app.py

import gc
import json
import os
import re
import shutil
import subprocess
import tempfile
import traceback
from collections import Counter
from pathlib import Path
from typing import Any, Dict, List, Tuple

import gradio as gr
import numpy as np
import pandas as pd
import soundfile as sf
import torch
from faster_whisper import WhisperModel
from pyannote.audio import Pipeline

GPU_AVAILABLE = torch.cuda.is_available()
ASR_DEVICE = "cuda" if GPU_AVAILABLE else "cpu"
DIAR_DEVICE = "cuda" if GPU_AVAILABLE else "cpu"

BEAM_SIZE = 5
BEST_OF = 5
PATIENCE = 1.0
TEMPERATURES = [0.0, 0.2, 0.4]
WINDOW_SECONDS = 28.0
WINDOW_GAP_SECONDS = 1.2
WINDOW_PAD_SECONDS = 0.35
MIN_SPEECH_SECONDS = 0.18
MIN_SILENCE_SECONDS = 0.35
MAX_SEGMENT_SECONDS = 7.0
MAX_SEGMENT_WORDS = 30

BAD_PHRASES = [
    "transcribe exactly",
    "hindi must be written only in devanagari script",
    "english must be written only in latin script",
    "never use urdu arabic or perso arabic script",
    "thank you for watching",
    "subscribe",
]
URDU_ARABIC_SCRIPT_RE = re.compile(r"[\u0600-\u06FF\u0750-\u077F\u08A0-\u08FF]")

def cleanup_torch():
    gc.collect()
    if torch.cuda.is_available():
        torch.cuda.empty_cache()
        try:
            torch.cuda.ipc_collect()
        except Exception:
            pass

def compute_type_for_model(asr_model_name: str) -> str:
    if ASR_DEVICE != "cuda":
        return "int8"
    if asr_model_name == "large-v3":
        return "int8_float16"
    return "float16"

def run_cmd(cmd):
    result = subprocess.run(cmd, stdout=subprocess.PIPE, stderr=subprocess.PIPE, text=True)
    if result.returncode != 0:
        raise RuntimeError(f"Command failed:\n{' '.join(cmd)}\n\nSTDERR:\n{result.stderr}")
    return result

def ffprobe_duration(input_path: Path):
    cmd = ["ffprobe", "-v", "error", "-show_entries", "format=duration", "-of", "default=noprint_wrappers=1:nokey=1", str(input_path)]
    result = subprocess.run(cmd, stdout=subprocess.PIPE, stderr=subprocess.PIPE, text=True)
    if result.returncode != 0:
        return None
    try:
        return float(result.stdout.strip())
    except Exception:
        return None

def to_wav_16k(input_path: Path, output_path: Path, enhance_audio: bool):
    af = ["aresample=async=1:first_pts=0"]
    if enhance_audio:
        af = [
            "highpass=f=80",
            "lowpass=f=7600",
            "dynaudnorm=f=150:g=15:p=0.90",
            "aresample=async=1:first_pts=0",
        ]
    cmd = ["ffmpeg", "-y", "-i", str(input_path), "-vn", "-ac", "1", "-ar", "16000", "-c:a", "pcm_s16le", "-af", ",".join(af), str(output_path)]
    run_cmd(cmd)
    return output_path

def load_waveform_for_pyannote(wav_path: Path):
    audio, sample_rate = sf.read(str(wav_path), dtype="float32")
    if audio.ndim > 1:
        audio = audio.mean(axis=1)
    waveform = torch.from_numpy(audio).unsqueeze(0)
    return {"waveform": waveform, "sample_rate": int(sample_rate)}

def load_audio_np(audio_path: Path):
    audio, sample_rate = sf.read(str(audio_path), dtype="float32")
    if audio.ndim > 1:
        audio = np.mean(audio, axis=1).astype(np.float32)
    audio = np.asarray(audio, dtype=np.float32)
    if sample_rate != 16000:
        raise ValueError(f"Expected 16k WAV after ffmpeg conversion, got {sample_rate}")
    if len(audio) == 0:
        raise ValueError("Audio file is empty")
    return audio, sample_rate

def normalize_spaces(text):
    text = (text or "").replace("\n", " ").replace("\r", " ")
    text = re.sub(r"\s+", " ", text).strip()
    return text

def normalize_for_compare(text):
    text = normalize_spaces(text).casefold()
    text = re.sub(r"[\W_]+", " ", text, flags=re.UNICODE)
    return re.sub(r"\s+", " ", text).strip()

def looks_bad_text(text):
    norm = normalize_for_compare(text)
    if not norm:
        return True
    return any(p in norm for p in BAD_PHRASES)

def contains_urdu_or_arabic_script(text):
    return bool(URDU_ARABIC_SCRIPT_RE.search(text or ""))

def similarity(a: str, b: str) -> float:
    from difflib import SequenceMatcher
    if not a and not b:
        return 1.0
    if not a or not b:
        return 0.0
    return SequenceMatcher(None, a, b).ratio()

def text_has_bad_repetition(text):
    norm = normalize_for_compare(text)
    words = norm.split()
    if len(words) < 8:
        return False
    for n in range(1, min(6, len(words) // 2 + 1)):
        run = 1
        prev = None
        for i in range(0, len(words) - n + 1, n):
            gram = tuple(words[i:i + n])
            if len(gram) != n:
                continue
            if gram == prev:
                run += 1
                if run >= 3:
                    return True
            else:
                run = 1
                prev = gram
    counts = Counter(words)
    if len(words) >= 12 and counts and max(counts.values()) / max(1, len(words)) >= 0.45:
        return True
    return False

def safe_float(value: Any, default: float = 0.0) -> float:
    try:
        return float(value)
    except Exception:
        return default

def format_hhmmss_mmm(seconds):
    seconds = max(0.0, float(seconds))
    total_ms = int(round(seconds * 1000.0))
    ms = total_ms % 1000
    total_s = total_ms // 1000
    s = total_s % 60
    total_m = total_s // 60
    m = total_m % 60
    h = total_m // 60
    return f"{h:02d}:{m:02d}:{s:02d}.{ms:03d}"

def preflight(media_file, asr_model_name, language, enhance_audio, num_speakers, min_speakers, max_speakers):
    lines = [
        "=== PREFLIGHT ===",
        f"GPU available: {GPU_AVAILABLE}",
        f"ASR device: {ASR_DEVICE}",
        f"Diarization device: {DIAR_DEVICE}",
        "Diarization model: pyannote/speaker-diarization-community-1",
        f"ASR model: {asr_model_name}",
        f"ASR compute type: {compute_type_for_model(asr_model_name)}",
        f"Language: {language}",
        f"Enhance audio: {enhance_audio}",
        f"HF_TOKEN present: {bool(os.getenv('HF_TOKEN'))}",
        f"ffmpeg found: {shutil.which('ffmpeg') is not None}",
        f"ffprobe found: {shutil.which('ffprobe') is not None}",
        f"torch version: {torch.__version__}",
        f"Speaker controls -> num:{num_speakers} min:{min_speakers} max:{max_speakers}",
        "Repo-style transcription logic is active.",
    ]
    if media_file is None:
        lines.append("No media file uploaded yet.")
        return "\n".join(lines)
    try:
        p = Path(media_file)
        size_mb = p.stat().st_size / (1024 * 1024)
        dur = ffprobe_duration(p)
        lines.append(f"Uploaded file: {p.name}")
        lines.append(f"File size: {size_mb:.2f} MB")
        if dur is not None:
            lines.append(f"Estimated duration: {dur:.2f} sec")
            if dur > 1800:
                lines.append("Warning: long file on T4 small. Start with medium.")
    except Exception as e:
        lines.append(f"File inspection failed: {e}")
    return "\n".join(lines)

# ===== repo-style speech windows =====
def frame_rms(audio: np.ndarray, sample_rate: int, frame_ms: float = 30.0, hop_ms: float = 10.0) -> Tuple[np.ndarray, np.ndarray]:
    frame = max(1, int(sample_rate * frame_ms / 1000.0))
    hop = max(1, int(sample_rate * hop_ms / 1000.0))
    if len(audio) < frame:
        padded = np.pad(audio, (0, frame - len(audio)))
        return np.array([0.0], dtype=np.float32), np.array([float(np.sqrt(np.mean(padded * padded) + 1e-12))], dtype=np.float32)
    starts = np.arange(0, len(audio) - frame + 1, hop, dtype=np.int64)
    rms = np.empty(len(starts), dtype=np.float32)
    for i, start in enumerate(starts):
        chunk = audio[start:start + frame]
        rms[i] = float(np.sqrt(np.mean(chunk * chunk) + 1e-12))
    times = starts.astype(np.float32) / float(sample_rate)
    return times, rms

def fill_short_silences(active: np.ndarray, max_gap_frames: int) -> np.ndarray:
    if max_gap_frames <= 0 or len(active) == 0:
        return active
    output = active.copy()
    i = 0
    n = len(output)
    while i < n:
        if output[i]:
            i += 1
            continue
        start = i
        while i < n and not output[i]:
            i += 1
        end = i
        left_active = start > 0 and output[start - 1]
        right_active = end < n and output[end]
        if left_active and right_active and (end - start) <= max_gap_frames:
            output[start:end] = True
    return output

def remove_short_speech(active: np.ndarray, min_speech_frames: int) -> np.ndarray:
    if min_speech_frames <= 1 or len(active) == 0:
        return active
    output = active.copy()
    i = 0
    n = len(output)
    while i < n:
        if not output[i]:
            i += 1
            continue
        start = i
        while i < n and output[i]:
            i += 1
        end = i
        if (end - start) < min_speech_frames:
            output[start:end] = False
    return output

def detect_speech_intervals(audio: np.ndarray, sample_rate: int, total_duration: float) -> List[Tuple[float, float]]:
    times, rms = frame_rms(audio, sample_rate)
    db = 20.0 * np.log10(np.maximum(rms, 1e-8))
    p20 = float(np.percentile(db, 20))
    p50 = float(np.percentile(db, 50))
    p75 = float(np.percentile(db, 75))
    p90 = float(np.percentile(db, 90))
    threshold = max(p20 + 6.0, p50 + 2.5)
    threshold = min(threshold, p75 - 2.0 if p75 > p20 + 8.0 else threshold)
    threshold = min(threshold, p90 - 8.0 if p90 > p20 + 12.0 else threshold)
    active = db >= threshold
    hop_seconds = 0.010
    active = fill_short_silences(active, max_gap_frames=int(MIN_SILENCE_SECONDS / hop_seconds))
    active = remove_short_speech(active, min_speech_frames=max(1, int(MIN_SPEECH_SECONDS / hop_seconds)))

    intervals = []
    i = 0
    n = len(active)
    while i < n:
        if not active[i]:
            i += 1
            continue
        start_idx = i
        while i < n and active[i]:
            i += 1
        end_idx = i
        start = max(0.0, float(times[start_idx]) - WINDOW_PAD_SECONDS)
        end = min(total_duration, float(times[min(end_idx - 1, len(times) - 1)]) + 0.03 + WINDOW_PAD_SECONDS)
        if end - start >= MIN_SPEECH_SECONDS:
            intervals.append((start, end))
    if not intervals:
        return [(0.0, total_duration)]

    merged = []
    for start, end in intervals:
        if not merged:
            merged.append((start, end))
            continue
        prev_start, prev_end = merged[-1]
        if start - prev_end <= WINDOW_GAP_SECONDS and (end - prev_start) <= WINDOW_SECONDS:
            merged[-1] = (prev_start, max(prev_end, end))
        else:
            merged.append((start, end))
    return merged

def split_long_intervals(intervals: List[Tuple[float, float]], total_duration: float) -> List[Tuple[float, float]]:
    windows = []
    for start, end in intervals:
        duration = end - start
        if duration <= WINDOW_SECONDS:
            windows.append((start, end))
            continue
        cursor = start
        overlap = min(1.0, max(0.0, WINDOW_PAD_SECONDS))
        while cursor < end:
            win_end = min(end, cursor + WINDOW_SECONDS)
            windows.append((max(0.0, cursor), min(total_duration, win_end)))
            if win_end >= end:
                break
            cursor = max(cursor + 1.0, win_end - overlap)
    cleaned = []
    for start, end in windows:
        if end - start < 0.08:
            continue
        if cleaned and abs(start - cleaned[-1][0]) < 0.05 and abs(end - cleaned[-1][1]) < 0.05:
            continue
        cleaned.append((round(start, 3), round(end, 3)))
    return cleaned

def word_list_from_segment(seg: Any, base_offset: float, window_start: float, window_end: float) -> List[Dict[str, Any]]:
    words = []
    raw_words = getattr(seg, "words", None) or []
    for w in raw_words:
        if getattr(w, "start", None) is None or getattr(w, "end", None) is None:
            continue
        start = float(w.start) + base_offset
        end = float(w.end) + base_offset
        if end < window_start - 0.20 or start > window_end + 0.20:
            continue
        words.append({"start": round(max(0.0, start), 2), "end": round(max(0.0, end), 2), "word": str(getattr(w, "word", "") or "")})
    return words

def transcribe_window(model: WhisperModel, audio: np.ndarray, sample_rate: int, start: float, end: float, language: str) -> List[Dict[str, Any]]:
    start_sample = max(0, int(start * sample_rate))
    end_sample = min(len(audio), int(end * sample_rate))
    chunk = audio[start_sample:end_sample]
    if len(chunk) < int(0.08 * sample_rate):
        return []

    prompt = (
        "This is an Indian meeting conversation containing only Hindi, Hinglish, and English. "
        "Transcribe exactly. Do not translate. "
        "Hindi must be written only in Devanagari script. "
        "English must be written only in Latin script. "
        "Never use Urdu, Arabic, or Perso-Arabic script. "
        "Preserve names, product terms, technical terms, repository names, GitHub terms, and code-mixed speech exactly."
    )
    kwargs: Dict[str, Any] = {
        "language": language,
        "beam_size": BEAM_SIZE,
        "best_of": BEST_OF,
        "patience": PATIENCE,
        "temperature": TEMPERATURES,
        "condition_on_previous_text": False,
        "vad_filter": False,
        "word_timestamps": True,
        "task": "transcribe",
        "initial_prompt": prompt,
        "no_speech_threshold": 0.82,
        "log_prob_threshold": -1.35,
        "compression_ratio_threshold": 2.55,
        "hallucination_silence_threshold": 1.2,
    }
    try:
        segments_iter, _ = model.transcribe(chunk, **kwargs)
    except TypeError:
        for key in ["hallucination_silence_threshold", "best_of", "patience", "initial_prompt"]:
            kwargs.pop(key, None)
        segments_iter, _ = model.transcribe(chunk, **kwargs)

    output = []
    for seg in segments_iter:
        text = normalize_spaces(str(getattr(seg, "text", "") or ""))
        if not text:
            continue
        if contains_urdu_or_arabic_script(text):
            continue
        seg_start = float(getattr(seg, "start", 0.0)) + start
        seg_end = float(getattr(seg, "end", 0.0)) + start
        if seg_end <= seg_start:
            continue
        seg_start = max(0.0, min(seg_start, end))
        seg_end = max(seg_start + 0.01, min(seg_end, end))
        output.append({
            "start": round(seg_start, 2),
            "end": round(seg_end, 2),
            "text": text,
            "words": word_list_from_segment(seg, start, start, end),
        })
    return output

def split_segment_by_words(seg: Dict[str, Any]) -> List[Dict[str, Any]]:
    words = seg.get("words") or []
    if not words:
        return [dict(seg)]
    start = safe_float(seg.get("start"))
    end = safe_float(seg.get("end"), start)
    if (end - start) <= MAX_SEGMENT_SECONDS and len(words) <= MAX_SEGMENT_WORDS:
        return [dict(seg)]

    pieces = []
    bucket = []

    def flush():
        nonlocal bucket
        if not bucket:
            return
        text = normalize_spaces("".join(str(w.get("word", "")) for w in bucket))
        if text:
            new_seg = dict(seg)
            new_seg["start"] = round(safe_float(bucket[0].get("start")), 2)
            new_seg["end"] = round(safe_float(bucket[-1].get("end")), 2)
            new_seg["text"] = text
            new_seg["words"] = [dict(w) for w in bucket]
            pieces.append(new_seg)
        bucket = []

    for idx, word in enumerate(words):
        bucket.append(word)
        bucket_start = safe_float(bucket[0].get("start"))
        bucket_end = safe_float(bucket[-1].get("end"))
        duration = bucket_end - bucket_start
        next_gap = 0.0
        if idx + 1 < len(words):
            next_gap = max(0.0, safe_float(words[idx + 1].get("start")) - safe_float(word.get("end")))
        token = str(word.get("word", "")).strip()
        boundary = token.endswith((".", "?", "!", ",", "।")) or next_gap >= 0.45
        too_long = duration >= MAX_SEGMENT_SECONDS or len(bucket) >= MAX_SEGMENT_WORDS
        if (boundary and duration >= 0.9) or too_long:
            flush()
    flush()
    return pieces or [dict(seg)]

def dedupe_transcript_segments(segments: List[Dict[str, Any]]) -> List[Dict[str, Any]]:
    if not segments:
        return []
    segments = sorted(segments, key=lambda x: (safe_float(x.get("start")), safe_float(x.get("end"))))
    cleaned = []
    for seg in segments:
        text = normalize_spaces(str(seg.get("text", "")))
        if not text:
            continue
        if contains_urdu_or_arabic_script(text):
            continue
        if text_has_bad_repetition(text):
            continue
        seg = dict(seg)
        seg["text"] = text
        start = safe_float(seg.get("start"))
        end = safe_float(seg.get("end"), start)
        if end <= start:
            continue

        curr_norm = normalize_for_compare(text)
        duplicate_idx = None
        for idx in range(max(0, len(cleaned) - 6), len(cleaned)):
            prev = cleaned[idx]
            prev_norm = normalize_for_compare(str(prev.get("text", "")))
            if not prev_norm or not curr_norm:
                continue
            prev_start = safe_float(prev.get("start"))
            prev_end = safe_float(prev.get("end"), prev_start)
            time_overlap = max(0.0, min(prev_end, end) - max(prev_start, start))
            min_duration = max(0.01, min(prev_end - prev_start, end - start))
            overlap_ratio = time_overlap / min_duration
            near_boundary = abs(start - prev_start) <= 1.25 or abs(end - prev_end) <= 1.25 or start - prev_end <= 0.8
            same_or_contained = curr_norm == prev_norm or curr_norm in prev_norm or prev_norm in curr_norm
            very_similar = similarity(curr_norm, prev_norm) >= 0.94
            if (overlap_ratio >= 0.35 or near_boundary) and (same_or_contained or very_similar):
                duplicate_idx = idx
                break

        if duplicate_idx is not None:
            prev = cleaned[duplicate_idx]
            if len(curr_norm) > len(normalize_for_compare(str(prev.get("text", "")))):
                prev["text"] = text
                if seg.get("words"):
                    prev["words"] = seg.get("words")
            prev["start"] = round(min(safe_float(prev.get("start")), start), 2)
            prev["end"] = round(max(safe_float(prev.get("end")), end), 2)
            continue

        cleaned.append(seg)

    final = []
    for seg in cleaned:
        item = {"start": round(safe_float(seg.get("start")), 2), "end": round(safe_float(seg.get("end")), 2), "text": normalize_spaces(str(seg.get("text", "")))}
        if seg.get("words"):
            item["words"] = seg.get("words")
        final.append(item)
    return final

def transcribe_audio_chunked_repo_style(wav_path: Path, asr_model_name: str, language_choice: str):
    audio, sample_rate = load_audio_np(wav_path)
    total_duration = len(audio) / float(sample_rate)
    intervals = detect_speech_intervals(audio, sample_rate, total_duration)
    windows = split_long_intervals(intervals, total_duration)

    model = WhisperModel(
        asr_model_name,
        device=ASR_DEVICE,
        compute_type=compute_type_for_model(asr_model_name),
        cpu_threads=4 if ASR_DEVICE == "cpu" else 2,
        num_workers=1,
    )

    # mimic attached repo default behavior
    lang = "hi" if language_choice == "auto" else language_choice

    all_segments = []
    for start, end in windows:
        try:
            window_segments = transcribe_window(model, audio, sample_rate, start, end, lang)
        except Exception:
            continue
        for seg in window_segments:
            all_segments.extend(split_segment_by_words(seg))

    del model
    cleanup_torch()

    results = dedupe_transcript_segments(all_segments)
    results.sort(key=lambda x: (float(x["start"]), float(x["end"])))
    return results, len(windows), lang

def choose_speaker_for_word(word_start, word_end, diar_df):
    if diar_df.empty:
        return "UNKNOWN_SPEAKER"
    tmp = diar_df.copy()
    tmp["overlap"] = tmp.apply(lambda r: max(0.0, min(word_end, r["end"]) - max(word_start, r["start"])), axis=1)
    hits = tmp[tmp["overlap"] > 0].copy()
    if not hits.empty:
        best = hits.sort_values("overlap", ascending=False).iloc[0]
        return str(best["speaker"])
    mid = (word_start + word_end) / 2.0
    tmp["dist"] = tmp.apply(lambda r: min(abs(mid - r["start"]), abs(mid - r["end"])), axis=1)
    best = tmp.sort_values("dist").iloc[0]
    return str(best["speaker"])

def assign_speaker_to_segment(segment, diar_df):
    speaker_counts = {}
    for w in segment.get("words", []):
        spk = choose_speaker_for_word(float(w["start"]), float(w["end"]), diar_df)
        speaker_counts[spk] = speaker_counts.get(spk, 0) + 1
    if speaker_counts:
        return max(speaker_counts, key=speaker_counts.get)
    return "UNKNOWN_SPEAKER"

def merge_adjacent_same_speaker(segments):
    if not segments:
        return []
    merged = [dict(segments[0])]
    for seg in segments[1:]:
        last = merged[-1]
        if seg["speaker"] == last["speaker"]:
            last["end"] = max(float(last["end"]), float(seg["end"]))
            if seg["text"]:
                last["text"] = normalize_spaces(last["text"] + " " + seg["text"])
        else:
            merged.append(dict(seg))
    return merged

def process_media(media_file, asr_model_name, language, enhance_audio, filter_known_bad, num_speakers, min_speakers, max_speakers, progress=gr.Progress(track_tqdm=False)):
    if media_file is None:
        raise gr.Error("Please upload a media file.")
    hf_token = (os.getenv("HF_TOKEN") or "").strip()
    if not hf_token:
        raise gr.Error("Missing HF_TOKEN Space Secret.")
    work_root = Path(tempfile.mkdtemp(prefix="diarized_c1_"))
    out_dir = work_root / "outputs"
    out_dir.mkdir(parents=True, exist_ok=True)
    input_path = Path(media_file)
    wav_path = out_dir / "input_16k.wav"

    try:
        progress(0.05, desc="Preparing audio")
        to_wav_16k(input_path, wav_path, enhance_audio=enhance_audio)

        progress(0.16, desc=f"Repo-style transcription: {asr_model_name}")
        raw_segments, window_count, used_language = transcribe_audio_chunked_repo_style(wav_path, asr_model_name, language)

        if filter_known_bad:
            filtered = []
            for seg in raw_segments:
                t = normalize_spaces(seg.get("text", ""))
                if not t:
                    continue
                if looks_bad_text(t):
                    continue
                if text_has_bad_repetition(t):
                    continue
                seg = dict(seg)
                seg["text"] = t
                filtered.append(seg)
            raw_segments = filtered

        word_count = sum(len(seg.get("words", []) or []) for seg in raw_segments)

        progress(0.56, desc="Loading diarization model")
        pipeline = Pipeline.from_pretrained("pyannote/speaker-diarization-community-1", token=hf_token)
        if DIAR_DEVICE == "cuda":
            pipeline.to(torch.device("cuda"))

        diar_kwargs = {}
        if num_speakers and int(num_speakers) > 0:
            diar_kwargs["num_speakers"] = int(num_speakers)
        else:
            if min_speakers and int(min_speakers) > 0:
                diar_kwargs["min_speakers"] = int(min_speakers)
            if max_speakers and int(max_speakers) > 0:
                diar_kwargs["max_speakers"] = int(max_speakers)

        progress(0.70, desc="Running diarization")
        media = load_waveform_for_pyannote(wav_path)
        output = pipeline(media, **diar_kwargs)
        if hasattr(output, "exclusive_speaker_diarization"):
            diarization = output.exclusive_speaker_diarization
        elif hasattr(output, "speaker_diarization"):
            diarization = output.speaker_diarization
        else:
            diarization = output

        del pipeline
        cleanup_torch()

        diar_rows = []
        for turn, _, speaker in diarization.itertracks(yield_label=True):
            diar_rows.append({"start": float(turn.start), "end": float(turn.end), "speaker": str(speaker)})
        diar_df = pd.DataFrame(diar_rows).sort_values(["start", "end"]).reset_index(drop=True)

        progress(0.84, desc="Assigning speakers to raw segments")
        assigned = []
        for seg in raw_segments:
            speaker = assign_speaker_to_segment(seg, diar_df)
            assigned.append({
                "speaker": speaker,
                "start": float(seg["start"]),
                "end": float(seg["end"]),
                "text": seg["text"],
            })

        cleaned = merge_adjacent_same_speaker(assigned)

        raw_speakers = []
        for r in cleaned:
            if r["speaker"] not in raw_speakers:
                raw_speakers.append(r["speaker"])
        speaker_map = {spk: f"Speaker {i:02d}" for i, spk in enumerate(raw_speakers, start=1)}

        final_rows = []
        for seg in cleaned:
            final_rows.append({
                "speaker": speaker_map[seg["speaker"]],
                "start": float(seg["start"]),
                "end": float(seg["end"]),
                "start_hhmmss": format_hhmmss_mmm(seg["start"]),
                "end_hhmmss": format_hhmmss_mmm(seg["end"]),
                "text": seg["text"],
            })

        df = pd.DataFrame(final_rows)
        txt_path = out_dir / "speaker_transcript.txt"
        json_path = out_dir / "speaker_transcript.json"
        csv_path = out_dir / "speaker_transcript.csv"

        df.to_csv(csv_path, index=False)
        with open(json_path, "w", encoding="utf-8") as f:
            json.dump(final_rows, f, ensure_ascii=False, indent=2)
        with open(txt_path, "w", encoding="utf-8") as f:
            for _, row in df.iterrows():
                f.write(f"{row['speaker']}: {row['start_hhmmss']} - {row['end_hhmmss']}\n")
                f.write(f"Text: {row['text']}\n\n")

        preview_lines = [
            "=== RUN SUMMARY ===",
            f"ASR model used: {asr_model_name}",
            f"Repo-style language used: {used_language}",
            f"ASR device used: {ASR_DEVICE}",
            f"Diarization device used: {DIAR_DEVICE}",
            f"Speech windows: {window_count}",
            f"Raw transcript segments: {len(raw_segments)}",
            f"Raw transcript words: {word_count}",
            f"Diarization segments: {len(diar_df)}",
            f"Final cleaned diarized segments: {len(df)}",
            f"Detected speakers: {len(raw_speakers)}",
            "",
        ]
        for _, row in df.head(20).iterrows():
            preview_lines.append(f"{row['speaker']}: {row['start_hhmmss']} - {row['end_hhmmss']}")
            preview_lines.append(f"Text: {row['text']}")
            preview_lines.append("")

        progress(1.0, desc="Done")
        return "\n".join(preview_lines), df, str(txt_path), str(json_path), str(csv_path)
    except Exception:
        return "=== FAILURE ===\n" + traceback.format_exc(), [], None, None, None

with gr.Blocks(title="Diarized Speaker Segments Community-1") as demo:
    gr.Markdown(
        """
        # Diarized Speaker Segments Community-1
        Uses **attached-repo transcription logic** plus **pyannote/speaker-diarization-community-1**.

        Cleanup rule:
        - if adjacent speaker segments are the same, merge them
        - otherwise do not touch them

        Notes:
        - default ASR model is **medium**
        - **large-v3** is available for comparison
        - default language is **hi** to mimic the attached repo behavior
        """
    )
    with gr.Row():
        with gr.Column():
            media_file = gr.File(label="Upload video/audio", type="filepath")
            asr_model_name = gr.Dropdown(
                choices=["medium", "large-v3"],
                value="medium",
                label="ASR model",
                info="Default is medium. large-v3 is available for comparison."
            )
            language = gr.Dropdown(
                choices=["hi", "auto", "en"],
                value="hi",
                label="Language",
                info="Default is hi to mimic the attached repo transcription behavior."
            )
            enhance_audio = gr.Checkbox(value=True, label="Enhance audio before transcription")
            filter_known_bad = gr.Checkbox(value=True, label="Filter obvious hallucination / prompt-leak phrases")
            with gr.Row():
                num_speakers = gr.Number(label="Exact number of speakers (optional)", value=None, precision=0)
                min_speakers = gr.Number(label="Min speakers (optional)", value=1, precision=0)
                max_speakers = gr.Number(label="Max speakers (optional)", value=8, precision=0)
            with gr.Row():
                preflight_btn = gr.Button("Run preflight")
                run_btn = gr.Button("Generate diarized transcript", variant="primary")
        with gr.Column():
            preview = gr.Textbox(label="Diagnostics / Preview", lines=24)
            table = gr.Dataframe(label="Diarized transcript segments", wrap=True, interactive=False)
            txt_file = gr.File(label="TXT output")
            json_file = gr.File(label="JSON output")
            csv_file = gr.File(label="CSV output")

    preflight_btn.click(
        fn=preflight,
        inputs=[media_file, asr_model_name, language, enhance_audio, num_speakers, min_speakers, max_speakers],
        outputs=[preview],
    )
    run_btn.click(
        fn=process_media,
        inputs=[media_file, asr_model_name, language, enhance_audio, filter_known_bad, num_speakers, min_speakers, max_speakers],
        outputs=[preview, table, txt_file, json_file, csv_file],
    )

if __name__ == "__main__":
    demo.launch(server_name="0.0.0.0", server_port=7860)