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  1. f5_tts/eval/utils_eval.py +405 -0
f5_tts/eval/utils_eval.py ADDED
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1
+ import math
2
+ import os
3
+ import random
4
+ import string
5
+
6
+ import torch
7
+ import torch.nn.functional as F
8
+ import torchaudio
9
+ from tqdm import tqdm
10
+
11
+ from f5_tts.eval.ecapa_tdnn import ECAPA_TDNN_SMALL
12
+ from f5_tts.model.modules import MelSpec
13
+ from f5_tts.model.utils import convert_char_to_pinyin
14
+
15
+
16
+ # seedtts testset metainfo: utt, prompt_text, prompt_wav, gt_text, gt_wav
17
+ def get_seedtts_testset_metainfo(metalst):
18
+ f = open(metalst)
19
+ lines = f.readlines()
20
+ f.close()
21
+ metainfo = []
22
+ for line in lines:
23
+ if len(line.strip().split("|")) == 5:
24
+ utt, prompt_text, prompt_wav, gt_text, gt_wav = line.strip().split("|")
25
+ elif len(line.strip().split("|")) == 4:
26
+ utt, prompt_text, prompt_wav, gt_text = line.strip().split("|")
27
+ gt_wav = os.path.join(os.path.dirname(metalst), "wavs", utt + ".wav")
28
+ if not os.path.isabs(prompt_wav):
29
+ prompt_wav = os.path.join(os.path.dirname(metalst), prompt_wav)
30
+ metainfo.append((utt, prompt_text, prompt_wav, gt_text, gt_wav))
31
+ return metainfo
32
+
33
+
34
+ # librispeech test-clean metainfo: gen_utt, ref_txt, ref_wav, gen_txt, gen_wav
35
+ def get_librispeech_test_clean_metainfo(metalst, librispeech_test_clean_path):
36
+ f = open(metalst)
37
+ lines = f.readlines()
38
+ f.close()
39
+ metainfo = []
40
+ for line in lines:
41
+ ref_utt, ref_dur, ref_txt, gen_utt, gen_dur, gen_txt = line.strip().split("\t")
42
+
43
+ # ref_txt = ref_txt[0] + ref_txt[1:].lower() + '.' # if use librispeech test-clean (no-pc)
44
+ ref_spk_id, ref_chaptr_id, _ = ref_utt.split("-")
45
+ ref_wav = os.path.join(librispeech_test_clean_path, ref_spk_id, ref_chaptr_id, ref_utt + ".flac")
46
+
47
+ # gen_txt = gen_txt[0] + gen_txt[1:].lower() + '.' # if use librispeech test-clean (no-pc)
48
+ gen_spk_id, gen_chaptr_id, _ = gen_utt.split("-")
49
+ gen_wav = os.path.join(librispeech_test_clean_path, gen_spk_id, gen_chaptr_id, gen_utt + ".flac")
50
+
51
+ metainfo.append((gen_utt, ref_txt, ref_wav, " " + gen_txt, gen_wav))
52
+
53
+ return metainfo
54
+
55
+
56
+ # padded to max length mel batch
57
+ def padded_mel_batch(ref_mels):
58
+ max_mel_length = torch.LongTensor([mel.shape[-1] for mel in ref_mels]).amax()
59
+ padded_ref_mels = []
60
+ for mel in ref_mels:
61
+ padded_ref_mel = F.pad(mel, (0, max_mel_length - mel.shape[-1]), value=0)
62
+ padded_ref_mels.append(padded_ref_mel)
63
+ padded_ref_mels = torch.stack(padded_ref_mels)
64
+ padded_ref_mels = padded_ref_mels.permute(0, 2, 1)
65
+ return padded_ref_mels
66
+
67
+
68
+ # get prompts from metainfo containing: utt, prompt_text, prompt_wav, gt_text, gt_wav
69
+
70
+
71
+ def get_inference_prompt(
72
+ metainfo,
73
+ speed=1.0,
74
+ tokenizer="pinyin",
75
+ polyphone=True,
76
+ target_sample_rate=24000,
77
+ n_fft=1024,
78
+ win_length=1024,
79
+ n_mel_channels=100,
80
+ hop_length=256,
81
+ mel_spec_type="vocos",
82
+ target_rms=0.1,
83
+ use_truth_duration=False,
84
+ infer_batch_size=1,
85
+ num_buckets=200,
86
+ min_secs=3,
87
+ max_secs=40,
88
+ ):
89
+ prompts_all = []
90
+
91
+ min_tokens = min_secs * target_sample_rate // hop_length
92
+ max_tokens = max_secs * target_sample_rate // hop_length
93
+
94
+ batch_accum = [0] * num_buckets
95
+ utts, ref_rms_list, ref_mels, ref_mel_lens, total_mel_lens, final_text_list = (
96
+ [[] for _ in range(num_buckets)] for _ in range(6)
97
+ )
98
+
99
+ mel_spectrogram = MelSpec(
100
+ n_fft=n_fft,
101
+ hop_length=hop_length,
102
+ win_length=win_length,
103
+ n_mel_channels=n_mel_channels,
104
+ target_sample_rate=target_sample_rate,
105
+ mel_spec_type=mel_spec_type,
106
+ )
107
+
108
+ for utt, prompt_text, prompt_wav, gt_text, gt_wav in tqdm(metainfo, desc="Processing prompts..."):
109
+ # Audio
110
+ ref_audio, ref_sr = torchaudio.load(prompt_wav)
111
+ ref_rms = torch.sqrt(torch.mean(torch.square(ref_audio)))
112
+ if ref_rms < target_rms:
113
+ ref_audio = ref_audio * target_rms / ref_rms
114
+ assert ref_audio.shape[-1] > 5000, f"Empty prompt wav: {prompt_wav}, or torchaudio backend issue."
115
+ if ref_sr != target_sample_rate:
116
+ resampler = torchaudio.transforms.Resample(ref_sr, target_sample_rate)
117
+ ref_audio = resampler(ref_audio)
118
+
119
+ # Text
120
+ if len(prompt_text[-1].encode("utf-8")) == 1:
121
+ prompt_text = prompt_text + " "
122
+ text = [prompt_text + gt_text]
123
+ if tokenizer == "pinyin":
124
+ text_list = convert_char_to_pinyin(text, polyphone=polyphone)
125
+ else:
126
+ text_list = text
127
+
128
+ # Duration, mel frame length
129
+ ref_mel_len = ref_audio.shape[-1] // hop_length
130
+ if use_truth_duration:
131
+ gt_audio, gt_sr = torchaudio.load(gt_wav)
132
+ if gt_sr != target_sample_rate:
133
+ resampler = torchaudio.transforms.Resample(gt_sr, target_sample_rate)
134
+ gt_audio = resampler(gt_audio)
135
+ total_mel_len = ref_mel_len + int(gt_audio.shape[-1] / hop_length / speed)
136
+
137
+ # # test vocoder resynthesis
138
+ # ref_audio = gt_audio
139
+ else:
140
+ ref_text_len = len(prompt_text.encode("utf-8"))
141
+ gen_text_len = len(gt_text.encode("utf-8"))
142
+ total_mel_len = ref_mel_len + int(ref_mel_len / ref_text_len * gen_text_len / speed)
143
+
144
+ # to mel spectrogram
145
+ ref_mel = mel_spectrogram(ref_audio)
146
+ ref_mel = ref_mel.squeeze(0)
147
+
148
+ # deal with batch
149
+ assert infer_batch_size > 0, "infer_batch_size should be greater than 0."
150
+ assert (
151
+ min_tokens <= total_mel_len <= max_tokens
152
+ ), f"Audio {utt} has duration {total_mel_len*hop_length//target_sample_rate}s out of range [{min_secs}, {max_secs}]."
153
+ bucket_i = math.floor((total_mel_len - min_tokens) / (max_tokens - min_tokens + 1) * num_buckets)
154
+
155
+ utts[bucket_i].append(utt)
156
+ ref_rms_list[bucket_i].append(ref_rms)
157
+ ref_mels[bucket_i].append(ref_mel)
158
+ ref_mel_lens[bucket_i].append(ref_mel_len)
159
+ total_mel_lens[bucket_i].append(total_mel_len)
160
+ final_text_list[bucket_i].extend(text_list)
161
+
162
+ batch_accum[bucket_i] += total_mel_len
163
+
164
+ if batch_accum[bucket_i] >= infer_batch_size:
165
+ # print(f"\n{len(ref_mels[bucket_i][0][0])}\n{ref_mel_lens[bucket_i]}\n{total_mel_lens[bucket_i]}")
166
+ prompts_all.append(
167
+ (
168
+ utts[bucket_i],
169
+ ref_rms_list[bucket_i],
170
+ padded_mel_batch(ref_mels[bucket_i]),
171
+ ref_mel_lens[bucket_i],
172
+ total_mel_lens[bucket_i],
173
+ final_text_list[bucket_i],
174
+ )
175
+ )
176
+ batch_accum[bucket_i] = 0
177
+ (
178
+ utts[bucket_i],
179
+ ref_rms_list[bucket_i],
180
+ ref_mels[bucket_i],
181
+ ref_mel_lens[bucket_i],
182
+ total_mel_lens[bucket_i],
183
+ final_text_list[bucket_i],
184
+ ) = [], [], [], [], [], []
185
+
186
+ # add residual
187
+ for bucket_i, bucket_frames in enumerate(batch_accum):
188
+ if bucket_frames > 0:
189
+ prompts_all.append(
190
+ (
191
+ utts[bucket_i],
192
+ ref_rms_list[bucket_i],
193
+ padded_mel_batch(ref_mels[bucket_i]),
194
+ ref_mel_lens[bucket_i],
195
+ total_mel_lens[bucket_i],
196
+ final_text_list[bucket_i],
197
+ )
198
+ )
199
+ # not only leave easy work for last workers
200
+ random.seed(666)
201
+ random.shuffle(prompts_all)
202
+
203
+ return prompts_all
204
+
205
+
206
+ # get wav_res_ref_text of seed-tts test metalst
207
+ # https://github.com/BytedanceSpeech/seed-tts-eval
208
+
209
+
210
+ def get_seed_tts_test(metalst, gen_wav_dir, gpus):
211
+ f = open(metalst)
212
+ lines = f.readlines()
213
+ f.close()
214
+
215
+ test_set_ = []
216
+ for line in tqdm(lines):
217
+ if len(line.strip().split("|")) == 5:
218
+ utt, prompt_text, prompt_wav, gt_text, gt_wav = line.strip().split("|")
219
+ elif len(line.strip().split("|")) == 4:
220
+ utt, prompt_text, prompt_wav, gt_text = line.strip().split("|")
221
+
222
+ if not os.path.exists(os.path.join(gen_wav_dir, utt + ".wav")):
223
+ continue
224
+ gen_wav = os.path.join(gen_wav_dir, utt + ".wav")
225
+ if not os.path.isabs(prompt_wav):
226
+ prompt_wav = os.path.join(os.path.dirname(metalst), prompt_wav)
227
+
228
+ test_set_.append((gen_wav, prompt_wav, gt_text))
229
+
230
+ num_jobs = len(gpus)
231
+ if num_jobs == 1:
232
+ return [(gpus[0], test_set_)]
233
+
234
+ wav_per_job = len(test_set_) // num_jobs + 1
235
+ test_set = []
236
+ for i in range(num_jobs):
237
+ test_set.append((gpus[i], test_set_[i * wav_per_job : (i + 1) * wav_per_job]))
238
+
239
+ return test_set
240
+
241
+
242
+ # get librispeech test-clean cross sentence test
243
+
244
+
245
+ def get_librispeech_test(metalst, gen_wav_dir, gpus, librispeech_test_clean_path, eval_ground_truth=False):
246
+ f = open(metalst)
247
+ lines = f.readlines()
248
+ f.close()
249
+
250
+ test_set_ = []
251
+ for line in tqdm(lines):
252
+ ref_utt, ref_dur, ref_txt, gen_utt, gen_dur, gen_txt = line.strip().split("\t")
253
+
254
+ if eval_ground_truth:
255
+ gen_spk_id, gen_chaptr_id, _ = gen_utt.split("-")
256
+ gen_wav = os.path.join(librispeech_test_clean_path, gen_spk_id, gen_chaptr_id, gen_utt + ".flac")
257
+ else:
258
+ if not os.path.exists(os.path.join(gen_wav_dir, gen_utt + ".wav")):
259
+ raise FileNotFoundError(f"Generated wav not found: {gen_utt}")
260
+ gen_wav = os.path.join(gen_wav_dir, gen_utt + ".wav")
261
+
262
+ ref_spk_id, ref_chaptr_id, _ = ref_utt.split("-")
263
+ ref_wav = os.path.join(librispeech_test_clean_path, ref_spk_id, ref_chaptr_id, ref_utt + ".flac")
264
+
265
+ test_set_.append((gen_wav, ref_wav, gen_txt))
266
+
267
+ num_jobs = len(gpus)
268
+ if num_jobs == 1:
269
+ return [(gpus[0], test_set_)]
270
+
271
+ wav_per_job = len(test_set_) // num_jobs + 1
272
+ test_set = []
273
+ for i in range(num_jobs):
274
+ test_set.append((gpus[i], test_set_[i * wav_per_job : (i + 1) * wav_per_job]))
275
+
276
+ return test_set
277
+
278
+
279
+ # load asr model
280
+
281
+
282
+ def load_asr_model(lang, ckpt_dir=""):
283
+ if lang == "zh":
284
+ from funasr import AutoModel
285
+
286
+ model = AutoModel(
287
+ model=os.path.join(ckpt_dir, "paraformer-zh"),
288
+ # vad_model = os.path.join(ckpt_dir, "fsmn-vad"),
289
+ # punc_model = os.path.join(ckpt_dir, "ct-punc"),
290
+ # spk_model = os.path.join(ckpt_dir, "cam++"),
291
+ disable_update=True,
292
+ ) # following seed-tts setting
293
+ elif lang == "en":
294
+ from faster_whisper import WhisperModel
295
+
296
+ model_size = "large-v3" if ckpt_dir == "" else ckpt_dir
297
+ model = WhisperModel(model_size, device="cuda", compute_type="float16")
298
+ return model
299
+
300
+
301
+ # WER Evaluation, the way Seed-TTS does
302
+
303
+
304
+ def run_asr_wer(args):
305
+ rank, lang, test_set, ckpt_dir = args
306
+
307
+ if lang == "zh":
308
+ import zhconv
309
+
310
+ torch.cuda.set_device(rank)
311
+ elif lang == "en":
312
+ os.environ["CUDA_VISIBLE_DEVICES"] = str(rank)
313
+ else:
314
+ raise NotImplementedError(
315
+ "lang support only 'zh' (funasr paraformer-zh), 'en' (faster-whisper-large-v3), for now."
316
+ )
317
+
318
+ asr_model = load_asr_model(lang, ckpt_dir=ckpt_dir)
319
+
320
+ from zhon.hanzi import punctuation
321
+
322
+ punctuation_all = punctuation + string.punctuation
323
+ wers = []
324
+
325
+ from jiwer import compute_measures
326
+
327
+ for gen_wav, prompt_wav, truth in tqdm(test_set):
328
+ if lang == "zh":
329
+ res = asr_model.generate(input=gen_wav, batch_size_s=300, disable_pbar=True)
330
+ hypo = res[0]["text"]
331
+ hypo = zhconv.convert(hypo, "zh-cn")
332
+ elif lang == "en":
333
+ segments, _ = asr_model.transcribe(gen_wav, beam_size=5, language="en")
334
+ hypo = ""
335
+ for segment in segments:
336
+ hypo = hypo + " " + segment.text
337
+
338
+ # raw_truth = truth
339
+ # raw_hypo = hypo
340
+
341
+ for x in punctuation_all:
342
+ truth = truth.replace(x, "")
343
+ hypo = hypo.replace(x, "")
344
+
345
+ truth = truth.replace(" ", " ")
346
+ hypo = hypo.replace(" ", " ")
347
+
348
+ if lang == "zh":
349
+ truth = " ".join([x for x in truth])
350
+ hypo = " ".join([x for x in hypo])
351
+ elif lang == "en":
352
+ truth = truth.lower()
353
+ hypo = hypo.lower()
354
+
355
+ measures = compute_measures(truth, hypo)
356
+ wer = measures["wer"]
357
+
358
+ # ref_list = truth.split(" ")
359
+ # subs = measures["substitutions"] / len(ref_list)
360
+ # dele = measures["deletions"] / len(ref_list)
361
+ # inse = measures["insertions"] / len(ref_list)
362
+
363
+ wers.append(wer)
364
+
365
+ return wers
366
+
367
+
368
+ # SIM Evaluation
369
+
370
+
371
+ def run_sim(args):
372
+ rank, test_set, ckpt_dir = args
373
+ device = f"cuda:{rank}"
374
+
375
+ model = ECAPA_TDNN_SMALL(feat_dim=1024, feat_type="wavlm_large", config_path=None)
376
+ state_dict = torch.load(ckpt_dir, weights_only=True, map_location=lambda storage, loc: storage)
377
+ model.load_state_dict(state_dict["model"], strict=False)
378
+
379
+ use_gpu = True if torch.cuda.is_available() else False
380
+ if use_gpu:
381
+ model = model.cuda(device)
382
+ model.eval()
383
+
384
+ sim_list = []
385
+ for wav1, wav2, truth in tqdm(test_set):
386
+ wav1, sr1 = torchaudio.load(wav1)
387
+ wav2, sr2 = torchaudio.load(wav2)
388
+
389
+ resample1 = torchaudio.transforms.Resample(orig_freq=sr1, new_freq=16000)
390
+ resample2 = torchaudio.transforms.Resample(orig_freq=sr2, new_freq=16000)
391
+ wav1 = resample1(wav1)
392
+ wav2 = resample2(wav2)
393
+
394
+ if use_gpu:
395
+ wav1 = wav1.cuda(device)
396
+ wav2 = wav2.cuda(device)
397
+ with torch.no_grad():
398
+ emb1 = model(wav1)
399
+ emb2 = model(wav2)
400
+
401
+ sim = F.cosine_similarity(emb1, emb2)[0].item()
402
+ # print(f"VSim score between two audios: {sim:.4f} (-1.0, 1.0).")
403
+ sim_list.append(sim)
404
+
405
+ return sim_list