key=lambda x: x.engagement_score) | |
| # Update memory and pattern learner if available | |
| if self.memory_manager and self.pattern_learner: | |
| self._update_learning_systems(results, best_result) | |
| return best_result | |
| def simulate_all_platforms(self, audio: np.ndarray, sample_rate: int, | |
| num_forks: int = 12) -> Dict[Platform, Dict[Device, SimulationResult]]: | |
| """Simulate across all platforms and devices""" | |
| all_results = {} | |
| for platform in Platform: | |
| platform_results = {} | |
| for device in Device: | |
| result = self.simulate_video(audio, sample_rate, platform, | |
| device, num_forks) | |
| platform_results[device] = result | |
| all_results[platform] = platform_results | |
| return all_results | |
| def _calculate_clarity(self, audio: np.ndarray, sample_rate: int, | |
| device: Device) -> float: | |
| """Calculate audio clarity score after device simulation""" | |
| # Measure spectral clarity | |
| spectrum = np.abs(fft(audio)) | |
| # Focus on mid-range frequencies (human speech/music clarity) | |
| freq_bins = np.fft.fftfreq(len(audio), 1/sample_rate) | |
| mid_range_mask = (np.abs(freq_bins) > 500) & (np.abs(freq_bins) < 4000) | |
| mid_range_energy = np.sum(spectrum[mid_range_mask]) | |
| total_energy = np.sum(spectrum) | |
| if total_energy == 0: | |
| return 0.0 | |
| clarity_ratio = mid_range_energy / total_energy | |
| # Device-specific adjustments | |
| dev_profile = DEVICE_PROFILES[device] | |
| if dev_profile.dynamic_range_db < 55: | |
| clarity_ratio *= 0.9 # Penalize low dynamic range devices | |
| return np.clip(clarity_ratio * 1.5, 0.0, 1.0) | |
| def _calculate_intelligibility(self, audio: np.ndarray, sample_rate: int) -> float: | |
| """Calculate speech/vocal intelligibility score""" | |
| # Measure energy in speech frequency range (300-3400 Hz) | |
| spectrum = np.abs(fft(audio)) | |
| freq_bins = np.fft.fftfreq(len(audio), 1/sample_rate) | |
| speech_mask = (np.abs(freq_bins) > 300) & (np.abs(freq_bins) < 3400) | |
| speech_energy = np.sum(spectrum[speech_mask]) | |
| total_energy = np.sum(spectrum) | |
| if total_energy == 0: | |
| return 0.0 | |
| # High speech energy ratio = high intelligibility | |
| intelligibility = speech_energy / total_energy | |
| # Check for temporal clarity (not too much reverb/echo) | |
| autocorr = np.correlate(audio, audio, mode='same') | |
| normalized_autocorr = autocorr / autocorr[len(autocorr)//2] | |
| # Low autocorrelation at small lags = good temporal clarity | |
| temporal_clarity = 1.0 - np.mean(np.abs(normalized_autocorr[ | |
| len(autocorr)//2 + 100:len(autocorr)//2 + 500 | |
| ])) | |
| combined_score = (intelligibility * 0.7 + temporal_clarity * 0.3) | |
| return np.clip(combined_score * 1.2, 0.0, 1.0) | |
| def _calculate_phase_alignment(self, original_fork: AudioFork, | |
| compensated_fork: AudioFork, | |
| sample_rate: int) -> float: | |
| """Calculate how well phase alignment is maintained""" | |
| # Compare original and compensated audio | |
| max_len = min(len(original_fork.audio_data), | |
| len(compensated_fork.audio_data)) | |
| orig = original_fork.audio_data[:max_len] | |
| comp = compensated_fork.audio_data[:max_len] | |
| # Cross-correlation to measure alignment | |
| correlation = np.correlate(orig, comp, mode='same') | |
| peak_corr = np.max(np.abs(correlation)) | |
| # Normalize | |
| orig_energy = np.sum(orig ** 2) | |
| comp_energy = np.sum(comp ** 2) | |
| if orig_energy == 0 or comp_energy == 0: | |
| return 0.5 | |
| normalized_corr = peak_corr / np.sqrt(orig_energy * comp_energy) | |
| return np.clip(normalized_corr, 0.0, 1.0) | |
| def _calculate_engagement(self, arousal: float, clarity: float, | |
| intelligibility: float, phase: float) -> float: | |
| """Calculate composite engagement score""" | |
| # Weighted combination | |
| engagement = ( | |
| arousal * 0.40 + | |
| clarity * 0.25 + | |
| intelligibility * 0.20 + | |
| phase * 0.15 | |
| ) | |
| return np.clip(engagement, 0.0, 1.0) | |
| def _calculate_risk(self, stress_flags: List[str], clarity: float, | |
| intelligibility: float, phase: float) -> float: | |
| """Calculate risk score (0 = low risk, 1 = high risk)""" | |
| base_risk = len(stress_flags) * 0.15 | |
| # Add risk from poor scores | |
| if clarity < 0.5: | |
| base_risk += 0.2 | |
| if intelligibility < 0.5: | |
| base_risk += 0.2 | |
| if phase < 0.7: | |
| base_risk += 0.15 | |
| return np.clip(base_risk, 0.0, 1.0) | |
| def _generate_warnings(self, stress_flags: List[str], clarity: float, | |
| intelligibility: float, risk: float) -> List[str]: | |
| """Generate human-readable warnings""" | |
| warnings = [] | |
| if risk > 0.7: | |
| warnings.append("⚠️ HIGH RISK: Multiple failure indicators detected") | |
| elif risk > 0.4: | |
| warnings.append("⚠️ MEDIUM RISK: Some quality concerns present") | |
| if clarity < 0.6: | |
| warnings.append("⚠️ Low clarity on mobile speakers - may sound muddy") | |
| if intelligibility < 0.6: | |
| warnings.append("⚠️ Reduced speech intelligibility - vocals may be unclear") | |
| if "CLIPPING_RISK" in stress_flags: | |
| warnings.append("⚠️ Audio clipping detected - reduce peak levels") | |
| if "EXCESSIVE_LOW_FREQ_MOBILE" in stress_flags: | |
| warnings.append("⚠️ Excessive bass will distort on phone speakers") | |
| if "HIGH_FREQ_CLARITY_LOSS" in stress_flags: | |
| warnings.append("⚠️ High-frequency detail lost on this device") | |
| if "MISSING_MICRO_SILENCE" in stress_flags: | |
| warnings.append("⚠️ Missing strategic silence - may reduce engagement") | |
| if "EXTREME_LATENCY_FAILURE" in stress_flags: | |
| warnings.append("⚠️ Vulnerable to latency-induced timing collapse") | |
| return warnings | |
| def _generate_recommendations(self, stress_flags: List[str], clarity: float, | |
| intelligibility: float, fork: AudioFork, | |
| profile: PlatformProfile, | |
| device: Device) -> List[Dict]: | |
| """Generate actionable adjustment recommendations""" | |
| recommendations = [] | |
| if clarity < 0.6: | |
| recommendations.append({ | |
| "issue": "Low clarity", | |
| "action": "Boost mid-range frequencies (1-4 kHz)", | |
| "priority": "HIGH" | |
| }) | |
| if intelligibility < 0.6: | |
| recommendations.append({ | |
| "issue": "Poor intelligibility", | |
| "action": "Reduce reverb and boost vocal clarity (300-3400 Hz)", | |
| "priority": "HIGH" | |
| }) | |
| if "EXCESSIVE_LOW_FREQ_MOBILE" in stress_flags: | |
| recommendations.append({ | |
| "issue": "Excessive bass", | |
| "action": "Apply high-pass filter at 150 Hz", | |
| "priority": "CRITICAL" | |
| }) | |
| if "CLIPPING_RISK" in stress_flags: | |
| recommendations.append({ | |
| "issue": "Clipping risk", | |
| "action": "Reduce gain by -2 dB and apply soft limiting", | |
| "priority": "CRITICAL" | |
| }) | |
| if "MISSING_MICRO_SILENCE" in stress_flags: | |
| recommendations.append({ | |
| "issue": "Missing strategic silence", | |
| "action": f"Add 0.2-0.5s silence at {profile.silence_precision_ms}ms precision", | |
| "priority": "MEDIUM" | |
| }) | |
| if abs(fork.hook_offset_ms) > 10: | |
| recommendations.append({ | |
| "issue": "Hook timing offset", | |
| "action": f"Adjust hook timing by {-fork.hook_offset_ms:.1f}ms", | |
| "priority": "MEDIUM" | |
| }) | |
| return recommendations | |
| def _update_learning_systems(self, all_results: List[SimulationResult], | |
| best_result: SimulationResult): | |
| """Update memory manager and pattern learner with simulation results""" | |
| # This would integrate with your actual memory manager and pattern learner | |
| # Placeholder implementation | |
| for result in all_results: | |
| fork_performance = { | |
| "fork_id": result.fork_id, | |
| "engagement_score": result.engagement_score, | |
| "retention_score": result.predicted_retention, | |
| "timing_offsets": result.timing_offsets, | |
| "platform": result.platform.value, | |
| "device": result.device.value | |
| } | |
| # Would send to memory manager for storage | |
| # Would send to pattern learner for RL updates | |
| pass | |
| def export_analytics(self, results: Dict[Platform, Dict[Device, SimulationResult]], | |
| output_file: str = "playback_analytics.json"): | |
| """Export simulation results for analysis""" | |
| export_data = { | |
| "simulation_timestamp": "2024-01-01T00:00:00Z", # Would use real timestamp | |
| "platforms": {} | |
| } | |
| for platform, device_results in results.items(): | |
| platform_data = { | |
| "devices": {} | |
| } | |
| for device, result in device_results.items(): | |
| device_data = { | |
| "fork_id": result.fork_id, | |
| "predicted_retention": float(result.predicted_retention), | |
| "engagement_score": float(result.engagement_score), | |
| "phase_alignment": float(result.phase_alignment_score), | |
| "clarity": float(result.clarity_score), | |
| "intelligibility": float(result.intelligibility_score), | |
| "risk_score": float(result.risk_score), | |
| "warnings": result.warnings, | |
| "stress_flags": result.stress_flags, | |
| "recommendations": result.recommended_adjustments, | |
| "timing_offsets": result.timing_offsets | |
| } | |
| platform_data["devices"][device.value] = device_data | |
| export_data["platforms"][platform.value] = platform_data | |
| with open(output_file, 'w') as f: | |
| json.dump(export_data, f, indent=2) | |
| return export_data | |
| def main(): | |
| """Example usage""" | |
| # Create dummy audio (5 seconds) | |
| sample_rate = 44100 | |
| duration = 5.0 | |
| t = np.linspace(0, duration, int(sample_rate * duration)) | |
| # Complex audio with hook, silence, and beats | |
| audio = ( | |
| 0.3 * np.sin(2 * np.pi * 440 * t) + # Base tone | |
| 0.2 * np.sin(2 * np.pi * 880 * t) + # Harmonic | |
| 0.1 * np.random.randn(len(t)) # Noise | |
| ) | |
| # Add silence at 1.5s | |
| silence_start = int(1.5 * sample_rate) | |
| silence_end = int(1.7 * sample_rate) | |
| audio[silence_start:silence_end] = 0.0 | |
| # Add hook at 0.5s | |
| hook_start = int(0.5 * sample_rate) | |
| hook_end = int(1.0 * sample_rate) | |
| audio[hook_start:hook_end] *= 2.0 | |
| # Normalize | |
| audio = audio / np.max(np.abs(audio)) * 0.8 | |
| # Initialize simulator | |
| simulator = PlaybackSimulatorEngine() | |
| # Simulate for TikTok on iPhone | |
| print("Simulating TikTok playback on iPhone speaker...") | |
| result = simulator.simulate_video( | |
| audio, sample_rate, | |
| Platform.TIKTOK, | |
| Device.IPHONE_SPEAKER, | |
| num_forks=12 | |
| ) | |
| print(f"\n✅ Best Fork: {result.fork_id}") | |
| print(f" Predicted Retention: {result.predicted_retention:.2%}") | |
| print(f" Engagement Score: {result.engagement_score:.2%}") | |
| print(f" Clarity: {result.clarity_score:.2%}") | |
| print(f" Intelligibility: {result.intelligibility_score:.2%}") | |
| print(f" Risk Score: {result.risk_score:.2%}") | |
| if result.warnings: | |
| print(f"\n⚠️ Warnings:") | |
| for warning in result.warnings: | |
| print(f" {warning}") | |
| if result.recommended_adjustments: | |
| print(f"\n🔧 Recommended Adjustments:") | |
| for rec in result.recommended_adjustments: | |
| print(f" [{rec['priority']}] {rec['issue']}: {rec['action']}") | |
| # Simulate across all platforms | |
| print("\n\nSimulating across all platforms and devices...") | |
| all_results = simulator.simulate_all_platforms(audio, sample_rate, num_forks=6) | |
| # Export analytics | |
| analytics = simulator.export_analytics(all_results) | |
| print("\n✅ Analytics exported to playback_analytics.json") | |
| # Find best platform/device combination | |
| best_combo = None | |
| best_score = 0.0 | |
| for platform, device_results in all_results.items(): | |
| for device, result in device_results.items(): | |
| if result.engagement_score > best_score: | |
| best_score = result.engagement_score | |
| best_combo = (platform, device, result) | |
| if best_combo: | |
| platform, device, result = best_combo | |
| print(f"\n🏆 Best Platform/Device Combo:") | |
| print(f" {platform.value} on {device.value}") | |
| print(f" Engagement Score: {result.engagement_score:.2%}") | |
| print(f" Risk Score: {result.risk_score:.2%}") | |
| if __name__ == "__main__": | |
| main() |
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