PyWhisperCpp API Reference
pywhispercpp.model
This module contains a simple Python API on-top of the C-style whisper.cpp API.
Segment
Segment(t0, t1, text)
A small class representing a transcription segment
Parameters:
-
t0
(int
) –start time
-
t1
(int
) –end time
-
text
(str
) –text
Source code in pywhispercpp/model.py
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Model
Model(
model="tiny",
models_dir=None,
params_sampling_strategy=0,
log_level=logging.INFO,
**params
)
This classes defines a Whisper.cpp model.
Example usage.
model = Model('base.en', n_threads=6)
segments = model.transcribe('file.mp3', speed_up=True)
for segment in segments:
print(segment.text)
Parameters:
-
model
(str
, default:'tiny'
) –The name of the model, one of the AVAILABLE_MODELS, (default to
tiny
), or a direct path to aggml
model. -
models_dir
(str
, default:None
) –The directory where the models are stored, or where they will be downloaded if they don't exist, default to MODELS_DIR
-
params_sampling_strategy
(int
, default:0
) –0 -> GREEDY, else BEAM_SEARCH
-
log_level
(int
, default:INFO
) –logging level, set to INFO by default
-
params
–keyword arguments for different whisper.cpp parameters, see PARAMS_SCHEMA
Source code in pywhispercpp/model.py
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transcribe
transcribe(
media,
n_processors=None,
new_segment_callback=None,
**params
)
Transcribes the media provided as input and returns list of Segment
objects.
Accepts a media_file path (audio/video) or a raw numpy array.
Parameters:
-
media
(Union[str, ndarray]
) –Media file path or a numpy array
-
n_processors
(int
, default:None
) –if not None, it will run the transcription on multiple processes binding to whisper.cpp/whisper_full_parallel > Split the input audio in chunks and process each chunk separately using whisper_full()
-
new_segment_callback
(Callable[[Segment], None]
, default:None
) –callback function that will be called when a new segment is generated
-
params
–keyword arguments for different whisper.cpp parameters, see ::: constants.PARAMS_SCHEMA
Returns:
-
List[Segment]
–List of transcription segments
Source code in pywhispercpp/model.py
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get_params
get_params()
Returns a dict
representation of the actual params
Returns:
-
dict
–params dict
Source code in pywhispercpp/model.py
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get_params_schema
staticmethod
get_params_schema()
A simple link to ::: constants.PARAMS_SCHEMA
Returns:
-
dict
–dict of params schema
Source code in pywhispercpp/model.py
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lang_max_id
staticmethod
lang_max_id()
Returns number of supported languages. Direct binding to whisper.cpp/lang_max_id
Returns:
-
int
–
Source code in pywhispercpp/model.py
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print_timings
print_timings()
Direct binding to whisper.cpp/whisper_print_timings
Returns:
-
None
–None
Source code in pywhispercpp/model.py
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system_info
staticmethod
system_info()
Direct binding to whisper.cpp/whisper_print_system_info
Returns:
-
None
–None
Source code in pywhispercpp/model.py
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available_languages
staticmethod
available_languages()
Returns a list of supported language codes
Returns:
-
list
–list of supported language codes
Source code in pywhispercpp/model.py
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pywhispercpp.constants
Constants
WHISPER_SAMPLE_RATE
module-attribute
WHISPER_SAMPLE_RATE = WHISPER_SAMPLE_RATE
MODELS_BASE_URL
module-attribute
MODELS_BASE_URL = (
"https://huggingface.co/ggerganov/whisper.cpp"
)
MODELS_PREFIX_URL
module-attribute
MODELS_PREFIX_URL = 'resolve/main/ggml'
PACKAGE_NAME
module-attribute
PACKAGE_NAME = 'pywhispercpp'
LOGGIN_LEVEL
module-attribute
LOGGIN_LEVEL = INFO
MODELS_DIR
module-attribute
MODELS_DIR = Path(user_data_dir(PACKAGE_NAME)) / 'models'
AVAILABLE_MODELS
module-attribute
AVAILABLE_MODELS = [
"base",
"base-q5_1",
"base.en",
"base.en-q5_1",
"large-v1",
"large-v2",
"large-v2-q5_0",
"large-v3",
"large-v3-q5_0",
"medium",
"medium-q5_0",
"medium.en",
"medium.en-q5_0",
"small",
"small-q5_1",
"small.en",
"small.en-q5_1",
"tiny",
"tiny-q5_1",
"tiny.en",
"tiny.en-q5_1",
"tiny.en-q8_0",
]
PARAMS_SCHEMA
module-attribute
PARAMS_SCHEMA = {
"n_threads": {
"type": int,
"description": "Number of threads to allocate for the inferencedefault to min(4, available hardware_concurrency)",
"options": None,
"default": None,
},
"n_max_text_ctx": {
"type": int,
"description": "max tokens to use from past text as prompt for the decoder",
"options": None,
"default": 16384,
},
"offset_ms": {
"type": int,
"description": "start offset in ms",
"options": None,
"default": 0,
},
"duration_ms": {
"type": int,
"description": "audio duration to process in ms",
"options": None,
"default": 0,
},
"translate": {
"type": bool,
"description": "whether to translate the audio to English",
"options": None,
"default": False,
},
"no_context": {
"type": bool,
"description": "do not use past transcription (if any) as initial prompt for the decoder",
"options": None,
"default": False,
},
"single_segment": {
"type": bool,
"description": "force single segment output (useful for streaming)",
"options": None,
"default": False,
},
"print_special": {
"type": bool,
"description": "print special tokens (e.g. <SOT>, <EOT>, <BEG>, etc.)",
"options": None,
"default": False,
},
"print_progress": {
"type": bool,
"description": "print progress information",
"options": None,
"default": True,
},
"print_realtime": {
"type": bool,
"description": "print results from within whisper.cpp (avoid it, use callback instead)",
"options": None,
"default": False,
},
"print_timestamps": {
"type": bool,
"description": "print timestamps for each text segment when printing realtime",
"options": None,
"default": True,
},
"token_timestamps": {
"type": bool,
"description": "enable token-level timestamps",
"options": None,
"default": False,
},
"thold_pt": {
"type": float,
"description": "timestamp token probability threshold (~0.01)",
"options": None,
"default": 0.01,
},
"thold_ptsum": {
"type": float,
"description": "timestamp token sum probability threshold (~0.01)",
"options": None,
"default": 0.01,
},
"max_len": {
"type": int,
"description": "max segment length in characters",
"options": None,
"default": 0,
},
"split_on_word": {
"type": bool,
"description": "split on word rather than on token (when used with max_len)",
"options": None,
"default": False,
},
"max_tokens": {
"type": int,
"description": "max tokens per segment (0 = no limit)",
"options": None,
"default": 0,
},
"speed_up": {
"type": bool,
"description": "speed-up the audio by 2x using Phase Vocoder",
"options": None,
"default": False,
},
"audio_ctx": {
"type": int,
"description": "overwrite the audio context size (0 = use default)",
"options": None,
"default": 0,
},
"initial_prompt": {
"type": str,
"description": "Initial prompt, these are prepended to any existing text context from a previous call",
"options": None,
"default": None,
},
"prompt_tokens": {
"type": Tuple,
"description": "tokens to provide to the whisper decoder as initial prompt",
"options": None,
"default": None,
},
"prompt_n_tokens": {
"type": int,
"description": "tokens to provide to the whisper decoder as initial prompt",
"options": None,
"default": 0,
},
"language": {
"type": str,
"description": 'for auto-detection, set to None, "" or "auto"',
"options": None,
"default": "en",
},
"suppress_blank": {
"type": bool,
"description": "common decoding parameters",
"options": None,
"default": True,
},
"suppress_non_speech_tokens": {
"type": bool,
"description": "common decoding parameters",
"options": None,
"default": False,
},
"temperature": {
"type": float,
"description": "initial decoding temperature",
"options": None,
"default": 0.0,
},
"max_initial_ts": {
"type": float,
"description": "max_initial_ts",
"options": None,
"default": 1.0,
},
"length_penalty": {
"type": float,
"description": "length_penalty",
"options": None,
"default": -1.0,
},
"temperature_inc": {
"type": float,
"description": "temperature_inc",
"options": None,
"default": 0.2,
},
"entropy_thold": {
"type": float,
"description": 'similar to OpenAI\'s "compression_ratio_threshold"',
"options": None,
"default": 2.4,
},
"logprob_thold": {
"type": float,
"description": "logprob_thold",
"options": None,
"default": -1.0,
},
"no_speech_thold": {
"type": float,
"description": "no_speech_thold",
"options": None,
"default": 0.6,
},
"greedy": {
"type": dict,
"description": "greedy",
"options": None,
"default": {"best_of": -1},
},
"beam_search": {
"type": dict,
"description": "beam_search",
"options": None,
"default": {"beam_size": -1, "patience": -1.0},
},
}
pywhispercpp.utils
Helper functions
download_model
download_model(
model_name, download_dir=None, chunk_size=1024
)
Helper function to download the ggml
models
Parameters:
-
model_name
(str
) –name of the model, one of ::: constants.AVAILABLE_MODELS
-
download_dir
–Where to store the models
-
chunk_size
–size of the download chunk
Returns:
-
str
–Absolute path of the downloaded model
Source code in pywhispercpp/utils.py
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to_timestamp
to_timestamp(t, separator=',')
376 -> 00:00:03,760 1344 -> 00:00:13,440
Implementation from whisper.cpp/examples/main
Parameters:
-
t
(int
) –input time from whisper timestamps
-
separator
–seprator between seconds and milliseconds
Returns:
-
str
–time representation in hh: mm: ss[separator]ms
Source code in pywhispercpp/utils.py
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output_txt
output_txt(segments, output_file_path)
Creates a raw text from a list of segments
Implementation from whisper.cpp/examples/main
Parameters:
-
segments
(list
) –list of segments
Returns:
-
str
–path of the file
Source code in pywhispercpp/utils.py
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output_vtt
output_vtt(segments, output_file_path)
Creates a vtt file from a list of segments
Implementation from whisper.cpp/examples/main
Parameters:
-
segments
(list
) –list of segments
Returns:
-
str
–Absolute path of the file
Source code in pywhispercpp/utils.py
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output_srt
output_srt(segments, output_file_path)
Creates a srt file from a list of segments
Parameters:
-
segments
(list
) –list of segments
Returns:
-
str
–Absolute path of the file
Source code in pywhispercpp/utils.py
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output_csv
output_csv(segments, output_file_path)
Creates a srt file from a list of segments
Parameters:
-
segments
(list
) –list of segments
Returns:
-
str
–Absolute path of the file
Source code in pywhispercpp/utils.py
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pywhispercpp.examples
assistant
A simple example showcasing the use of pywhispercpp
as an assistant.
The idea is to use a VAD
to detect speech (in this example we used webrtcvad), and when speech is detected
we run the inference.
Assistant
Assistant(
model="tiny",
input_device=None,
silence_threshold=8,
q_threshold=16,
block_duration=30,
commands_callback=None,
model_log_level=logging.INFO,
**model_params
)
Assistant class
Example usage
from pywhispercpp.examples.assistant import Assistant
my_assistant = Assistant(commands_callback=print, n_threads=8)
my_assistant.start()
Parameters:
-
model
–whisper.cpp model name or a direct path to a
ggml
model -
input_device
(int
, default:None
) –The input device (aka microphone), keep it None to take the default
-
silence_threshold
(int
, default:8
) –The duration of silence after which the inference will be running
-
q_threshold
(int
, default:16
) –The inference won't be running until the data queue is having at least
q_threshold
elements -
block_duration
(int
, default:30
) –minimum time audio updates in ms
-
commands_callback
(Callable[[str], None]
, default:None
) –The callback to run when a command is received
-
model_log_level
(int
, default:INFO
) –Logging level
-
model_params
–any other parameter to pass to the whsiper.cpp model see ::: pywhispercpp.constants.PARAMS_SCHEMA
Source code in pywhispercpp/examples/assistant.py
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start
start()
Use this function to start the assistant
Returns:
-
None
–None
Source code in pywhispercpp/examples/assistant.py
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livestream
Quick and dirty realtime livestream transcription.
Not fully satisfying though :) You are welcome to make it better.
LiveStream
LiveStream(
url,
model="tiny.en",
block_size=1024,
buffer_size=20,
sample_size=4,
output_device=None,
model_log_level=logging.CRITICAL,
**model_params
)
LiveStream class
Note
It heavily depends on the machine power, the processor will jump quickly to 100% with the wrong parameters.
Example usage
from pywhispercpp.examples.livestream import LiveStream
url = "" # Make sure it is a direct stream URL
ls = LiveStream(url=url, n_threads=4)
ls.start()
Parameters:
-
url
– -
model
–whisper.cpp model
-
block_size
(int
, default:1024
) –block size, default to 1024
-
buffer_size
(int
, default:20
) –number of blocks used for buffering, default to 20
-
sample_size
(int
, default:4
) –sample size
-
output_device
(int
, default:None
) –the output device, aka the speaker, leave it None to take the default
-
model_log_level
–logging level
-
model_params
–any other whisper.cpp params
Source code in pywhispercpp/examples/livestream.py
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main
A simple Command Line Interface to test the package
recording
A simple example showcasing how to use pywhispercpp to transcribe a recording.
Recording
Recording(duration, model='tiny.en', **model_params)
Recording class
Example usage
from pywhispercpp.examples.recording import Recording
myrec = Recording(5)
myrec.start()
Source code in pywhispercpp/examples/recording.py
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