RealTimeSTT 0.3.9__tar.gz → 0.3.81__tar.gz

{realtimestt-0.3.9 → RealtimeSTT-0.3.81}/PKG-INFO

@@ -1,6 +1,6 @@
 Metadata-Version: 2.1
 Name: RealtimeSTT
-Version: 0.3.9
+Version: 0.3.81
 Summary: A fast Voice Activity Detection and Transcription System
 Home-page: https://github.com/KoljaB/RealTimeSTT
 Author: Kolja Beigel
@@ -12,18 +12,6 @@ Classifier: Operating System :: OS Independent
 Requires-Python: >=3.6
 Description-Content-Type: text/markdown
 License-File: LICENSE
-Requires-Dist: PyAudio==0.2.14
-Requires-Dist: faster-whisper==1.0.3
-Requires-Dist: pvporcupine==1.9.5
-Requires-Dist: webrtcvad-wheels==2.0.14
-Requires-Dist: halo==0.0.31
-Requires-Dist: torch
-Requires-Dist: torchaudio
-Requires-Dist: scipy==1.14.1
-Requires-Dist: openwakeword>=0.4.0
-Requires-Dist: websockets==v12.0
-Requires-Dist: websocket-client==1.8.0
-Requires-Dist: numpy<2.0.0
 
 # RealtimeSTT
 [![PyPI](https://img.shields.io/pypi/v/RealtimeSTT)](https://pypi.org/project/RealtimeSTT/)
@@ -55,7 +43,7 @@ https://github.com/user-attachments/assets/797e6552-27cd-41b1-a7f3-e5cbc72094f5
 
 ### Updates
 
-Latest Version: v0.3.9
+Latest Version: v0.3.8
 
 See [release history](https://github.com/KoljaB/RealtimeSTT/releases).
 
@@ -129,26 +117,11 @@ pip install RealtimeSTT
 
 This will install all the necessary dependencies, including a **CPU support only** version of PyTorch.
 
-Although it is possible to run RealtimeSTT with a CPU installation only (use a small model like "tiny" or "base" in this case) you will get way better experience using CUDA (please scroll down).
-
-### Linux Installation
-
-Before installing RealtimeSTT please execute:
-
-```bash
-sudo apt-get install portaudio19-dev
-```
-
-### MacOS Installation
-
-Before installing RealtimeSTT please execute:
-
-```bash
-brew install portaudio
-```
+Although it is possible to run RealtimeSTT with a CPU installation only (use a small model like "tiny" or "base" in this case) you will get way better experience using:
 
 ### GPU Support with CUDA (recommended)
 
+
 ### Updating PyTorch for CUDA Support
 
 To upgrade your PyTorch installation to enable GPU support with CUDA, follow these instructions based on your specific CUDA version. This is useful if you wish to enhance the performance of RealtimeSTT with CUDA capabilities.
@@ -477,8 +450,6 @@ When you initialize the `AudioToTextRecorder` class, you have various options to
 
 - **level** (int, default=logging.WARNING): Logging level.
 
-- **batch_size** (int, default=16): Batch size for the main transcription. Set to 0 to deactivate.
-
 - **init_logging** (bool, default=True): Whether to initialize the logging framework. Set to False to manage this yourself.
 
 - **handle_buffer_overflow** (bool, default=True): If set, the system will log a warning when an input overflow occurs during recording and remove the data from the buffer.
@@ -518,8 +489,6 @@ When you initialize the `AudioToTextRecorder` class, you have various options to
 
 - **on_realtime_transcription_stabilized**: A callback function that is triggered whenever there's an update in the real-time transcription and returns a higher quality, stabilized text as its argument.
 
-- **realtime_batch_size**: (int, default=16): Batch size for the real-time transcription model. Set to 0 to deactivate.
-
 - **beam_size_realtime** (int, default=3): The beam size to use for real-time transcription beam search decoding.
 
 #### Voice Activation Parameters

{realtimestt-0.3.9 → RealtimeSTT-0.3.81}/README.md

@@ -28,7 +28,7 @@ https://github.com/user-attachments/assets/797e6552-27cd-41b1-a7f3-e5cbc72094f5
 
 ### Updates
 
-Latest Version: v0.3.9
+Latest Version: v0.3.8
 
 See [release history](https://github.com/KoljaB/RealtimeSTT/releases).
 
@@ -102,26 +102,11 @@ pip install RealtimeSTT
 
 This will install all the necessary dependencies, including a **CPU support only** version of PyTorch.
 
-Although it is possible to run RealtimeSTT with a CPU installation only (use a small model like "tiny" or "base" in this case) you will get way better experience using CUDA (please scroll down).
-
-### Linux Installation
-
-Before installing RealtimeSTT please execute:
-
-```bash
-sudo apt-get install portaudio19-dev
-```
-
-### MacOS Installation
-
-Before installing RealtimeSTT please execute:
-
-```bash
-brew install portaudio
-```
+Although it is possible to run RealtimeSTT with a CPU installation only (use a small model like "tiny" or "base" in this case) you will get way better experience using:
 
 ### GPU Support with CUDA (recommended)
 
+
 ### Updating PyTorch for CUDA Support
 
 To upgrade your PyTorch installation to enable GPU support with CUDA, follow these instructions based on your specific CUDA version. This is useful if you wish to enhance the performance of RealtimeSTT with CUDA capabilities.
@@ -450,8 +435,6 @@ When you initialize the `AudioToTextRecorder` class, you have various options to
 
 - **level** (int, default=logging.WARNING): Logging level.
 
-- **batch_size** (int, default=16): Batch size for the main transcription. Set to 0 to deactivate.
-
 - **init_logging** (bool, default=True): Whether to initialize the logging framework. Set to False to manage this yourself.
 
 - **handle_buffer_overflow** (bool, default=True): If set, the system will log a warning when an input overflow occurs during recording and remove the data from the buffer.
@@ -491,8 +474,6 @@ When you initialize the `AudioToTextRecorder` class, you have various options to
 
 - **on_realtime_transcription_stabilized**: A callback function that is triggered whenever there's an update in the real-time transcription and returns a higher quality, stabilized text as its argument.
 
-- **realtime_batch_size**: (int, default=16): Batch size for the real-time transcription model. Set to 0 to deactivate.
-
 - **beam_size_realtime** (int, default=3): The beam size to use for real-time transcription beam search decoding.
 
 #### Voice Activation Parameters

{realtimestt-0.3.9 → RealtimeSTT-0.3.81}/RealtimeSTT/audio_recorder.py

@@ -34,7 +34,6 @@ from openwakeword.model import Model
 from scipy.signal import resample
 from scipy import signal
 import signal as system_signal
-from faster_whisper import WhisperModel, BatchedInferencePipeline
 import faster_whisper
 import openwakeword
 import collections
@@ -88,12 +87,11 @@ if platform.system() != 'Darwin':
 
 
 class TranscriptionWorker:
-    def __init__(self, conn, stdout_pipe, model_path, download_root, compute_type, gpu_device_index, device,
-                 ready_event, shutdown_event, interrupt_stop_event, beam_size, initial_prompt, suppress_tokens, batch_size):
+    def __init__(self, conn, stdout_pipe, model_path, compute_type, gpu_device_index, device,
+                 ready_event, shutdown_event, interrupt_stop_event, beam_size, initial_prompt, suppress_tokens):
         self.conn = conn
         self.stdout_pipe = stdout_pipe
         self.model_path = model_path
-        self.download_root = download_root
         self.compute_type = compute_type
         self.gpu_device_index = gpu_device_index
         self.device = device
@@ -103,7 +101,6 @@ class TranscriptionWorker:
         self.beam_size = beam_size
         self.initial_prompt = initial_prompt
         self.suppress_tokens = suppress_tokens
-        self.batch_size = batch_size
         self.queue = queue.Queue()
 
     def custom_print(self, *args, **kwargs):
@@ -137,10 +134,7 @@ class TranscriptionWorker:
                 device=self.device,
                 compute_type=self.compute_type,
                 device_index=self.gpu_device_index,
-                download_root=self.download_root,
             )
-            if self.batch_size > 0:
-                model = BatchedInferencePipeline(model=model)
         except Exception as e:
             logging.exception(f"Error initializing main faster_whisper transcription model: {e}")
             raise
@@ -157,24 +151,13 @@ class TranscriptionWorker:
                 try:
                     audio, language = self.queue.get(timeout=0.1)
                     try:
-                        if self.batch_size > 0:
-                            segments, info = model.transcribe(
-                                audio,
-                                language=language if language else None,
-                                beam_size=self.beam_size,
-                                initial_prompt=self.initial_prompt,
-                                suppress_tokens=self.suppress_tokens,
-                                batch_size=self.batch_size
-                            )
-                        else:
-                            segments, info = model.transcribe(
-                                audio,
-                                language=language if language else None,
-                                beam_size=self.beam_size,
-                                initial_prompt=self.initial_prompt,
-                                suppress_tokens=self.suppress_tokens
-                            )
-
+                        segments, info = model.transcribe(
+                            audio,
+                            language=language if language else None,
+                            beam_size=self.beam_size,
+                            initial_prompt=self.initial_prompt,
+                            suppress_tokens=self.suppress_tokens
+                        )
                         transcription = " ".join(seg.text for seg in segments).strip()
                         logging.debug(f"Final text detected with main model: {transcription}")
                         self.conn.send(('success', (transcription, info)))
@@ -212,7 +195,6 @@ class AudioToTextRecorder:
 
     def __init__(self,
                  model: str = INIT_MODEL_TRANSCRIPTION,
-                 download_root: str = None, 
                  language: str = "",
                  compute_type: str = "default",
                  input_device_index: int = None,
@@ -226,7 +208,6 @@ class AudioToTextRecorder:
                  use_microphone=True,
                  spinner=True,
                  level=logging.WARNING,
-                 batch_size: int = 16,
 
                  # Realtime transcription parameters
                  enable_realtime_transcription=False,
@@ -236,7 +217,6 @@ class AudioToTextRecorder:
                  init_realtime_after_seconds=INIT_REALTIME_INITIAL_PAUSE,
                  on_realtime_transcription_update=None,
                  on_realtime_transcription_stabilized=None,
-                 realtime_batch_size: int = 16,
 
                  # Voice activation parameters
                  silero_sensitivity: float = INIT_SILERO_SENSITIVITY,
@@ -300,8 +280,6 @@ class AudioToTextRecorder:
                 'large-v2'.
                 If a specific size is provided, the model is downloaded
                 from the Hugging Face Hub.
-        - download_root (str, default=None): Specifies the root path were the Whisper models 
-          are downloaded to. When empty, the default is used. 
         - language (str, default=""): Language code for speech-to-text engine.
             If not specified, the model will attempt to detect the language
             automatically.
@@ -336,7 +314,6 @@ class AudioToTextRecorder:
         - spinner (bool, default=True): Show spinner animation with current
             state.
         - level (int, default=logging.WARNING): Logging level.
-        - batch_size (int, default=16): Batch size for the main transcription
         - enable_realtime_transcription (bool, default=False): Enables or
             disables real-time transcription of audio. When set to True, the
             audio will be transcribed continuously as it is being recorded.
@@ -368,8 +345,6 @@ class AudioToTextRecorder:
             triggered when the transcribed text stabilizes in quality. The
             stabilized text is generally more accurate but may arrive with a
             slight delay compared to the regular real-time updates.
-        - realtime_batch_size (int, default=16): Batch size for the real-time
-            transcription model.
         - silero_sensitivity (float, default=SILERO_SENSITIVITY): Sensitivity
             for the Silero Voice Activity Detection model ranging from 0
             (least sensitive) to 1 (most sensitive). Default is 0.5.
@@ -533,7 +508,6 @@ class AudioToTextRecorder:
         self.enable_realtime_transcription = enable_realtime_transcription
         self.use_main_model_for_realtime = use_main_model_for_realtime
         self.main_model_type = model
-        self.download_root = download_root
         self.realtime_model_type = realtime_model_type
         self.realtime_processing_pause = realtime_processing_pause
         self.init_realtime_after_seconds = init_realtime_after_seconds
@@ -548,8 +522,6 @@ class AudioToTextRecorder:
         self.beam_size = beam_size
         self.beam_size_realtime = beam_size_realtime
         self.allowed_latency_limit = allowed_latency_limit
-        self.batch_size = batch_size
-        self.realtime_batch_size = realtime_batch_size
 
         self.level = level
         self.audio_queue = mp.Queue()
@@ -658,8 +630,7 @@ class AudioToTextRecorder:
             args=(
                 child_transcription_pipe,
                 child_stdout_pipe,
-                self.main_model_type,
-                self.download_root,
+                model,
                 self.compute_type,
                 self.gpu_device_index,
                 self.device,
@@ -668,8 +639,7 @@ class AudioToTextRecorder:
                 self.interrupt_stop_event,
                 self.beam_size,
                 self.initial_prompt,
-                self.suppress_tokens,
-                self.batch_size
+                self.suppress_tokens
             )
         )
 
@@ -703,11 +673,8 @@ class AudioToTextRecorder:
                     model_size_or_path=self.realtime_model_type,
                     device=self.device,
                     compute_type=self.compute_type,
-                    device_index=self.gpu_device_index,
-                    download_root=self.download_root,
+                    device_index=self.gpu_device_index
                 )
-                if self.realtime_batch_size > 0:
-                    self.realtime_model_type = BatchedInferencePipeline(model=self.realtime_model_type)
 
             except Exception as e:
                 logging.exception("Error initializing faster_whisper "
@@ -2099,23 +2066,13 @@ class AudioToTextRecorder:
                                 continue
                     else:
                         # Perform transcription and assemble the text
-                        if self.realtime_batch_size > 0:
-                            segments, info = self.realtime_model_type.transcribe(
-                                audio_array,
-                                language=self.language if self.language else None,
-                                beam_size=self.beam_size_realtime,
-                                initial_prompt=self.initial_prompt,
-                                suppress_tokens=self.suppress_tokens,
-                                batch_size=self.realtime_batch_size
-                            )
-                        else:
-                            segments, info = self.realtime_model_type.transcribe(
-                                audio_array,
-                                language=self.language if self.language else None,
-                                beam_size=self.beam_size_realtime,
-                                initial_prompt=self.initial_prompt,
-                                suppress_tokens=self.suppress_tokens
-                            )
+                        segments, info = self.realtime_model_type.transcribe(
+                            audio_array,
+                            language=self.language if self.language else None,
+                            beam_size=self.beam_size_realtime,
+                            initial_prompt=self.initial_prompt,
+                            suppress_tokens=self.suppress_tokens,
+                        )
 
                         self.detected_realtime_language = info.language if info.language_probability > 0 else None
                         self.detected_realtime_language_probability = info.language_probability

{realtimestt-0.3.9 → RealtimeSTT-0.3.81}/RealtimeSTT.egg-info/PKG-INFO

@@ -1,6 +1,6 @@
 Metadata-Version: 2.1
 Name: RealtimeSTT
-Version: 0.3.9
+Version: 0.3.81
 Summary: A fast Voice Activity Detection and Transcription System
 Home-page: https://github.com/KoljaB/RealTimeSTT
 Author: Kolja Beigel
@@ -12,18 +12,6 @@ Classifier: Operating System :: OS Independent
 Requires-Python: >=3.6
 Description-Content-Type: text/markdown
 License-File: LICENSE
-Requires-Dist: PyAudio==0.2.14
-Requires-Dist: faster-whisper==1.0.3
-Requires-Dist: pvporcupine==1.9.5
-Requires-Dist: webrtcvad-wheels==2.0.14
-Requires-Dist: halo==0.0.31
-Requires-Dist: torch
-Requires-Dist: torchaudio
-Requires-Dist: scipy==1.14.1
-Requires-Dist: openwakeword>=0.4.0
-Requires-Dist: websockets==v12.0
-Requires-Dist: websocket-client==1.8.0
-Requires-Dist: numpy<2.0.0
 
 # RealtimeSTT
 [![PyPI](https://img.shields.io/pypi/v/RealtimeSTT)](https://pypi.org/project/RealtimeSTT/)
@@ -55,7 +43,7 @@ https://github.com/user-attachments/assets/797e6552-27cd-41b1-a7f3-e5cbc72094f5
 
 ### Updates
 
-Latest Version: v0.3.9
+Latest Version: v0.3.8
 
 See [release history](https://github.com/KoljaB/RealtimeSTT/releases).
 
@@ -129,26 +117,11 @@ pip install RealtimeSTT
 
 This will install all the necessary dependencies, including a **CPU support only** version of PyTorch.
 
-Although it is possible to run RealtimeSTT with a CPU installation only (use a small model like "tiny" or "base" in this case) you will get way better experience using CUDA (please scroll down).
-
-### Linux Installation
-
-Before installing RealtimeSTT please execute:
-
-```bash
-sudo apt-get install portaudio19-dev
-```
-
-### MacOS Installation
-
-Before installing RealtimeSTT please execute:
-
-```bash
-brew install portaudio
-```
+Although it is possible to run RealtimeSTT with a CPU installation only (use a small model like "tiny" or "base" in this case) you will get way better experience using:
 
 ### GPU Support with CUDA (recommended)
 
+
 ### Updating PyTorch for CUDA Support
 
 To upgrade your PyTorch installation to enable GPU support with CUDA, follow these instructions based on your specific CUDA version. This is useful if you wish to enhance the performance of RealtimeSTT with CUDA capabilities.
@@ -477,8 +450,6 @@ When you initialize the `AudioToTextRecorder` class, you have various options to
 
 - **level** (int, default=logging.WARNING): Logging level.
 
-- **batch_size** (int, default=16): Batch size for the main transcription. Set to 0 to deactivate.
-
 - **init_logging** (bool, default=True): Whether to initialize the logging framework. Set to False to manage this yourself.
 
 - **handle_buffer_overflow** (bool, default=True): If set, the system will log a warning when an input overflow occurs during recording and remove the data from the buffer.
@@ -518,8 +489,6 @@ When you initialize the `AudioToTextRecorder` class, you have various options to
 
 - **on_realtime_transcription_stabilized**: A callback function that is triggered whenever there's an update in the real-time transcription and returns a higher quality, stabilized text as its argument.
 
-- **realtime_batch_size**: (int, default=16): Batch size for the real-time transcription model. Set to 0 to deactivate.
-
 - **beam_size_realtime** (int, default=3): The beam size to use for real-time transcription beam search decoding.
 
 #### Voice Activation Parameters

{realtimestt-0.3.9 → RealtimeSTT-0.3.81}/setup.py

@@ -1,34 +1,34 @@
-import setuptools
-
-with open("README.md", "r") as fh:
-    long_description = fh.read()
-
-# Read requirements.txt
-with open('requirements.txt') as f:
-    requirements = f.read().splitlines()
-
-setuptools.setup(
-    name="RealtimeSTT",
-    version="0.3.9",
-    author="Kolja Beigel",
-    author_email="kolja.beigel@web.de",
-    description="A fast Voice Activity Detection and Transcription System",
-    long_description=long_description,
-    long_description_content_type="text/markdown",
-    url="https://github.com/KoljaB/RealTimeSTT",
-    packages=setuptools.find_packages(include=["RealtimeSTT", "RealtimeSTT_server"]),
-    classifiers=[
-        "Programming Language :: Python :: 3",
-        "License :: OSI Approved :: MIT License",
-        "Operating System :: OS Independent",
-    ],
-    python_requires='>=3.6',
-    install_requires=requirements,
-    keywords="real-time, audio, transcription, speech-to-text, voice-activity-detection, VAD, real-time-transcription, ambient-noise-detection, microphone-input, faster_whisper, speech-recognition, voice-assistants, audio-processing, buffered-transcription, pyaudio, ambient-noise-level, voice-deactivity",
-    entry_points={
-        'console_scripts': [
-            'stt-server=RealtimeSTT_server.stt_server:main',
-            'stt=RealtimeSTT_server.stt_cli_client:main',
-        ],
-    },
-)
+import setuptools
+
+with open("README.md", "r") as fh:
+    long_description = fh.read()
+
+# Read requirements.txt
+with open('requirements.txt') as f:
+    requirements = f.read().splitlines()
+
+setuptools.setup(
+    name="RealtimeSTT",
+    version="0.3.81",
+    author="Kolja Beigel",
+    author_email="kolja.beigel@web.de",
+    description="A fast Voice Activity Detection and Transcription System",
+    long_description=long_description,
+    long_description_content_type="text/markdown",
+    url="https://github.com/KoljaB/RealTimeSTT",
+    packages=setuptools.find_packages(include=["RealtimeSTT", "RealtimeSTT_server"]),
+    classifiers=[
+        "Programming Language :: Python :: 3",
+        "License :: OSI Approved :: MIT License",
+        "Operating System :: OS Independent",
+    ],
+    python_requires='>=3.6',
+    install_requires=requirements,
+    keywords="real-time, audio, transcription, speech-to-text, voice-activity-detection, VAD, real-time-transcription, ambient-noise-detection, microphone-input, faster_whisper, speech-recognition, voice-assistants, audio-processing, buffered-transcription, pyaudio, ambient-noise-level, voice-deactivity",
+    entry_points={
+        'console_scripts': [
+            'stt-server=RealtimeSTT_server.stt_server:main',
+            'stt=RealtimeSTT_server.stt_cli_client:main',
+        ],
+    },
+)