fix app.py to remove redundant generate_kwargs in transcription calls and enable demo sharing during launch

fix app.py to enable demo sharing during launch

refactor app.py to update demo launch logic; ensure proper execution within main guard

fix app.py to correct dtype parameter usage in model initialization and pipeline; remove redundant torch_dtype argument

fix app.py to correct variable name from dtype to torch_dtype in model and pipeline initialization

fix app.py to correct variable name for dtype in model and pipeline initialization

refactor app.py to remove flash attention installation logic and simplify attention implementation; enhance error handling in transcription functions

refactor requirements.txt to remove build isolation for flash attention installation

refactor requirements.txt to ensure flash attention installation without build isolation

refactor requirements.txt to add gradio and numpy dependencies for enhanced functionality

refactor app.py to move flash attention installation into a try-except block; ensure proper fallback to sdpa if installation fails

refactor app.py to streamline flash attention installation and model loading; remove fallback mechanisms and enhance transcription parameters

refactor app.py to remove condition_on_previous_text parameter from transcription functions; enhance output consistency

refactor app.py to enhance pipeline configuration; adjust chunk length for better context and simplify demo interface

refactor app.py to correct parameter name for model loading; streamline microphone input handling

refactor app.py to improve flash attention installation and model loading; add fallback mechanism for model loading

refactor app.py to streamline flash attention installation and model loading; enhance voice activity detection and transcription accuracy

refactor app.py to enhance transcription pipeline by adding condition_on_previous_text parameter; improve transcription accuracy

refactor app.py to specify language parameter in transcription pipeline; enhance accuracy for English transcriptions

refactor app.py to update model loading parameters for flash attention; improve code clarity and maintainability

refactor app.py to remove concurrency limit from audio stream; improve streaming performance

refactor app.py to remove unnecessary time limit from audio stream; enhance streaming functionality

refactor app.py to improve flash-attn installation and model loading with error handling; update demo launch condition

refactor app.py to streamline flash-attn installation and model loading; update requirements.txt to remove unnecessary dependencies

refactor app.py to improve model loading with flash attention and enhance streaming transcription; add packages.txt for dependencies

refactor requirements.txt to remove duplicate entries and ensure proper package versions

add flash-attn to requirements for improved attention mechanisms

install Flash Attention 2 and optimize Whisper model loading; enhance streaming transcription with pipeline approach and latency tracking

update demo launch settings for Spaces to share=False by default

adjust max_new_tokens in transcription functions to leave room for special tokens

enhance audio transcription accuracy with librosa resampling and voice activity detection; increase buffer duration for better context retention

refactor streaming transcription to only emit complete sentences and clear buffer after capture, removing incremental text accumulation logic

add incremental text accumulation to prevent duplicate transcription of buffered audio

simplify streaming transcription by removing VAD, diarization, and complex buffering logic

add speaker diarization with pyannote and optimize streaming pipeline with static caching and deque-based buffering

increase buffer duration and adjust transcription parameters to prioritize accuracy over latency

reduce buffer duration and adjust transcription parameters for lower latency

improve sentence splitting to avoid breaking on common abbreviations like Mr., Ms., Dr., etc.

remove live subtitle display and only show full transcript in the UI

improve near-duplicate detection to compare against all existing sentences instead of only the last one

add near-duplicate detection to suppress streaming variants of the same sentence

update live subtitle to persist last sentence when no incomplete audio

Merge branch 'asr-1'

refactor to use per-session state instead of global variables

update the instruction(copy) for ui

update the instruction(copy) for ui

edit the instruction for ui

uodate the new words modification

optimize the words for better display on ui

change to small whisper model