File size: 12,052 Bytes
d722140
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
#!/usr/bin/env python3
"""
Dataset Preparation Script for Indian Language TTS Training

This script prepares speech datasets for training VITS models on Indian languages.
It handles data from multiple sources and creates a unified format.

Supported Datasets:
- OpenSLR Indian Language Datasets
- Mozilla Common Voice (Indian subsets)
- IndicTTS Dataset (IIT Madras)
- Custom recordings

Output Format:
- audio/: Normalized WAV files (22050Hz, mono, 16-bit)
- metadata.csv: text|audio_path|speaker_id|duration
"""

import os
import sys
import csv
import json
import argparse
import logging
from pathlib import Path
from typing import List, Tuple, Optional
from dataclasses import dataclass
from concurrent.futures import ProcessPoolExecutor

import numpy as np

# Try to import audio processing libraries
try:
    import librosa
    import soundfile as sf

    HAS_AUDIO = True
except ImportError:
    HAS_AUDIO = False
    print("Warning: librosa/soundfile not installed. Audio processing disabled.")


# Dataset configurations
DATASET_CONFIGS = {
    "openslr_hindi": {
        "url": "https://www.openslr.org/resources/103/",
        "name": "OpenSLR Hindi ASR Corpus",
        "language": "hindi",
        "sample_rate": 16000,
    },
    "openslr_bengali": {
        "url": "https://www.openslr.org/resources/37/",
        "name": "OpenSLR Bengali Multi-speaker",
        "language": "bengali",
        "sample_rate": 16000,
    },
    "openslr_marathi": {
        "url": "https://www.openslr.org/resources/64/",
        "name": "OpenSLR Marathi",
        "language": "marathi",
        "sample_rate": 16000,
    },
    "openslr_telugu": {
        "url": "https://www.openslr.org/resources/66/",
        "name": "OpenSLR Telugu",
        "language": "telugu",
        "sample_rate": 16000,
    },
    "openslr_kannada": {
        "url": "https://www.openslr.org/resources/79/",
        "name": "OpenSLR Kannada",
        "language": "kannada",
        "sample_rate": 16000,
    },
    "openslr_gujarati": {
        "url": "https://www.openslr.org/resources/78/",
        "name": "OpenSLR Gujarati",
        "language": "gujarati",
        "sample_rate": 16000,
    },
    "commonvoice_hindi": {
        "url": "https://commonvoice.mozilla.org/en/datasets",
        "name": "Mozilla Common Voice Hindi",
        "language": "hindi",
        "sample_rate": 48000,
    },
    "indictts": {
        "url": "https://www.iitm.ac.in/donlab/tts/",
        "name": "IndicTTS Dataset (IIT Madras)",
        "languages": ["hindi", "bengali", "marathi", "telugu", "kannada", "gujarati"],
        "sample_rate": 22050,
    },
}


@dataclass
class AudioSample:
    """Represents a single audio sample"""

    audio_path: Path
    text: str
    speaker_id: str
    language: str
    duration: float = 0.0
    sample_rate: int = 22050


class DatasetProcessor:
    """Process and prepare datasets for TTS training"""

    TARGET_SAMPLE_RATE = 22050
    MIN_DURATION = 0.5  # seconds
    MAX_DURATION = 15.0  # seconds

    def __init__(self, output_dir: Path, language: str):
        self.output_dir = output_dir
        self.language = language
        self.audio_dir = output_dir / "audio"
        self.audio_dir.mkdir(parents=True, exist_ok=True)

        logging.basicConfig(level=logging.INFO)
        self.logger = logging.getLogger(__name__)

    def process_audio(self, input_path: Path, output_path: Path) -> Optional[float]:
        """
        Process a single audio file:
        - Resample to target sample rate
        - Convert to mono
        - Normalize volume
        - Trim silence
        """
        if not HAS_AUDIO:
            return None

        try:
            # Load audio
            audio, sr = librosa.load(input_path, sr=None, mono=True)

            # Resample if necessary
            if sr != self.TARGET_SAMPLE_RATE:
                audio = librosa.resample(
                    audio, orig_sr=sr, target_sr=self.TARGET_SAMPLE_RATE
                )

            # Trim silence
            audio, _ = librosa.effects.trim(audio, top_db=20)

            # Normalize
            audio = audio / np.abs(audio).max() * 0.95

            # Calculate duration
            duration = len(audio) / self.TARGET_SAMPLE_RATE

            # Filter by duration
            if duration < self.MIN_DURATION or duration > self.MAX_DURATION:
                return None

            # Save processed audio
            sf.write(output_path, audio, self.TARGET_SAMPLE_RATE)

            return duration

        except Exception as e:
            self.logger.warning(f"Error processing {input_path}: {e}")
            return None

    def process_openslr(self, data_dir: Path) -> List[AudioSample]:
        """Process OpenSLR format dataset"""
        samples = []

        # OpenSLR typically has transcripts.txt or similar
        transcript_file = data_dir / "transcripts.txt"
        if not transcript_file.exists():
            transcript_file = data_dir / "text"

        if transcript_file.exists():
            with open(transcript_file, "r", encoding="utf-8") as f:
                for line in f:
                    parts = line.strip().split("|")
                    if len(parts) >= 2:
                        audio_id, text = parts[0], parts[1]
                        audio_path = data_dir / "audio" / f"{audio_id}.wav"

                        if audio_path.exists():
                            output_path = self.audio_dir / f"{audio_id}.wav"
                            duration = self.process_audio(audio_path, output_path)

                            if duration:
                                samples.append(
                                    AudioSample(
                                        audio_path=output_path,
                                        text=text,
                                        speaker_id="spk_001",
                                        language=self.language,
                                        duration=duration,
                                    )
                                )

        return samples

    def process_commonvoice(self, data_dir: Path) -> List[AudioSample]:
        """Process Mozilla Common Voice format"""
        samples = []

        # Common Voice uses validated.tsv
        tsv_file = data_dir / "validated.tsv"
        clips_dir = data_dir / "clips"

        if tsv_file.exists():
            with open(tsv_file, "r", encoding="utf-8") as f:
                reader = csv.DictReader(f, delimiter="\t")
                for row in reader:
                    audio_path = clips_dir / row["path"]
                    text = row["sentence"]
                    speaker_id = row.get("client_id", "unknown")[:8]

                    if audio_path.exists():
                        output_name = f"cv_{audio_path.stem}.wav"
                        output_path = self.audio_dir / output_name
                        duration = self.process_audio(audio_path, output_path)

                        if duration:
                            samples.append(
                                AudioSample(
                                    audio_path=output_path,
                                    text=text,
                                    speaker_id=speaker_id,
                                    language=self.language,
                                    duration=duration,
                                )
                            )

        return samples

    def process_indictts(self, data_dir: Path) -> List[AudioSample]:
        """Process IndicTTS format dataset"""
        samples = []

        # IndicTTS has wav/ folder and txt/ folder
        wav_dir = data_dir / "wav"
        txt_dir = data_dir / "txt"

        if wav_dir.exists() and txt_dir.exists():
            for wav_file in wav_dir.glob("*.wav"):
                txt_file = txt_dir / f"{wav_file.stem}.txt"

                if txt_file.exists():
                    with open(txt_file, "r", encoding="utf-8") as f:
                        text = f.read().strip()

                    output_path = self.audio_dir / wav_file.name
                    duration = self.process_audio(wav_file, output_path)

                    if duration:
                        samples.append(
                            AudioSample(
                                audio_path=output_path,
                                text=text,
                                speaker_id="indic_001",
                                language=self.language,
                                duration=duration,
                            )
                        )

        return samples

    def save_metadata(self, samples: List[AudioSample]):
        """Save processed samples to metadata CSV"""
        metadata_path = self.output_dir / "metadata.csv"

        with open(metadata_path, "w", encoding="utf-8", newline="") as f:
            writer = csv.writer(f, delimiter="|")
            writer.writerow(["audio_path", "text", "speaker_id", "duration"])

            for sample in samples:
                writer.writerow(
                    [
                        sample.audio_path.name,
                        sample.text,
                        sample.speaker_id,
                        f"{sample.duration:.3f}",
                    ]
                )

        self.logger.info(f"Saved {len(samples)} samples to {metadata_path}")

        # Save statistics
        stats = {
            "total_samples": len(samples),
            "total_duration_hours": sum(s.duration for s in samples) / 3600,
            "language": self.language,
            "speakers": len(set(s.speaker_id for s in samples)),
        }

        with open(self.output_dir / "stats.json", "w") as f:
            json.dump(stats, f, indent=2)

        self.logger.info(f"Dataset stats: {stats}")


def create_train_val_split(metadata_path: Path, train_ratio: float = 0.95):
    """Split metadata into train and validation sets"""
    with open(metadata_path, "r", encoding="utf-8") as f:
        reader = csv.reader(f, delimiter="|")
        header = next(reader)
        rows = list(reader)

    # Shuffle
    np.random.shuffle(rows)

    # Split
    split_idx = int(len(rows) * train_ratio)
    train_rows = rows[:split_idx]
    val_rows = rows[split_idx:]

    # Save splits
    for name, data in [("train", train_rows), ("val", val_rows)]:
        output_path = metadata_path.parent / f"metadata_{name}.csv"
        with open(output_path, "w", encoding="utf-8", newline="") as f:
            writer = csv.writer(f, delimiter="|")
            writer.writerow(header)
            writer.writerows(data)

        print(f"Saved {len(data)} samples to {output_path}")


def main():
    parser = argparse.ArgumentParser(description="Prepare datasets for TTS training")
    parser.add_argument(
        "--input", type=str, required=True, help="Input dataset directory"
    )
    parser.add_argument("--output", type=str, required=True, help="Output directory")
    parser.add_argument("--language", type=str, required=True, help="Target language")
    parser.add_argument(
        "--format",
        type=str,
        default="openslr",
        choices=["openslr", "commonvoice", "indictts"],
        help="Dataset format",
    )
    parser.add_argument("--split", action="store_true", help="Create train/val split")

    args = parser.parse_args()

    processor = DatasetProcessor(
        output_dir=Path(args.output),
        language=args.language,
    )

    # Process based on format
    if args.format == "openslr":
        samples = processor.process_openslr(Path(args.input))
    elif args.format == "commonvoice":
        samples = processor.process_commonvoice(Path(args.input))
    elif args.format == "indictts":
        samples = processor.process_indictts(Path(args.input))

    # Save metadata
    processor.save_metadata(samples)

    # Create train/val split if requested
    if args.split:
        create_train_val_split(Path(args.output) / "metadata.csv")


if __name__ == "__main__":
    main()