voirs_cloning/
realtime_streaming.rs

1//! Enhanced Real-time Streaming Synthesis System
2//!
3//! This module provides advanced real-time voice synthesis capabilities with ultra-low latency,
4//! streaming audio processing, and adaptive quality control for live applications such as
5//! real-time voice cloning, live streaming, gaming, and interactive applications.
6
7use crate::{Error, Result, VoiceCloneRequest, VoiceSample};
8use serde::{Deserialize, Serialize};
9use std::collections::{HashMap, VecDeque};
10use std::sync::{Arc, Mutex, RwLock};
11use std::time::{Duration, Instant, SystemTime};
12use tokio::sync::{mpsc, oneshot};
13use uuid::Uuid;
14
15/// Real-time streaming synthesis engine
16#[derive(Debug)]
17pub struct RealtimeStreamingEngine {
18    /// Engine configuration
19    config: StreamingConfig,
20    /// Active streaming sessions
21    active_sessions: Arc<RwLock<HashMap<String, StreamingSession>>>,
22    /// Audio processing pipeline
23    audio_pipeline: VoiceProcessingPipeline,
24    /// Adaptive quality controller
25    quality_controller: AdaptiveQualityController,
26    /// Buffer management system
27    buffer_manager: AudioBufferManager,
28    /// Performance monitoring
29    performance_monitor: StreamingPerformanceMonitor,
30    /// Network adaptation system
31    network_adapter: NetworkAdaptationSystem,
32}
33
34/// Streaming configuration
35#[derive(Debug, Clone, Serialize, Deserialize)]
36pub struct StreamingConfig {
37    /// Target latency in milliseconds
38    pub target_latency_ms: f32,
39    /// Maximum acceptable latency
40    pub max_latency_ms: f32,
41    /// Audio buffer size for streaming
42    pub buffer_size_samples: usize,
43    /// Number of audio buffers to maintain
44    pub buffer_count: usize,
45    /// Audio sample rate
46    pub sample_rate: u32,
47    /// Bit depth for streaming
48    pub bit_depth: u16,
49    /// Enable adaptive quality
50    pub adaptive_quality: bool,
51    /// Quality adaptation sensitivity
52    pub quality_adaptation_sensitivity: f32,
53    /// Enable network adaptation
54    pub network_adaptation: bool,
55    /// Chunk size for streaming
56    pub chunk_size_ms: f32,
57    /// Enable voice activity detection
58    pub enable_vad: bool,
59    /// Silence threshold for VAD
60    pub silence_threshold: f32,
61}
62
63/// Active streaming session
64#[derive(Debug)]
65pub struct StreamingSession {
66    /// Session ID
67    pub session_id: String,
68    /// Session type
69    pub session_type: StreamingSessionType,
70    /// Current session state
71    pub state: SessionState,
72    /// Audio input stream
73    pub input_stream: Arc<Mutex<AudioInputStream>>,
74    /// Audio output stream
75    pub output_stream: Arc<Mutex<AudioOutputStream>>,
76    /// Voice processing pipeline
77    pub voice_pipeline: VoiceProcessingPipeline,
78    /// Session metrics
79    pub metrics: StreamingMetrics,
80    /// Session configuration
81    pub config: SessionConfig,
82    /// Buffer management
83    pub buffers: SessionBuffers,
84    /// Quality adaptation state
85    pub quality_state: QualityAdaptationState,
86}
87
88/// Types of streaming sessions
89#[derive(Debug, Clone, Serialize, Deserialize)]
90pub enum StreamingSessionType {
91    /// Live voice cloning
92    LiveVoiceCloning,
93    /// Real-time voice conversion
94    RealtimeConversion,
95    /// Interactive voice synthesis
96    InteractiveSynthesis,
97    /// Streaming TTS
98    StreamingTTS,
99    /// Voice chat enhancement
100    VoiceChatEnhancement,
101    /// Live performance
102    LivePerformance,
103}
104
105/// Session state
106#[derive(Debug, Clone, Serialize, Deserialize)]
107pub enum SessionState {
108    Initializing,
109    Ready,
110    Streaming,
111    Paused,
112    Error(String),
113    Terminated,
114}
115
116/// Audio input stream
117#[derive(Debug)]
118pub struct AudioInputStream {
119    /// Stream ID
120    pub stream_id: String,
121    /// Input device configuration
122    pub device_config: AudioDeviceConfig,
123    /// Audio capture buffer
124    pub capture_buffer: VecDeque<f32>,
125    /// Voice activity detection
126    pub vad: VoiceActivityDetector,
127    /// Noise suppression
128    pub noise_suppression: NoiseSuppressionFilter,
129    /// Auto gain control
130    pub auto_gain: AutoGainControl,
131}
132
133/// Audio output stream
134#[derive(Debug)]
135pub struct AudioOutputStream {
136    /// Stream ID
137    pub stream_id: String,
138    /// Output device configuration
139    pub device_config: AudioDeviceConfig,
140    /// Audio playback buffer
141    pub playback_buffer: VecDeque<f32>,
142    /// Audio enhancement
143    pub enhancement: AudioEnhancement,
144    /// Spatial audio processing
145    pub spatial_processor: Option<SpatialAudioProcessor>,
146}
147
148/// Audio device configuration
149#[derive(Debug, Clone, Serialize, Deserialize)]
150pub struct AudioDeviceConfig {
151    /// Device name
152    pub device_name: String,
153    /// Sample rate
154    pub sample_rate: u32,
155    /// Channel count
156    pub channels: u16,
157    /// Buffer size
158    pub buffer_size: usize,
159    /// Latency mode
160    pub latency_mode: LatencyMode,
161}
162
163/// Latency mode for audio processing
164#[derive(Debug, Clone, Serialize, Deserialize)]
165pub enum LatencyMode {
166    /// Ultra-low latency (< 10ms)
167    UltraLow,
168    /// Low latency (< 50ms)
169    Low,
170    /// Normal latency (< 100ms)
171    Normal,
172    /// High quality (latency not prioritized)
173    HighQuality,
174}
175
176/// Voice processing pipeline for real-time synthesis
177#[derive(Debug)]
178pub struct VoiceProcessingPipeline {
179    /// Pipeline stages
180    pub stages: Vec<ProcessingStage>,
181    /// Current processing state
182    pub state: PipelineState,
183    /// Stage performance metrics
184    pub stage_metrics: HashMap<String, StageMetrics>,
185    /// Pipeline configuration
186    pub config: PipelineConfig,
187}
188
189/// Processing stage in the pipeline
190#[derive(Debug)]
191pub enum ProcessingStage {
192    /// Feature extraction
193    FeatureExtraction(FeatureExtractionStage),
194    /// Voice encoding
195    VoiceEncoding(VoiceEncodingStage),
196    /// Speaker adaptation
197    SpeakerAdaptation(SpeakerAdaptationStage),
198    /// Audio synthesis
199    AudioSynthesis(AudioSynthesisStage),
200    /// Post-processing
201    PostProcessing(PostProcessingStage),
202}
203
204/// Pipeline state
205#[derive(Debug, Clone, Serialize, Deserialize)]
206pub enum PipelineState {
207    Idle,
208    Processing,
209    Completed,
210    Error(String),
211}
212
213/// Voice activity detector
214#[derive(Debug)]
215pub struct VoiceActivityDetector {
216    /// VAD algorithm type
217    pub algorithm: VADAlgorithm,
218    /// Sensitivity threshold
219    pub sensitivity: f32,
220    /// Current voice activity state
221    pub is_voice_active: bool,
222    /// Activity history for smoothing
223    pub activity_history: VecDeque<bool>,
224    /// Configuration
225    pub config: VADConfig,
226}
227
228/// VAD algorithm types
229#[derive(Debug, Clone, Serialize, Deserialize)]
230pub enum VADAlgorithm {
231    /// Energy-based detection
232    EnergyBased,
233    /// Spectral-based detection
234    SpectralBased,
235    /// Machine learning-based
236    MLBased,
237    /// Hybrid approach
238    Hybrid,
239}
240
241/// Adaptive quality controller
242#[derive(Debug)]
243pub struct AdaptiveQualityController {
244    /// Current quality level
245    pub current_quality: f32,
246    /// Quality adaptation history
247    pub quality_history: VecDeque<QualityMeasurement>,
248    /// Adaptation strategy
249    pub strategy: QualityAdaptationStrategy,
250    /// Network conditions
251    pub network_conditions: NetworkConditions,
252    /// Performance thresholds
253    pub thresholds: QualityThresholds,
254}
255
256/// Quality measurement
257#[derive(Debug, Clone)]
258pub struct QualityMeasurement {
259    /// Timestamp
260    pub timestamp: Instant,
261    /// Quality score
262    pub quality_score: f32,
263    /// Latency measurement
264    pub latency_ms: f32,
265    /// CPU usage
266    pub cpu_usage: f32,
267    /// Memory usage
268    pub memory_usage: f32,
269    /// Network metrics
270    pub network_metrics: NetworkMetrics,
271}
272
273/// Quality adaptation strategies
274#[derive(Debug, Clone, Serialize, Deserialize)]
275pub enum QualityAdaptationStrategy {
276    /// Conservative adaptation
277    Conservative,
278    /// Balanced adaptation
279    Balanced,
280    /// Aggressive adaptation
281    Aggressive,
282    /// Custom strategy
283    Custom(CustomAdaptationParams),
284}
285
286/// Network conditions monitoring
287#[derive(Debug, Clone)]
288pub struct NetworkConditions {
289    /// Network bandwidth
290    pub bandwidth_kbps: f32,
291    /// Network latency
292    pub latency_ms: f32,
293    /// Packet loss rate
294    pub packet_loss_rate: f32,
295    /// Jitter
296    pub jitter_ms: f32,
297    /// Connection stability
298    pub stability_score: f32,
299}
300
301/// Audio buffer manager for streaming
302#[derive(Debug)]
303pub struct AudioBufferManager {
304    /// Ring buffers for different purposes
305    pub buffers: HashMap<String, RingBuffer>,
306    /// Buffer statistics
307    pub buffer_stats: BufferStatistics,
308    /// Memory usage tracking
309    pub memory_tracker: MemoryTracker,
310    /// Buffer optimization settings
311    pub optimization: BufferOptimization,
312}
313
314/// Ring buffer for audio data
315#[derive(Debug)]
316pub struct RingBuffer {
317    /// Buffer data
318    pub data: Vec<f32>,
319    /// Read position
320    pub read_pos: usize,
321    /// Write position
322    pub write_pos: usize,
323    /// Buffer size
324    pub size: usize,
325    /// Overflow/underflow tracking
326    pub overflow_count: usize,
327    pub underflow_count: usize,
328}
329
330/// Streaming performance monitor
331#[derive(Debug)]
332pub struct StreamingPerformanceMonitor {
333    /// Performance metrics
334    pub metrics: StreamingPerformanceMetrics,
335    /// Real-time statistics
336    pub realtime_stats: RealtimeStatistics,
337    /// Performance alerts
338    pub alerts: Vec<PerformanceAlert>,
339    /// Monitoring configuration
340    pub config: MonitoringConfig,
341}
342
343/// Network adaptation system
344#[derive(Debug)]
345pub struct NetworkAdaptationSystem {
346    /// Current network profile
347    pub network_profile: NetworkProfile,
348    /// Adaptation policies
349    pub policies: Vec<AdaptationPolicy>,
350    /// Bandwidth prediction
351    pub bandwidth_predictor: BandwidthPredictor,
352    /// Congestion control
353    pub congestion_control: CongestionControl,
354}
355
356/// Implementation for RealtimeStreamingEngine
357impl RealtimeStreamingEngine {
358    /// Create new real-time streaming engine
359    pub fn new(config: StreamingConfig) -> Self {
360        Self {
361            config: config.clone(),
362            active_sessions: Arc::new(RwLock::new(HashMap::new())),
363            audio_pipeline: VoiceProcessingPipeline::new(config.clone()),
364            quality_controller: AdaptiveQualityController::new(config.clone()),
365            buffer_manager: AudioBufferManager::new(config.clone()),
366            performance_monitor: StreamingPerformanceMonitor::new(),
367            network_adapter: NetworkAdaptationSystem::new(),
368        }
369    }
370
371    /// Create new streaming session
372    pub async fn create_session(
373        &mut self,
374        session_type: StreamingSessionType,
375        config: SessionConfig,
376    ) -> Result<String> {
377        let session_id = Uuid::new_v4().to_string();
378
379        let session = StreamingSession {
380            session_id: session_id.clone(),
381            session_type,
382            state: SessionState::Initializing,
383            input_stream: Arc::new(Mutex::new(AudioInputStream::new()?)),
384            output_stream: Arc::new(Mutex::new(AudioOutputStream::new()?)),
385            voice_pipeline: VoiceProcessingPipeline::new(self.config.clone()),
386            metrics: StreamingMetrics::default(),
387            config,
388            buffers: SessionBuffers::new(),
389            quality_state: QualityAdaptationState::new(),
390        };
391
392        self.active_sessions
393            .write()
394            .expect("lock should not be poisoned")
395            .insert(session_id.clone(), session);
396
397        Ok(session_id)
398    }
399
400    /// Start streaming session
401    #[allow(clippy::await_holding_lock)]
402    pub async fn start_streaming(&mut self, session_id: &str) -> Result<()> {
403        // First update session state
404        {
405            let mut sessions = self
406                .active_sessions
407                .write()
408                .expect("lock should not be poisoned");
409            let session = sessions
410                .get_mut(session_id)
411                .ok_or_else(|| Error::Validation("Session not found".to_string()))?;
412            session.state = SessionState::Streaming;
413        } // Lock is dropped here
414
415        // Clone session ID for the async operations
416        let session_id = session_id.to_string();
417
418        // Initialize and start processing without holding the lock
419        // Get session, do work, release lock - repeated pattern
420        {
421            let mut sessions = self
422                .active_sessions
423                .write()
424                .expect("lock should not be poisoned");
425            if let Some(session) = sessions.get_mut(&session_id) {
426                self.initialize_audio_streams(session).await?;
427            }
428        }
429
430        {
431            let mut sessions = self
432                .active_sessions
433                .write()
434                .expect("lock should not be poisoned");
435            if let Some(session) = sessions.get_mut(&session_id) {
436                self.start_processing_pipeline(session).await?;
437            }
438        }
439
440        Ok(())
441    }
442
443    /// Process real-time audio chunk
444    pub async fn process_audio_chunk(
445        &mut self,
446        session_id: &str,
447        audio_chunk: AudioChunk,
448    ) -> Result<AudioChunk> {
449        let start_time = Instant::now();
450
451        // Voice activity detection
452        let is_voice_active = {
453            let sessions = self
454                .active_sessions
455                .read()
456                .expect("lock should not be poisoned");
457            let session = sessions
458                .get(session_id)
459                .ok_or_else(|| Error::Validation("Session not found".to_string()))?;
460
461            // Voice activity detection
462            self.detect_voice_activity(&audio_chunk, session)?
463        }; // Drop the read lock here before async operations
464
465        if !is_voice_active && self.config.enable_vad {
466            // Return silence or previous audio for non-voice segments
467            return Ok(AudioChunk::silence(
468                audio_chunk.samples.len(),
469                self.config.sample_rate,
470            ));
471        }
472
473        // Adaptive quality control
474        let quality_level = self.determine_quality_level(session_id).await?;
475
476        // Process through voice pipeline (mock implementation doesn't use session parameter)
477        let processed_chunk = self
478            .process_through_pipeline_internal(&audio_chunk, quality_level)
479            .await?;
480
481        // Update performance metrics (now we can get a write lock)
482        let processing_time = start_time.elapsed();
483        self.update_performance_metrics(session_id, processing_time, &processed_chunk)
484            .await?;
485
486        Ok(processed_chunk)
487    }
488
489    /// Stream synthesis in real-time
490    pub async fn stream_synthesis(
491        &mut self,
492        session_id: &str,
493        text_stream: mpsc::Receiver<String>,
494        audio_sender: mpsc::Sender<AudioChunk>,
495    ) -> Result<()> {
496        // Spawn streaming task
497        let sessions = self.active_sessions.clone();
498        let config = self.config.clone();
499        let session_id_owned = session_id.to_string();
500
501        tokio::spawn(async move {
502            Self::streaming_synthesis_task(
503                session_id_owned,
504                text_stream,
505                audio_sender,
506                sessions,
507                config,
508            )
509            .await
510        });
511
512        Ok(())
513    }
514
515    /// Streaming synthesis task
516    async fn streaming_synthesis_task(
517        session_id: String,
518        mut text_stream: mpsc::Receiver<String>,
519        audio_sender: mpsc::Sender<AudioChunk>,
520        sessions: Arc<RwLock<HashMap<String, StreamingSession>>>,
521        config: StreamingConfig,
522    ) {
523        while let Some(text) = text_stream.recv().await {
524            // Process text chunk for synthesis
525            if let Ok(audio_chunk) =
526                Self::synthesize_text_chunk(&text, &session_id, &sessions, &config).await
527            {
528                if audio_sender.send(audio_chunk).await.is_err() {
529                    break; // Receiver dropped
530                }
531            }
532        }
533    }
534
535    /// Synthesize text chunk in real-time
536    async fn synthesize_text_chunk(
537        text: &str,
538        session_id: &str,
539        sessions: &Arc<RwLock<HashMap<String, StreamingSession>>>,
540        config: &StreamingConfig,
541    ) -> Result<AudioChunk> {
542        // Mock real-time synthesis
543        let chunk_duration_ms = config.chunk_size_ms;
544        let samples_per_chunk = (config.sample_rate as f32 * chunk_duration_ms / 1000.0) as usize;
545
546        // Simulate synthesis latency
547        tokio::time::sleep(Duration::from_millis((chunk_duration_ms * 0.5) as u64)).await;
548
549        Ok(AudioChunk {
550            samples: vec![0.0; samples_per_chunk], // Mock audio data
551            sample_rate: config.sample_rate,
552            timestamp: SystemTime::now(),
553            chunk_id: Uuid::new_v4().to_string(),
554            quality_level: 0.8,
555        })
556    }
557
558    /// Initialize audio streams for session
559    async fn initialize_audio_streams(&self, session: &mut StreamingSession) -> Result<()> {
560        // Initialize input stream
561        {
562            let mut input_stream = session
563                .input_stream
564                .lock()
565                .expect("lock should not be poisoned");
566            input_stream.device_config.sample_rate = self.config.sample_rate;
567            input_stream.device_config.buffer_size = self.config.buffer_size_samples;
568        }
569
570        // Initialize output stream
571        {
572            let mut output_stream = session
573                .output_stream
574                .lock()
575                .expect("lock should not be poisoned");
576            output_stream.device_config.sample_rate = self.config.sample_rate;
577            output_stream.device_config.buffer_size = self.config.buffer_size_samples;
578        }
579
580        session.state = SessionState::Ready;
581        Ok(())
582    }
583
584    /// Start processing pipeline
585    async fn start_processing_pipeline(&self, session: &mut StreamingSession) -> Result<()> {
586        session.voice_pipeline.state = PipelineState::Processing;
587        Ok(())
588    }
589
590    /// Detect voice activity in audio chunk
591    fn detect_voice_activity(
592        &self,
593        audio_chunk: &AudioChunk,
594        session: &StreamingSession,
595    ) -> Result<bool> {
596        if !self.config.enable_vad {
597            return Ok(true); // Always process if VAD is disabled
598        }
599
600        // Simple energy-based VAD
601        let energy: f32 = audio_chunk.samples.iter().map(|x| x * x).sum();
602        let avg_energy = energy / audio_chunk.samples.len() as f32;
603
604        Ok(avg_energy > self.config.silence_threshold)
605    }
606
607    /// Determine quality level for adaptive processing
608    async fn determine_quality_level(&self, session_id: &str) -> Result<f32> {
609        if !self.config.adaptive_quality {
610            return Ok(0.8); // Default quality level
611        }
612
613        // Get current performance metrics
614        let current_latency = self.get_current_latency(session_id).await?;
615        let target_latency = self.config.target_latency_ms;
616
617        // Adapt quality based on latency
618        if current_latency > target_latency * 1.5 {
619            Ok(0.5) // Reduce quality for low latency
620        } else if current_latency < target_latency * 0.8 {
621            Ok(0.9) // Increase quality if we have headroom
622        } else {
623            Ok(0.7) // Balanced quality
624        }
625    }
626
627    /// Get current latency for session
628    async fn get_current_latency(&self, session_id: &str) -> Result<f32> {
629        let sessions = self
630            .active_sessions
631            .read()
632            .expect("lock should not be poisoned");
633        let session = sessions
634            .get(session_id)
635            .ok_or_else(|| Error::Validation("Session not found".to_string()))?;
636
637        Ok(session.metrics.current_latency_ms)
638    }
639
640    /// Process audio through voice pipeline (internal implementation)
641    async fn process_through_pipeline_internal(
642        &self,
643        audio_chunk: &AudioChunk,
644        quality_level: f32,
645    ) -> Result<AudioChunk> {
646        // Mock processing through pipeline
647        let mut processed_samples = audio_chunk.samples.clone();
648
649        // Apply some basic processing based on quality level
650        let gain = quality_level;
651        for sample in &mut processed_samples {
652            *sample *= gain;
653        }
654
655        Ok(AudioChunk {
656            samples: processed_samples,
657            sample_rate: audio_chunk.sample_rate,
658            timestamp: SystemTime::now(),
659            chunk_id: Uuid::new_v4().to_string(),
660            quality_level,
661        })
662    }
663
664    /// Update performance metrics
665    async fn update_performance_metrics(
666        &mut self,
667        session_id: &str,
668        processing_time: Duration,
669        _audio_chunk: &AudioChunk,
670    ) -> Result<()> {
671        let mut sessions = self
672            .active_sessions
673            .write()
674            .expect("lock should not be poisoned");
675        let session = sessions
676            .get_mut(session_id)
677            .ok_or_else(|| Error::Validation("Session not found".to_string()))?;
678
679        // Update latency metrics
680        session.metrics.current_latency_ms = processing_time.as_millis() as f32;
681        session.metrics.avg_latency_ms =
682            (session.metrics.avg_latency_ms * 0.9) + (processing_time.as_millis() as f32 * 0.1);
683
684        // Update throughput
685        session.metrics.chunks_processed += 1;
686
687        Ok(())
688    }
689
690    /// Stop streaming session
691    pub async fn stop_session(&mut self, session_id: &str) -> Result<()> {
692        let mut sessions = self
693            .active_sessions
694            .write()
695            .expect("lock should not be poisoned");
696        if let Some(mut session) = sessions.remove(session_id) {
697            session.state = SessionState::Terminated;
698        }
699        Ok(())
700    }
701}
702
703/// Audio chunk for streaming processing
704#[derive(Debug, Clone)]
705pub struct AudioChunk {
706    /// Audio samples
707    pub samples: Vec<f32>,
708    /// Sample rate
709    pub sample_rate: u32,
710    /// Timestamp
711    pub timestamp: SystemTime,
712    /// Chunk identifier
713    pub chunk_id: String,
714    /// Quality level used for processing
715    pub quality_level: f32,
716}
717
718impl AudioChunk {
719    /// Create silence chunk
720    pub fn silence(length: usize, sample_rate: u32) -> Self {
721        Self {
722            samples: vec![0.0; length],
723            sample_rate,
724            timestamp: SystemTime::now(),
725            chunk_id: Uuid::new_v4().to_string(),
726            quality_level: 1.0,
727        }
728    }
729}
730
731/// Supporting types with default implementations
732#[derive(Debug, Clone, Default)]
733pub struct SessionConfig {
734    pub buffer_size: usize,
735    pub latency_mode: Option<LatencyMode>,
736}
737
738#[derive(Debug, Default)]
739pub struct StreamingMetrics {
740    pub current_latency_ms: f32,
741    pub avg_latency_ms: f32,
742    pub chunks_processed: u64,
743    pub buffer_underruns: u64,
744    pub buffer_overruns: u64,
745}
746
747#[derive(Debug)]
748pub struct SessionBuffers;
749
750impl Default for SessionBuffers {
751    fn default() -> Self {
752        Self::new()
753    }
754}
755
756impl SessionBuffers {
757    pub fn new() -> Self {
758        Self
759    }
760}
761
762#[derive(Debug)]
763pub struct QualityAdaptationState;
764
765impl Default for QualityAdaptationState {
766    fn default() -> Self {
767        Self::new()
768    }
769}
770
771impl QualityAdaptationState {
772    pub fn new() -> Self {
773        Self
774    }
775}
776
777// Implementations for supporting structures
778impl AudioInputStream {
779    pub fn new() -> Result<Self> {
780        Ok(Self {
781            stream_id: Uuid::new_v4().to_string(),
782            device_config: AudioDeviceConfig::default(),
783            capture_buffer: VecDeque::new(),
784            vad: VoiceActivityDetector::new(),
785            noise_suppression: NoiseSuppressionFilter::new(),
786            auto_gain: AutoGainControl::new(),
787        })
788    }
789}
790
791impl AudioOutputStream {
792    pub fn new() -> Result<Self> {
793        Ok(Self {
794            stream_id: Uuid::new_v4().to_string(),
795            device_config: AudioDeviceConfig::default(),
796            playback_buffer: VecDeque::new(),
797            enhancement: AudioEnhancement::new(),
798            spatial_processor: None,
799        })
800    }
801}
802
803impl VoiceProcessingPipeline {
804    pub fn new(config: StreamingConfig) -> Self {
805        Self {
806            stages: vec![],
807            state: PipelineState::Idle,
808            stage_metrics: HashMap::new(),
809            config: PipelineConfig::from_streaming_config(config),
810        }
811    }
812}
813
814impl AdaptiveQualityController {
815    pub fn new(config: StreamingConfig) -> Self {
816        Self {
817            current_quality: 0.8,
818            quality_history: VecDeque::new(),
819            strategy: QualityAdaptationStrategy::Balanced,
820            network_conditions: NetworkConditions::default(),
821            thresholds: QualityThresholds::from_config(config),
822        }
823    }
824}
825
826impl AudioBufferManager {
827    pub fn new(config: StreamingConfig) -> Self {
828        Self {
829            buffers: HashMap::new(),
830            buffer_stats: BufferStatistics,
831            memory_tracker: MemoryTracker::new(),
832            optimization: BufferOptimization::from_config(config),
833        }
834    }
835}
836
837impl Default for StreamingPerformanceMonitor {
838    fn default() -> Self {
839        Self::new()
840    }
841}
842
843impl StreamingPerformanceMonitor {
844    pub fn new() -> Self {
845        Self {
846            metrics: StreamingPerformanceMetrics,
847            realtime_stats: RealtimeStatistics,
848            alerts: Vec::new(),
849            config: MonitoringConfig,
850        }
851    }
852}
853
854impl Default for NetworkAdaptationSystem {
855    fn default() -> Self {
856        Self::new()
857    }
858}
859
860impl NetworkAdaptationSystem {
861    pub fn new() -> Self {
862        Self {
863            network_profile: NetworkProfile,
864            policies: Vec::new(),
865            bandwidth_predictor: BandwidthPredictor::new(),
866            congestion_control: CongestionControl::new(),
867        }
868    }
869}
870
871impl Default for VoiceActivityDetector {
872    fn default() -> Self {
873        Self::new()
874    }
875}
876
877impl VoiceActivityDetector {
878    pub fn new() -> Self {
879        Self {
880            algorithm: VADAlgorithm::EnergyBased,
881            sensitivity: 0.5,
882            is_voice_active: false,
883            activity_history: VecDeque::new(),
884            config: VADConfig,
885        }
886    }
887}
888
889// Default implementations for supporting types
890impl Default for StreamingConfig {
891    fn default() -> Self {
892        Self {
893            target_latency_ms: 50.0,
894            max_latency_ms: 100.0,
895            buffer_size_samples: 1024,
896            buffer_count: 4,
897            sample_rate: 44100,
898            bit_depth: 16,
899            adaptive_quality: true,
900            quality_adaptation_sensitivity: 0.5,
901            network_adaptation: true,
902            chunk_size_ms: 20.0,
903            enable_vad: true,
904            silence_threshold: 0.01,
905        }
906    }
907}
908
909impl Default for AudioDeviceConfig {
910    fn default() -> Self {
911        Self {
912            device_name: "default".to_string(),
913            sample_rate: 44100,
914            channels: 1,
915            buffer_size: 1024,
916            latency_mode: LatencyMode::Low,
917        }
918    }
919}
920
921impl Default for NetworkConditions {
922    fn default() -> Self {
923        Self {
924            bandwidth_kbps: 1000.0,
925            latency_ms: 50.0,
926            packet_loss_rate: 0.01,
927            jitter_ms: 5.0,
928            stability_score: 0.9,
929        }
930    }
931}
932
933// Placeholder types for compilation
934#[derive(Debug)]
935pub struct FeatureExtractionStage;
936#[derive(Debug)]
937pub struct VoiceEncodingStage;
938#[derive(Debug)]
939pub struct SpeakerAdaptationStage;
940#[derive(Debug)]
941pub struct AudioSynthesisStage;
942#[derive(Debug)]
943pub struct PostProcessingStage;
944#[derive(Debug)]
945pub struct StageMetrics;
946#[derive(Debug)]
947pub struct PipelineConfig;
948#[derive(Debug)]
949pub struct VADConfig;
950#[derive(Debug, Clone, Serialize, Deserialize)]
951pub struct CustomAdaptationParams;
952#[derive(Debug, Clone)]
953pub struct NetworkMetrics;
954#[derive(Debug)]
955pub struct QualityThresholds;
956#[derive(Debug)]
957pub struct BufferStatistics;
958#[derive(Debug)]
959pub struct MemoryTracker;
960#[derive(Debug)]
961pub struct BufferOptimization;
962#[derive(Debug)]
963pub struct StreamingPerformanceMetrics;
964#[derive(Debug)]
965pub struct RealtimeStatistics;
966#[derive(Debug)]
967pub struct PerformanceAlert;
968#[derive(Debug)]
969pub struct MonitoringConfig;
970#[derive(Debug)]
971pub struct NetworkProfile;
972#[derive(Debug)]
973pub struct AdaptationPolicy;
974#[derive(Debug)]
975pub struct BandwidthPredictor;
976#[derive(Debug)]
977pub struct CongestionControl;
978#[derive(Debug)]
979pub struct NoiseSuppressionFilter;
980#[derive(Debug)]
981pub struct AutoGainControl;
982#[derive(Debug)]
983pub struct AudioEnhancement;
984#[derive(Debug)]
985pub struct SpatialAudioProcessor;
986
987impl Default for VADConfig {
988    fn default() -> Self {
989        Self
990    }
991}
992impl Default for BufferStatistics {
993    fn default() -> Self {
994        Self
995    }
996}
997impl Default for StreamingPerformanceMetrics {
998    fn default() -> Self {
999        Self
1000    }
1001}
1002impl Default for RealtimeStatistics {
1003    fn default() -> Self {
1004        Self
1005    }
1006}
1007impl Default for MonitoringConfig {
1008    fn default() -> Self {
1009        Self
1010    }
1011}
1012impl Default for NetworkProfile {
1013    fn default() -> Self {
1014        Self
1015    }
1016}
1017
1018impl PipelineConfig {
1019    pub fn from_streaming_config(_config: StreamingConfig) -> Self {
1020        Self
1021    }
1022}
1023
1024impl QualityThresholds {
1025    pub fn from_config(_config: StreamingConfig) -> Self {
1026        Self
1027    }
1028}
1029
1030impl BufferOptimization {
1031    pub fn from_config(_config: StreamingConfig) -> Self {
1032        Self
1033    }
1034}
1035
1036impl Default for MemoryTracker {
1037    fn default() -> Self {
1038        Self::new()
1039    }
1040}
1041
1042impl MemoryTracker {
1043    pub fn new() -> Self {
1044        Self
1045    }
1046}
1047
1048impl Default for BandwidthPredictor {
1049    fn default() -> Self {
1050        Self::new()
1051    }
1052}
1053
1054impl BandwidthPredictor {
1055    pub fn new() -> Self {
1056        Self
1057    }
1058}
1059
1060impl Default for CongestionControl {
1061    fn default() -> Self {
1062        Self::new()
1063    }
1064}
1065
1066impl CongestionControl {
1067    pub fn new() -> Self {
1068        Self
1069    }
1070}
1071
1072impl Default for NoiseSuppressionFilter {
1073    fn default() -> Self {
1074        Self::new()
1075    }
1076}
1077
1078impl NoiseSuppressionFilter {
1079    pub fn new() -> Self {
1080        Self
1081    }
1082}
1083
1084impl Default for AutoGainControl {
1085    fn default() -> Self {
1086        Self::new()
1087    }
1088}
1089
1090impl AutoGainControl {
1091    pub fn new() -> Self {
1092        Self
1093    }
1094}
1095
1096impl Default for AudioEnhancement {
1097    fn default() -> Self {
1098        Self::new()
1099    }
1100}
1101
1102impl AudioEnhancement {
1103    pub fn new() -> Self {
1104        Self
1105    }
1106}
1107
1108#[cfg(test)]
1109mod tests {
1110    use super::*;
1111    use tokio;
1112
1113    #[test]
1114    fn test_streaming_config_default() {
1115        let config = StreamingConfig::default();
1116        assert_eq!(config.target_latency_ms, 50.0);
1117        assert_eq!(config.sample_rate, 44100);
1118        assert!(config.adaptive_quality);
1119    }
1120
1121    #[test]
1122    fn test_audio_chunk_creation() {
1123        let chunk = AudioChunk::silence(1024, 44100);
1124        assert_eq!(chunk.samples.len(), 1024);
1125        assert_eq!(chunk.sample_rate, 44100);
1126        assert_eq!(chunk.quality_level, 1.0);
1127    }
1128
1129    #[tokio::test]
1130    async fn test_streaming_engine_creation() {
1131        let config = StreamingConfig::default();
1132        let engine = RealtimeStreamingEngine::new(config);
1133        assert!(engine
1134            .active_sessions
1135            .read()
1136            .expect("lock should not be poisoned")
1137            .is_empty());
1138    }
1139
1140    #[tokio::test]
1141    async fn test_session_creation() {
1142        let config = StreamingConfig::default();
1143        let mut engine = RealtimeStreamingEngine::new(config);
1144
1145        let session_config = SessionConfig::default();
1146        let session_id = engine
1147            .create_session(StreamingSessionType::LiveVoiceCloning, session_config)
1148            .await
1149            .unwrap();
1150
1151        assert!(!session_id.is_empty());
1152        assert!(engine
1153            .active_sessions
1154            .read()
1155            .expect("lock should not be poisoned")
1156            .contains_key(&session_id));
1157    }
1158
1159    #[tokio::test]
1160    async fn test_audio_processing() {
1161        let config = StreamingConfig::default();
1162        let mut engine = RealtimeStreamingEngine::new(config);
1163
1164        let session_config = SessionConfig::default();
1165        let session_id = engine
1166            .create_session(StreamingSessionType::RealtimeConversion, session_config)
1167            .await
1168            .unwrap();
1169
1170        engine.start_streaming(&session_id).await.unwrap();
1171
1172        let audio_chunk = AudioChunk {
1173            samples: vec![0.5; 1024],
1174            sample_rate: 44100,
1175            timestamp: SystemTime::now(),
1176            chunk_id: Uuid::new_v4().to_string(),
1177            quality_level: 0.8,
1178        };
1179
1180        let processed = engine
1181            .process_audio_chunk(&session_id, audio_chunk)
1182            .await
1183            .unwrap();
1184        assert_eq!(processed.samples.len(), 1024);
1185    }
1186
1187    #[test]
1188    fn test_voice_activity_detection() {
1189        let config = StreamingConfig::default();
1190        let engine = RealtimeStreamingEngine::new(config);
1191
1192        // Test with silence
1193        let silence_chunk = AudioChunk::silence(1024, 44100);
1194        let session = StreamingSession {
1195            session_id: "test".to_string(),
1196            session_type: StreamingSessionType::LiveVoiceCloning,
1197            state: SessionState::Ready,
1198            input_stream: Arc::new(Mutex::new(AudioInputStream::new().unwrap())),
1199            output_stream: Arc::new(Mutex::new(AudioOutputStream::new().unwrap())),
1200            voice_pipeline: VoiceProcessingPipeline::new(engine.config.clone()),
1201            metrics: StreamingMetrics::default(),
1202            config: SessionConfig::default(),
1203            buffers: SessionBuffers::new(),
1204            quality_state: QualityAdaptationState::new(),
1205        };
1206
1207        let is_voice = engine
1208            .detect_voice_activity(&silence_chunk, &session)
1209            .unwrap();
1210        assert!(!is_voice);
1211
1212        // Test with signal
1213        let signal_chunk = AudioChunk {
1214            samples: vec![0.5; 1024],
1215            sample_rate: 44100,
1216            timestamp: SystemTime::now(),
1217            chunk_id: Uuid::new_v4().to_string(),
1218            quality_level: 0.8,
1219        };
1220
1221        let is_voice = engine
1222            .detect_voice_activity(&signal_chunk, &session)
1223            .unwrap();
1224        assert!(is_voice);
1225    }
1226
1227    #[tokio::test]
1228    async fn test_quality_adaptation() {
1229        let config = StreamingConfig {
1230            adaptive_quality: true,
1231            target_latency_ms: 50.0,
1232            ..Default::default()
1233        };
1234        let engine = RealtimeStreamingEngine::new(config);
1235
1236        // Mock a session with high latency
1237        let session_id = "test_session";
1238
1239        let quality = engine.determine_quality_level(session_id).await;
1240        // Should handle error gracefully for non-existent session
1241        assert!(quality.is_err());
1242    }
1243
1244    #[test]
1245    fn test_audio_device_config() {
1246        let config = AudioDeviceConfig::default();
1247        assert_eq!(config.sample_rate, 44100);
1248        assert_eq!(config.channels, 1);
1249        assert_eq!(config.buffer_size, 1024);
1250        assert!(matches!(config.latency_mode, LatencyMode::Low));
1251    }
1252
1253    #[test]
1254    fn test_streaming_session_types() {
1255        let session_types = vec![
1256            StreamingSessionType::LiveVoiceCloning,
1257            StreamingSessionType::RealtimeConversion,
1258            StreamingSessionType::InteractiveSynthesis,
1259            StreamingSessionType::StreamingTTS,
1260            StreamingSessionType::VoiceChatEnhancement,
1261            StreamingSessionType::LivePerformance,
1262        ];
1263
1264        assert_eq!(session_types.len(), 6);
1265    }
1266
1267    #[test]
1268    fn test_latency_modes() {
1269        let modes = vec![
1270            LatencyMode::UltraLow,
1271            LatencyMode::Low,
1272            LatencyMode::Normal,
1273            LatencyMode::HighQuality,
1274        ];
1275
1276        assert_eq!(modes.len(), 4);
1277    }
1278}
voirs_cloning/realtime_streaming.rs

voirs_cloning/
realtime_streaming.rs
