to 0.2.11

DEV_NOTES.md

@@ -18,8 +18,29 @@ Decoder weights: 59110771 bytes
 Encoder weights: 15268874 bytes
 
 
+# 2. Translation: Faster model for each system
 
-# 2. SortFormer Diarization: 4-to-2 Speaker Constraint Algorithm
+## Benchmark Results
+
+Testing on MacBook M3 with NLLB-200-distilled-600M model:
+
+### Standard Transformers vs CTranslate2
+
+| Test Text | Standard Inference Time | CTranslate2 Inference Time | Speedup |
+|-----------|-------------------------|---------------------------|---------|
+| UN Chief says there is no military solution in Syria | 0.9395s | 2.0472s | 0.5x |
+| The rapid advancement of AI technology is transforming various industries | 0.7171s | 1.7516s | 0.4x |
+| Climate change poses a significant threat to global ecosystems | 0.8533s | 1.8323s | 0.5x |
+| International cooperation is essential for addressing global challenges | 0.7209s | 1.3575s | 0.5x |
+| The development of renewable energy sources is crucial for a sustainable future | 0.8760s | 1.5589s | 0.6x |
+
+**Results:**
+- Total Standard time: 4.1068s
+- Total CTranslate2 time: 8.5476s
+- CTranslate2 is slower on this system --> Use Transformers, and ideally we would have an mlx implementation.
+
+
+# 3. SortFormer Diarization: 4-to-2 Speaker Constraint Algorithm
 
 Transform a diarization model that predicts up to 4 speakers into one that predicts up to 2 speakers by mapping the output predictions.
 

README.md

@@ -18,9 +18,9 @@ Real-time speech transcription directly to your browser, with a ready-to-use bac
 
 #### Powered by Leading Research:
 
-- [SimulStreaming](https://github.com/ufal/SimulStreaming) (SOTA 2025) - Ultra-low latency transcription with AlignAtt policy
+- [SimulStreaming](https://github.com/ufalSimulStreaming) (SOTA 2025) - Ultra-low latency transcription using [AlignAtt policy](https://arxiv.org/pdf/2305.11408)
 - [NLLB](https://arxiv.org/abs/2207.04672), ([distilled](https://huggingface.co/entai2965/nllb-200-distilled-600M-ctranslate2)) (2024) - Translation to more than 100 languages.
-- [WhisperStreaming](https://github.com/ufal/whisper_streaming) (SOTA 2023) - Low latency transcription with LocalAgreement policy
+- [WhisperStreaming](https://github.com/ufal/whisper_streaming) (SOTA 2023) - Low latency transcription using [LocalAgreement policy](https://www.isca-archive.org/interspeech_2020/liu20s_interspeech.pdf)
 - [Streaming Sortformer](https://arxiv.org/abs/2507.18446) (SOTA 2025) - Advanced real-time speaker diarization
 - [Diart](https://github.com/juanmc2005/diart) (SOTA 2021) - Real-time speaker diarization
 - [Silero VAD](https://github.com/snakers4/silero-vad) (2024) - Enterprise-grade Voice Activity Detection
@@ -42,15 +42,6 @@ pip install whisperlivekit
 ```
 > You can also clone the repo and `pip install -e .` for the latest version.
 
-
->  **FFmpeg is required** and must be installed before using WhisperLiveKit
-> 
-> | OS | How to install |
-> |-----------|-------------|
->  | Ubuntu/Debian | `sudo apt install ffmpeg` |
-> | MacOS | `brew install ffmpeg` |
-> | Windows | Download .exe from https://ffmpeg.org/download.html and add to PATH |
-
 #### Quick Start
 1. **Start the transcription server:**
    ```bash
@@ -86,11 +77,11 @@ See  **Parameters & Configuration** below on how to use them.
 **Command-line Interface**: Start the transcription server with various options:
 
 ```bash
-# Use better model than default (small)
-whisperlivekit-server --model large-v3
+# Large model and translate from french to danish
+whisperlivekit-server --model large-v3 --language fr --target-language da
 
-# Advanced configuration with diarization and language
-whisperlivekit-server --host 0.0.0.0 --port 8000 --model medium --diarization --language fr
+# Diarization and server listening on */80 
+whisperlivekit-server --host 0.0.0.0 --port 80 --model medium --diarization --language fr
 ```
 
 
@@ -137,26 +128,15 @@ async def websocket_endpoint(websocket: WebSocket):
 
 ## Parameters & Configuration
 
-An important list of parameters can be changed. But what *should* you change?
-- the `--model` size. List and recommandations [here](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/available_models.md)
-- the `--language`.  List [here](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/whisperlivekit/simul_whisper/whisper/tokenizer.py). If you use `auto`, the model attempts to detect the language automatically, but it tends to bias towards English.
-- the `--backend` ? you can switch to `--backend faster-whisper` if  `simulstreaming` does not work correctly or if you prefer to avoid the dual-license requirements.
-- `--warmup-file`, if you have one
-- `--task translate`, to translate in english
-- `--host`, `--port`, `--ssl-certfile`, `--ssl-keyfile`, if you set up a server
-- `--diarization`, if you want to use it.
-- [BETA] `--target-language`, to translate using NLLB. [118 languages available](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/whisperlivekit/translation/mapping_languages.py). If you want to translate to english, you should rather use `--task translate`, since Whisper can do it directly.
-
-### Full list of parameters :
 
 | Parameter | Description | Default |
 |-----------|-------------|---------|
-| `--model` | Whisper model size. | `small` |
-| `--language` | Source language code or `auto` | `auto` |
-| `--task` | Set to `translate` to translate to english | `transcribe` |
-| `--target-language` | [BETA] Translation language target. Ex: `fr` | `None` |
-| `--backend` | Processing backend | `simulstreaming` |
-| `--min-chunk-size` | Minimum audio chunk size (seconds) | `1.0` |
+| `--model` | Whisper model size. List and recommandations [here](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/available_models.md) | `small` |
+| `--language` | List [here](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/whisperlivekit/simul_whisper/whisper/tokenizer.py). If you use `auto`, the model attempts to detect the language automatically, but it tends to bias towards English. | `auto` |
+| `--target-language` | If sets, activates translation using NLLB. Ex: `fr`. [118 languages available](https://github.com/QuentinFuxa/WhisperLiveKit/blob/main/whisperlivekit/translation/mapping_languages.py). If you want to translate to english, you should rather use `--task translate`, since Whisper can do it directly. | `None` |
+| `--task` | Set to `translate` to translate *only* to english, using Whisper translation. | `transcribe` |
+| `--diarization` | Enable speaker identification | `False` |
+| `--backend` | Processing backend. You can switch to `faster-whisper` if  `simulstreaming` does not work correctly | `simulstreaming` |
 | `--no-vac` | Disable Voice Activity Controller | `False` |
 | `--no-vad` | Disable Voice Activity Detection | `False` |
 | `--warmup-file` | Audio file path for model warmup | `jfk.wav` |
@@ -164,8 +144,19 @@ An important list of parameters can be changed. But what *should* you change?
 | `--port` | Server port | `8000` |
 | `--ssl-certfile` | Path to the SSL certificate file (for HTTPS support) | `None` |
 | `--ssl-keyfile` | Path to the SSL private key file (for HTTPS support) | `None` |
-| `--pcm-input` | raw PCM (s16le) data is expected as input and FFmpeg will be bypassed. | `False` |
+| `--pcm-input` | raw PCM (s16le) data is expected as input and FFmpeg will be bypassed. Frontend will use AudioWorklet instead of MediaRecorder | `False` |
 
+| Translation options | Description | Default |
+|-----------|-------------|---------|
+| `--nllb-backend` | `transformers` or `ctranslate2` | `ctranslate2` |
+| `--nllb-size` | `600M` or `1.3B` | `600M` |
+
+| Diarization options | Description | Default |
+|-----------|-------------|---------|
+| `--diarization-backend` |  `diart` or `sortformer` | `sortformer` |
+| `--disable-punctuation-split` |  Disable punctuation based splits. See #214 | `False` |
+| `--segmentation-model` | Hugging Face model ID for Diart segmentation model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `pyannote/segmentation-3.0` |
+| `--embedding-model` | Hugging Face model ID for Diart embedding model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `speechbrain/spkrec-ecapa-voxceleb` |
 
 | SimulStreaming backend options | Description | Default |
 |-----------|-------------|---------|
@@ -184,25 +175,16 @@ An important list of parameters can be changed. But what *should* you change?
 | `--preload-model-count` | Optional. Number of models to preload in memory to speed up loading (set up to the expected number of concurrent users) | `1` |
 
 
+
 | WhisperStreaming backend options | Description | Default |
 |-----------|-------------|---------|
 | `--confidence-validation` | Use confidence scores for faster validation | `False` |
 | `--buffer_trimming` | Buffer trimming strategy (`sentence` or `segment`) | `segment` |
 
-| Diarization options | Description | Default |
-|-----------|-------------|---------|
-| `--diarization` | Enable speaker identification | `False` |
-| `--diarization-backend` |  `diart` or `sortformer` | `sortformer` |
-| `--disable-punctuation-split` |  Disable punctuation based splits. See #214 | `False` |
-| `--segmentation-model` | Hugging Face model ID for Diart segmentation model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `pyannote/segmentation-3.0` |
-| `--embedding-model` | Hugging Face model ID for Diart embedding model. [Available models](https://github.com/juanmc2005/diart/tree/main?tab=readme-ov-file#pre-trained-models) | `speechbrain/spkrec-ecapa-voxceleb` |
 
 
-> For diarization using Diart, you need access to pyannote.audio models:
-> 1. [Accept user conditions](https://huggingface.co/pyannote/segmentation) for the `pyannote/segmentation` model
-> 2. [Accept user conditions](https://huggingface.co/pyannote/segmentation-3.0) for the `pyannote/segmentation-3.0` model
-> 3. [Accept user conditions](https://huggingface.co/pyannote/embedding) for the `pyannote/embedding` model
->4. Login with HuggingFace: `huggingface-cli login`
+
+> For diarization using Diart, you need to accept user conditions [here](https://huggingface.co/pyannote/segmentation) for the `pyannote/segmentation` model, [here](https://huggingface.co/pyannote/segmentation-3.0) for the `pyannote/segmentation-3.0` model and [here](https://huggingface.co/pyannote/embedding) for the `pyannote/embedding` model. **Then**, login to HuggingFace: `huggingface-cli login`
 
 ### 🚀 Deployment Guide
 

available_models.md

@@ -1,4 +1,4 @@
-# Available model sizes:
+# Available Whisper model sizes:
 
 - tiny.en (english only)
 - tiny
@@ -71,3 +71,39 @@
 3. Good hardware and want best quality? → `large-v3`
 4. Need fast, high-quality transcription without translation? → `large-v3-turbo`
 5. Need translation capabilities? → `large-v2` or `large-v3` (avoid turbo)
+
+
+_______________________
+
+# Translation Models and Backend
+
+**Language Support**: ~200 languages
+
+## Distilled Model Sizes Available
+
+| Model | Size | Parameters | VRAM (FP16) | VRAM (INT8) | Quality |
+|-------|------|------------|-------------|-------------|---------|
+| 600M | 2.46 GB | 600M | ~1.5GB | ~800MB | Good, understandable |
+| 1.3B | 5.48 GB | 1.3B | ~3GB | ~1.5GB | Better accuracy, context |
+
+**Quality Impact**: 1.3B has ~15-25% better BLEU scores vs 600M across language pairs.
+
+## Backend Performance
+
+| Backend | Speed vs Base | Memory Usage | Quality Loss |
+|---------|---------------|--------------|--------------|
+| CTranslate2 | 6-10x faster | 40-60% less | ~5% BLEU drop |
+| Transformers | Baseline | High | None |
+| Transformers + MPS (on Apple Silicon) | 2x faster | Medium | None |
+
+**Metrics**:
+- CTranslate2: 50-100+ tokens/sec
+- Transformers: 10-30 tokens/sec
+- Apple Silicon with MPS: Up to 2x faster than CTranslate2
+
+## Quick Decision Matrix
+
+**Choose 600M**: Limited resources, close to 0 lag
+**Choose 1.3B**: Quality matters
+**Choose Transformers**: On Apple Silicon
+

pyproject.toml

@@ -4,7 +4,7 @@ build-backend = "setuptools.build_meta"
 
 [project]
 name = "whisperlivekit"
-version = "0.2.9"
+version = "0.2.11"
 description = "Real-time speech-to-text with speaker diarization using Whisper"
 readme = "README.md"
 authors = [

whisperlivekit/audio_processor.py

@@ -4,10 +4,11 @@ from time import time, sleep
 import math
 import logging
 import traceback
-from whisperlivekit.timed_objects import ASRToken, Silence, Line, FrontData, State
+from whisperlivekit.timed_objects import ASRToken, Silence, Line, FrontData, State, Transcript, ChangeSpeaker
 from whisperlivekit.core import TranscriptionEngine, online_factory, online_diarization_factory, online_translation_factory
 from whisperlivekit.silero_vad_iterator import FixedVADIterator
 from whisperlivekit.results_formater import format_output
+from whisperlivekit.ffmpeg_manager import FFmpegManager, FFmpegState
 # Set up logging once
 logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
 logger = logging.getLogger(__name__)
@@ -48,7 +49,7 @@ class AudioProcessor:
         self.bytes_per_sample = 2
         self.bytes_per_sec = self.samples_per_sec * self.bytes_per_sample
         self.max_bytes_per_sec = 32000 * 5  # 5 seconds of audio at 32 kHz
-        self.is_pcm_input = True
+        self.is_pcm_input = self.args.pcm_input
         self.debug = False
 
         # State management
@@ -57,14 +58,17 @@ class AudioProcessor:
         self.silence_duration = 0.0
         self.tokens = []
         self.translated_segments = []
-        self.buffer_transcription = ""
-        self.buffer_diarization = ""
+        self.buffer_transcription = Transcript()
         self.end_buffer = 0
         self.end_attributed_speaker = 0
         self.lock = asyncio.Lock()
         self.beg_loop = None #to deal with a potential little lag at the websocket initialization, this is now set in process_audio
         self.sep = " "  # Default separator
         self.last_response_content = FrontData()
+        self.last_detected_speaker = None
+        self.speaker_languages = {}
+        self.cumulative_pcm_len = 0
+        self.diarization_before_transcription = False
 
         # Models and processing
         self.asr = models.asr
@@ -75,6 +79,20 @@ class AudioProcessor:
         else:
             self.vac = None
                          
+        self.ffmpeg_manager = None
+        self.ffmpeg_reader_task = None
+        self._ffmpeg_error = None
+
+        if not self.is_pcm_input:
+            self.ffmpeg_manager = FFmpegManager(
+                sample_rate=self.sample_rate,
+                channels=self.channels
+            )
+            async def handle_ffmpeg_error(error_type: str):
+                logger.error(f"FFmpeg error: {error_type}")
+                self._ffmpeg_error = error_type
+            self.ffmpeg_manager.on_error_callback = handle_ffmpeg_error
+             
         self.transcription_queue = asyncio.Queue() if self.args.transcription else None
         self.diarization_queue = asyncio.Queue() if self.args.diarization else None
         self.translation_queue = asyncio.Queue() if self.args.target_language else None
@@ -84,33 +102,20 @@ class AudioProcessor:
         self.diarization_task = None
         self.watchdog_task = None
         self.all_tasks_for_cleanup = []
+        self.online_translation = None
         
         if self.args.transcription:
             self.online = online_factory(self.args, models.asr, models.tokenizer)        
             self.sep = self.online.asr.sep   
         if self.args.diarization:
             self.diarization = online_diarization_factory(self.args, models.diarization_model)
-        if self.args.target_language:
+        if models.translation_model:
             self.online_translation = online_translation_factory(self.args, models.translation_model)
 
     def convert_pcm_to_float(self, pcm_buffer):
         """Convert PCM buffer in s16le format to normalized NumPy array."""
         return np.frombuffer(pcm_buffer, dtype=np.int16).astype(np.float32) / 32768.0
 
-    async def update_transcription(self, new_tokens, buffer, end_buffer):
-        """Thread-safe update of transcription with new data."""
-        async with self.lock:
-            self.tokens.extend(new_tokens)
-            self.buffer_transcription = buffer
-            self.end_buffer = end_buffer
-            
-    async def update_diarization(self, end_attributed_speaker, buffer_diarization=""):
-        """Thread-safe update of diarization with new data."""
-        async with self.lock:
-            self.end_attributed_speaker = end_attributed_speaker
-            if buffer_diarization:
-                self.buffer_diarization = buffer_diarization
-            
     async def add_dummy_token(self):
         """Placeholder token when no transcription is available."""
         async with self.lock:
@@ -139,7 +144,6 @@ class AudioProcessor:
                 tokens=self.tokens.copy(),
                 translated_segments=self.translated_segments.copy(),
                 buffer_transcription=self.buffer_transcription,
-                buffer_diarization=self.buffer_diarization,
                 end_buffer=self.end_buffer,
                 end_attributed_speaker=self.end_attributed_speaker,
                 remaining_time_transcription=remaining_transcription,
@@ -151,10 +155,60 @@ class AudioProcessor:
         async with self.lock:
             self.tokens = []
             self.translated_segments = []
-            self.buffer_transcription = self.buffer_diarization = ""
+            self.buffer_transcription = Transcript()
             self.end_buffer = self.end_attributed_speaker = 0
             self.beg_loop = time()
 
+    async def ffmpeg_stdout_reader(self):
+        """Read audio data from FFmpeg stdout and process it into the PCM pipeline."""
+        beg = time()
+        while True:
+            try:
+                if self.is_stopping:
+                    logger.info("Stopping ffmpeg_stdout_reader due to stopping flag.")
+                    break
+
+                state = await self.ffmpeg_manager.get_state() if self.ffmpeg_manager else FFmpegState.STOPPED
+                if state == FFmpegState.FAILED:
+                    logger.error("FFmpeg is in FAILED state, cannot read data")
+                    break
+                elif state == FFmpegState.STOPPED:
+                    logger.info("FFmpeg is stopped")
+                    break
+                elif state != FFmpegState.RUNNING:
+                    await asyncio.sleep(0.1)
+                    continue
+
+                current_time = time()
+                elapsed_time = max(0.0, current_time - beg)
+                buffer_size = max(int(32000 * elapsed_time), 4096)  # dynamic read
+                beg = current_time
+
+                chunk = await self.ffmpeg_manager.read_data(buffer_size)
+                if not chunk:
+                    # No data currently available
+                    await asyncio.sleep(0.05)
+                    continue
+
+                self.pcm_buffer.extend(chunk)
+                await self.handle_pcm_data()
+
+            except asyncio.CancelledError:
+                logger.info("ffmpeg_stdout_reader cancelled.")
+                break
+            except Exception as e:
+                logger.warning(f"Exception in ffmpeg_stdout_reader: {e}")
+                logger.debug(f"Traceback: {traceback.format_exc()}")
+                await asyncio.sleep(0.2)
+
+        logger.info("FFmpeg stdout processing finished. Signaling downstream processors if needed.")
+        if not self.diarization_before_transcription and self.transcription_queue:
+            await self.transcription_queue.put(SENTINEL)
+        if self.args.diarization and self.diarization_queue:
+            await self.diarization_queue.put(SENTINEL)
+        if self.online_translation:
+            await self.translation_queue.put(SENTINEL)
+
     async def transcription_processor(self):
         """Process audio chunks for transcription."""
         cumulative_pcm_duration_stream_time = 0.0
@@ -167,11 +221,6 @@ class AudioProcessor:
                     self.transcription_queue.task_done()
                     break
 
-                if not self.online:
-                    logger.warning("Transcription processor: self.online not initialized.")
-                    self.transcription_queue.task_done()
-                    continue
-
                 asr_internal_buffer_duration_s = len(getattr(self.online, 'audio_buffer', [])) / self.online.SAMPLING_RATE
                 transcription_lag_s = max(0.0, time() - self.beg_loop - self.end_buffer)
                 asr_processing_logs = f"internal_buffer={asr_internal_buffer_duration_s:.2f}s | lag={transcription_lag_s:.2f}s |"
@@ -179,17 +228,16 @@ class AudioProcessor:
                     asr_processing_logs += f" + Silence of = {item.duration:.2f}s"
                     if self.tokens:
                         asr_processing_logs += f" | last_end = {self.tokens[-1].end} |"
-                logger.info(asr_processing_logs)
-                
-                if type(item) is Silence:
+                    logger.info(asr_processing_logs)
                     cumulative_pcm_duration_stream_time += item.duration
                     self.online.insert_silence(item.duration, self.tokens[-1].end if self.tokens else 0)
                     continue
-                
-                if isinstance(item, np.ndarray):
+                elif isinstance(item, ChangeSpeaker):
+                    self.online.new_speaker(item)
+                elif isinstance(item, np.ndarray):
                     pcm_array = item
-                else:
-                    raise Exception('item should be pcm_array')
+                
+                logger.info(asr_processing_logs)
                 
                 duration_this_chunk = len(pcm_array) / self.sample_rate
                 cumulative_pcm_duration_stream_time += duration_this_chunk
@@ -198,32 +246,30 @@ class AudioProcessor:
                 self.online.insert_audio_chunk(pcm_array, stream_time_end_of_current_pcm)
                 new_tokens, current_audio_processed_upto = await asyncio.to_thread(self.online.process_iter)
                 
-                # Get buffer information
-                _buffer_transcript_obj = self.online.get_buffer()
-                buffer_text = _buffer_transcript_obj.text
+                _buffer_transcript = self.online.get_buffer()
+                buffer_text = _buffer_transcript.text
 
                 if new_tokens:
                     validated_text = self.sep.join([t.text for t in new_tokens])
                     if buffer_text.startswith(validated_text):
-                        buffer_text = buffer_text[len(validated_text):].lstrip()
+                        _buffer_transcript.text = buffer_text[len(validated_text):].lstrip()
 
                 candidate_end_times = [self.end_buffer]
 
                 if new_tokens:
                     candidate_end_times.append(new_tokens[-1].end)
                 
-                if _buffer_transcript_obj.end is not None:
-                    candidate_end_times.append(_buffer_transcript_obj.end)
+                if _buffer_transcript.end is not None:
+                    candidate_end_times.append(_buffer_transcript.end)
                 
                 candidate_end_times.append(current_audio_processed_upto)
                 
-                new_end_buffer = max(candidate_end_times)
+                async with self.lock:
+                    self.tokens.extend(new_tokens)
+                    self.buffer_transcription = _buffer_transcript
+                    self.end_buffer = max(candidate_end_times)
                 
-                await self.update_transcription(
-                    new_tokens, buffer_text, new_end_buffer
-                )
-                
-                if new_tokens and self.args.target_language and self.translation_queue:
+                if self.translation_queue:
                     for token in new_tokens:
                         await self.translation_queue.put(token)
                         
@@ -247,8 +293,7 @@ class AudioProcessor:
 
     async def diarization_processor(self, diarization_obj):
         """Process audio chunks for speaker diarization."""
-        buffer_diarization = ""
-        cumulative_pcm_duration_stream_time = 0.0
+        self.current_speaker = 0
         while True:
             try:
                 item = await self.diarization_queue.get()
@@ -256,30 +301,36 @@ class AudioProcessor:
                     logger.debug("Diarization processor received sentinel. Finishing.")
                     self.diarization_queue.task_done()
                     break
-                
-                if type(item) is Silence:
-                    cumulative_pcm_duration_stream_time += item.duration
+                elif type(item) is Silence:
                     diarization_obj.insert_silence(item.duration)
                     continue
-    
-                if isinstance(item, np.ndarray):
+                elif isinstance(item, np.ndarray):
                     pcm_array = item
                 else:
                     raise Exception('item should be pcm_array') 
                 
                 # Process diarization
                 await diarization_obj.diarize(pcm_array)
+                segments = diarization_obj.get_segments()
                 
-                async with self.lock:
-                    self.tokens = diarization_obj.assign_speakers_to_tokens(
-                        self.tokens,
-                        use_punctuation_split=self.args.punctuation_split
-                    )
-                    if len(self.tokens) > 0:
-                        self.end_attributed_speaker = max(self.tokens[-1].end, self.end_attributed_speaker)
-                    if buffer_diarization:
-                        self.buffer_diarization = buffer_diarization
-                
+                if self.diarization_before_transcription:
+                    if segments and segments[-1].speaker != self.current_speaker:
+                        self.current_speaker = segments[-1].speaker
+                        cut_at = int(segments[-1].start*16000 - (self.cumulative_pcm_len))
+                        await self.transcription_queue.put(pcm_array[cut_at:])
+                        await self.transcription_queue.put(ChangeSpeaker(speaker=self.current_speaker, start=cut_at))                        
+                        await self.transcription_queue.put(pcm_array[:cut_at])
+                    else:
+                        await self.transcription_queue.put(pcm_array)
+                else:           
+                    async with self.lock:
+                        self.tokens = diarization_obj.assign_speakers_to_tokens(
+                            self.tokens,
+                            use_punctuation_split=self.args.punctuation_split
+                        )
+                self.cumulative_pcm_len += len(pcm_array)            
+                if len(self.tokens) > 0:
+                    self.end_attributed_speaker = max(self.tokens[-1].end, self.end_attributed_speaker)
                 self.diarization_queue.task_done()
                 
             except Exception as e:
@@ -289,20 +340,23 @@ class AudioProcessor:
                     self.diarization_queue.task_done()
         logger.info("Diarization processor task finished.")
 
-    async def translation_processor(self, online_translation):
+    async def translation_processor(self):
         # the idea is to ignore diarization for the moment. We use only transcription tokens. 
         # And the speaker is attributed given the segments used for the translation
         # in the future we want to have different languages for each speaker etc, so it will be more complex.
         while True:
             try:
-                token = await self.translation_queue.get() #block until at least 1 token
-                if token is SENTINEL:
+                item = await self.translation_queue.get() #block until at least 1 token
+                if item is SENTINEL:
                     logger.debug("Translation processor received sentinel. Finishing.")
                     self.translation_queue.task_done()
                     break
+                elif type(item) is Silence:
+                    self.online_translation.insert_silence(item.duration)
+                    continue
                 
                 # get all the available tokens for translation. The more words, the more precise
-                tokens_to_process = [token]
+                tokens_to_process = [item]
                 additional_tokens = await get_all_from_queue(self.translation_queue)
                 
                 sentinel_found = False
@@ -312,9 +366,8 @@ class AudioProcessor:
                         break
                     tokens_to_process.append(additional_token)                
                 if tokens_to_process:
-                    online_translation.insert_tokens(tokens_to_process)
-                    self.translated_segments = await asyncio.to_thread(online_translation.process)
-                
+                    self.online_translation.insert_tokens(tokens_to_process)
+                    self.translated_segments = await asyncio.to_thread(self.online_translation.process)
                 self.translation_queue.task_done()
                 for _ in additional_tokens:
                     self.translation_queue.task_done()
@@ -326,7 +379,7 @@ class AudioProcessor:
             except Exception as e:
                 logger.warning(f"Exception in translation_processor: {e}")
                 logger.warning(f"Traceback: {traceback.format_exc()}")
-                if 'token' in locals() and token is not SENTINEL:
+                if 'token' in locals() and item is not SENTINEL:
                     self.translation_queue.task_done()
                 if 'additional_tokens' in locals():
                     for _ in additional_tokens:
@@ -337,6 +390,16 @@ class AudioProcessor:
         """Format processing results for output."""
         while True:
             try:
+                # If FFmpeg error occurred, notify front-end
+                if self._ffmpeg_error:
+                    yield FrontData(
+                        status="error",
+                        error=f"FFmpeg error: {self._ffmpeg_error}"
+                    )
+                    self._ffmpeg_error = None
+                    await asyncio.sleep(1)
+                    continue
+
                 # Get current state
                 state = await self.get_current_state()
                                 
@@ -347,7 +410,7 @@ class AudioProcessor:
                     state = await self.get_current_state()
                 
                 # Format output
-                lines, undiarized_text, buffer_transcription, buffer_diarization = format_output(
+                lines, undiarized_text, end_w_silence = format_output(
                     state,
                     self.silence,
                     current_time = time() - self.beg_loop if self.beg_loop else None,
@@ -355,30 +418,34 @@ class AudioProcessor:
                     debug = self.debug,
                     sep=self.sep
                 )
-                # Handle undiarized text
+                if end_w_silence:
+                    buffer_transcription = Transcript()
+                else:
+                    buffer_transcription = state.buffer_transcription
+
+                buffer_diarization = ''
                 if undiarized_text:
-                    combined = self.sep.join(undiarized_text)
-                    if buffer_transcription:
-                        combined += self.sep
-                    await self.update_diarization(state.end_attributed_speaker, combined)
-                    buffer_diarization = combined
+                    buffer_diarization = self.sep.join(undiarized_text)
+
+                    async with self.lock:
+                        self.end_attributed_speaker = state.end_attributed_speaker
                 
                 response_status = "active_transcription"
                 if not state.tokens and not buffer_transcription and not buffer_diarization:
                     response_status = "no_audio_detected"
                     lines = []
-                elif response_status == "active_transcription" and not lines:
+                elif not lines:
                     lines = [Line(
                         speaker=1,
-                        start=state.get("end_buffer", 0),
-                        end=state.get("end_buffer", 0)        
+                        start=state.end_buffer,
+                        end=state.end_buffer
                     )]
                 
                 response = FrontData(
                     status=response_status,
                     lines=lines,
-                    buffer_transcription=buffer_transcription,
-                    buffer_diarization=buffer_diarization,
+                    buffer_transcription=buffer_transcription.text.strip(),
+                    buffer_diarization=buffer_diarization.strip(),
                     remaining_time_transcription=state.remaining_time_transcription,
                     remaining_time_diarization=state.remaining_time_diarization if self.args.diarization else 0
                 )
@@ -412,6 +479,21 @@ class AudioProcessor:
         self.all_tasks_for_cleanup = []
         processing_tasks_for_watchdog = []
 
+        # If using FFmpeg (non-PCM input), start it and spawn stdout reader
+        if not self.is_pcm_input:
+            success = await self.ffmpeg_manager.start()
+            if not success:
+                logger.error("Failed to start FFmpeg manager")
+                async def error_generator():
+                    yield FrontData(
+                        status="error",
+                        error="FFmpeg failed to start. Please check that FFmpeg is installed."
+                    )
+                return error_generator()
+            self.ffmpeg_reader_task = asyncio.create_task(self.ffmpeg_stdout_reader())
+            self.all_tasks_for_cleanup.append(self.ffmpeg_reader_task)
+            processing_tasks_for_watchdog.append(self.ffmpeg_reader_task)
+
         if self.args.transcription and self.online:
             self.transcription_task = asyncio.create_task(self.transcription_processor())
             self.all_tasks_for_cleanup.append(self.transcription_task)
@@ -422,8 +504,8 @@ class AudioProcessor:
             self.all_tasks_for_cleanup.append(self.diarization_task)
             processing_tasks_for_watchdog.append(self.diarization_task)
         
-        if self.args.target_language and self.args.lan != 'auto':
-            self.translation_task = asyncio.create_task(self.translation_processor(self.online_translation))
+        if self.online_translation:
+            self.translation_task = asyncio.create_task(self.translation_processor())
             self.all_tasks_for_cleanup.append(self.translation_task)
             processing_tasks_for_watchdog.append(self.translation_task)
         
@@ -462,13 +544,20 @@ class AudioProcessor:
             if task and not task.done():
                 task.cancel()
             
-            created_tasks = [t for t in self.all_tasks_for_cleanup if t]
-            if created_tasks:
-                await asyncio.gather(*created_tasks, return_exceptions=True)
-            logger.info("All processing tasks cancelled or finished.")
-            if self.args.diarization and hasattr(self, 'diarization') and hasattr(self.diarization, 'close'):
-                self.diarization.close()
-            logger.info("AudioProcessor cleanup complete.")
+        created_tasks = [t for t in self.all_tasks_for_cleanup if t]
+        if created_tasks:
+            await asyncio.gather(*created_tasks, return_exceptions=True)
+        logger.info("All processing tasks cancelled or finished.")
+
+        if not self.is_pcm_input and self.ffmpeg_manager:
+            try:
+                await self.ffmpeg_manager.stop()
+                logger.info("FFmpeg manager stopped.")
+            except Exception as e:
+                logger.warning(f"Error stopping FFmpeg manager: {e}")
+        if self.args.diarization and hasattr(self, 'dianization') and hasattr(self.diarization, 'close'):
+            self.diarization.close()
+        logger.info("AudioProcessor cleanup complete.")
 
 
     async def process_audio(self, message):
@@ -484,6 +573,9 @@ class AudioProcessor:
             if self.transcription_queue:
                 await self.transcription_queue.put(SENTINEL)
 
+            if not self.is_pcm_input and self.ffmpeg_manager:
+                await self.ffmpeg_manager.stop()
+
             return
 
         if self.is_stopping:
@@ -493,6 +585,17 @@ class AudioProcessor:
         if self.is_pcm_input:
             self.pcm_buffer.extend(message)
             await self.handle_pcm_data()
+        else:
+            if not self.ffmpeg_manager:
+                logger.error("FFmpeg manager not initialized for non-PCM input.")
+                return
+            success = await self.ffmpeg_manager.write_data(message)
+            if not success:
+                ffmpeg_state = await self.ffmpeg_manager.get_state()
+                if ffmpeg_state == FFmpegState.FAILED:
+                    logger.error("FFmpeg is in FAILED state, cannot process audio")
+                else:
+                    logger.warning("Failed to write audio data to FFmpeg")
 
     async def handle_pcm_data(self):
         # Process when enough data
@@ -524,13 +627,15 @@ class AudioProcessor:
                 silence_buffer = Silence(duration=time() - self.start_silence)
 
         if silence_buffer:
-            if self.args.transcription and self.transcription_queue:
+            if not self.diarization_before_transcription and self.transcription_queue:
                 await self.transcription_queue.put(silence_buffer)
             if self.args.diarization and self.diarization_queue:
                 await self.diarization_queue.put(silence_buffer)
+            if self.translation_queue:
+                await self.translation_queue.put(silence_buffer)
 
         if not self.silence:
-            if self.args.transcription and self.transcription_queue:
+            if not self.diarization_before_transcription and self.transcription_queue:
                 await self.transcription_queue.put(pcm_array.copy())
 
             if self.args.diarization and self.diarization_queue:

whisperlivekit/basic_server.py

@@ -5,9 +5,6 @@ from fastapi.middleware.cors import CORSMiddleware
 from whisperlivekit import TranscriptionEngine, AudioProcessor, get_inline_ui_html, parse_args
 import asyncio
 import logging
-from starlette.staticfiles import StaticFiles
-import pathlib
-import whisperlivekit.web as webpkg
 
 logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
 logging.getLogger().setLevel(logging.WARNING)
@@ -19,15 +16,6 @@ transcription_engine = None
 
 @asynccontextmanager
 async def lifespan(app: FastAPI):    
-    
-    #to remove after 0.2.8
-    if args.backend == "simulstreaming" and not args.disable_fast_encoder:
-        logger.warning(f"""
-{'='*50}
-WhisperLiveKit 0.2.8 has introduced a new fast encoder feature using MLX Whisper or Faster Whisper for improved speed. Use --disable-fast-encoder to disable if you encounter issues.
-{'='*50}
-    """)
-    
     global transcription_engine
     transcription_engine = TranscriptionEngine(
         **vars(args),
@@ -42,8 +30,6 @@ app.add_middleware(
     allow_methods=["*"],
     allow_headers=["*"],
 )
-web_dir = pathlib.Path(webpkg.__file__).parent
-app.mount("/web", StaticFiles(directory=str(web_dir)), name="web")
 
 @app.get("/")
 async def get():
@@ -73,6 +59,11 @@ async def websocket_endpoint(websocket: WebSocket):
     await websocket.accept()
     logger.info("WebSocket connection opened.")
 
+    try:
+        await websocket.send_json({"type": "config", "useAudioWorklet": bool(args.pcm_input)})
+    except Exception as e:
+        logger.warning(f"Failed to send config to client: {e}")
+            
     results_generator = await audio_processor.create_tasks()
     websocket_task = asyncio.create_task(handle_websocket_results(websocket, results_generator))
 

whisperlivekit/core.py

@@ -43,10 +43,12 @@ class TranscriptionEngine:
             "transcription": True,
             "vad": True,
             "pcm_input": False,
+            
             # whisperstreaming params:
             "buffer_trimming": "segment",
             "confidence_validation": False,
             "buffer_trimming_sec": 15,
+            
             # simulstreaming params:
             "disable_fast_encoder": False,
             "frame_threshold": 25,
@@ -61,10 +63,15 @@ class TranscriptionEngine:
             "max_context_tokens": None,
             "model_path": './base.pt',
             "diarization_backend": "sortformer",
+            
             # diarization params:
             "disable_punctuation_split" : False,
             "segmentation_model": "pyannote/segmentation-3.0",
             "embedding_model": "pyannote/embedding",  
+            
+            # translation params:
+            "nllb_backend": "ctranslate2",
+            "nllb_size": "600M"
         }
 
         config_dict = {**defaults, **kwargs}
@@ -138,12 +145,11 @@ class TranscriptionEngine:
         
         self.translation_model = None
         if self.args.target_language:
-            if self.args.lan == 'auto':
-                raise Exception('Translation cannot be set with language auto')
+            if self.args.lan == 'auto' and self.args.backend != "simulstreaming":
+                raise Exception('Translation cannot be set with language auto when transcription backend is not simulstreaming')
             else:
                 from whisperlivekit.translation.translation import load_model
-                self.translation_model = load_model([self.args.lan]) #in the future we want to handle different languages for different speakers
-            
+                self.translation_model = load_model([self.args.lan], backend=self.args.nllb_backend, model_size=self.args.nllb_size) #in the future we want to handle different languages for different speakers
         TranscriptionEngine._initialized = True
 
 

whisperlivekit/diarization/sortformer_backend.py

@@ -289,6 +289,7 @@ class SortformerDiarizationOnline:
             
         Returns:
             List of tokens with speaker assignments
+            Last speaker_segment
         """
         with self.segment_lock:
             segments = self.speaker_segments.copy()

whisperlivekit/ffmpeg_manager.py

@@ -0,0 +1,197 @@
+import asyncio
+import logging
+from enum import Enum
+from typing import Optional, Callable
+import contextlib
+
+logger = logging.getLogger(__name__)
+logging.basicConfig(level=logging.INFO)
+
+ERROR_INSTALL_INSTRUCTIONS = f"""
+{'='*50}
+FFmpeg is not installed or not found in your system's PATH.
+Alternative Solution: You can still use WhisperLiveKit without FFmpeg by adding the --pcm-input parameter. Note that when using this option, audio will not be compressed between the frontend and backend, which may result in higher bandwidth usage.
+
+If you want to install FFmpeg:
+
+# Ubuntu/Debian:
+sudo apt update && sudo apt install ffmpeg
+
+# macOS (using Homebrew):
+brew install ffmpeg
+
+# Windows:
+# 1. Download the latest static build from https://ffmpeg.org/download.html
+# 2. Extract the archive (e.g., to C:\\FFmpeg).
+# 3. Add the 'bin' directory (e.g., C:\\FFmpeg\\bin) to your system's PATH environment variable.
+
+After installation, please restart the application.
+{'='*50}
+"""
+
+class FFmpegState(Enum):
+    STOPPED = "stopped"
+    STARTING = "starting"
+    RUNNING = "running"
+    RESTARTING = "restarting"
+    FAILED = "failed"
+
+class FFmpegManager:
+    def __init__(self, sample_rate: int = 16000, channels: int = 1):
+        self.sample_rate = sample_rate
+        self.channels = channels
+
+        self.process: Optional[asyncio.subprocess.Process] = None
+        self._stderr_task: Optional[asyncio.Task] = None
+
+        self.on_error_callback: Optional[Callable[[str], None]] = None
+
+        self.state = FFmpegState.STOPPED
+        self._state_lock = asyncio.Lock()
+
+    async def start(self) -> bool:
+        async with self._state_lock:
+            if self.state != FFmpegState.STOPPED:
+                logger.warning(f"FFmpeg already running in state: {self.state}")
+                return False
+            self.state = FFmpegState.STARTING
+
+        try:
+            cmd = [
+                "ffmpeg",
+                "-hide_banner",
+                "-loglevel", "error",
+                "-i", "pipe:0",
+                "-f", "s16le",
+                "-acodec", "pcm_s16le",
+                "-ac", str(self.channels),
+                "-ar", str(self.sample_rate),
+                "pipe:1"
+            ]
+
+            self.process = await asyncio.create_subprocess_exec(
+                *cmd,
+                stdin=asyncio.subprocess.PIPE,
+                stdout=asyncio.subprocess.PIPE,
+                stderr=asyncio.subprocess.PIPE
+            )
+
+            self._stderr_task = asyncio.create_task(self._drain_stderr())
+
+            async with self._state_lock:
+                self.state = FFmpegState.RUNNING
+
+            logger.info("FFmpeg started.")
+            return True
+
+        except FileNotFoundError:
+            logger.error(ERROR_INSTALL_INSTRUCTIONS)
+            async with self._state_lock:
+                self.state = FFmpegState.FAILED
+            if self.on_error_callback:
+                await self.on_error_callback("ffmpeg_not_found")
+            return False
+
+        except Exception as e:
+            logger.error(f"Error starting FFmpeg: {e}")
+            async with self._state_lock:
+                self.state = FFmpegState.FAILED
+            if self.on_error_callback:
+                await self.on_error_callback("start_failed")
+            return False
+
+    async def stop(self):
+        async with self._state_lock:
+            if self.state == FFmpegState.STOPPED:
+                return
+            self.state = FFmpegState.STOPPED
+
+        if self.process:
+            if self.process.stdin and not self.process.stdin.is_closing():
+                self.process.stdin.close()
+                await self.process.stdin.wait_closed()
+            await self.process.wait()
+            self.process = None
+
+        if self._stderr_task:
+            self._stderr_task.cancel()
+            with contextlib.suppress(asyncio.CancelledError):
+                await self._stderr_task
+
+        logger.info("FFmpeg stopped.")
+
+    async def write_data(self, data: bytes) -> bool:
+        async with self._state_lock:
+            if self.state != FFmpegState.RUNNING:
+                logger.warning(f"Cannot write, FFmpeg state: {self.state}")
+                return False
+
+        try:
+            self.process.stdin.write(data)
+            await self.process.stdin.drain()
+            return True
+        except Exception as e:
+            logger.error(f"Error writing to FFmpeg: {e}")
+            if self.on_error_callback:
+                await self.on_error_callback("write_error")
+            return False
+
+    async def read_data(self, size: int) -> Optional[bytes]:
+        async with self._state_lock:
+            if self.state != FFmpegState.RUNNING:
+                logger.warning(f"Cannot read, FFmpeg state: {self.state}")
+                return None
+
+        try:
+            data = await asyncio.wait_for(
+                self.process.stdout.read(size),
+                timeout=20.0
+            )
+            return data
+        except asyncio.TimeoutError:
+            logger.warning("FFmpeg read timeout.")
+            return None
+        except Exception as e:
+            logger.error(f"Error reading from FFmpeg: {e}")
+            if self.on_error_callback:
+                await self.on_error_callback("read_error")
+            return None
+
+    async def get_state(self) -> FFmpegState:
+        async with self._state_lock:
+            return self.state
+
+    async def restart(self) -> bool:
+        async with self._state_lock:
+            if self.state == FFmpegState.RESTARTING:
+                logger.warning("Restart already in progress.")
+                return False
+            self.state = FFmpegState.RESTARTING
+
+        logger.info("Restarting FFmpeg...")
+
+        try:
+            await self.stop()
+            await asyncio.sleep(1)  # short delay before restarting
+            return await self.start()
+        except Exception as e:
+            logger.error(f"Error during FFmpeg restart: {e}")
+            async with self._state_lock:
+                self.state = FFmpegState.FAILED
+            if self.on_error_callback:
+                await self.on_error_callback("restart_failed")
+            return False
+
+    async def _drain_stderr(self):
+        try:
+            while True:
+                if not self.process or not self.process.stderr:
+                    break
+                line = await self.process.stderr.readline()
+                if not line:
+                    break
+                logger.debug(f"FFmpeg stderr: {line.decode(errors='ignore').strip()}")
+        except asyncio.CancelledError:
+            logger.info("FFmpeg stderr drain task cancelled.")
+        except Exception as e:
+            logger.error(f"Error draining FFmpeg stderr: {e}")

whisperlivekit/parse_args.py

@@ -177,7 +177,7 @@ def parse_args():
         "--pcm-input",
         action="store_true",
         default=False,
-        help="If set, raw PCM (s16le) data is expected as input and FFmpeg will be bypassed."
+        help="If set, raw PCM (s16le) data is expected as input and FFmpeg will be bypassed. Frontend will use AudioWorklet instead of MediaRecorder."
     )
     # SimulStreaming-specific arguments
     simulstreaming_group = parser.add_argument_group('SimulStreaming arguments (only used with --backend simulstreaming)')
@@ -287,6 +287,20 @@ def parse_args():
         help="Optional. Number of models to preload in memory to speed up loading (set up to the expected number of concurrent instances).",
     )
 
+    simulstreaming_group.add_argument(
+        "--nllb-backend",
+        type=str,
+        default="ctranslate2",
+        help="transformers or ctranslate2",
+    )
+    
+    simulstreaming_group.add_argument(
+        "--nllb-size",
+        type=str,
+        default="600M",
+        help="600M or 1.3B",
+    )
+
     args = parser.parse_args()
     
     args.transcription = not args.no_transcription

whisperlivekit/remove_silences.py

@@ -39,7 +39,7 @@ def blank_to_silence(tokens):
                         )
                 else:
                     if silence_token: #there was silence but no more
-                        if silence_token.end - silence_token.start >= MIN_SILENCE_DURATION:
+                        if silence_token.duration() >= MIN_SILENCE_DURATION:
                             cleaned_tokens.append(
                                 silence_token
                             )
@@ -77,15 +77,17 @@ def no_token_to_silence(tokens):
             new_tokens.append(token)
     return new_tokens
             
-def ends_with_silence(tokens, buffer_transcription, buffer_diarization, current_time, vac_detected_silence):
+def ends_with_silence(tokens, current_time, vac_detected_silence):
+    end_w_silence = False
     if not tokens:
-        return [], buffer_transcription, buffer_diarization
+        return [], end_w_silence
     last_token = tokens[-1]
     if tokens and current_time and (
         current_time - last_token.end >= END_SILENCE_DURATION 
         or
         (current_time - last_token.end >= 3 and vac_detected_silence)
         ):
+        end_w_silence = True
         if last_token.speaker == -2:
             last_token.end = current_time
         else:
@@ -97,14 +99,12 @@ def ends_with_silence(tokens, buffer_transcription, buffer_diarization, current_
                     probability=0.95
                 )
             )
-        buffer_transcription = "" # for whisperstreaming backend, we should probably validate the buffer has because of the silence
-        buffer_diarization  = ""
-    return tokens, buffer_transcription, buffer_diarization
+    return tokens, end_w_silence
     
 
-def handle_silences(tokens, buffer_transcription, buffer_diarization, current_time, vac_detected_silence):
+def handle_silences(tokens, current_time, vac_detected_silence):
     tokens = blank_to_silence(tokens) #useful for simulstreaming backend which tends to generate [BLANK_AUDIO] text
     tokens = no_token_to_silence(tokens)
-    tokens, buffer_transcription, buffer_diarization = ends_with_silence(tokens, buffer_transcription, buffer_diarization, current_time, vac_detected_silence)
-    return tokens, buffer_transcription, buffer_diarization
+    tokens, end_w_silence = ends_with_silence(tokens, current_time, vac_detected_silence)
+    return tokens, end_w_silence
      

whisperlivekit/results_formater.py

@@ -6,11 +6,10 @@ from whisperlivekit.timed_objects import Line, format_time
 logger = logging.getLogger(__name__)
 logger.setLevel(logging.DEBUG)
 
-PUNCTUATION_MARKS = {'.', '!', '?', '。', '！', '？'}
 CHECK_AROUND = 4
 
 def is_punctuation(token):
-    if token.text.strip() in PUNCTUATION_MARKS:
+    if token.is_punctuation():
         return True
     return False
 
@@ -39,26 +38,28 @@ def new_line(
         text = token.text + debug_info,
         start = token.start,
         end = token.end,
+        detected_language=token.detected_language
     )
 
 def append_token_to_last_line(lines, sep, token, debug_info):
     if token.text:
         lines[-1].text += sep + token.text + debug_info
         lines[-1].end = token.end
+    if not lines[-1].detected_language and token.detected_language:
+        lines[-1].detected_language = token.detected_language
+         
 
 def format_output(state, silence, current_time, args, debug, sep):
     diarization = args.diarization
     disable_punctuation_split = args.disable_punctuation_split
     tokens = state.tokens
     translated_segments = state.translated_segments # Here we will attribute the speakers only based on the timestamps of the segments
-    buffer_transcription = state.buffer_transcription
-    buffer_diarization = state.buffer_diarization
     end_attributed_speaker = state.end_attributed_speaker
     
     previous_speaker = -1
     lines = []
     undiarized_text = []
-    tokens, buffer_transcription, buffer_diarization = handle_silences(tokens, buffer_transcription, buffer_diarization, current_time, silence)
+    tokens, end_w_silence = handle_silences(tokens, current_time, silence)
     last_punctuation = None
     for i, token in enumerate(tokens):
         speaker = token.speaker
@@ -122,15 +123,39 @@ def format_output(state, silence, current_time, args, debug, sep):
                 pass
             
         append_token_to_last_line(lines, sep, token, debug_info)
-    if lines and translated_segments:
-        cts_idx = 0 # current_translated_segment_idx
-        for line in lines:
-            while cts_idx < len(translated_segments):
-                ts = translated_segments[cts_idx]
-                if ts and ts.start and ts.start >= line.start and ts.end <= line.end:
-                    line.translation += ts.text + ' '
-                    cts_idx += 1
-                else:
-                    break
-    return lines, undiarized_text, buffer_transcription, '' 
 
+    if lines and translated_segments:
+        unassigned_translated_segments = []
+        for ts in translated_segments:
+            assigned = False
+            for line in lines:
+                if ts and ts.overlaps_with(line):
+                    if ts.is_within(line):
+                        line.translation += ts.text + ' '
+                        assigned = True
+                        break
+                    else:
+                        ts0, ts1 = ts.approximate_cut_at(line.end)
+                        if ts0 and line.overlaps_with(ts0):
+                            line.translation += ts0.text + ' '
+                        if ts1:
+                            unassigned_translated_segments.append(ts1)
+                        assigned = True
+                        break
+            if not assigned:
+                unassigned_translated_segments.append(ts)
+        
+        if unassigned_translated_segments:
+            for line in lines:
+                remaining_segments = []
+                for ts in unassigned_translated_segments:
+                    if ts and ts.overlaps_with(line):
+                        line.translation += ts.text + ' '
+                    else:
+                        remaining_segments.append(ts)
+                unassigned_translated_segments = remaining_segments #maybe do smth in the future about that
+    
+    if state.buffer_transcription and lines:
+        lines[-1].end = max(state.buffer_transcription.end, lines[-1].end)
+        
+    return lines, undiarized_text, end_w_silence

whisperlivekit/simul_whisper/backend.py

@@ -4,9 +4,8 @@ import logging
 from typing import List, Tuple, Optional
 import logging
 import platform
-from whisperlivekit.timed_objects import ASRToken, Transcript
+from whisperlivekit.timed_objects import ASRToken, Transcript, ChangeSpeaker
 from whisperlivekit.warmup import load_file
-from whisperlivekit.simul_whisper.license_simulstreaming import SIMULSTREAMING_LICENSE
 from .whisper import load_model, tokenizer
 from .whisper.audio import TOKENS_PER_SECOND
 import os
@@ -23,7 +22,11 @@ try:
     HAS_MLX_WHISPER = True
 except ImportError:
     if platform.system() == "Darwin" and platform.machine() == "arm64":
-        print('MLX Whisper not found but you are on Apple Silicon. Consider installing mlx-whisper for better performance: pip install mlx-whisper')
+        print(f"""
+            {"="*50}
+            MLX Whisper not found but you are on Apple Silicon. Consider installing mlx-whisper for better performance: pip install mlx-whisper
+            {"="*50}
+            """)
     HAS_MLX_WHISPER = False
 if HAS_MLX_WHISPER:
     HAS_FASTER_WHISPER = False
@@ -49,8 +52,7 @@ class SimulStreamingOnlineProcessor:
         self.asr = asr
         self.logfile = logfile
         self.end = 0.0
-        self.global_time_offset = 0.0
-        
+        self.buffer = []
         self.committed: List[ASRToken] = []
         self.last_result_tokens: List[ASRToken] = []
         self.load_new_backend()
@@ -79,7 +81,7 @@ class SimulStreamingOnlineProcessor:
         else:
             self.process_iter(is_last=True) #we want to totally process what remains in the buffer.
             self.model.refresh_segment(complete=True)
-            self.global_time_offset = silence_duration + offset
+            self.model.global_time_offset = silence_duration + offset
 
 
         
@@ -91,63 +93,15 @@ class SimulStreamingOnlineProcessor:
         self.end = audio_stream_end_time #Only to be aligned with what happens in whisperstreaming backend.
         self.model.insert_audio(audio_tensor)
 
+    def new_speaker(self, change_speaker: ChangeSpeaker):
+            self.process_iter(is_last=True)
+            self.model.refresh_segment(complete=True)
+            self.model.speaker = change_speaker.speaker
+            self.global_time_offset = change_speaker.start
+            
     def get_buffer(self):
-        return Transcript(
-            start=None, 
-            end=None, 
-            text='', 
-            probability=None
-        )
-
-    def timestamped_text(self, tokens, generation):
-        """
-        generate timestamped text from tokens and generation data.
-        
-        args:
-            tokens: List of tokens to process
-            generation: Dictionary containing generation progress and optionally results
-            
-        returns:
-            List of tuples containing (start_time, end_time, word) for each word
-        """
-        FRAME_DURATION = 0.02    
-        if "result" in generation:
-            split_words = generation["result"]["split_words"]
-            split_tokens = generation["result"]["split_tokens"]
-        else:
-            split_words, split_tokens = self.model.tokenizer.split_to_word_tokens(tokens)
-        progress = generation["progress"]
-        frames = [p["most_attended_frames"][0] for p in progress]
-        absolute_timestamps = [p["absolute_timestamps"][0] for p in progress]
-        tokens_queue = tokens.copy()
-        timestamped_words = []
-        
-        for word, word_tokens in zip(split_words, split_tokens):
-            # start_frame = None
-            # end_frame = None
-            for expected_token in word_tokens:
-                if not tokens_queue or not frames:
-                    raise ValueError(f"Insufficient tokens or frames for word '{word}'")
-                    
-                actual_token = tokens_queue.pop(0)
-                current_frame = frames.pop(0)
-                current_timestamp = absolute_timestamps.pop(0)
-                if actual_token != expected_token:
-                    raise ValueError(
-                        f"Token mismatch: expected '{expected_token}', "
-                        f"got '{actual_token}' at frame {current_frame}"
-                    )
-                # if start_frame is None:
-                #     start_frame = current_frame
-                # end_frame = current_frame
-            # start_time = start_frame * FRAME_DURATION
-            # end_time = end_frame * FRAME_DURATION
-            start_time = current_timestamp
-            end_time = current_timestamp + 0.1
-            timestamp_entry = (start_time, end_time, word)
-            timestamped_words.append(timestamp_entry)
-            logger.debug(f"TS-WORD:\t{start_time:.2f}\t{end_time:.2f}\t{word}")
-        return timestamped_words
+        concat_buffer = Transcript.from_tokens(tokens= self.buffer, sep='')
+        return concat_buffer
 
     def process_iter(self, is_last=False) -> Tuple[List[ASRToken], float]:
         """
@@ -156,47 +110,14 @@ class SimulStreamingOnlineProcessor:
         Returns a tuple: (list of committed ASRToken objects, float representing the audio processed up to time).
         """
         try:
-            tokens, generation_progress = self.model.infer(is_last=is_last)
-            ts_words = self.timestamped_text(tokens, generation_progress)
+            timestamped_words = self.model.infer(is_last=is_last)
+            if timestamped_words and timestamped_words[0].detected_language == None:
+                self.buffer.extend(timestamped_words)
+                return [], self.end
             
-            new_tokens = []
-            for ts_word in ts_words:
-                
-                start, end, word = ts_word
-                token = ASRToken(
-                    start=start,
-                    end=end,
-                    text=word,
-                    probability=0.95  # fake prob. Maybe we can extract it from the model?
-                ).with_offset(
-                    self.global_time_offset
-                )
-                new_tokens.append(token)
-                
-            # identical_tokens = 0
-            # n_new_tokens = len(new_tokens)
-            # if n_new_tokens:
-            
-            self.committed.extend(new_tokens)
-            
-            # if token in self.committed:
-            #     pos = len(self.committed) - 1 - self.committed[::-1].index(token)
-            # if pos:
-            #     for i in range(len(self.committed) - n_new_tokens, -1, -n_new_tokens):
-            #         commited_segment = self.committed[i:i+n_new_tokens]
-            #         if commited_segment == new_tokens:
-            #             identical_segments +=1
-            #             if identical_tokens >= TOO_MANY_REPETITIONS:
-            #                 logger.warning('Too many repetition, model is stuck. Load a new one')
-            #                 self.committed = self.committed[:i]
-            #                 self.load_new_backend()
-            #                 return [], self.end
-
-            # pos = self.committed.rindex(token)
-
-            
-            
-            return new_tokens, self.end
+            self.committed.extend(timestamped_words)
+            self.buffer = []
+            return timestamped_words, self.end
 
             
         except Exception as e:
@@ -226,7 +147,6 @@ class SimulStreamingASR():
     sep = ""
 
     def __init__(self, lan, modelsize=None, cache_dir=None, model_dir=None, logfile=sys.stderr, **kwargs):
-        logger.warning(SIMULSTREAMING_LICENSE)
         self.logfile = logfile
         self.transcribe_kargs = {}
         self.original_language = lan

whisperlivekit/simul_whisper/simul_whisper.py

@@ -8,6 +8,7 @@ import torch.nn.functional as F
 
 from .whisper import load_model, DecodingOptions, tokenizer
 from .config import AlignAttConfig
+from whisperlivekit.timed_objects import ASRToken
 from .whisper.audio import log_mel_spectrogram, TOKENS_PER_SECOND, pad_or_trim, N_SAMPLES, N_FRAMES
 from .whisper.timing import median_filter
 from .whisper.decoding import GreedyDecoder, BeamSearchDecoder, SuppressTokens, detect_language
@@ -18,6 +19,7 @@ from time import time
 from .token_buffer import TokenBuffer
 
 import numpy as np
+from ..timed_objects import PUNCTUATION_MARKS
 from .generation_progress import *
 
 DEC_PAD = 50257
@@ -40,12 +42,6 @@ else:
     except ImportError:
         HAS_FASTER_WHISPER = False
 
-# New features added to the original version of Simul-Whisper: 
-# - large-v3 model support
-# - translation support
-# - beam search
-# - prompt -- static vs. non-static
-# - context
 class PaddedAlignAttWhisper:
     def __init__(
             self, 
@@ -70,7 +66,7 @@ class PaddedAlignAttWhisper:
             self.fw_feature_extractor = FeatureExtractor(feature_size=self.model.dims.n_mels)
             
         logger.info(f"Model dimensions: {self.model.dims}")
-
+        self.speaker = -1
         self.decode_options = DecodingOptions(
             language = cfg.language, 
             without_timestamps = True,
@@ -78,7 +74,10 @@ class PaddedAlignAttWhisper:
         )
         self.tokenizer_is_multilingual = not model_name.endswith(".en")
         self.create_tokenizer(cfg.language if cfg.language != "auto" else None)
+        # self.create_tokenizer('en')
         self.detected_language = cfg.language if cfg.language != "auto" else None
+        self.global_time_offset = 0.0
+        self.reset_tokenizer_to_auto_next_call = False
         
         self.max_text_len = self.model.dims.n_text_ctx
         self.num_decoder_layers = len(self.model.decoder.blocks)
@@ -153,6 +152,7 @@ class PaddedAlignAttWhisper:
         
         self.last_attend_frame = -self.cfg.rewind_threshold
         self.cumulative_time_offset = 0.0
+        self.first_timestamp = None
 
         if self.cfg.max_context_tokens is None:
             self.max_context_tokens = self.max_text_len
@@ -260,7 +260,6 @@ class PaddedAlignAttWhisper:
         self.init_context()
         logger.debug(f"Context: {self.context}")
         if not complete and len(self.segments) > 2:
-            logger.debug("keeping last two segments because they are and it is not complete.")
             self.segments = self.segments[-2:]
         else:
             logger.debug("removing all segments.")
@@ -382,11 +381,11 @@ class PaddedAlignAttWhisper:
         new_segment = True
         if len(self.segments) == 0:
             logger.debug("No segments, nothing to do")
-            return [], {}
+            return []
         if not self._apply_minseglen():
             logger.debug(f"applied minseglen {self.cfg.audio_min_len} > {self.segments_len()}.")
             input_segments = torch.cat(self.segments, dim=0)
-            return [], {}
+            return []
 
         # input_segments is concatenation of audio, it's one array
         if len(self.segments) > 1:
@@ -394,6 +393,13 @@ class PaddedAlignAttWhisper:
         else:
             input_segments = self.segments[0]
 
+        # if self.cfg.language == "auto" and self.reset_tokenizer_to_auto_next_call:
+        #     logger.debug("Resetting tokenizer to auto for new sentence.")
+        #     self.create_tokenizer(None)
+        #     self.detected_language = None
+        #     self.init_tokens()
+        #     self.reset_tokenizer_to_auto_next_call = False
+
         # NEW : we can use a different encoder, before using standart whisper for cross attention with the hooks on the decoder
         beg_encode = time()
         if self.mlx_encoder:
@@ -426,58 +432,38 @@ class PaddedAlignAttWhisper:
         end_encode = time()
         # print('Encoder duration:', end_encode-beg_encode)
                 
-#        logger.debug(f"Encoder feature shape: {encoder_feature.shape}")
-#        if mel.shape[-2:] != (self.model.dims.n_audio_ctx, self.model.dims.n_audio_state):
-#            logger.debug("mel ")
-        if self.cfg.language == "auto" and self.detected_language is None:
-            language_tokens, language_probs = self.lang_id(encoder_feature) 
-            logger.debug(f"Language tokens: {language_tokens}, probs: {language_probs}")
-            top_lan, p = max(language_probs[0].items(), key=lambda x: x[1])
-            logger.info(f"Detected language: {top_lan} with p={p:.4f}")
-            #self.tokenizer.language = top_lan
-            #self.tokenizer.__post_init__()
-            self.create_tokenizer(top_lan)
-            self.detected_language = top_lan
-            self.init_tokens()
-            logger.info(f"Tokenizer language: {self.tokenizer.language}, {self.tokenizer.sot_sequence_including_notimestamps}")
+        if self.cfg.language == "auto" and self.detected_language is None and self.first_timestamp:
+            seconds_since_start = self.segments_len() - self.first_timestamp
+            if seconds_since_start >= 2.0:
+                language_tokens, language_probs = self.lang_id(encoder_feature) 
+                top_lan, p = max(language_probs[0].items(), key=lambda x: x[1])
+                print(f"Detected language: {top_lan} with p={p:.4f}")
+                self.create_tokenizer(top_lan)
+                self.last_attend_frame = -self.cfg.rewind_threshold
+                self.cumulative_time_offset = 0.0
+                self.init_tokens()
+                self.init_context()
+                self.detected_language = top_lan
+                logger.info(f"Tokenizer language: {self.tokenizer.language}, {self.tokenizer.sot_sequence_including_notimestamps}")
 
         self.trim_context()
         current_tokens = self._current_tokens()
-#        
+   
         fire_detected = self.fire_at_boundary(encoder_feature[:, :content_mel_len, :])
 
 
-        ####################### Decoding loop
-        logger.info("Decoding loop starts\n")
-
         sum_logprobs = torch.zeros(self.cfg.beam_size, device=self.device)
         completed = False
+        # punctuation_stop = False
 
         attn_of_alignment_heads = None
         most_attended_frame = None
 
         token_len_before_decoding = current_tokens.shape[1]
         
-        generation_progress = []
-        generation = {
-            "starting_tokens": BeamTokens(current_tokens[0,:].clone(), self.cfg.beam_size),
-            "token_len_before_decoding": token_len_before_decoding,
-            #"fire_detected": fire_detected,
-            "frames_len": content_mel_len,
-            "frames_threshold": 4 if is_last else self.cfg.frame_threshold,
+        l_absolute_timestamps = []
         
-            # to be filled later
-            "logits_starting": None,
-
-            # to be filled later
-            "no_speech_prob": None,
-            "no_speech": False,
-
-            # to be filled in the loop
-            "progress": generation_progress,
-        }
         while not completed and current_tokens.shape[1] < self.max_text_len: # bos is 3 tokens
-            generation_progress_loop = []
 
             if new_segment:
                 tokens_for_logits = current_tokens
@@ -486,50 +472,26 @@ class PaddedAlignAttWhisper:
                 tokens_for_logits = current_tokens[:,-1:]
 
             logits = self.logits(tokens_for_logits, encoder_feature) # B, len(tokens), token dict size
-            if new_segment:
-                generation["logits_starting"] = Logits(logits[:,:,:])
 
             if new_segment and self.tokenizer.no_speech is not None:
                 probs_at_sot = logits[:, self.sot_index, :].float().softmax(dim=-1)
                 no_speech_probs = probs_at_sot[:, self.tokenizer.no_speech].tolist()
-                generation["no_speech_prob"] = no_speech_probs[0]
                 if no_speech_probs[0] > self.cfg.nonspeech_prob:
-                    generation["no_speech"] = True
                     logger.info("no speech, stop")
                     break
 
             logits = logits[:, -1, :] # logits for the last token
-            generation_progress_loop.append(("logits_before_suppress",Logits(logits)))
 
             # supress blank tokens only at the beginning of the segment
             if new_segment:
                 logits[:, self.tokenizer.encode(" ") + [self.tokenizer.eot]] = -np.inf
             new_segment = False
             self.suppress_tokens(logits)
-            #generation_progress_loop.append(("logits_after_suppres",BeamLogits(logits[0,:].clone(), self.cfg.beam_size)))
-            generation_progress_loop.append(("logits_after_suppress",Logits(logits)))
-
             current_tokens, completed = self.token_decoder.update(current_tokens, logits, sum_logprobs)
-            generation_progress_loop.append(("beam_tokens",Tokens(current_tokens[:,-1].clone())))
-            generation_progress_loop.append(("sum_logprobs",sum_logprobs.tolist()))
-            generation_progress_loop.append(("completed",completed))
 
             logger.debug(f"Decoding completed: {completed}, sum_logprobs: {sum_logprobs.tolist()}, tokens: ")
             self.debug_print_tokens(current_tokens)
 
-
-            # if self.decoder_type == "beam":
-            #     logger.debug(f"Finished sequences: {self.token_decoder.finished_sequences}")
-
-            #     logprobs = F.log_softmax(logits.float(), dim=-1)
-            #     idx = 0
-            #     logger.debug(f"Beam search topk: {logprobs[idx].topk(self.cfg.beam_size + 1)}")
-            #     logger.debug(f"Greedy search argmax: {logits.argmax(dim=-1)}")
-            # if completed:
-            #     self.debug_print_tokens(current_tokens)
-
-            #     logger.debug("decode stopped because decoder completed")
-
             attn_of_alignment_heads = [[] for _ in range(self.num_align_heads)]
             for i, attn_mat in enumerate(self.dec_attns):
                 layer_rank = int(i % len(self.model.decoder.blocks))
@@ -548,30 +510,24 @@ class PaddedAlignAttWhisper:
                 t = torch.cat(mat, dim=1)
                 tmp.append(t) 
             attn_of_alignment_heads = torch.stack(tmp, dim=1)
-#            logger.debug(str(attn_of_alignment_heads.shape) + " tttady")
             std, mean = torch.std_mean(attn_of_alignment_heads, dim=-2, keepdim=True, unbiased=False)
             attn_of_alignment_heads = (attn_of_alignment_heads - mean) / std
             attn_of_alignment_heads = median_filter(attn_of_alignment_heads, 7) # from whisper.timing
             attn_of_alignment_heads = attn_of_alignment_heads.mean(dim=1)
-#            logger.debug(str(attn_of_alignment_heads.shape) + " po mean")
             attn_of_alignment_heads = attn_of_alignment_heads[:,:, :content_mel_len]
-#            logger.debug(str(attn_of_alignment_heads.shape) + " pak ")
 
             # for each beam, the most attended frame is:
             most_attended_frames = torch.argmax(attn_of_alignment_heads[:,-1,:], dim=-1)
-            generation_progress_loop.append(("most_attended_frames",most_attended_frames.clone().tolist()))
             
             # Calculate absolute timestamps accounting for cumulative offset
             absolute_timestamps = [(frame * 0.02 + self.cumulative_time_offset) for frame in most_attended_frames.tolist()]
-            generation_progress_loop.append(("absolute_timestamps", absolute_timestamps))
             
             logger.debug(str(most_attended_frames.tolist()) + " most att frames")
             logger.debug(f"Absolute timestamps: {absolute_timestamps} (offset: {self.cumulative_time_offset:.2f}s)")
 
             most_attended_frame = most_attended_frames[0].item()
+            l_absolute_timestamps.append(absolute_timestamps[0])
 
-
-            generation_progress.append(dict(generation_progress_loop))
             logger.debug("current tokens" + str(current_tokens.shape))
             if completed:
             #    # stripping the last token, the eot
@@ -609,66 +565,53 @@ class PaddedAlignAttWhisper:
                     self.tokenizer.decode([current_tokens[i, -1].item()])
                 ))
 
-#        for k,v in generation.items():
-#            print(k,v,file=sys.stderr)
-#        for x in generation_progress:
-#            for y in x.items():
-#                print("\t\t",*y,file=sys.stderr)
-#            print("\t","----", file=sys.stderr)
-#        print("\t", "end of generation_progress_loop", file=sys.stderr)
-        #    sys.exit(1)
-        ####################### End of decoding loop
-
-        logger.info("End of decoding loop")
-
-        # if attn_of_alignment_heads is not None:
-        #     seg_len = int(segment.shape[0] / 16000 * TOKENS_PER_SECOND)
-
-        #     # Lets' now consider only the top hypothesis in the beam search
-        #     top_beam_attn_of_alignment_heads = attn_of_alignment_heads[0]
-
-        #     # debug print: how is the new token attended?
-        #     new_token_attn = top_beam_attn_of_alignment_heads[token_len_before_decoding:, -seg_len:]
-        #     logger.debug(f"New token attention shape: {new_token_attn.shape}")
-        #     if new_token_attn.shape[0] == 0:  # it's not attended in the current audio segment
-        #         logger.debug("no token generated")
-        #     else:  # it is, and the max attention is:
-        #         new_token_max_attn, _ = new_token_attn.max(dim=-1)
-        #         logger.debug(f"segment max attention: {new_token_max_attn.mean().item()/len(self.segments)}")
-
-
-        # let's now operate only with the top beam hypothesis
         tokens_to_split = current_tokens[0, token_len_before_decoding:]
-        if fire_detected or is_last:
+
+        if fire_detected or is_last: #or punctuation_stop:
             new_hypothesis = tokens_to_split.flatten().tolist()
+            split_words, split_tokens = self.tokenizer.split_to_word_tokens(new_hypothesis)
         else:
             # going to truncate the tokens after the last space
             split_words, split_tokens = self.tokenizer.split_to_word_tokens(tokens_to_split.tolist())
-            generation["result"] = {"split_words": split_words[:-1], "split_tokens": split_tokens[:-1]}
-            generation["result_truncated"] = {"split_words": split_words[-1:], "split_tokens": split_tokens[-1:]}
-
-#            text_to_split = self.tokenizer.decode(tokens_to_split)
-#            logger.debug(f"text_to_split: {text_to_split}")
-#            logger.debug("text at current step: {}".format(text_to_split.replace(" ", "<space>")))
-#            text_before_space = " ".join(text_to_split.split(" ")[:-1])
-#            logger.debug("before the last space: {}".format(text_before_space.replace(" ", "<space>")))
             if len(split_words) > 1:
                 new_hypothesis = [i for sublist in split_tokens[:-1] for i in sublist]  
             else:
                 new_hypothesis = []
 
 
-        ### new hypothesis
         logger.debug(f"new_hypothesis: {new_hypothesis}")
         new_tokens = torch.tensor([new_hypothesis], dtype=torch.long).repeat_interleave(self.cfg.beam_size, dim=0).to(
             device=self.device,
         )
         self.tokens.append(new_tokens)
-        # TODO: test if this is redundant or not
-#        ret = ret[ret<DEC_PAD]
 
         logger.info(f"Output: {self.tokenizer.decode(new_hypothesis)}")
         
         self._clean_cache()
 
-        return new_hypothesis, generation
+        if len(l_absolute_timestamps) >=2 and self.first_timestamp is None:
+            self.first_timestamp = l_absolute_timestamps[0]
+            
+
+        timestamped_words = []
+        timestamp_idx = 0
+        for word, word_tokens in zip(split_words, split_tokens):
+            try:
+                current_timestamp = l_absolute_timestamps[timestamp_idx]
+            except:
+                pass
+            timestamp_idx += len(word_tokens)
+
+            timestamp_entry = ASRToken(
+                    start=current_timestamp,
+                    end=current_timestamp + 0.1,
+                    text= word,
+                    probability=0.95,
+                    speaker=self.speaker,
+                    detected_language=self.detected_language
+                ).with_offset(
+                    self.global_time_offset
+            )
+            timestamped_words.append(timestamp_entry)
+        
+        return timestamped_words

whisperlivekit/timed_objects.py

@@ -1,7 +1,9 @@
 from dataclasses import dataclass, field
-from typing import Optional
+from typing import Optional, Any, List
 from datetime import timedelta
 
+PUNCTUATION_MARKS = {'.', '!', '?', '。', '！', '？'}
+
 def format_time(seconds: float) -> str:
     """Format seconds as HH:MM:SS."""
     return str(timedelta(seconds=int(seconds)))
@@ -15,12 +17,35 @@ class TimedText:
     speaker: Optional[int] = -1
     probability: Optional[float] = None
     is_dummy: Optional[bool] = False
+    detected_language: Optional[str] = None
     
-@dataclass
+    def is_punctuation(self):
+        return self.text.strip() in PUNCTUATION_MARKS
+    
+    def overlaps_with(self, other: 'TimedText') -> bool:
+        return not (self.end <= other.start or other.end <= self.start)
+    
+    def is_within(self, other: 'TimedText') -> bool:
+        return other.contains_timespan(self)
+
+    def duration(self) -> float:
+        return self.end - self.start
+
+    def contains_time(self, time: float) -> bool:
+        return self.start <= time <= self.end
+
+    def contains_timespan(self, other: 'TimedText') -> bool:
+        return self.start <= other.start and self.end >= other.end
+    
+    def __bool__(self):
+        return bool(self.text)
+
+
+@dataclass()
 class ASRToken(TimedText):
     def with_offset(self, offset: float) -> "ASRToken":
         """Return a new token with the time offset added."""
-        return ASRToken(self.start + offset, self.end + offset, self.text, self.speaker, self.probability)
+        return ASRToken(self.start + offset, self.end + offset, self.text, self.speaker, self.probability, detected_language=self.detected_language)
 
 @dataclass
 class Sentence(TimedText):
@@ -28,7 +53,28 @@ class Sentence(TimedText):
 
 @dataclass
 class Transcript(TimedText):
-    pass
+    """
+    represents a concatenation of several ASRToken
+    """
+
+    @classmethod
+    def from_tokens(
+        cls,
+        tokens: List[ASRToken],
+        sep: Optional[str] = None,
+        offset: float = 0
+    ) -> "Transcript":
+        sep = sep if sep is not None else ' '
+        text = sep.join(token.text for token in tokens)
+        probability = sum(token.probability for token in tokens if token.probability) / len(tokens) if tokens else None
+        if tokens:
+            start = offset + tokens[0].start
+            end = offset + tokens[-1].end
+        else:
+            start = None
+            end = None
+        return cls(start, end, text, probability=probability)
+
 
 @dataclass
 class SpeakerSegment(TimedText):
@@ -41,6 +87,34 @@ class SpeakerSegment(TimedText):
 class Translation(TimedText):
     pass
 
+    def approximate_cut_at(self, cut_time):
+        """
+        Each word in text is considered to be of duration (end-start)/len(words in text)
+        """
+        if not self.text or not self.contains_time(cut_time):
+            return self, None
+
+        words = self.text.split()
+        num_words = len(words)
+        if num_words == 0:
+            return self, None
+
+        duration_per_word = self.duration() / num_words
+        
+        cut_word_index = int((cut_time - self.start) / duration_per_word)
+        
+        if cut_word_index >= num_words:
+            cut_word_index = num_words -1
+        
+        text0 = " ".join(words[:cut_word_index])
+        text1 = " ".join(words[cut_word_index:])
+
+        segment0 = Translation(start=self.start, end=cut_time, text=text0)
+        segment1 = Translation(start=cut_time, end=self.end, text=text1)
+
+        return segment0, segment1
+        
+
 @dataclass
 class Silence():
     duration: float
@@ -51,13 +125,18 @@ class Line(TimedText):
     translation: str = ''
     
     def to_dict(self):
-        return {
+        _dict = {
             'speaker': int(self.speaker),
             'text': self.text,
-            'translation': self.translation,
             'start': format_time(self.start),
             'end': format_time(self.end),
         }
+        if self.translation:
+            _dict['translation'] = self.translation
+        if self.detected_language:
+            _dict['detected_language'] = self.detected_language
+        return _dict
+    
 
 @dataclass  
 class FrontData():
@@ -82,12 +161,16 @@ class FrontData():
             _dict['error'] = self.error
         return _dict
 
+@dataclass  
+class ChangeSpeaker:
+    speaker: int
+    start: int
+
 @dataclass  
 class State():
     tokens: list
     translated_segments: list
     buffer_transcription: str
-    buffer_diarization: str
     end_buffer: float
     end_attributed_speaker: float
     remaining_time_transcription: float

whisperlivekit/translation/translation.py

@@ -1,42 +1,63 @@
+import logging
+import time
 import ctranslate2
 import torch
 import transformers
-from dataclasses import dataclass
+from dataclasses import dataclass, field
 import huggingface_hub
 from whisperlivekit.translation.mapping_languages import get_nllb_code
 from whisperlivekit.timed_objects import Translation
 
+logger = logging.getLogger(__name__)
 
 #In diarization case, we may want to translate just one speaker, or at least start the sentences there
 
-PUNCTUATION_MARKS = {'.', '!', '?', '。', '！', '？'}
-
+MIN_SILENCE_DURATION_DEL_BUFFER = 3 #After a silence of x seconds, we consider the model should not use the buffer, even if the previous
+# sentence is not finished.
 
 @dataclass
 class TranslationModel():
     translator: ctranslate2.Translator
-    tokenizer: dict
+    device: str
+    tokenizer: dict = field(default_factory=dict)
+    backend_type: str = 'ctranslate2'
+    model_size: str = '600M'
     
-def load_model(src_langs):
-    MODEL = 'nllb-200-distilled-600M-ctranslate2'
-    MODEL_GUY = 'entai2965'
-    huggingface_hub.snapshot_download(MODEL_GUY + '/' + MODEL,local_dir=MODEL)
+    def get_tokenizer(self, input_lang):
+        if not self.tokenizer.get(input_lang, False):
+            self.tokenizer[input_lang] = transformers.AutoTokenizer.from_pretrained(
+                f"facebook/nllb-200-distilled-{self.model_size}",
+                src_lang=input_lang,
+                clean_up_tokenization_spaces=True
+            )
+        return self.tokenizer[input_lang]
+            
+
+def load_model(src_langs, backend='ctranslate2', model_size='600M'):
     device = "cuda" if torch.cuda.is_available() else "cpu"
-    translator = ctranslate2.Translator(MODEL,device=device)
+    MODEL = f'nllb-200-distilled-{model_size}-ctranslate2'
+    if backend=='ctranslate2':
+        MODEL_GUY = 'entai2965'
+        huggingface_hub.snapshot_download(MODEL_GUY + '/' + MODEL,local_dir=MODEL)
+        translator = ctranslate2.Translator(MODEL,device=device)
+    elif backend=='transformers':
+        translator = transformers.AutoModelForSeq2SeqLM.from_pretrained(f"facebook/nllb-200-distilled-{model_size}")
     tokenizer = dict()
     for src_lang in src_langs:
-        tokenizer[src_lang] = transformers.AutoTokenizer.from_pretrained(MODEL, src_lang=src_lang, clean_up_tokenization_spaces=True)
-    return TranslationModel(
-        translator=translator,
-        tokenizer=tokenizer
-    )
+        if src_lang != 'auto':
+            tokenizer[src_lang] = transformers.AutoTokenizer.from_pretrained(MODEL, src_lang=src_lang, clean_up_tokenization_spaces=True)
 
-def translate(input, translation_model, tgt_lang):
-    source = translation_model.tokenizer.convert_ids_to_tokens(translation_model.tokenizer.encode(input))
-    target_prefix = [tgt_lang]
-    results = translation_model.translator.translate_batch([source], target_prefix=[target_prefix])
-    target = results[0].hypotheses[0][1:]
-    return translation_model.tokenizer.decode(translation_model.tokenizer.convert_tokens_to_ids(target))
+    translation_model = TranslationModel(
+        translator=translator,
+        tokenizer=tokenizer,
+        backend_type=backend,
+        device = device,
+        model_size = model_size
+    )
+    for src_lang in src_langs:
+        if src_lang != 'auto':
+            translation_model.get_tokenizer(src_lang)
+    return translation_model
 
 class OnlineTranslation:
     def __init__(self, translation_model: TranslationModel, input_languages: list, output_languages: list):
@@ -59,27 +80,38 @@ class OnlineTranslation:
                     self.commited.extend(self.buffer[:i])
                     self.buffer = results[i:]
 
-    def translate(self, input, input_lang=None, output_lang=None):
+    def translate(self, input, input_lang, output_lang):
         if not input:
             return ""
-        if input_lang is None:
-            input_lang = self.input_languages[0]
-        if output_lang is None:
-            output_lang = self.output_languages[0]
         nllb_output_lang = get_nllb_code(output_lang)
             
-        source = self.translation_model.tokenizer[input_lang].convert_ids_to_tokens(self.translation_model.tokenizer[input_lang].encode(input))   
-        results = self.translation_model.translator.translate_batch([source], target_prefix=[[nllb_output_lang]]) #we can use return_attention=True to try to optimize the stuff.
-        target = results[0].hypotheses[0][1:]
-        results = self.translation_model.tokenizer[input_lang].decode(self.translation_model.tokenizer[input_lang].convert_tokens_to_ids(target))
-        return results
+        tokenizer = self.translation_model.get_tokenizer(input_lang)
+        tokenizer_output = tokenizer(input, return_tensors="pt").to(self.translation_model.device)
+        
+        if self.translation_model.backend_type == 'ctranslate2':
+            source = tokenizer.convert_ids_to_tokens(tokenizer_output['input_ids'][0])    
+            results = self.translation_model.translator.translate_batch([source], target_prefix=[[nllb_output_lang]])
+            target = results[0].hypotheses[0][1:]
+            result = tokenizer.decode(tokenizer.convert_tokens_to_ids(target))
+        else:
+            translated_tokens = self.translation_model.translator.generate(**tokenizer_output, forced_bos_token_id=tokenizer.convert_tokens_to_ids(nllb_output_lang))
+            result = tokenizer.batch_decode(translated_tokens, skip_special_tokens=True)[0]
+        return result
     
     def translate_tokens(self, tokens):
         if tokens:
             text = ' '.join([token.text for token in tokens])
             start = tokens[0].start
             end = tokens[-1].end
-            translated_text = self.translate(text)
+            if self.input_languages[0] == 'auto':
+                input_lang = tokens[0].detected_language
+            else:
+                input_lang = self.input_languages[0]
+                
+            translated_text = self.translate(text,
+                                            input_lang,
+                                            self.output_languages[0]
+                                            )
             translation = Translation(
                 text=translated_text,
                 start=start,
@@ -89,7 +121,6 @@ class OnlineTranslation:
         return None
             
 
-
     def insert_tokens(self, tokens):
         self.buffer.extend(tokens)
         pass
@@ -99,7 +130,7 @@ class OnlineTranslation:
         if len(self.buffer) < self.len_processed_buffer + 3: #nothing new to process
             return self.validated + [self.translation_remaining]
         while i < len(self.buffer):
-            if self.buffer[i].text in PUNCTUATION_MARKS:
+            if self.buffer[i].is_punctuation():
                 translation_sentence = self.translate_tokens(self.buffer[:i+1])
                 self.validated.append(translation_sentence)
                 self.buffer = self.buffer[i+1:]
@@ -110,6 +141,10 @@ class OnlineTranslation:
         self.len_processed_buffer = len(self.buffer)
         return self.validated + [self.translation_remaining]
 
+    def insert_silence(self, silence_duration: float):
+        if silence_duration >= MIN_SILENCE_DURATION_DEL_BUFFER:
+            self.buffer = []
+            self.validated += [self.translation_remaining]
 
 if __name__ == '__main__':
     output_lang = 'fr'
@@ -122,16 +157,13 @@ if __name__ == '__main__':
     test = test_string.split(' ')
     step = len(test) // 3
     
-    shared_model = load_model([input_lang])
+    shared_model = load_model([input_lang], backend='ctranslate2')
     online_translation = OnlineTranslation(shared_model, input_languages=[input_lang], output_languages=[output_lang])
     
+    beg_inference = time.time()    
     for id in range(5):
         val = test[id*step : (id+1)*step]
         val_str = ' '.join(val)
         result = online_translation.translate(val_str)
         print(result)
-    
-    
-    
-    
-    # print(result)
+    print('inference time:', time.time() - beg_inference)

whisperlivekit/web/live_transcription.css

@@ -346,7 +346,7 @@ label {
 
 .label_diarization {
   background-color: var(--chip-bg);
-  border-radius: 8px 8px 8px 8px;
+  border-radius: 100px;
   padding: 2px 10px;
   margin-left: 10px;
   display: inline-block;
@@ -358,7 +358,7 @@ label {
 
 .label_transcription {
   background-color: var(--chip-bg);
-  border-radius: 8px 8px 8px 8px;
+  border-radius: 100px;
   padding: 2px 10px;
   display: inline-block;
   white-space: nowrap;
@@ -370,16 +370,20 @@ label {
 
 .label_translation {
   background-color: var(--chip-bg);
+  display: inline-flex;
   border-radius: 10px;
   padding: 4px 8px;
   margin-top: 4px;
   font-size: 14px;
   color: var(--text);
-  display: flex;
   align-items: flex-start;
   gap: 4px;
 }
 
+.lag-diarization-value {
+    margin-left: 10px;
+}
+
 .label_translation img {
   margin-top: 2px;
 }
@@ -391,7 +395,7 @@ label {
 
 #timeInfo {
   color: var(--muted);
-  margin-left: 10px;
+  margin-left: 0px;
 }
 
 .textcontent {
@@ -438,7 +442,6 @@ label {
   font-size: 13px;
   border-radius: 30px;
   padding: 2px 10px;
-  display: none;
 }
 
 .loading {
@@ -515,3 +518,33 @@ label {
     padding: 10px;
   }
 }
+
+.label_language {
+  background-color: var(--chip-bg);
+  margin-bottom: 0px;
+  margin-top: 5px;
+  height: 18.5px;
+  border-radius: 100px;
+  padding: 2px 8px;
+  margin-left: 10px;
+  display: inline-flex;
+  align-items: center;
+  gap: 4px;
+  font-size: 14px;
+  color: var(--muted);
+}
+
+
+.speaker-badge {
+  display: inline-flex;
+  align-items: center;
+  justify-content: center;
+  width: 16px;
+  height: 16px;
+  margin-left: -5px;
+  border-radius: 50%;
+  font-size: 11px;
+  line-height: 1;
+  font-weight: 800;
+  color: var(--muted);
+}

whisperlivekit/web/live_transcription.js

@@ -22,6 +22,9 @@ let lastReceivedData = null;
 let lastSignature = null;
 let availableMicrophones = [];
 let selectedMicrophoneId = null;
+let serverUseAudioWorklet = null;
+let configReadyResolve;
+const configReady = new Promise((r) => (configReadyResolve = r));
 
 waveCanvas.width = 60 * (window.devicePixelRatio || 1);
 waveCanvas.height = 30 * (window.devicePixelRatio || 1);
@@ -37,6 +40,11 @@ const timerElement = document.querySelector(".timer");
 const themeRadios = document.querySelectorAll('input[name="theme"]');
 const microphoneSelect = document.getElementById("microphoneSelect");
 
+const translationIcon = `<svg xmlns="http://www.w3.org/2000/svg" height="12px" viewBox="0 -960 960 960" width="12px" fill="#5f6368"><path d="m603-202-34 97q-4 11-14 18t-22 7q-20 0-32.5-16.5T496-133l152-402q5-11 15-18t22-7h30q12 0 22 7t15 18l152 403q8 19-4 35.5T868-80q-13 0-22.5-7T831-106l-34-96H603ZM362-401 188-228q-11 11-27.5 11.5T132-228q-11-11-11-28t11-28l174-174q-35-35-63.5-80T190-640h84q20 39 40 68t48 58q33-33 68.5-92.5T484-720H80q-17 0-28.5-11.5T40-760q0-17 11.5-28.5T80-800h240v-40q0-17 11.5-28.5T360-880q17 0 28.5 11.5T400-840v40h240q17 0 28.5 11.5T680-760q0 17-11.5 28.5T640-720h-76q-21 72-63 148t-83 116l96 98-30 82-122-125Zm266 129h144l-72-204-72 204Z"/></svg>`
+const silenceIcon = `<svg xmlns="http://www.w3.org/2000/svg" style="vertical-align: text-bottom;" height="14px" viewBox="0 -960 960 960" width="14px" fill="#5f6368"><path d="M514-556 320-752q9-3 19-5.5t21-2.5q66 0 113 47t47 113q0 11-1.5 22t-4.5 22ZM40-200v-32q0-33 17-62t47-44q51-26 115-44t141-18q26 0 49.5 2.5T456-392l-56-54q-9 3-19 4.5t-21 1.5q-66 0-113-47t-47-113q0-11 1.5-21t4.5-19L84-764q-11-11-11-28t11-28q12-12 28.5-12t27.5 12l675 685q11 11 11.5 27.5T816-80q-11 13-28 12.5T759-80L641-200h39q0 33-23.5 56.5T600-120H120q-33 0-56.5-23.5T40-200Zm80 0h480v-32q0-14-4.5-19.5T580-266q-36-18-92.5-36T360-320q-71 0-127.5 18T140-266q-9 5-14.5 14t-5.5 20v32Zm240 0Zm560-400q0 69-24.5 131.5T829-355q-12 14-30 15t-32-13q-13-13-12-31t12-33q30-38 46.5-85t16.5-98q0-51-16.5-97T767-781q-12-15-12.5-33t12.5-32q13-14 31.5-13.5T829-845q42 51 66.5 113.5T920-600Zm-182 0q0 32-10 61.5T700-484q-11 15-29.5 15.5T638-482q-13-13-13.5-31.5T633-549q6-11 9.5-24t3.5-27q0-14-3.5-27t-9.5-25q-9-17-8.5-35t13.5-31q14-14 32.5-13.5T700-716q18 25 28 54.5t10 61.5Z"/></svg>`;
+const languageIcon = `<svg xmlns="http://www.w3.org/2000/svg" height="12" viewBox="0 -960 960 960" width="12" fill="#5f6368"><path d="M480-80q-82 0-155-31.5t-127.5-86Q143-252 111.5-325T80-480q0-83 31.5-155.5t86-127Q252-817 325-848.5T480-880q83 0 155.5 31.5t127 86q54.5 54.5 86 127T880-480q0 82-31.5 155t-86 127.5q-54.5 54.5-127 86T480-80Zm0-82q26-36 45-75t31-83H404q12 44 31 83t45 75Zm-104-16q-18-33-31.5-68.5T322-320H204q29 50 72.5 87t99.5 55Zm208 0q56-18 99.5-55t72.5-87H638q-9 38-22.5 73.5T584-178ZM170-400h136q-3-20-4.5-39.5T300-480q0-21 1.5-40.5T306-560H170q-5 20-7.5 39.5T160-480q0 21 2.5 40.5T170-400Zm216 0h188q3-20 4.5-39.5T580-480q0-21-1.5-40.5T574-560H386q-3 20-4.5 39.5T380-480q0 21 1.5 40.5T386-400Zm268 0h136q5-20 7.5-39.5T800-480q0-21-2.5-40.5T790-560H654q3 20 4.5 39.5T660-480q0 21-1.5 40.5T654-400Zm-16-240h118q-29-50-72.5-87T584-782q18 33 31.5 68.5T638-640Zm-234 0h152q-12-44-31-83t-45-75q-26 36-45 75t-31 83Zm-200 0h118q9-38 22.5-73.5T376-782q-56 18-99.5 55T204-640Z"/></svg>`
+const speakerIcon = `<svg xmlns="http://www.w3.org/2000/svg" height="16px" style="vertical-align: text-bottom;" viewBox="0 -960 960 960" width="16px" fill="#5f6368"><path d="M480-480q-66 0-113-47t-47-113q0-66 47-113t113-47q66 0 113 47t47 113q0 66-47 113t-113 47ZM160-240v-32q0-34 17.5-62.5T224-378q62-31 126-46.5T480-440q66 0 130 15.5T736-378q29 15 46.5 43.5T800-272v32q0 33-23.5 56.5T720-160H240q-33 0-56.5-23.5T160-240Zm80 0h480v-32q0-11-5.5-20T700-306q-54-27-109-40.5T480-360q-56 0-111 13.5T260-306q-9 5-14.5 14t-5.5 20v32Zm240-320q33 0 56.5-23.5T560-640q0-33-23.5-56.5T480-720q-33 0-56.5 23.5T400-640q0 33 23.5 56.5T480-560Zm0-80Zm0 400Z"/></svg>`;
+
 function getWaveStroke() {
   const styles = getComputedStyle(document.documentElement);
   const v = styles.getPropertyValue("--wave-stroke").trim();
@@ -228,6 +236,14 @@ function setupWebSocket() {
 
     websocket.onmessage = (event) => {
       const data = JSON.parse(event.data);
+      if (data.type === "config") {
+        serverUseAudioWorklet = !!data.useAudioWorklet;
+        statusText.textContent = serverUseAudioWorklet
+          ? "Connected. Using AudioWorklet (PCM)."
+          : "Connected. Using MediaRecorder (WebM).";
+        if (configReadyResolve) configReadyResolve();
+        return;
+      }
 
       if (data.type === "ready_to_stop") {
         console.log("Ready to stop received, finalizing display and closing WebSocket.");
@@ -295,7 +311,7 @@ function renderLinesWithBuffer(
   const showTransLag = !isFinalizing && remaining_time_transcription > 0;
   const showDiaLag = !isFinalizing && !!buffer_diarization && remaining_time_diarization > 0;
   const signature = JSON.stringify({
-    lines: (lines || []).map((it) => ({ speaker: it.speaker, text: it.text, start: it.start, end: it.end })),
+    lines: (lines || []).map((it) => ({ speaker: it.speaker, text: it.text, start: it.start, end: it.end, detected_language: it.detected_language })),
     buffer_transcription: buffer_transcription || "",
     buffer_diarization: buffer_diarization || "",
     status: current_status,
@@ -324,24 +340,22 @@ function renderLinesWithBuffer(
 
       let speakerLabel = "";
       if (item.speaker === -2) {
-        speakerLabel = `<span class="silence">Silence<span id='timeInfo'>${timeInfo}</span></span>`;
+        speakerLabel = `<span class="silence">${silenceIcon}<span id='timeInfo'>${timeInfo}</span></span>`;
       } else if (item.speaker == 0 && !isFinalizing) {
         speakerLabel = `<span class='loading'><span class="spinner"></span><span id='timeInfo'><span class="loading-diarization-value">${fmt1(
           remaining_time_diarization
         )}</span> second(s) of audio are undergoing diarization</span></span>`;
       } else if (item.speaker !== 0) {
-        speakerLabel = `<span id="speaker">Speaker ${item.speaker}<span id='timeInfo'>${timeInfo}</span></span>`;
+        const speakerNum = `<span class="speaker-badge">${item.speaker}</span>`;
+        speakerLabel = `<span id="speaker">${speakerIcon}${speakerNum}<span id='timeInfo'>${timeInfo}</span></span>`;
+
+        if (item.detected_language) {
+          speakerLabel += `<span class="label_language">${languageIcon}<span>${item.detected_language}</span></span>`;
+        }
       }
 
       let currentLineText = item.text || "";
 
-      if (item.translation) {
-        currentLineText += `<div class="label_translation">
-          <img src="/web/src/translate.svg" alt="Translation" width="12" height="12" />
-          <span>${item.translation}</span>
-        </div>`;
-      }
-
       if (idx === lines.length - 1) {
         if (!isFinalizing && item.speaker !== -2) {
           if (remaining_time_transcription > 0) {
@@ -375,6 +389,13 @@ function renderLinesWithBuffer(
         }
       }
       
+      if (item.translation) {
+        currentLineText += `<div class="label_translation">
+          ${translationIcon}
+          <span>${item.translation}</span>
+        </div>`;
+      }
+
       return currentLineText.trim().length > 0 || speakerLabel.length > 0
         ? `<p>${speakerLabel}<br/><div class='textcontent'>${currentLineText}</div></p>`
         : `<p>${speakerLabel}<br/></p>`;
@@ -459,38 +480,54 @@ async function startRecording() {
     microphone = audioContext.createMediaStreamSource(stream);
     microphone.connect(analyser);
 
-    if (!audioContext.audioWorklet) {
-      throw new Error("AudioWorklet is not supported in this browser");
-    }
-    await audioContext.audioWorklet.addModule("/web/pcm_worklet.js");
-    workletNode = new AudioWorkletNode(audioContext, "pcm-forwarder", { numberOfInputs: 1, numberOfOutputs: 0, channelCount: 1 });
-    microphone.connect(workletNode);
-
-    recorderWorker = new Worker("/web/recorder_worker.js");
-    recorderWorker.postMessage({
-      command: "init",
-      config: {
-        sampleRate: audioContext.sampleRate,
-      },
-    });
-
-    recorderWorker.onmessage = (e) => {
-      if (websocket && websocket.readyState === WebSocket.OPEN) {
-        websocket.send(e.data.buffer);
+    if (serverUseAudioWorklet) {
+      if (!audioContext.audioWorklet) {
+        throw new Error("AudioWorklet is not supported in this browser");
       }
-    };
+      await audioContext.audioWorklet.addModule("/web/pcm_worklet.js");
+      workletNode = new AudioWorkletNode(audioContext, "pcm-forwarder", { numberOfInputs: 1, numberOfOutputs: 0, channelCount: 1 });
+      microphone.connect(workletNode);
 
-    workletNode.port.onmessage = (e) => {
-      const data = e.data;
-      const ab = data instanceof ArrayBuffer ? data : data.buffer;
-      recorderWorker.postMessage(
-        {
-          command: "record",
-          buffer: ab,
+      recorderWorker = new Worker("/web/recorder_worker.js");
+      recorderWorker.postMessage({
+        command: "init",
+        config: {
+          sampleRate: audioContext.sampleRate,
         },
-        [ab]
-      );
-    };
+      });
+
+      recorderWorker.onmessage = (e) => {
+        if (websocket && websocket.readyState === WebSocket.OPEN) {
+          websocket.send(e.data.buffer);
+        }
+      };
+
+      workletNode.port.onmessage = (e) => {
+        const data = e.data;
+        const ab = data instanceof ArrayBuffer ? data : data.buffer;
+        recorderWorker.postMessage(
+          {
+            command: "record",
+            buffer: ab,
+          },
+          [ab]
+        );
+      };
+    } else {
+      try {
+        recorder = new MediaRecorder(stream, { mimeType: "audio/webm" });
+      } catch (e) {
+        recorder = new MediaRecorder(stream);
+      }
+      recorder.ondataavailable = (e) => {
+        if (websocket && websocket.readyState === WebSocket.OPEN) {
+          if (e.data && e.data.size > 0) {
+            websocket.send(e.data);
+          }
+        }
+      };
+      recorder.start(chunkDuration);
+    }
 
     startTime = Date.now();
     timerInterval = setInterval(updateTimer, 1000);
@@ -528,6 +565,14 @@ async function stopRecording() {
     statusText.textContent = "Recording stopped. Processing final audio...";
   }
 
+  if (recorder) {
+    try {
+      recorder.stop();
+    } catch (e) {
+    }
+    recorder = null;
+  }
+
   if (recorderWorker) {
     recorderWorker.terminate();
     recorderWorker = null;
@@ -586,9 +631,11 @@ async function toggleRecording() {
     console.log("Connecting to WebSocket");
     try {
       if (websocket && websocket.readyState === WebSocket.OPEN) {
+        await configReady;
         await startRecording();
       } else {
         await setupWebSocket();
+        await configReady;
         await startRecording();
       }
     } catch (err) {

whisperlivekit/web/src/language.svg

@@ -0,0 +1 @@
+<svg xmlns="http://www.w3.org/2000/svg" height="24px" viewBox="0 -960 960 960" width="24px" fill="#5f6368"><path d="M480-80q-82 0-155-31.5t-127.5-86Q143-252 111.5-325T80-480q0-83 31.5-155.5t86-127Q252-817 325-848.5T480-880q83 0 155.5 31.5t127 86q54.5 54.5 86 127T880-480q0 82-31.5 155t-86 127.5q-54.5 54.5-127 86T480-80Zm0-82q26-36 45-75t31-83H404q12 44 31 83t45 75Zm-104-16q-18-33-31.5-68.5T322-320H204q29 50 72.5 87t99.5 55Zm208 0q56-18 99.5-55t72.5-87H638q-9 38-22.5 73.5T584-178ZM170-400h136q-3-20-4.5-39.5T300-480q0-21 1.5-40.5T306-560H170q-5 20-7.5 39.5T160-480q0 21 2.5 40.5T170-400Zm216 0h188q3-20 4.5-39.5T580-480q0-21-1.5-40.5T574-560H386q-3 20-4.5 39.5T380-480q0 21 1.5 40.5T386-400Zm268 0h136q5-20 7.5-39.5T800-480q0-21-2.5-40.5T790-560H654q3 20 4.5 39.5T660-480q0 21-1.5 40.5T654-400Zm-16-240h118q-29-50-72.5-87T584-782q18 33 31.5 68.5T638-640Zm-234 0h152q-12-44-31-83t-45-75q-26 36-45 75t-31 83Zm-200 0h118q9-38 22.5-73.5T376-782q-56 18-99.5 55T204-640Z"/></svg>

whisperlivekit/web/src/silence.svg

@@ -0,0 +1 @@
+<svg xmlns="http://www.w3.org/2000/svg" height="24px" viewBox="0 -960 960 960" width="24px" fill="#5f6368"><path d="M514-556 320-752q9-3 19-5.5t21-2.5q66 0 113 47t47 113q0 11-1.5 22t-4.5 22ZM40-200v-32q0-33 17-62t47-44q51-26 115-44t141-18q26 0 49.5 2.5T456-392l-56-54q-9 3-19 4.5t-21 1.5q-66 0-113-47t-47-113q0-11 1.5-21t4.5-19L84-764q-11-11-11-28t11-28q12-12 28.5-12t27.5 12l675 685q11 11 11.5 27.5T816-80q-11 13-28 12.5T759-80L641-200h39q0 33-23.5 56.5T600-120H120q-33 0-56.5-23.5T40-200Zm80 0h480v-32q0-14-4.5-19.5T580-266q-36-18-92.5-36T360-320q-71 0-127.5 18T140-266q-9 5-14.5 14t-5.5 20v32Zm240 0Zm560-400q0 69-24.5 131.5T829-355q-12 14-30 15t-32-13q-13-13-12-31t12-33q30-38 46.5-85t16.5-98q0-51-16.5-97T767-781q-12-15-12.5-33t12.5-32q13-14 31.5-13.5T829-845q42 51 66.5 113.5T920-600Zm-182 0q0 32-10 61.5T700-484q-11 15-29.5 15.5T638-482q-13-13-13.5-31.5T633-549q6-11 9.5-24t3.5-27q0-14-3.5-27t-9.5-25q-9-17-8.5-35t13.5-31q14-14 32.5-13.5T700-716q18 25 28 54.5t10 61.5Z"/></svg>

whisperlivekit/web/src/speaker.svg

@@ -0,0 +1 @@
+<svg xmlns="http://www.w3.org/2000/svg" height="24px" viewBox="0 -960 960 960" width="24px" fill="#5f6368"><path d="M480-480q-66 0-113-47t-47-113q0-66 47-113t113-47q66 0 113 47t47 113q0 66-47 113t-113 47ZM160-240v-32q0-34 17.5-62.5T224-378q62-31 126-46.5T480-440q66 0 130 15.5T736-378q29 15 46.5 43.5T800-272v32q0 33-23.5 56.5T720-160H240q-33 0-56.5-23.5T160-240Zm80 0h480v-32q0-11-5.5-20T700-306q-54-27-109-40.5T480-360q-56 0-111 13.5T260-306q-9 5-14.5 14t-5.5 20v32Zm240-320q33 0 56.5-23.5T560-640q0-33-23.5-56.5T480-720q-33 0-56.5 23.5T400-640q0 33 23.5 56.5T480-560Zm0-80Zm0 400Z"/></svg>