feat: apply new agent naming standards

Expanding abbreviations to remove ambiguity, simplifying agent names to reduce repetition.

ref: N25B-257
This commit is contained in:
Twirre Meulenbelt
2025-11-19 15:56:09 +01:00
parent f4dbca5b94
commit efe49c219c
39 changed files with 218 additions and 222 deletions

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@@ -0,0 +1,107 @@
import abc
import sys
if sys.platform == "darwin":
import mlx.core as mx
import mlx_whisper
from mlx_whisper.transcribe import ModelHolder
import numpy as np
import torch
import whisper
class SpeechRecognizer(abc.ABC):
def __init__(self, limit_output_length=True):
"""
:param limit_output_length: When `True`, the length of the generated speech will be limited
by the length of the input audio and some heuristics.
"""
self.limit_output_length = limit_output_length
@abc.abstractmethod
def load_model(self): ...
@abc.abstractmethod
def recognize_speech(self, audio: np.ndarray) -> str:
"""
Recognize speech from the given audio sample.
:param audio: A full utterance sample. Audio must be 16 kHz, mono, np.float32, values in the
range [-1.0, 1.0].
:return: Recognized speech.
"""
@staticmethod
def _estimate_max_tokens(audio: np.ndarray) -> int:
"""
Estimate the maximum length of a given audio sample in tokens. Assumes a maximum speaking
rate of 450 words per minute (3x average), and assumes that 3 words is 4 tokens.
:param audio: The audio sample (16 kHz) to use for length estimation.
:return: The estimated length of the transcribed audio in tokens.
"""
length_seconds = len(audio) / 16_000
length_minutes = length_seconds / 60
word_count = length_minutes * 450
token_count = word_count / 3 * 4
return int(token_count) + 10
def _get_decode_options(self, audio: np.ndarray) -> dict:
"""
:param audio: The audio sample (16 kHz) to use to determine options like max decode length.
:return: A dict that can be used to construct `whisper.DecodingOptions`.
"""
options = {}
if self.limit_output_length:
options["sample_len"] = self._estimate_max_tokens(audio)
return options
@staticmethod
def best_type():
"""Get the best type of SpeechRecognizer based on system capabilities."""
if torch.mps.is_available():
print("Choosing MLX Whisper model.")
return MLXWhisperSpeechRecognizer()
else:
print("Choosing reference Whisper model.")
return OpenAIWhisperSpeechRecognizer()
class MLXWhisperSpeechRecognizer(SpeechRecognizer):
def __init__(self, limit_output_length=True):
super().__init__(limit_output_length)
self.was_loaded = False
self.model_name = "mlx-community/whisper-small.en-mlx"
def load_model(self):
if self.was_loaded:
return
# There appears to be no dedicated mechanism to preload a model, but this `get_model` does
# store it in memory for later usage
ModelHolder.get_model(self.model_name, mx.float16)
self.was_loaded = True
def recognize_speech(self, audio: np.ndarray) -> str:
self.load_model()
return mlx_whisper.transcribe(
audio,
path_or_hf_repo=self.model_name,
**self._get_decode_options(audio),
)["text"].strip()
class OpenAIWhisperSpeechRecognizer(SpeechRecognizer):
def __init__(self, limit_output_length=True):
super().__init__(limit_output_length)
self.model = None
def load_model(self):
if self.model is not None:
return
device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
self.model = whisper.load_model("small.en", device=device)
def recognize_speech(self, audio: np.ndarray) -> str:
self.load_model()
return whisper.transcribe(self.model, audio, **self._get_decode_options(audio))["text"]