new structure for transcription engine

- found problems with mps
- integrated jinja2 templates for rendering (.tex?)
- raw ui & bad structure
This commit is contained in:
2026-02-24 02:58:28 +03:00
parent 9e67b36842
commit 3d48f473b0
24 changed files with 584 additions and 403 deletions
+17
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@@ -0,0 +1,17 @@
from dataclasses import dataclass
import torchaudio
import torch
class Audio:
waveform: torch.Tensor
sr: int
def load(self, filepath):
"""
Loads audio from file's path
"""
self.waveform, self.sr = torchaudio.load(
filepath,
backend="ffmpeg",
)
return self
+36 -232
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@@ -1,246 +1,50 @@
import logging
import math
import time
import sys
import gc
from transcription.audio import Audio
from transcription.preprocessing.audio_preprocessor import AudioPreprocessor
from transcription.preprocessing.splitter import Splitter
from transcription.engines.whisper import WhisperEngine
from transcription.configuration import Configuration
import torch
import torchaudio
from tqdm import tqdm
from transformers import WhisperForConditionalGeneration, WhisperProcessor
from transcription.device_configuration import DeviceConfiguration
from ui.ui_log_handler import UILogHandler
# TODO: implement transcription with shift
# maybe inherit from AudioTranscription and rename to something like WhisperTranscription?
class AudioTranscription:
"""
Class for automatical audio transcription using Whisper.
Provides audio file loading, resampling, conversion to mono,
splitting into chunks and batches, running the model, and compiling the final text.
Attributes:
model_name (str): Whisper model name in HuggingFace.
filepath (str): Input filepath.
waveform (torch.Tensor): Audiosignal in tensor form.
sampling_rate (int): Input file's sampling frequency.
chunks (list): Audio's chunks list.
batches (list): Batches combined from chunks.
chunk_size (int): Chunk size in samples.
custom_chunk_length (int): Custom chunk length (in seconds).
custom_batch_length (int): Custom batch length (in chunks).
device (str): Inference device ("cuda", "cpu", "mps").
processor (WhisperProcessor | None): Tokenizator/preprocessor.
model (WhisperForConditionalGeneration | None): Whisper model.
logger (logging.Logger): Logger.
torch_dtype (torch.dtype): Data type for calculations (fp16/fp32).
language (str): Transcription language (default "ru").
all_transcription (list): List of strings with transcription.
Args:
filepath (str): Input filepath.
device_configuration (DeviceConfiguration): Device configuration
(GPU/CPU/MPS, model, chunk length, batch etc.).
logger (logging.Logger): Logger.
language (str, optional): Transcription language. Default "ru".
Example:
>>> from transcription.device_configuration import DeviceConfiguration
>>> import logging
>>> config = DeviceConfiguration(device="cuda", model_name="openai/whisper-large-v3-turbo") # recheck this, not true i think
>>> logger = logging.getLogger("transcription")
>>> transcriber = AudioTranscription("audio.wav", config, logger, language="en")
>>> text = transcriber.transcribe_audio()
>>> print(text)
"This is a test audio transcription."
Methods:
transcribe_audio() -> str:
Starts full transcription pipeline: model loading,
file loading and preprocessing, splitting into chunks/batches,
inference and model unloading. Returns the final transcription.
"""
model_name = "openai/whisper-large-v3-turbo"
filepath: str
waveform: torch.Tensor
sampling_rate: int
file_format: str
chunks: list = []
batches: list = []
chunk_size: int
custom_chunk_length: int
custom_batch_length: int
device = "cuda"
processor: WhisperProcessor | None
model: WhisperForConditionalGeneration | None
logger: logging.Logger
torch_dtype: torch.dtype
language = "ru"
all_transcription: list = []
# add multimodel ability
def __init__(
self,
filepath: str,
device_configuration: DeviceConfiguration,
logger: logging.Logger,
language: str = "ru",
config: Configuration,
language,
# logger
) -> None:
# TODO: add pretty docs here
self.filepath = filepath
self.language = language
self.logger = logger
# self.logger = logger
# setting file extension
self.file_format = filepath.split(".")[-1]
# extracting configuration
self.device = device_configuration.device
self.model_name = device_configuration.model_name
self.custom_chunk_length = device_configuration.chunk_length_s
self.custom_batch_length = device_configuration.batch_size
self.torch_dtype = device_configuration.torch_dtype
self.chunks: list = []
self.batches: list = []
self.all_transcription: list = []
def _load_model(self) -> None:
self.logger.info("Loading model WhisperProcessor...")
try:
start_time = time.time()
self.processor = WhisperProcessor.from_pretrained(self.model_name)
self.model = WhisperForConditionalGeneration.from_pretrained(
self.model_name, torch_dtype=self.torch_dtype
).to(self.device)
end_time = time.time()
self.logger.info(
f"Model loaded successfully in {end_time - start_time:.2f} seconds."
)
except Exception as e:
self.logger.error(f"Error while loading model: {e}")
def _unload_model(self) -> None:
self.logger.info("Unloading model...")
self.model = None
self.processor = None
if self.device == "cuda":
torch.cuda.empty_cache()
# TODO: maybe do something here for MPS
self.logger.info("Model unloaded successfully.")
def _load_file(self) -> None:
self.logger.info(f"Loading file {self.filepath}")
try:
# self.waveform, self.sampling_rate = torchaudio.load(
# self.filepath, format=self.file_format, backend="ffmpeg"
# )
self.waveform, self.sampling_rate = torchaudio.load(self.filepath)
self.logger.info(f"Successfully loaded file {self.filepath}.")
except Exception as e:
self.logger.error(f"Unable to load file {self.filepath}: {e}")
# raise RuntimeError(f"Failed to load file {self.filepath}: {e}")
def _resample(self) -> None:
self.waveform = torchaudio.functional.resample(
self.waveform, self.sampling_rate, 16000
self.audio = Audio()
self.preprocessor = AudioPreprocessor()
self.splitter = Splitter(
chunkSize=config.chunkSize,
batchSize=config.batchSize,
)
def _to_mono(self):
if self.waveform.shape[0] > 1:
self.waveform = self.waveform.mean(dim=0, keepdim=True)
self.waveform = self.waveform.squeeze(0)
def _split_to_chunks(self, shift: bool = False) -> None:
self.logger.info(f"Splitting audio on chunks...")
self.chunk_size = self.custom_chunk_length * 16000 # 16kHz after resampling
total_samples = self.waveform.shape[0]
chunks_count = (total_samples + self.chunk_size - 1) // self.chunk_size
self.logger.info(
f"File length - {total_samples / 16000:.1f} seconds, splitting on {chunks_count} chunks by {self.custom_chunk_length} seconds per chunk."
self.engine = WhisperEngine(
modelName=config.modelName,
language=self.language,
dType=config.dType,
device=config.device,
)
# maybe add something like temperature here?
def transcribeAudio(self) -> str:
transcription: list = []
self.engine.loadModel()
self.preprocessor.prepare(self.audio.load(self.filepath))
self.chunks = []
for idx in tqdm(range(chunks_count)):
start = idx * self.chunk_size
end = min((idx + 1) * self.chunk_size, total_samples)
chunk = self.waveform[start:end].cpu().numpy().astype("float32")
self.chunks.append(chunk)
batches = self.splitter.split(self.audio.waveform)
def _resplit_chunks_to_batches(self) -> None:
self.logger.info(f"Splitting chunks into batches...")
self.batches = []
for i in range(0, len(self.chunks), self.custom_batch_length):
batch = self.chunks[i : i + self.custom_batch_length]
self.batches.append(batch)
self.logger.info(f"Total: {len(self.batches)} batches, weight = {sys.getsizeof(self.batches)}")
for batch in batches:
batchText: str = self.engine.transcribeBatch(batch)
transcription.append(batchText)
def _process_all_batches(self) -> None:
start_time = time.time()
try:
assert self.processor is not None
assert self.model is not None
self.all_transcription = []
for idx in tqdm(range(len(self.batches))):
inputs = self.processor(
self.batches[idx],
sampling_rate=16000,
return_tensors="pt",
padding=True,
)
input_features = inputs.input_features.to(self.device).to(
self.torch_dtype
)
with torch.no_grad():
predicted_ids = self.model.generate(
input_features,
language=self.language,
task="transcribe",
temperature=0.0,
)
texts = self.processor.batch_decode(
predicted_ids, skip_special_tokens=True
)
self.all_transcription.extend(texts)
inputs = None
input_features = None
predicted_ids = None
gc.collect()
if self.device.startswith("cuda"):
torch.cuda.empty_cache()
torch.cuda.ipc_collect()
end_time = time.time()
self.logger.info(
f"Transcription completed in {end_time - start_time:.2f} seconds"
)
except Exception as e:
self.logger.error(f"Errors occured while processing chunks: {e}")
def transcribe_audio(self) -> str:
self._load_model()
self._load_file()
self._resample()
self._to_mono()
self._split_to_chunks()
self._resplit_chunks_to_batches()
self._process_all_batches()
self._unload_model()
return " ".join(self.all_transcription)
self.engine.unloadModel()
return str(" ".join(transcription))
+23
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@@ -0,0 +1,23 @@
from dataclasses import dataclass
import torch
@dataclass
class Configuration:
# add new models
device: str = "cuda"
modelName: str = "openai/whisper-large-v2"
chunkSize: int = 30
batchSize: int = 16
dataType: str = "torch.float16"
_dtype_map = {
"torch.float16": torch.float16,
"torch.float32": torch.float32,
"torch.bfloat16": torch.bfloat16,
}
dType: torch.dtype = None
def __post_init__(self):
self.dType = self._dtype_map[self.dataType]
-47
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@@ -1,47 +0,0 @@
from dataclasses import dataclass
import torch
@dataclass
class DeviceConfiguration:
"""
Configurations for Whisper model on different devices.
Attributes:
device (str): Type of device. Possible options: "cuda", "cpu", "mps".
model_name (str): Whisper models. Possible options:
- "openai/whisper-tiny"
- "openai/whisper-small"
- "openai/whisper-medium"
- "openai/whisper-large"
- "openai/whisper-large-v2"
- "openai/whisper-large-v3-turbo"
batch_size (int): Chunks in one batch. Selected for VRAM.
chunk_length_s (int): Length of one audio chunk in seconds. Smaller -> less VRAM.
data_type (str): custom data type of model. Variants:
- torch.float16 - for GPUs
- torch.float32 - for CPU
- torch.bfloat16 - for GPUs which has BF16 support (usually RTX 40XX+)
"""
device: str = "cuda"
model_name: str = "openai/whisper-large-v2"
batch_size: int = 16
chunk_length_s: int = 30
data_type: str = "torch.float16"
_dtype_map = {
"torch.float16": torch.float16,
"torch.float32": torch.float32,
"torch.bfloat16": torch.bfloat16,
}
torch_dtype: torch.dtype = None
def __post_init__(self):
self.torch_dtype = self._dtype_map[self.data_type]
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+30
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@@ -0,0 +1,30 @@
import torch
from abc import ABC, abstractmethod
class BaseEngine(ABC):
def __init__(
self,
modelName: str,
language: str,
dType: torch.dtype,
device: str
):
self.modelName = modelName
self.device = device
self.language = language
self.dType = dType
@abstractmethod
def loadModel(self) -> None:
pass
@abstractmethod
def unloadModel(self) -> None:
pass
@abstractmethod
def transcribeBatch(
self,
batch
) -> str:
pass
+65
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@@ -0,0 +1,65 @@
# from logging import Logger
import time
import torch
import gc
from transformers import WhisperForConditionalGeneration, WhisperProcessor
from transcription.engines.base_engine import BaseEngine
class WhisperEngine(BaseEngine):
def loadModel(self) -> None:
self.processor = WhisperProcessor.from_pretrained(self.modelName)
self.model = WhisperForConditionalGeneration.from_pretrained(
self.modelName,
torch_dtype = self.dType # check twice
).to(self.device) # ??? recheck
def unloadModel(self) -> None:
self.model = None
self.processor = None
# TODO: MPS?
if self.device == "cuda":
torch.cuda.empty_cache()
def transcribeBatch(
self,
batch,
) -> str:
assert self.processor is not None
assert self.model is not None
inputs = self.processor(
batch,
sampling_rate=16000,
return_tensors="pt",
padding=True,
)
input_features = inputs.input_features.to(self.device).to(self.dType)
with torch.no_grad():
predicted_ids = self.model.generate(
input_features,
language=self.language,
task="transcribe",
temperature=0.0,
)
batchText = self.processor.batch_decode(
predicted_ids,
skip_special_tokens=True,
)
inputs = None
input_features = None
predicted_ids = None
gc.collect()
# maybe do here something with MPS?
if self.device.startswith("cuda"):
torch.cuda.empty_cache()
torch.cuda.ipc_collect()
return " ".join(batchText)
@@ -0,0 +1,35 @@
from transcription.audio import Audio
import torchaudio
class AudioPreprocessor:
TARGET_SAMPLING_RATE: int = 16000
# for different models in future
# def __init__(self, model):
# pass
def _resample(
self,
audio: Audio
) -> None:
if audio.sr != self.TARGET_SAMPLING_RATE:
audio.waveform = torchaudio.functional.resample(
audio.waveform,
audio.sr,
self.TARGET_SAMPLING_RATE
)
def _to_mono(
self,
audio: Audio
) -> None:
if audio.waveform.shape[0] > 1:
audio.waveform = audio.waveform.mean(dim=0, keepdim=True)
audio.waveform = audio.waveform.squeeze(0)
def prepare(
self,
audio: Audio
):
self._resample(audio)
self._to_mono(audio)
+50
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@@ -0,0 +1,50 @@
import torch
from typing import List
class Splitter:
def __init__(
self,
chunkSize: int,
batchSize: int,
) -> None:
self.chunkSize = chunkSize * 16000 # 16 kHz after resampling
self.batchSize = batchSize
# maybe raise some exceptions here?
def _split_to_chunks(
self,
waveform: torch.Tensor,
) -> List:
totalSamples = waveform.shape[0]
chunksCount = (totalSamples + self.chunkSize - 1) // self.chunkSize
chunks: List = []
# tqdm or something here?
for chunkNum in range(chunksCount):
start = chunkNum * self.chunkSize
end = min((chunkNum + 1) * self.chunkSize, totalSamples)
chunk = waveform[start : end].cpu().numpy().astype("float32")
chunks.append(chunk)
return chunks
def _split_to_batches(
self,
chunks: List,
) -> List:
batches: List = []
for i in range(0, len(chunks), self.batchSize):
batch = chunks[i : i + self.batchSize]
batches.append(batch)
return batches
def split(
self,
waveform: torch.Tensor
) -> List:
chunks = self._split_to_chunks(waveform)
batches = self._split_to_batches(chunks)
return batches
+1 -3
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@@ -1,10 +1,8 @@
import logging
from dataclasses import dataclass
import torch
def check_torch(logger: logging.Logger) -> None:
def checkTorch(logger: logging.Logger) -> None:
logger.info("=== Checking PyTorch ===")
logger.info(f"Torch version: {torch.__version__}")
+1 -1
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@@ -1,4 +1,4 @@
from transcription.device_configuration import DeviceConfiguration
from transcription.configuration import Configuration
# TODO: implement saving & removing configuration
+36
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@@ -0,0 +1,36 @@
import tkinter as tk
import torch
class State:
def __init__(self, root):
# transcription
self.model = tk.StringVar(root, "openai/whisper-large-v3-turbo")
self.batch = tk.StringVar(root, "32")
self.chunk = tk.StringVar(root, "30")
self.dtype = tk.StringVar(root, "torch.float16")
self.language = tk.StringVar(root, "ru")
# llm
self.api_key = tk.StringVar(root, "")
self.api_model = tk.StringVar(root, "")
self.base_url = tk.StringVar(root, "")
self.conspect_lang = tk.StringVar(root, "Russian")
# flags
self.create_conspect = tk.BooleanVar(root, False)
self.remove_transcription = tk.BooleanVar(root, False)
# files
self.input_file = tk.StringVar(root)
self.output_file = tk.StringVar(root)
# device
devices = []
if torch.cuda.is_available():
devices.append("cuda")
if torch.backends.mps.is_available():
devices.append("mps")
devices.append("cpu")
self.device_opts = devices
self.device = tk.StringVar(root, devices[0])
+1
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@@ -0,0 +1 @@
# new structure coming soon
+1
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@@ -0,0 +1 @@
# new structure coming soon
+1
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@@ -0,0 +1 @@
# new structure coming soon
+49 -35
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@@ -11,8 +11,8 @@ import customtkinter as ctk
import torch
from transcription.audio_transcription import AudioTranscription
from transcription.device_configuration import DeviceConfiguration
from transcription.torch_checker import check_torch
from transcription.configuration import Configuration
from transcription.torch_checker import checkTorch
from ui.tooltip import ToolTip
from ui.ui_log_handler import setup_ui_logger
from utils.requests_to_api import LLMrequest
@@ -20,7 +20,6 @@ from utils.requests_to_api import LLMrequest
WINDOW_WIDTH = 1000
WINDOW_HEIGHT = 725
class TranscriberApp(ctk.CTk):
def __init__(self):
super().__init__()
@@ -54,6 +53,7 @@ class TranscriberApp(ctk.CTk):
# checkboxes
self.create_conspect = tk.BooleanVar(value=False)
self.remove_transcription = tk.BooleanVar(value=False)
self.latex_compiler = tk.StringVar(value="")
# settings device options
device_opts = []
@@ -187,7 +187,7 @@ class TranscriberApp(ctk.CTk):
row=0,
col=0,
text="Model:",
tooltip="Choose model for speed recognition",
tooltip="Choose model for speech recognition",
variable=self.model_var,
values=[
"openai/whisper-large-v3-turbo",
@@ -296,6 +296,20 @@ class TranscriberApp(ctk.CTk):
variable=self.conspect_transcription_lang_var,
values=None,
)
add_setting(
parent=grid,
row=4,
col=0,
text="LaTeX compiler:",
tooltip="Choose LaTeX compiler",
variable=self.latex_compiler,
values=[
"latexmk",
"lualatex",
"xelatex",
"bibtex"
]
)
### Custom Prompt setting
customPromptFrame = ctk.CTkFrame(grid)
@@ -350,13 +364,13 @@ class TranscriberApp(ctk.CTk):
input_name = os.path.basename(self.input_file_var.get())
name, _ = os.path.splitext(input_name)
# TODO: redo output_name logic maybe?
output_name = f"{"".join(name.split(".").pop())}.txt"
output_name = f"{"".join(name.split(".")[:-1:])}.txt"
path = os.path.join(directory, output_name)
self.output_file_var.set(path)
def _check_torch(self):
check_torch(self.ui_logger)
checkTorch(self.ui_logger)
self.ui_logger.info(f"==== Transcription ====")
self.ui_logger.info(f"Transcription model: {self.model_var.get()}")
@@ -405,21 +419,21 @@ class TranscriberApp(ctk.CTk):
def _transcribe_worker(self, infile: str):
try:
config = DeviceConfiguration(
config = Configuration(
device=self.device_var.get(),
model_name=self.model_var.get(),
batch_size=int(self.batch_var.get()),
chunk_length_s=int(self.chunk_var.get()),
modelName=self.model_var.get(),
batchSize=int(self.batch_var.get()),
chunkSize=int(self.chunk_var.get()),
data_type=self.dtype_var.get(),
)
Audio = AudioTranscription(
filepath=infile,
device_configuration=config,
logger=self.ui_logger,
config=config,
language=self.transcription_lang_var.get(),
)
transcription = Audio.transcribe_audio()
transcription = Audio.transcribeAudio()
# remove from here
outfile = self.output_file_var.get().strip()
if not outfile:
outfile = infile
@@ -431,32 +445,32 @@ class TranscriberApp(ctk.CTk):
f.write(transcription)
self.ui_logger.info(f"Transcription saved to {outfile}.")
if self.create_conspect.get():
# TODO: add custom prompt ability here
# TODO: add logging here
# TODO: add progressbar instead of tqdm
self.ui_logger.info(f"Starting creating conspect via {self.api_model_var.get()}...")
with open("utils/default_prompt.txt", "r", encoding="utf-8") as f:
default_prompt = "\n".join(f.readlines())
# if self.create_conspect.get():
# # TODO: add custom prompt ability here
# # TODO: add logging here
# # TODO: add progressbar instead of tqdm
# self.ui_logger.info(f"Starting creating conspect via {self.api_model_var.get()}...")
# with open("utils/prompts/default_prompt.txt", "r", encoding="utf-8") as f:
# default_prompt = "\n".join(f.readlines())
# if self.custom_prompt_textbox.get(): # <-- issue here
# prompt = transcription + "\n" + self.custom_prompt_textbox.get()
# else:
# prompt = transcription + "\n" + default_prompt
# # if self.custom_prompt_textbox.get(): # <-- issue here
# # prompt = transcription + "\n" + self.custom_prompt_textbox.get()
# # else:
# # prompt = transcription + "\n" + default_prompt
prompt = transcription + "\n" + default_prompt
# prompt = transcription + "\n" + default_prompt
request = LLMrequest(
api_key=self.api_key_var.get(),
model_name=self.api_model_var.get(),
base_url=self.base_url_var.get(),
)
response = request.get_response(prompt=prompt)
outfile += ".md"
with open(outfile, "w", encoding="utf-8") as f:
f.write(response)
# request = LLMrequest(
# api_key=self.api_key_var.get(),
# model_name=self.api_model_var.get(),
# base_url=self.base_url_var.get(),
# )
# response = request.get_response(prompt=prompt)
# outfile += ".md"
# with open(outfile, "w", encoding="utf-8") as f:
# f.write(response)
self.ui_logger.info(f"Conspect saved to {outfile}.")
# self.ui_logger.info(f"Conspect saved to {outfile}.")
except Exception as e:
self.ui_logger.error(f"Error: {e}")
+13
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@@ -24,3 +24,16 @@ def setup_ui_logger(text_widget: tk.Text, level=logging.INFO):
logger.addHandler(handler)
return logger
# def setup_ui_logger(
# text_widget: tk.Text,
# base_logger: logging.Logger,
# level=logging.INFO,
# ):
# handler = UILogHandler(text_widget)
# handler.setLevel(level)
# handler.setFormatter(
# logging.Formatter("%(asctime)s - %(levelname)s - %(message)s")
# )
# base_logger.addHandler(handler)
-84
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@@ -1,84 +0,0 @@
Используй **исключительно текст из транскрибации** как первоисточник.
Разрешается расширять тему и объяснять глубже, но **дополнительная информация должна быть строго отделена и явно помечена**.
### Цель
Создать структурированную, полный образовательный конспект лекции **формата Obsidian Markdown**, который:
* воспроизводит информацию из транскрибата максимально точно
* дополняет и разъясняет сложные моменты
* превращает речь в ясный учебный материал
* сохраняет смысл, порядок, и логическую структуру лектора
* улучшает академичность и связность
### Правила работы с транскриптом
* Используй только факты из транскрибации как основу.
* **Не выдумывай факты**, которых не было.
* **Дополнительные пояснения допустимы**, но строго в спец-блоках (см. ниже).
* Цитаты из оригинального текста — **дослівно**, максимум **25 слов подряд**.
* Нельзя писать «лектор сказал», «в тексте было» — пишем как учебный материал.
### Блоки для расширений
Если нужно объяснить термин, дополнить концепцию, исправить недосказанность — используй:
```
> AI clarification:
> Ваше дополнение, объяснение, аналогия или расширение темы.
```
Для коротких определений:
```
**AI note:** краткое пояснение
```
### Обязательная структура результата
```
# Название темы
## Ключевые идеи
- маркеры
- …
## Основные понятия и определения
- **Термин** — определение
- …
## Подробный конспект
(структура, логика, абзацы, тезисы)
> Цитаты (≤25 слов)
## Примеры и объяснения
- собственные пересказанные примеры из транскрибации
- при необходимости:
> AI clarification
## Выводы и ключевые инсайты
- …
## Вопросы для самопроверки
- …
```
### Обязательный формат Obsidian Markdown
Используй следующие возможности Obsidian:
| Элемент | Формат |
| ----------------- | ----------------------------------------- |
| Заголовки | `#`, `##`, `###` |
| Цитаты | `>` |
| Списки | `-` и `1.` |
| Врезка/внимание | `> [!note]` `> [!tip]` |
| Код/формулы | `\`…`` и блоки ``` |
| Выделение | `**жирный**`, `_курсив_`, `==выделение==` |
| Внутренние ссылки | `[[Термин]]` если термин важный |
| Таблицы | стандартные markdown таблицы |
| Чек-лист | `- [ ]` |
| Формулы | Для однострочных формул используй $ formula $. Для мультистрочных формул используй $$ formula $$ |
**Не добавляй** приветствий, заключений, обращений, пояснений формата. Только готовый md-текст.
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# not sure, but why not?
class AudioPreparationError(Exception):
"""Base exception for audio preparation"""
pass
class ResamplingError(AudioPreparationError):
"""Raises when resampling fails"""
pass
class DownmixingError(AudioPreparationError):
"""Raises when downmixing to mono fails"""
pass
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Create a complete, structured **Obsidian Markdown** educational summary based on the transcript.
The final document must be written in **{{ language }}**.
Use the transcript strictly as the factual basis:
- Do NOT invent facts.
- You may add clarifications and explanations, but they MUST be placed in special blocks.
- Preserve the logical order and meaning of the original speaker.
- Transform speech into clear academic exposition.
- Quotes must be accurate and **<= 25 words**.
- Do not reference the transcript or the speaker. Write as a stand-alone educational text.
You must follow the template below exactly.
---
# {{ title }}
## Key Ideas
- bullet points summarizing the main concepts
## Core Concepts and Definitions
- **Term** — definition
- **Another Term** — definition
## Detailed Summary
Write a logically structured, academically clear exposition.
Break it into paragraphs, subsections, lists, and quotes.
To include quotes from the transcript, use:
> "{{ quote }}"
(Ensure that every quote is ≤ 25 words.)
## Examples and Explanations
Use examples that appear in the transcript.
Additional clarifications must use the following block:
> AI clarification:
> Your extended explanation, analogy, or contextual detail.
Short notes should use:
**AI note:** short clarification
## Conclusions and Key Insights
- summary points
- what the reader should remember
## Self-Assessment Questions
- question 1
- question 2
- question 3
---
### Transcript (for your reference only):
{{ transcription }}
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Create a clean, structured **LaTeX educational summary** based strictly on the transcript.
The final document must be written in **{{ language }}**.
Guidelines:
- Use ONLY facts from the transcript as the base.
- You MAY add explanations, but they MUST appear in designated blocks.
- Do NOT reference the transcript or the speaker.
- Maintain the speaker`s logical order.
- Produce academically clear, structured LaTeX text.
- Quotes must be accurate and <= 25 words.
Use the structure below.
Assume that stylistic environments are already defined in the `.sty` file.
---
\section*{ {{ title }} }
\subsection*{Key Ideas}
\begin{itemize}
\item main idea
\item …
\end{itemize}
\subsection*{Core Concepts and Definitions}
\begin{description}
\item[\textbf{Term}] definition
\item[\textbf{Another Term}] definition
\end{description}
\subsection*{Detailed Summary}
Write a clear, logically structured academic explanation.
Break content into paragraphs, subsections, lists.
Quotes from the transcript must be included as:
\begin{quote}
"{{ quote }}"
\end{quote}
(Ensure each quote is ≤ 25 words.)
\subsection*{Examples and Explanations}
Examples based on the transcript.
Additional clarifications must use the following custom environment (assumed to exist in the .sty):
\begin{aiclarification}
Your additional explanation, analogy, or contextual expansion.
\end{aiclarification}
Short notes should use:
\ainote{short clarification}
\subsection*{Conclusions and Key Insights}
\begin{itemize}
\item conclusion
\item conclusion
\end{itemize}
\subsection*{Self-Assessment Questions}
\begin{enumerate}
\item question
\item question
\item question
\end{enumerate}
---
% Transcript included only for reference (model must not copy or analyze or mention this directly)
% ======================================================
% {{ transcription | replace("%", "\\%") }}
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from __future__ import annotations
import os
from jinja2 import Environment, FileSystemLoader, StrictUndefined
class PromptManager:
"""
Self-written class for Jinja2 templates especially for summarization.
Supports:
- md
- latex
Usage examples:
```
from prompts.prompt_manager import PromptManager
pm = PromptManager("notecast/prompts")
markdown_text = pm.render(
"markdown_default_prompt.md.j2",
language="Russian",
title="Calculus - Lecture 1",
transcription=raw_transcription_text
)
latex_text = pm.render(
"latex_default_prompt.j2",
language="Russian",
title="Calculus - Lecture 1",
transcription=raw_transcription_text
)
```
"""
def __init__(self, templates_dir: str) -> None:
if not os.path.isdir(templates_dir):
raise NotADirectoryError(f"Templates directory not found: {templates_dir}")
self.templates_dir = templates_dir
self.env = Environment(
loader=FileSystemLoader(self.templates_dir),
autoescape=False,
trim_blocks=True,
lstrip_blocks=True,
undefined=StrictUndefined
)
def list_templates(self) -> list[str]:
templates = []
for file in os.listdir(self.templates_dir):
if file.endswith(".j2"):
templates.append(file)
return templates
def load(self, template_name: str):
try:
return self.env.get_template(template_name)
except Exception as e:
raise RuntimeError(f"Failed to load template '{template_name}': {e}")
def render(
self,
template_name: str,
*,
language: str,
title: str,
transcription: str
) -> str:
template = self.load(template_name)
return template.render(
language=language,
title=title,
transcription=transcription
)
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import openai
# maybe make it asynchronous?
class LLMrequest:
def __init__(self, api_key: str, model_name: str, base_url: str = None):
if base_url: