new structure for transcription engine

- found problems with mps
- integrated jinja2 templates for rendering (.tex?)
- raw ui & bad structure

transcription/audio.py

@@ -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

transcription/audio_transcription.py

@@ -1,246 +1,50 @@
-import logging
+from transcription.audio import Audio
-import math
+from transcription.preprocessing.audio_preprocessor import AudioPreprocessor
-import time
+from transcription.preprocessing.splitter import Splitter
-import sys
+from transcription.engines.whisper import WhisperEngine
-import gc
+from transcription.configuration import Configuration
 
-import torch
+# maybe inherit from AudioTranscription and rename to something like WhisperTranscription?
-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
 class AudioTranscription:
-    """
+    # add multimodel ability
-    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 = []
-
     def __init__(
         self,
         filepath: str,
-        device_configuration: DeviceConfiguration,
+        config: Configuration,
-        logger: logging.Logger,
+        language,
-        language: str = "ru",
+        # logger
     ) -> None:
-        # TODO: add pretty docs here
         self.filepath = filepath
         self.language = language
-        self.logger = logger
+        # self.logger = logger
 
-        # setting file extension
+        self.audio = Audio()
-        self.file_format = filepath.split(".")[-1]
+        self.preprocessor = AudioPreprocessor()
-
+        self.splitter = Splitter(
-        # extracting configuration
+            chunkSize=config.chunkSize,
-        self.device = device_configuration.device
+            batchSize=config.batchSize,
-        self.model_name = device_configuration.model_name
+        )
-        self.custom_chunk_length = device_configuration.chunk_length_s
+        self.engine = WhisperEngine(
-        self.custom_batch_length = device_configuration.batch_size
+            modelName=config.modelName,
-        self.torch_dtype = device_configuration.torch_dtype
+            language=self.language,
-
+            dType=config.dType,
-        self.chunks: list = []
+            device=config.device,
-        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
         )
     
-    def _to_mono(self):
+    # maybe add something like temperature here?
-        if self.waveform.shape[0] > 1:
+    def transcribeAudio(self) -> str:
-            self.waveform = self.waveform.mean(dim=0, keepdim=True)
+        transcription: list = []
-        self.waveform = self.waveform.squeeze(0)
         
-    def _split_to_chunks(self, shift: bool = False) -> None:
+        self.engine.loadModel()
-        self.logger.info(f"Splitting audio on chunks...")
         
-        self.chunk_size = self.custom_chunk_length * 16000  # 16kHz after resampling
+        self.preprocessor.prepare(self.audio.load(self.filepath))
-        total_samples = self.waveform.shape[0]
-        chunks_count = (total_samples + self.chunk_size - 1) // self.chunk_size
 
-        self.logger.info(
+        batches = self.splitter.split(self.audio.waveform)
-            f"File length - {total_samples / 16000:.1f} seconds, splitting on {chunks_count} chunks by {self.custom_chunk_length} seconds per chunk."
-        )
 
-        self.chunks = []
+        for batch in batches:
-        for idx in tqdm(range(chunks_count)):
+            batchText: str = self.engine.transcribeBatch(batch)
-            start = idx * self.chunk_size
+            transcription.append(batchText)
-            end = min((idx + 1) * self.chunk_size, total_samples)
-            chunk = self.waveform[start:end].cpu().numpy().astype("float32")
-            self.chunks.append(chunk)
 
-    def _resplit_chunks_to_batches(self) -> None:
+        self.engine.unloadModel()
-        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)}")
         
-    def _process_all_batches(self) -> None:
+        return str(" ".join(transcription))
-        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)

transcription/configuration.py

@@ -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]

transcription/device_configuration.py

@@ -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]

transcription/engines/base_engine.py

@@ -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

transcription/engines/whisper.py

@@ -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)
+        

transcription/preprocessing/audio_preprocessor.py

@@ -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)

transcription/preprocessing/splitter.py

@@ -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

transcription/torch_checker.py

@@ -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__}")
 

ui/configuration_actions.py

@@ -1,4 +1,4 @@
-from transcription.device_configuration import DeviceConfiguration
+from transcription.configuration import Configuration
 
 
 # TODO: implement saving & removing configuration

ui/state.py

@@ -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])

ui/tabs/file_selector.py

@@ -0,0 +1 @@
+# new structure coming soon

ui/tabs/main.py

@@ -0,0 +1 @@
+# new structure coming soon

ui/tabs/settings.py

@@ -0,0 +1 @@
+# new structure coming soon

ui/ui.py

@@ -11,8 +11,8 @@ import customtkinter as ctk
 import torch
 
 from transcription.audio_transcription import AudioTranscription
-from transcription.device_configuration import DeviceConfiguration
+from transcription.configuration import Configuration
-from transcription.torch_checker import check_torch
+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(),
+                modelName=self.model_var.get(),
-                batch_size=int(self.batch_var.get()),
+                batchSize=int(self.batch_var.get()),
-                chunk_length_s=int(self.chunk_var.get()),
+                chunkSize=int(self.chunk_var.get()),
                 data_type=self.dtype_var.get(),
             )
             Audio = AudioTranscription(
                 filepath=infile,
-                device_configuration=config,
+                config=config,
-                logger=self.ui_logger,
                 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():
+            # if self.create_conspect.get():
-                # TODO: add custom prompt ability here
+            #     # TODO: add custom prompt ability here
-                # TODO: add logging here
+            #     # TODO: add logging here
-                # TODO: add progressbar instead of tqdm
+            #     # TODO: add progressbar instead of tqdm
-                self.ui_logger.info(f"Starting creating conspect via {self.api_model_var.get()}...")
+            #     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:
+            #     with open("utils/prompts/default_prompt.txt", "r", encoding="utf-8") as f:
-                    default_prompt = "\n".join(f.readlines())
+            #         default_prompt = "\n".join(f.readlines())
                 
-                # if self.custom_prompt_textbox.get(): # <-- issue here
+            #     # if self.custom_prompt_textbox.get(): # <-- issue here
-                #     prompt = transcription + "\n" + self.custom_prompt_textbox.get()
+            #     #     prompt = transcription + "\n" + self.custom_prompt_textbox.get()
-                # else:
+            #     # else:
-                #     prompt = transcription + "\n" + default_prompt
+            #     #     prompt = transcription + "\n" + default_prompt
                 
-                prompt = transcription + "\n" + default_prompt
+            #     prompt = transcription + "\n" + default_prompt
 
-                request = LLMrequest(
+            #     request = LLMrequest(
-                    api_key=self.api_key_var.get(),
+            #         api_key=self.api_key_var.get(),
-                    model_name=self.api_model_var.get(),
+            #         model_name=self.api_model_var.get(),
-                    base_url=self.base_url_var.get(),
+            #         base_url=self.base_url_var.get(),
-                )
+            #     )
-                response = request.get_response(prompt=prompt)
+            #     response = request.get_response(prompt=prompt)
-                outfile += ".md"
+            #     outfile += ".md"
-                with open(outfile, "w", encoding="utf-8") as f:
+            #     with open(outfile, "w", encoding="utf-8") as f:
-                    f.write(response)
+            #         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}")

ui/ui_log_handler.py

@@ -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)

utils/default_prompt.txt

@@ -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.

utils/exceptions.py

@@ -0,0 +1,13 @@
+# 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

utils/prompts/default_prompt_md.j2

@@ -0,0 +1,58 @@
+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 }}

utils/prompts/default_prompt_tex.j2

@@ -0,0 +1,75 @@
+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("%", "\\%") }}

utils/prompts/prompt_manager.py

@@ -0,0 +1,78 @@
+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
+        )

utils/requests_to_api.py

@@ -1,6 +1,6 @@
 import openai
 
-
+# maybe make it asynchronous?
 class LLMrequest:
     def __init__(self, api_key: str, model_name: str, base_url: str = None):
         if base_url: