Update pyproject.toml

Fixes and refines - use_max_poss_sil_at_max_speech arg

README.md

@@ -1,6 +1,6 @@
 [![Mailing list : test](http://img.shields.io/badge/Email-gray.svg?style=for-the-badge&logo=gmail)](mailto:hello@silero.ai) [![Mailing list : test](http://img.shields.io/badge/Telegram-blue.svg?style=for-the-badge&logo=telegram)](https://t.me/silero_speech) [![License: CC BY-NC 4.0](https://img.shields.io/badge/License-MIT-lightgrey.svg?style=for-the-badge)](https://github.com/snakers4/silero-vad/blob/master/LICENSE) [![downloads](https://img.shields.io/pypi/dm/silero-vad?style=for-the-badge)](https://pypi.org/project/silero-vad/) 
 
-[![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/github/snakers4/silero-vad/blob/master/silero-vad.ipynb) [![Test Package](https://github.com/snakers4/silero-vad/actions/workflows/test.yml/badge.svg)](https://github.com/snakers4/silero-vad/actions/workflows/test.yml)
+[![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/github/snakers4/silero-vad/blob/master/silero-vad.ipynb) [![Test Package](https://github.com/snakers4/silero-vad/actions/workflows/test.yml/badge.svg)](https://github.com/snakers4/silero-vad/actions/workflows/test.yml) [![Pypi version](https://img.shields.io/pypi/v/silero-vad)](https://pypi.org/project/silero-vad/) [![Python version](https://img.shields.io/pypi/pyversions/silero-vad)](https://pypi.org/project/silero-vad)
 
 ![header](https://user-images.githubusercontent.com/12515440/89997349-b3523080-dc94-11ea-9906-ca2e8bc50535.png)
 

examples/java-example/pom.xml

@@ -1,30 +1,31 @@
 <project xmlns="http://maven.apache.org/POM/4.0.0" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
-  xsi:schemaLocation="http://maven.apache.org/POM/4.0.0 http://maven.apache.org/xsd/maven-4.0.0.xsd">
+         xsi:schemaLocation="http://maven.apache.org/POM/4.0.0 http://maven.apache.org/xsd/maven-4.0.0.xsd">
-  <modelVersion>4.0.0</modelVersion>
+    <modelVersion>4.0.0</modelVersion>
 
-  <groupId>org.example</groupId>
+    <groupId>org.example</groupId>
-  <artifactId>java-example</artifactId>
+    <artifactId>java-example</artifactId>
-  <version>1.0-SNAPSHOT</version>
+    <version>1.0-SNAPSHOT</version>
-  <packaging>jar</packaging>
+    <packaging>jar</packaging>
 
-  <name>sliero-vad-example</name>
+    <name>sliero-vad-example</name>
-  <url>http://maven.apache.org</url>
+    <url>http://maven.apache.org</url>
 
-  <properties>
+    <properties>
-    <project.build.sourceEncoding>UTF-8</project.build.sourceEncoding>
+        <project.build.sourceEncoding>UTF-8</project.build.sourceEncoding>
-  </properties>
+    </properties>
 
-  <dependencies>
+    <dependencies>
-    <dependency>
+        <dependency>
-      <groupId>junit</groupId>
+            <groupId>junit</groupId>
-      <artifactId>junit</artifactId>
+            <artifactId>junit</artifactId>
-      <version>3.8.1</version>
+            <version>3.8.1</version>
-      <scope>test</scope>
+            <scope>test</scope>
-    </dependency>
+        </dependency>
-    <dependency>
+        <!-- https://mvnrepository.com/artifact/com.microsoft.onnxruntime/onnxruntime -->
-      <groupId>com.microsoft.onnxruntime</groupId>
+        <dependency>
-      <artifactId>onnxruntime</artifactId>
+            <groupId>com.microsoft.onnxruntime</groupId>
-      <version>1.16.0-rc1</version>
+            <artifactId>onnxruntime</artifactId>
-    </dependency>
+            <version>1.23.1</version>
-  </dependencies>
+        </dependency>
+    </dependencies>
 </project>

examples/java-example/src/main/java/org/example/App.java

@@ -2,68 +2,263 @@ package org.example;
 
 import ai.onnxruntime.OrtException;
 import javax.sound.sampled.*;
+import java.io.File;
+import java.io.IOException;
+import java.util.ArrayList;
+import java.util.HashMap;
+import java.util.List;
 import java.util.Map;
 
+/**
+ * Silero VAD Java Example
+ * Voice Activity Detection using ONNX model
+ * 
+ * @author VvvvvGH
+ */
 public class App {
 
-    private static final String MODEL_PATH = "src/main/resources/silero_vad.onnx";
+    // ONNX model path - using the model file from the project
+    private static final String MODEL_PATH = "../../src/silero_vad/data/silero_vad.onnx";
+    // Test audio file path
+    private static final String AUDIO_FILE_PATH = "../../en_example.wav";
+    // Sampling rate
     private static final int SAMPLE_RATE = 16000;
-    private static final float START_THRESHOLD = 0.6f;
+    // Speech threshold (consistent with Python default)
-    private static final float END_THRESHOLD = 0.45f;
+    private static final float THRESHOLD = 0.5f;
-    private static final int MIN_SILENCE_DURATION_MS = 600;
+    // Negative threshold (used to determine speech end)
-    private static final int SPEECH_PAD_MS = 500;
+    private static final float NEG_THRESHOLD = 0.35f; // threshold - 0.15
-    private static final int WINDOW_SIZE_SAMPLES = 2048;
+    // Minimum speech duration (milliseconds)
+    private static final int MIN_SPEECH_DURATION_MS = 250;
+    // Minimum silence duration (milliseconds)
+    private static final int MIN_SILENCE_DURATION_MS = 100;
+    // Speech padding (milliseconds)
+    private static final int SPEECH_PAD_MS = 30;
+    // Window size (samples) - 512 samples for 16kHz
+    private static final int WINDOW_SIZE_SAMPLES = 512;
 
     public static void main(String[] args) {
-        // Initialize the Voice Activity Detector
+        System.out.println("=".repeat(60));
-        SlieroVadDetector vadDetector;
+        System.out.println("Silero VAD Java ONNX Example");
+        System.out.println("=".repeat(60));
+        
+        // Load ONNX model
+        SlieroVadOnnxModel model;
         try {
-            vadDetector = new SlieroVadDetector(MODEL_PATH, START_THRESHOLD, END_THRESHOLD, SAMPLE_RATE, MIN_SILENCE_DURATION_MS, SPEECH_PAD_MS);
+            System.out.println("Loading ONNX model: " + MODEL_PATH);
+            model = new SlieroVadOnnxModel(MODEL_PATH);
+            System.out.println("Model loaded successfully!");
         } catch (OrtException e) {
-            System.err.println("Error initializing the VAD detector: " + e.getMessage());
+            System.err.println("Failed to load model: " + e.getMessage());
+            e.printStackTrace();
             return;
         }
 
-        // Set audio format
+        // Read WAV file
-        AudioFormat format = new AudioFormat(SAMPLE_RATE, 16, 1, true, false);
+        float[] audioData;
-        DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
-
-        // Get the target data line and open it with the specified format
-        TargetDataLine targetDataLine;
         try {
-            targetDataLine = (TargetDataLine) AudioSystem.getLine(info);
+            System.out.println("\nReading audio file: " + AUDIO_FILE_PATH);
-            targetDataLine.open(format);
+            audioData = readWavFileAsFloatArray(AUDIO_FILE_PATH);
-            targetDataLine.start();
+            System.out.println("Audio file read successfully, samples: " + audioData.length);
-        } catch (LineUnavailableException e) {
+            System.out.println("Audio duration: " + String.format("%.2f", (audioData.length / (float) SAMPLE_RATE)) + " seconds");
-            System.err.println("Error opening target data line: " + e.getMessage());
+        } catch (Exception e) {
+            System.err.println("Failed to read audio file: " + e.getMessage());
+            e.printStackTrace();
             return;
         }
 
-        // Main loop to continuously read data and apply Voice Activity Detection
+        // Get speech timestamps (batch mode, consistent with Python's get_speech_timestamps)
-        while (targetDataLine.isOpen()) {
+        System.out.println("\nDetecting speech segments...");
-            byte[] data = new byte[WINDOW_SIZE_SAMPLES];
+        List<Map<String, Integer>> speechTimestamps;
-
+        try {
-            int numBytesRead = targetDataLine.read(data, 0, data.length);
+            speechTimestamps = getSpeechTimestamps(
-            if (numBytesRead <= 0) {
+                audioData,
-                System.err.println("Error reading data from target data line.");
+                model,
-                continue;
+                THRESHOLD,
-            }
+                SAMPLE_RATE,
-
+                MIN_SPEECH_DURATION_MS,
-            // Apply the Voice Activity Detector to the data and get the result
+                MIN_SILENCE_DURATION_MS,
-            Map<String, Double> detectResult;
+                SPEECH_PAD_MS,
-            try {
+                NEG_THRESHOLD
-                detectResult = vadDetector.apply(data, true);
+            );
-            } catch (Exception e) {
+        } catch (OrtException e) {
-                System.err.println("Error applying VAD detector: " + e.getMessage());
+            System.err.println("Failed to detect speech timestamps: " + e.getMessage());
-                continue;
+            e.printStackTrace();
-            }
+            return;
-
-            if (!detectResult.isEmpty()) {
-                System.out.println(detectResult);
-            }
         }
 
-        // Close the target data line to release audio resources
+        // Output detection results
-        targetDataLine.close();
+        System.out.println("\nDetected speech timestamps (in samples):");
+        for (Map<String, Integer> timestamp : speechTimestamps) {
+            System.out.println(timestamp);
+        }
+
+        // Output summary
+        System.out.println("\n" + "=".repeat(60));
+        System.out.println("Detection completed!");
+        System.out.println("Total detected " + speechTimestamps.size() + " speech segments");
+        System.out.println("=".repeat(60));
+        
+        // Close model
+        try {
+            model.close();
+        } catch (OrtException e) {
+            System.err.println("Error closing model: " + e.getMessage());
+        }
     }
+
+    /**
+     * Get speech timestamps
+     * Implements the same logic as Python's get_speech_timestamps
+     * 
+     * @param audio Audio data (float array)
+     * @param model ONNX model
+     * @param threshold Speech threshold
+     * @param samplingRate Sampling rate
+     * @param minSpeechDurationMs Minimum speech duration (milliseconds)
+     * @param minSilenceDurationMs Minimum silence duration (milliseconds)
+     * @param speechPadMs Speech padding (milliseconds)
+     * @param negThreshold Negative threshold (used to determine speech end)
+     * @return List of speech timestamps
+     */
+    private static List<Map<String, Integer>> getSpeechTimestamps(
+            float[] audio,
+            SlieroVadOnnxModel model,
+            float threshold,
+            int samplingRate,
+            int minSpeechDurationMs,
+            int minSilenceDurationMs,
+            int speechPadMs,
+            float negThreshold) throws OrtException {
+        
+        // Reset model states
+        model.resetStates();
+        
+        // Calculate parameters
+        int minSpeechSamples = samplingRate * minSpeechDurationMs / 1000;
+        int speechPadSamples = samplingRate * speechPadMs / 1000;
+        int minSilenceSamples = samplingRate * minSilenceDurationMs / 1000;
+        int windowSizeSamples = samplingRate == 16000 ? 512 : 256;
+        int audioLengthSamples = audio.length;
+        
+        // Calculate speech probabilities for all audio chunks
+        List<Float> speechProbs = new ArrayList<>();
+        for (int currentStart = 0; currentStart < audioLengthSamples; currentStart += windowSizeSamples) {
+            float[] chunk = new float[windowSizeSamples];
+            int chunkLength = Math.min(windowSizeSamples, audioLengthSamples - currentStart);
+            System.arraycopy(audio, currentStart, chunk, 0, chunkLength);
+            
+            // Pad with zeros if chunk is shorter than window size
+            if (chunkLength < windowSizeSamples) {
+                for (int i = chunkLength; i < windowSizeSamples; i++) {
+                    chunk[i] = 0.0f;
+                }
+            }
+            
+            float speechProb = model.call(new float[][]{chunk}, samplingRate)[0];
+            speechProbs.add(speechProb);
+        }
+        
+        // Detect speech segments using the same algorithm as Python
+        boolean triggered = false;
+        List<Map<String, Integer>> speeches = new ArrayList<>();
+        Map<String, Integer> currentSpeech = null;
+        int tempEnd = 0;
+        
+        for (int i = 0; i < speechProbs.size(); i++) {
+            float speechProb = speechProbs.get(i);
+            
+            // Reset temporary end if speech probability exceeds threshold
+            if (speechProb >= threshold && tempEnd != 0) {
+                tempEnd = 0;
+            }
+            
+            // Detect speech start
+            if (speechProb >= threshold && !triggered) {
+                triggered = true;
+                currentSpeech = new HashMap<>();
+                currentSpeech.put("start", windowSizeSamples * i);
+                continue;
+            }
+            
+            // Detect speech end
+            if (speechProb < negThreshold && triggered) {
+                if (tempEnd == 0) {
+                    tempEnd = windowSizeSamples * i;
+                }
+                if (windowSizeSamples * i - tempEnd < minSilenceSamples) {
+                    continue;
+                } else {
+                    currentSpeech.put("end", tempEnd);
+                    if (currentSpeech.get("end") - currentSpeech.get("start") > minSpeechSamples) {
+                        speeches.add(currentSpeech);
+                    }
+                    currentSpeech = null;
+                    tempEnd = 0;
+                    triggered = false;
+                }
+            }
+        }
+        
+        // Handle the last speech segment
+        if (currentSpeech != null && 
+            (audioLengthSamples - currentSpeech.get("start")) > minSpeechSamples) {
+            currentSpeech.put("end", audioLengthSamples);
+            speeches.add(currentSpeech);
+        }
+        
+        // Add speech padding - same logic as Python
+        for (int i = 0; i < speeches.size(); i++) {
+            Map<String, Integer> speech = speeches.get(i);
+            if (i == 0) {
+                speech.put("start", Math.max(0, speech.get("start") - speechPadSamples));
+            }
+            if (i != speeches.size() - 1) {
+                int silenceDuration = speeches.get(i + 1).get("start") - speech.get("end");
+                if (silenceDuration < 2 * speechPadSamples) {
+                    speech.put("end", speech.get("end") + silenceDuration / 2);
+                    speeches.get(i + 1).put("start", 
+                        Math.max(0, speeches.get(i + 1).get("start") - silenceDuration / 2));
+                } else {
+                    speech.put("end", Math.min(audioLengthSamples, speech.get("end") + speechPadSamples));
+                    speeches.get(i + 1).put("start", 
+                        Math.max(0, speeches.get(i + 1).get("start") - speechPadSamples));
+                }
+            } else {
+                speech.put("end", Math.min(audioLengthSamples, speech.get("end") + speechPadSamples));
+            }
+        }
+        
+        return speeches;
+    }
+
+    /**
+     * Read WAV file and return as float array
+     * 
+     * @param filePath WAV file path
+     * @return Audio data as float array (normalized to -1.0 to 1.0)
+     */
+    private static float[] readWavFileAsFloatArray(String filePath) 
+            throws UnsupportedAudioFileException, IOException {
+        File audioFile = new File(filePath);
+        AudioInputStream audioStream = AudioSystem.getAudioInputStream(audioFile);
+        
+        // Get audio format information
+        AudioFormat format = audioStream.getFormat();
+        System.out.println("Audio format: " + format);
+        
+        // Read all audio data
+        byte[] audioBytes = audioStream.readAllBytes();
+        audioStream.close();
+        
+        // Convert to float array
+        float[] audioData = new float[audioBytes.length / 2];
+        for (int i = 0; i < audioData.length; i++) {
+            // 16-bit PCM: two bytes per sample (little-endian)
+            short sample = (short) ((audioBytes[i * 2] & 0xff) | (audioBytes[i * 2 + 1] << 8));
+            audioData[i] = sample / 32768.0f; // Normalize to -1.0 to 1.0
+        }
+        
+        return audioData;
+    }
+
 }

examples/java-example/src/main/java/org/example/SlieroVadDetector.java

@@ -8,25 +8,30 @@ import java.util.Collections;
 import java.util.HashMap;
 import java.util.Map;
 
-
+/**
+ * Silero VAD Detector
+ * Real-time voice activity detection
+ * 
+ * @author VvvvvGH
+ */
 public class SlieroVadDetector {
-    // OnnxModel model used for speech processing
+    // ONNX model for speech processing
     private final SlieroVadOnnxModel model;
-    // Threshold for speech start
+    // Speech start threshold
     private final float startThreshold;
-    // Threshold for speech end
+    // Speech end threshold
     private final float endThreshold;
     // Sampling rate
     private final int samplingRate;
-    // Minimum number of silence samples to determine the end threshold of speech
+    // Minimum silence samples to determine speech end
     private final float minSilenceSamples;
-    // Additional number of samples for speech start or end to calculate speech start or end time
+    // Speech padding samples for calculating speech boundaries
     private final float speechPadSamples;
-    // Whether in the triggered state (i.e. whether speech is being detected)
+    // Triggered state (whether speech is being detected)
     private boolean triggered;
-    // Temporarily stored number of speech end samples
+    // Temporary speech end sample position
     private int tempEnd;
-    // Number of samples currently being processed
+    // Current sample position
     private int currentSample;
 
 
@@ -36,23 +41,25 @@ public class SlieroVadDetector {
                              int samplingRate,
                              int minSilenceDurationMs,
                              int speechPadMs) throws OrtException {
-        // Check if the sampling rate is 8000 or 16000, if not, throw an exception
+        // Validate sampling rate
         if (samplingRate != 8000 && samplingRate != 16000) {
-            throw new IllegalArgumentException("does not support sampling rates other than [8000, 16000]");
+            throw new IllegalArgumentException("Does not support sampling rates other than [8000, 16000]");
         }
 
-        // Initialize the parameters
+        // Initialize parameters
         this.model = new SlieroVadOnnxModel(modelPath);
         this.startThreshold = startThreshold;
         this.endThreshold = endThreshold;
         this.samplingRate = samplingRate;
         this.minSilenceSamples = samplingRate * minSilenceDurationMs / 1000f;
         this.speechPadSamples = samplingRate * speechPadMs / 1000f;
-        // Reset the state
+        // Reset state
         reset();
     }
 
-    // Method to reset the state, including the model state, trigger state, temporary end time, and current sample count
+    /**
+     * Reset detector state
+     */
     public void reset() {
         model.resetStates();
         triggered = false;
@@ -60,21 +67,27 @@ public class SlieroVadDetector {
         currentSample = 0;
     }
 
-    // apply method for processing the audio array, returning possible speech start or end times
+    /**
+     * Process audio data and detect speech events
+     * 
+     * @param data Audio data as byte array
+     * @param returnSeconds Whether to return timestamps in seconds
+     * @return Speech event (start or end) or empty map if no event
+     */
     public Map<String, Double> apply(byte[] data, boolean returnSeconds) {
 
-        // Convert the byte array to a float array
+        // Convert byte array to float array
         float[] audioData = new float[data.length / 2];
         for (int i = 0; i < audioData.length; i++) {
             audioData[i] = ((data[i * 2] & 0xff) | (data[i * 2 + 1] << 8)) / 32767.0f;
         }
 
-        // Get the length of the audio array as the window size
+        // Get window size from audio data length
         int windowSizeSamples = audioData.length;
-        // Update the current sample count
+        // Update current sample position
         currentSample += windowSizeSamples;
 
-        // Call the model to get the prediction probability of speech
+        // Get speech probability from model
         float speechProb = 0;
         try {
             speechProb = model.call(new float[][]{audioData}, samplingRate)[0];
@@ -82,19 +95,18 @@ public class SlieroVadDetector {
             throw new RuntimeException(e);
         }
 
-        // If the speech probability is greater than the threshold and the temporary end time is not 0, reset the temporary end time
+        // Reset temporary end if speech probability exceeds threshold
-        // This indicates that the speech duration has exceeded expectations and needs to recalculate the end time
         if (speechProb >= startThreshold && tempEnd != 0) {
             tempEnd = 0;
         }
 
-        // If the speech probability is greater than the threshold and not in the triggered state, set to triggered state and calculate the speech start time
+        // Detect speech start
         if (speechProb >= startThreshold && !triggered) {
             triggered = true;
             int speechStart = (int) (currentSample - speechPadSamples);
             speechStart = Math.max(speechStart, 0);
             Map<String, Double> result = new HashMap<>();
-            // Decide whether to return the result in seconds or sample count based on the returnSeconds parameter
+            // Return in seconds or samples based on returnSeconds parameter
             if (returnSeconds) {
                 double speechStartSeconds = speechStart / (double) samplingRate;
                 double roundedSpeechStart = BigDecimal.valueOf(speechStartSeconds).setScale(1, RoundingMode.HALF_UP).doubleValue();
@@ -106,18 +118,17 @@ public class SlieroVadDetector {
             return result;
         }
 
-        // If the speech probability is less than a certain threshold and in the triggered state, calculate the speech end time
+        // Detect speech end
         if (speechProb < endThreshold && triggered) {
-            // Initialize or update the temporary end time
+            // Initialize or update temporary end position
             if (tempEnd == 0) {
                 tempEnd = currentSample;
             }
-            // If the number of silence samples between the current sample and the temporary end time is less than the minimum silence samples, return null
+            // Wait for minimum silence duration before confirming speech end
-            // This indicates that it is not yet possible to determine whether the speech has ended
             if (currentSample - tempEnd < minSilenceSamples) {
                 return Collections.emptyMap();
             } else {
-                // Calculate the speech end time, reset the trigger state and temporary end time
+                // Calculate speech end time and reset state
                 int speechEnd = (int) (tempEnd + speechPadSamples);
                 tempEnd = 0;
                 triggered = false;
@@ -134,7 +145,7 @@ public class SlieroVadDetector {
             }
         }
 
-        // If the above conditions are not met, return null by default
+        // No speech event detected
         return Collections.emptyMap();
     }
 

examples/java-example/src/main/java/org/example/SlieroVadOnnxModel.java

@@ -9,42 +9,58 @@ import java.util.HashMap;
 import java.util.List;
 import java.util.Map;
 
+/**
+ * Silero VAD ONNX Model Wrapper
+ * 
+ * @author VvvvvGH
+ */
 public class SlieroVadOnnxModel {
-    // Define private variable OrtSession
+    // ONNX runtime session
     private final OrtSession session;
-    private float[][][] h;
+    // Model state - dimensions: [2, batch_size, 128]
-    private float[][][] c;
+    private float[][][] state;
-    // Define the last sample rate
+    // Context - stores the tail of the previous audio chunk
+    private float[][] context;
+    // Last sample rate
     private int lastSr = 0;
-    // Define the last batch size
+    // Last batch size
     private int lastBatchSize = 0;
-    // Define a list of supported sample rates
+    // Supported sample rates
     private static final List<Integer> SAMPLE_RATES = Arrays.asList(8000, 16000);
 
     // Constructor
     public SlieroVadOnnxModel(String modelPath) throws OrtException {
         // Get the ONNX runtime environment
         OrtEnvironment env = OrtEnvironment.getEnvironment();
-        // Create an ONNX session options object
+        // Create ONNX session options
         OrtSession.SessionOptions opts = new OrtSession.SessionOptions();
-        // Set the InterOp thread count to 1, InterOp threads are used for parallel processing of different computation graph operations
+        // Set InterOp thread count to 1 (for parallel processing of different graph operations)
         opts.setInterOpNumThreads(1);
-        // Set the IntraOp thread count to 1, IntraOp threads are used for parallel processing within a single operation
+        // Set IntraOp thread count to 1 (for parallel processing within a single operation)
         opts.setIntraOpNumThreads(1);
-        // Add a CPU device, setting to false disables CPU execution optimization
+        // Enable CPU execution optimization
         opts.addCPU(true);
-        // Create an ONNX session using the environment, model path, and options
+        // Create ONNX session with the environment, model path, and options
         session = env.createSession(modelPath, opts);
         // Reset states
         resetStates();
     }
 
     /**
-     * Reset states
+     * Reset states with default batch size
      */
     void resetStates() {
-        h = new float[2][1][64];
+        resetStates(1);
-        c = new float[2][1][64];
+    }
+    
+    /**
+     * Reset states with specific batch size
+     * 
+     * @param batchSize Batch size for state initialization
+     */
+    void resetStates(int batchSize) {
+        state = new float[2][batchSize][128];
+        context = new float[0][]; // Empty context
         lastSr = 0;
         lastBatchSize = 0;
     }
@@ -54,13 +70,12 @@ public class SlieroVadOnnxModel {
     }
 
     /**
-     * Define inner class ValidationResult
+     * Inner class for validation result
      */
     public static class ValidationResult {
         public final float[][] x;
         public final int sr;
 
-        // Constructor
         public ValidationResult(float[][] x, int sr) {
             this.x = x;
             this.sr = sr;
@@ -68,19 +83,23 @@ public class SlieroVadOnnxModel {
     }
 
     /**
-     * Function to validate input data
+     * Validate input data
+     * 
+     * @param x Audio data array
+     * @param sr Sample rate
+     * @return Validated input data and sample rate
      */
     private ValidationResult validateInput(float[][] x, int sr) {
-        // Process the input data with dimension 1
+        // Ensure input is at least 2D
         if (x.length == 1) {
             x = new float[][]{x[0]};
         }
-        // Throw an exception when the input data dimension is greater than 2
+        // Check if input dimension is valid
         if (x.length > 2) {
             throw new IllegalArgumentException("Incorrect audio data dimension: " + x[0].length);
         }
 
-        // Process the input data when the sample rate is not equal to 16000 and is a multiple of 16000
+        // Downsample if sample rate is a multiple of 16000
         if (sr != 16000 && (sr % 16000 == 0)) {
             int step = sr / 16000;
             float[][] reducedX = new float[x.length][];
@@ -100,22 +119,26 @@ public class SlieroVadOnnxModel {
             sr = 16000;
         }
 
-        // If the sample rate is not in the list of supported sample rates, throw an exception
+        // Validate sample rate
         if (!SAMPLE_RATES.contains(sr)) {
             throw new IllegalArgumentException("Only supports sample rates " + SAMPLE_RATES + " (or multiples of 16000)");
         }
 
-        // If the input audio block is too short, throw an exception
+        // Check if audio chunk is too short
         if (((float) sr) / x[0].length > 31.25) {
             throw new IllegalArgumentException("Input audio is too short");
         }
 
-        // Return the validated result
         return new ValidationResult(x, sr);
     }
 
     /**
-     * Method to call the ONNX model
+     * Call the ONNX model for inference
+     * 
+     * @param x Audio data array
+     * @param sr Sample rate
+     * @return Speech probability output
+     * @throws OrtException If ONNX runtime error occurs
      */
     public float[] call(float[][] x, int sr) throws OrtException {
         ValidationResult result = validateInput(x, sr);
@@ -123,38 +146,62 @@ public class SlieroVadOnnxModel {
         sr = result.sr;
 
         int batchSize = x.length;
+        int numSamples = sr == 16000 ? 512 : 256;
+        int contextSize = sr == 16000 ? 64 : 32;
 
-        if (lastBatchSize == 0 || lastSr != sr || lastBatchSize != batchSize) {
+        // Reset states only when sample rate or batch size changes
-            resetStates();
+        if (lastSr != 0 && lastSr != sr) {
+            resetStates(batchSize);
+        } else if (lastBatchSize != 0 && lastBatchSize != batchSize) {
+            resetStates(batchSize);
+        } else if (lastBatchSize == 0) {
+            // First call - state is already initialized, just set batch size
+            lastBatchSize = batchSize;
+        }
+
+        // Initialize context if needed
+        if (context.length == 0) {
+            context = new float[batchSize][contextSize];
+        }
+
+        // Concatenate context and input
+        float[][] xWithContext = new float[batchSize][contextSize + numSamples];
+        for (int i = 0; i < batchSize; i++) {
+            // Copy context
+            System.arraycopy(context[i], 0, xWithContext[i], 0, contextSize);
+            // Copy input
+            System.arraycopy(x[i], 0, xWithContext[i], contextSize, numSamples);
         }
 
         OrtEnvironment env = OrtEnvironment.getEnvironment();
 
         OnnxTensor inputTensor = null;
-        OnnxTensor hTensor = null;
+        OnnxTensor stateTensor = null;
-        OnnxTensor cTensor = null;
         OnnxTensor srTensor = null;
         OrtSession.Result ortOutputs = null;
 
         try {
             // Create input tensors
-            inputTensor = OnnxTensor.createTensor(env, x);
+            inputTensor = OnnxTensor.createTensor(env, xWithContext);
-            hTensor = OnnxTensor.createTensor(env, h);
+            stateTensor = OnnxTensor.createTensor(env, state);
-            cTensor = OnnxTensor.createTensor(env, c);
             srTensor = OnnxTensor.createTensor(env, new long[]{sr});
 
             Map<String, OnnxTensor> inputs = new HashMap<>();
             inputs.put("input", inputTensor);
             inputs.put("sr", srTensor);
-            inputs.put("h", hTensor);
+            inputs.put("state", stateTensor);
-            inputs.put("c", cTensor);
 
-            // Call the ONNX model for calculation
+            // Run ONNX model inference
             ortOutputs = session.run(inputs);
-            // Get the output results
+            // Get output results
             float[][] output = (float[][]) ortOutputs.get(0).getValue();
-            h = (float[][][]) ortOutputs.get(1).getValue();
+            state = (float[][][]) ortOutputs.get(1).getValue();
-            c = (float[][][]) ortOutputs.get(2).getValue();
+
+            // Update context - save the last contextSize samples from input
+            for (int i = 0; i < batchSize; i++) {
+                System.arraycopy(xWithContext[i], xWithContext[i].length - contextSize, 
+                               context[i], 0, contextSize);
+            }
 
             lastSr = sr;
             lastBatchSize = batchSize;
@@ -163,11 +210,8 @@ public class SlieroVadOnnxModel {
             if (inputTensor != null) {
                 inputTensor.close();
             }
-            if (hTensor != null) {
+            if (stateTensor != null) {
-                hTensor.close();
+                stateTensor.close();
-            }
-            if (cTensor != null) {
-                cTensor.close();
             }
             if (srTensor != null) {
                 srTensor.close();

pyproject.toml

@@ -3,7 +3,7 @@ requires = ["hatchling"]
 build-backend = "hatchling.build"
 [project]
 name = "silero-vad"
-version = "6.0.0"
+version = "6.1.0"
 authors = [
   {name="Silero Team", email="hello@silero.ai"},
 ]
@@ -28,6 +28,7 @@ classifiers = [
     "Topic :: Scientific/Engineering",
 ]
 dependencies = [
+  "packaging",
   "torch>=1.12.0",
   "torchaudio>=0.12.0",
   "onnxruntime>=1.16.1",

src/silero_vad/utils_vad.py

@@ -2,6 +2,7 @@ import torch
 import torchaudio
 from typing import Callable, List
 import warnings
+from packaging import version
 
 languages = ['ru', 'en', 'de', 'es']
 
@@ -134,40 +135,60 @@ class Validator():
         return outs
 
 
-def read_audio(path: str,
+def read_audio(path: str, sampling_rate: int = 16000) -> torch.Tensor:
-               sampling_rate: int = 16000):
+    ta_ver = version.parse(torchaudio.__version__)
-    list_backends = torchaudio.list_audio_backends()
+    if ta_ver < version.parse("2.9"):
+        try:
+            effects = [['channels', '1'],['rate', str(sampling_rate)]]
+            wav, sr = torchaudio.sox_effects.apply_effects_file(path, effects=effects)
+        except:
+            wav, sr = torchaudio.load(path)
+    else:
+        try:
+            wav, sr = torchaudio.load(path)
+        except:
+            try:
+                from torchcodec.decoders import AudioDecoder
+                samples = AudioDecoder(path).get_all_samples()
+                wav = samples.data
+                sr = samples.sample_rate
+            except ImportError:
+                raise RuntimeError(
+                    f"torchaudio version {torchaudio.__version__} requires torchcodec for audio I/O. "
+                    + "Install torchcodec or pin torchaudio < 2.9"
+                )
 
-    assert len(list_backends) > 0, 'The list of available backends is empty, please install backend manually. \
+    if wav.ndim > 1 and wav.size(0) > 1:
-                                    \n Recommendations: \n \tSox (UNIX OS) \n \tSoundfile (Windows OS, UNIX OS) \n \tffmpeg (Windows OS, UNIX OS)'
+        wav = wav.mean(dim=0, keepdim=True)
 
-    try:
+    if sr != sampling_rate:
-        effects = [
+        wav = torchaudio.transforms.Resample(sr, sampling_rate)(wav)
-            ['channels', '1'],
-            ['rate', str(sampling_rate)]
-        ]
 
-        wav, sr = torchaudio.sox_effects.apply_effects_file(path, effects=effects)
-    except:
-        wav, sr = torchaudio.load(path)
-
-        if wav.size(0) > 1:
-            wav = wav.mean(dim=0, keepdim=True)
-
-        if sr != sampling_rate:
-            transform = torchaudio.transforms.Resample(orig_freq=sr,
-                                                       new_freq=sampling_rate)
-            wav = transform(wav)
-            sr = sampling_rate
-
-    assert sr == sampling_rate
     return wav.squeeze(0)
 
 
-def save_audio(path: str,
+def save_audio(path: str, tensor: torch.Tensor, sampling_rate: int = 16000):
-               tensor: torch.Tensor,
+    tensor = tensor.detach().cpu()
-               sampling_rate: int = 16000):
+    if tensor.ndim == 1:
-    torchaudio.save(path, tensor.unsqueeze(0), sampling_rate, bits_per_sample=16)
+        tensor = tensor.unsqueeze(0)
+
+    ta_ver = version.parse(torchaudio.__version__)
+
+    try:
+        torchaudio.save(path, tensor, sampling_rate, bits_per_sample=16)
+    except Exception:
+        if ta_ver >= version.parse("2.9"):
+            try:
+                from torchcodec.encoders import AudioEncoder
+                encoder = AudioEncoder(tensor, sample_rate=16000)
+                encoder.to_file(path)
+            except ImportError:
+                raise RuntimeError(
+                    f"torchaudio version {torchaudio.__version__} requires torchcodec for saving. "
+                    + "Install torchcodec or pin torchaudio < 2.9"
+                )
+        else:
+            raise
 
 
 def init_jit_model(model_path: str,
@@ -227,7 +248,7 @@ def get_speech_timestamps(audio: torch.Tensor,
 
     max_speech_duration_s: int (default -  inf)
         Maximum duration of speech chunks in seconds
-        Chunks longer than max_speech_duration_s will be split at the timestamp of the last silence that lasts more than 100ms (if any), to prevent agressive cutting.
+        Chunks longer than max_speech_duration_s will be split at the timestamp of the last silence that lasts more than 100ms (if any), to prevent aggressive cutting.
         Otherwise, they will be split aggressively just before max_speech_duration_s.
 
     min_silence_duration_ms: int (default - 100 milliseconds)
@@ -289,7 +310,6 @@ def get_speech_timestamps(audio: torch.Tensor,
         raise ValueError("Currently silero VAD models support 8000 and 16000 (or multiply of 16000) sample rates")
 
     window_size_samples = 512 if sampling_rate == 16000 else 256
-    hop_size_samples = int(window_size_samples)
 
     model.reset_states()
     min_speech_samples = sampling_rate * min_speech_duration_ms / 1000
@@ -301,17 +321,14 @@ def get_speech_timestamps(audio: torch.Tensor,
     audio_length_samples = len(audio)
 
     speech_probs = []
-    for current_start_sample in range(0, audio_length_samples, hop_size_samples):
+    for current_start_sample in range(0, audio_length_samples, window_size_samples):
         chunk = audio[current_start_sample: current_start_sample + window_size_samples]
         if len(chunk) < window_size_samples:
             chunk = torch.nn.functional.pad(chunk, (0, int(window_size_samples - len(chunk))))
-        try:
+        speech_prob = model(chunk, sampling_rate).item()
-            speech_prob = model(chunk, sampling_rate).item()
-        except Exception as e:
-            import ipdb; ipdb.set_trace()
         speech_probs.append(speech_prob)
-        # caculate progress and seng it to callback function
+        # calculate progress and send it to callback function
-        progress = current_start_sample + hop_size_samples
+        progress = current_start_sample + window_size_samples
         if progress > audio_length_samples:
             progress = audio_length_samples
         progress_percent = (progress / audio_length_samples) * 100
@@ -329,53 +346,70 @@ def get_speech_timestamps(audio: torch.Tensor,
     possible_ends = []
 
     for i, speech_prob in enumerate(speech_probs):
-        if (speech_prob >= threshold) and temp_end:
+        cur_sample = window_size_samples * i
-            if temp_end != 0:
-                sil_dur = (hop_size_samples * i) - temp_end
-                if sil_dur > min_silence_samples_at_max_speech:
-                    possible_ends.append((temp_end, sil_dur))
-                temp_end = 0
-            if next_start < prev_end:
-                next_start = hop_size_samples * i
 
+        # If speech returns after a temp_end, record candidate silence if long enough and clear temp_end
+        if (speech_prob >= threshold) and temp_end:
+            sil_dur = cur_sample - temp_end
+            if sil_dur > min_silence_samples_at_max_speech:
+                possible_ends.append((temp_end, sil_dur))
+            temp_end = 0
+            if next_start < prev_end:
+                next_start = cur_sample
+
+        # Start of speech
         if (speech_prob >= threshold) and not triggered:
             triggered = True
-            current_speech['start'] = hop_size_samples * i
+            current_speech['start'] = cur_sample
             continue
 
-        if triggered and (hop_size_samples * i) - current_speech['start'] > max_speech_samples:
+        # Max speech length reached: decide where to cut
-            if possible_ends:
+        if triggered and (cur_sample - current_speech['start'] > max_speech_samples):
-                if use_max_poss_sil_at_max_speech:
+            if use_max_poss_sil_at_max_speech and possible_ends:
-                    prev_end, dur = max(possible_ends, key=lambda x: x[1])  # use the longest possible silence segment in the current speech chunk
+                prev_end, dur = max(possible_ends, key=lambda x: x[1])  # use the longest possible silence segment in the current speech chunk
-                else:
-                    prev_end, dur = possible_ends[-1]   # use the last possible silence segement
                 current_speech['end'] = prev_end
                 speeches.append(current_speech)
                 current_speech = {}
                 next_start = prev_end + dur
-                if next_start < prev_end + hop_size_samples * i:  # previously reached silence (< neg_thres) and is still not speech (< thres)
+
-                    #triggered = False
+                if next_start < prev_end + cur_sample:  # previously reached silence (< neg_thres) and is still not speech (< thres)
                     current_speech['start'] = next_start
                 else:
                     triggered = False
-                    #current_speech['start'] = next_start
                 prev_end = next_start = temp_end = 0
                 possible_ends = []
             else:
-                current_speech['end'] = hop_size_samples * i
+                # Legacy max-speech cut (use_max_poss_sil_at_max_speech=False): prefer last valid silence (prev_end) if available
-                speeches.append(current_speech)
+                if prev_end:
-                current_speech = {}
+                    current_speech['end'] = prev_end
-                prev_end = next_start = temp_end = 0
+                    speeches.append(current_speech)
-                triggered = False
+                    current_speech = {}
-                possible_ends = []
+                    if next_start < prev_end:
-                continue
+                        triggered = False
+                    else:
+                        current_speech['start'] = next_start
+                    prev_end = next_start = temp_end = 0
+                    possible_ends = []
+                else:
+                    # No prev_end -> fallback to cutting at current sample
+                    current_speech['end'] = cur_sample
+                    speeches.append(current_speech)
+                    current_speech = {}
+                    prev_end = next_start = temp_end = 0
+                    triggered = False
+                    possible_ends = []
+                    continue
 
+        # Silence detection while in speech
         if (speech_prob < neg_threshold) and triggered:
             if not temp_end:
-                temp_end = hop_size_samples * i
+                temp_end = cur_sample
-            # if ((hop_size_samples * i) - temp_end) > min_silence_samples_at_max_speech:  # condition to avoid cutting in very short silence
+            sil_dur_now = cur_sample - temp_end
-            #     prev_end = temp_end
+
-            if (hop_size_samples * i) - temp_end < min_silence_samples:
+            if not use_max_poss_sil_at_max_speech and sil_dur_now > min_silence_samples_at_max_speech:  # condition to avoid cutting in very short silence
+                prev_end = temp_end
+
+            if sil_dur_now < min_silence_samples:
                 continue
             else:
                 current_speech['end'] = temp_end
@@ -416,7 +450,7 @@ def get_speech_timestamps(audio: torch.Tensor,
             speech_dict['end'] *= step
 
     if visualize_probs:
-        make_visualization(speech_probs, hop_size_samples / sampling_rate)
+        make_visualization(speech_probs, window_size_samples / sampling_rate)
 
     return speeches
 
@@ -607,6 +641,8 @@ def drop_chunks(tss: List[dict],
         chunks.append((wav[cur_start: i['start']]))
         cur_start = i['end']
 
+    chunks.append(wav[cur_start:])
+
     return torch.cat(chunks)
 
 

tuning/utils.py

@@ -118,8 +118,6 @@ class SileroVadDataset(Dataset):
 
         assert len(gt) == len(wav) / self.num_samples
 
-        mask[gt == 0]
-
         return wav, gt, mask
 
     def get_ground_truth_annotated(self, annotation, audio_length_samples):
@@ -240,6 +238,7 @@ def train(config,
 
             loss = criterion(stacked, targets)
             loss = (loss * masks).mean()
+            optimizer.zero_grad()
             loss.backward()
             optimizer.step()
             losses.update(loss.item(), masks.numel())