返回 F5-TTS
utils_eval.py
根目录 / src / f5_tts / eval / utils_eval.py
1 import math
2 import os
3 import random
4 import string
5 from pathlib import Path
6
7 import torch
8 import torch.nn.functional as F
9 import torchaudio
10 from tqdm import tqdm
11
12 from f5_tts.eval.ecapa_tdnn import ECAPA_TDNN_SMALL
13 from f5_tts.model.modules import MelSpec
14 from f5_tts.model.utils import convert_char_to_pinyin
15
16
17 # seedtts testset metainfo: utt, prompt_text, prompt_wav, gt_text, gt_wav
18 def get_seedtts_testset_metainfo(metalst):
19 f = open(metalst)
20 lines = f.readlines()
21 f.close()
22 metainfo = []
23 for line in lines:
24 if len(line.strip().split("|")) == 5:
25 utt, prompt_text, prompt_wav, gt_text, gt_wav = line.strip().split("|")
26 elif len(line.strip().split("|")) == 4:
27 utt, prompt_text, prompt_wav, gt_text = line.strip().split("|")
28 gt_wav = os.path.join(os.path.dirname(metalst), "wavs", utt + ".wav")
29 if not os.path.isabs(prompt_wav):
30 prompt_wav = os.path.join(os.path.dirname(metalst), prompt_wav)
31 metainfo.append((utt, prompt_text, prompt_wav, gt_text, gt_wav))
32 return metainfo
33
34
35 # librispeech test-clean metainfo: gen_utt, ref_txt, ref_wav, gen_txt, gen_wav
36 def get_librispeech_test_clean_metainfo(metalst, librispeech_test_clean_path):
37 f = open(metalst)
38 lines = f.readlines()
39 f.close()
40 metainfo = []
41 for line in lines:
42 ref_utt, ref_dur, ref_txt, gen_utt, gen_dur, gen_txt = line.strip().split("\t")
43
44 # ref_txt = ref_txt[0] + ref_txt[1:].lower() + '.' # if use librispeech test-clean (no-pc)
45 ref_spk_id, ref_chaptr_id, _ = ref_utt.split("-")
46 ref_wav = os.path.join(librispeech_test_clean_path, ref_spk_id, ref_chaptr_id, ref_utt + ".flac")
47
48 # gen_txt = gen_txt[0] + gen_txt[1:].lower() + '.' # if use librispeech test-clean (no-pc)
49 gen_spk_id, gen_chaptr_id, _ = gen_utt.split("-")
50 gen_wav = os.path.join(librispeech_test_clean_path, gen_spk_id, gen_chaptr_id, gen_utt + ".flac")
51
52 metainfo.append((gen_utt, ref_txt, ref_wav, " " + gen_txt, gen_wav))
53
54 return metainfo
55
56
57 # padded to max length mel batch
58 def padded_mel_batch(ref_mels):
59 max_mel_length = torch.LongTensor([mel.shape[-1] for mel in ref_mels]).amax()
60 padded_ref_mels = []
61 for mel in ref_mels:
62 padded_ref_mel = F.pad(mel, (0, max_mel_length - mel.shape[-1]), value=0)
63 padded_ref_mels.append(padded_ref_mel)
64 padded_ref_mels = torch.stack(padded_ref_mels)
65 padded_ref_mels = padded_ref_mels.permute(0, 2, 1)
66 return padded_ref_mels
67
68
69 # get prompts from metainfo containing: utt, prompt_text, prompt_wav, gt_text, gt_wav
70
71
72 def get_inference_prompt(
73 metainfo,
74 speed=1.0,
75 tokenizer="pinyin",
76 polyphone=True,
77 target_sample_rate=24000,
78 n_fft=1024,
79 win_length=1024,
80 n_mel_channels=100,
81 hop_length=256,
82 mel_spec_type="vocos",
83 target_rms=0.1,
84 use_truth_duration=False,
85 infer_batch_size=1,
86 num_buckets=200,
87 min_secs=3,
88 max_secs=40,
89 ):
90 prompts_all = []
91
92 min_tokens = min_secs * target_sample_rate // hop_length
93 max_tokens = max_secs * target_sample_rate // hop_length
94
95 batch_accum = [0] * num_buckets
96 utts, ref_rms_list, ref_mels, ref_mel_lens, total_mel_lens, final_text_list = (
97 [[] for _ in range(num_buckets)] for _ in range(6)
98 )
99
100 mel_spectrogram = MelSpec(
101 n_fft=n_fft,
102 hop_length=hop_length,
103 win_length=win_length,
104 n_mel_channels=n_mel_channels,
105 target_sample_rate=target_sample_rate,
106 mel_spec_type=mel_spec_type,
107 )
108
109 for utt, prompt_text, prompt_wav, gt_text, gt_wav in tqdm(metainfo, desc="Processing prompts..."):
110 # Audio
111 ref_audio, ref_sr = torchaudio.load(prompt_wav)
112 ref_rms = torch.sqrt(torch.mean(torch.square(ref_audio)))
113 if ref_rms < target_rms:
114 ref_audio = ref_audio * target_rms / ref_rms
115 assert ref_audio.shape[-1] > 5000, f"Empty prompt wav: {prompt_wav}, or torchaudio backend issue."
116 if ref_sr != target_sample_rate:
117 resampler = torchaudio.transforms.Resample(ref_sr, target_sample_rate)
118 ref_audio = resampler(ref_audio)
119
120 # Text
121 if len(prompt_text[-1].encode("utf-8")) == 1:
122 prompt_text = prompt_text + " "
123 text = [prompt_text + gt_text]
124 if tokenizer == "pinyin":
125 text_list = convert_char_to_pinyin(text, polyphone=polyphone)
126 else:
127 text_list = text
128
129 # to mel spectrogram
130 ref_mel = mel_spectrogram(ref_audio)
131 ref_mel = ref_mel.squeeze(0)
132
133 # Duration, mel frame length
134 ref_mel_len = ref_mel.shape[-1]
135
136 if use_truth_duration:
137 gt_audio, gt_sr = torchaudio.load(gt_wav)
138 if gt_sr != target_sample_rate:
139 resampler = torchaudio.transforms.Resample(gt_sr, target_sample_rate)
140 gt_audio = resampler(gt_audio)
141 total_mel_len = ref_mel_len + int(gt_audio.shape[-1] / hop_length / speed)
142
143 # # test vocoder resynthesis
144 # ref_audio = gt_audio
145 else:
146 ref_text_len = len(prompt_text.encode("utf-8"))
147 gen_text_len = len(gt_text.encode("utf-8"))
148 total_mel_len = ref_mel_len + int(ref_mel_len / ref_text_len * gen_text_len / speed)
149
150 # deal with batch
151 assert infer_batch_size > 0, "infer_batch_size should be greater than 0."
152 assert min_tokens <= total_mel_len <= max_tokens, (
153 f"Audio {utt} has duration {total_mel_len * hop_length // target_sample_rate}s out of range [{min_secs}, {max_secs}]."
154 )
155 bucket_i = math.floor((total_mel_len - min_tokens) / (max_tokens - min_tokens + 1) * num_buckets)
156
157 utts[bucket_i].append(utt)
158 ref_rms_list[bucket_i].append(ref_rms)
159 ref_mels[bucket_i].append(ref_mel)
160 ref_mel_lens[bucket_i].append(ref_mel_len)
161 total_mel_lens[bucket_i].append(total_mel_len)
162 final_text_list[bucket_i].extend(text_list)
163
164 batch_accum[bucket_i] += total_mel_len
165
166 if batch_accum[bucket_i] >= infer_batch_size:
167 # print(f"\n{len(ref_mels[bucket_i][0][0])}\n{ref_mel_lens[bucket_i]}\n{total_mel_lens[bucket_i]}")
168 prompts_all.append(
169 (
170 utts[bucket_i],
171 ref_rms_list[bucket_i],
172 padded_mel_batch(ref_mels[bucket_i]),
173 ref_mel_lens[bucket_i],
174 total_mel_lens[bucket_i],
175 final_text_list[bucket_i],
176 )
177 )
178 batch_accum[bucket_i] = 0
179 (
180 utts[bucket_i],
181 ref_rms_list[bucket_i],
182 ref_mels[bucket_i],
183 ref_mel_lens[bucket_i],
184 total_mel_lens[bucket_i],
185 final_text_list[bucket_i],
186 ) = [], [], [], [], [], []
187
188 # add residual
189 for bucket_i, bucket_frames in enumerate(batch_accum):
190 if bucket_frames > 0:
191 prompts_all.append(
192 (
193 utts[bucket_i],
194 ref_rms_list[bucket_i],
195 padded_mel_batch(ref_mels[bucket_i]),
196 ref_mel_lens[bucket_i],
197 total_mel_lens[bucket_i],
198 final_text_list[bucket_i],
199 )
200 )
201 # not only leave easy work for last workers
202 random.seed(666)
203 random.shuffle(prompts_all)
204
205 return prompts_all
206
207
208 # get wav_res_ref_text of seed-tts test metalst
209 # https://github.com/BytedanceSpeech/seed-tts-eval
210
211
212 def get_seed_tts_test(metalst, gen_wav_dir, gpus):
213 f = open(metalst)
214 lines = f.readlines()
215 f.close()
216
217 test_set_ = []
218 for line in tqdm(lines):
219 if len(line.strip().split("|")) == 5:
220 utt, prompt_text, prompt_wav, gt_text, gt_wav = line.strip().split("|")
221 elif len(line.strip().split("|")) == 4:
222 utt, prompt_text, prompt_wav, gt_text = line.strip().split("|")
223
224 if not os.path.exists(os.path.join(gen_wav_dir, utt + ".wav")):
225 continue
226 gen_wav = os.path.join(gen_wav_dir, utt + ".wav")
227 if not os.path.isabs(prompt_wav):
228 prompt_wav = os.path.join(os.path.dirname(metalst), prompt_wav)
229
230 test_set_.append((gen_wav, prompt_wav, gt_text))
231
232 num_jobs = len(gpus)
233 if num_jobs == 1:
234 return [(gpus[0], test_set_)]
235
236 wav_per_job = len(test_set_) // num_jobs + 1
237 test_set = []
238 for i in range(num_jobs):
239 test_set.append((gpus[i], test_set_[i * wav_per_job : (i + 1) * wav_per_job]))
240
241 return test_set
242
243
244 # get librispeech test-clean cross sentence test
245
246
247 def get_librispeech_test(metalst, gen_wav_dir, gpus, librispeech_test_clean_path, eval_ground_truth=False):
248 f = open(metalst)
249 lines = f.readlines()
250 f.close()
251
252 test_set_ = []
253 for line in tqdm(lines):
254 ref_utt, ref_dur, ref_txt, gen_utt, gen_dur, gen_txt = line.strip().split("\t")
255
256 if eval_ground_truth:
257 gen_spk_id, gen_chaptr_id, _ = gen_utt.split("-")
258 gen_wav = os.path.join(librispeech_test_clean_path, gen_spk_id, gen_chaptr_id, gen_utt + ".flac")
259 else:
260 if not os.path.exists(os.path.join(gen_wav_dir, gen_utt + ".wav")):
261 raise FileNotFoundError(f"Generated wav not found: {gen_utt}")
262 gen_wav = os.path.join(gen_wav_dir, gen_utt + ".wav")
263
264 ref_spk_id, ref_chaptr_id, _ = ref_utt.split("-")
265 ref_wav = os.path.join(librispeech_test_clean_path, ref_spk_id, ref_chaptr_id, ref_utt + ".flac")
266
267 test_set_.append((gen_wav, ref_wav, gen_txt))
268
269 num_jobs = len(gpus)
270 if num_jobs == 1:
271 return [(gpus[0], test_set_)]
272
273 wav_per_job = len(test_set_) // num_jobs + 1
274 test_set = []
275 for i in range(num_jobs):
276 test_set.append((gpus[i], test_set_[i * wav_per_job : (i + 1) * wav_per_job]))
277
278 return test_set
279
280
281 # load asr model
282
283
284 def load_asr_model(lang, ckpt_dir=""):
285 if lang == "zh":
286 from funasr import AutoModel
287
288 model = AutoModel(
289 model=os.path.join(ckpt_dir, "paraformer-zh"),
290 # vad_model = os.path.join(ckpt_dir, "fsmn-vad"),
291 # punc_model = os.path.join(ckpt_dir, "ct-punc"),
292 # spk_model = os.path.join(ckpt_dir, "cam++"),
293 disable_update=True,
294 ) # following seed-tts setting
295 elif lang == "en":
296 from faster_whisper import WhisperModel
297
298 model_size = "large-v3" if ckpt_dir == "" else ckpt_dir
299 model = WhisperModel(model_size, device="cuda", compute_type="float16")
300 return model
301
302
303 # WER Evaluation, the way Seed-TTS does
304
305
306 def run_asr_wer(args):
307 rank, lang, test_set, ckpt_dir = args
308
309 if lang == "zh":
310 import zhconv
311
312 torch.cuda.set_device(rank)
313 elif lang == "en":
314 os.environ["CUDA_VISIBLE_DEVICES"] = str(rank)
315 else:
316 raise NotImplementedError(
317 "lang support only 'zh' (funasr paraformer-zh), 'en' (faster-whisper-large-v3), for now."
318 )
319
320 asr_model = load_asr_model(lang, ckpt_dir=ckpt_dir)
321
322 from zhon.hanzi import punctuation
323
324 punctuation_all = punctuation + string.punctuation
325 wer_results = []
326
327 from jiwer import process_words
328
329 for gen_wav, prompt_wav, truth in tqdm(test_set):
330 if lang == "zh":
331 res = asr_model.generate(input=gen_wav, batch_size_s=300, disable_pbar=True)
332 hypo = res[0]["text"]
333 hypo = zhconv.convert(hypo, "zh-cn")
334 elif lang == "en":
335 segments, _ = asr_model.transcribe(gen_wav, beam_size=5, language="en")
336 hypo = ""
337 for segment in segments:
338 hypo = hypo + " " + segment.text
339
340 raw_truth = truth
341 raw_hypo = hypo
342
343 for x in punctuation_all:
344 truth = truth.replace(x, "")
345 hypo = hypo.replace(x, "")
346
347 truth = truth.replace(" ", " ")
348 hypo = hypo.replace(" ", " ")
349
350 if lang == "zh":
351 truth = " ".join([x for x in truth])
352 hypo = " ".join([x for x in hypo])
353 elif lang == "en":
354 truth = truth.lower()
355 hypo = hypo.lower()
356
357 measures = process_words(truth, hypo)
358 wer = measures.wer
359
360 # ref_list = truth.split(" ")
361 # subs = measures.substitutions / len(ref_list)
362 # dele = measures.deletions / len(ref_list)
363 # inse = measures.insertions / len(ref_list)
364
365 wer_results.append(
366 {
367 "wav": Path(gen_wav).stem,
368 "truth": raw_truth,
369 "hypo": raw_hypo,
370 "wer": wer,
371 }
372 )
373
374 return wer_results
375
376
377 # SIM Evaluation
378
379
380 def run_sim(args):
381 rank, test_set, ckpt_dir = args
382 device = f"cuda:{rank}"
383
384 model = ECAPA_TDNN_SMALL(feat_dim=1024, feat_type="wavlm_large", config_path=None)
385 state_dict = torch.load(ckpt_dir, weights_only=True, map_location=lambda storage, loc: storage)
386 model.load_state_dict(state_dict["model"], strict=False)
387
388 use_gpu = True if torch.cuda.is_available() else False
389 if use_gpu:
390 model = model.cuda(device)
391 model.eval()
392
393 sim_results = []
394 for gen_wav, prompt_wav, truth in tqdm(test_set):
395 wav1, sr1 = torchaudio.load(gen_wav)
396 wav2, sr2 = torchaudio.load(prompt_wav)
397
398 if use_gpu:
399 wav1 = wav1.cuda(device)
400 wav2 = wav2.cuda(device)
401
402 if sr1 != 16000:
403 resample1 = torchaudio.transforms.Resample(orig_freq=sr1, new_freq=16000)
404 if use_gpu:
405 resample1 = resample1.cuda(device)
406 wav1 = resample1(wav1)
407 if sr2 != 16000:
408 resample2 = torchaudio.transforms.Resample(orig_freq=sr2, new_freq=16000)
409 if use_gpu:
410 resample2 = resample2.cuda(device)
411 wav2 = resample2(wav2)
412
413 with torch.no_grad():
414 emb1 = model(wav1)
415 emb2 = model(wav2)
416
417 sim = F.cosine_similarity(emb1, emb2)[0].item()
418 # print(f"VSim score between two audios: {sim:.4f} (-1.0, 1.0).")
419 sim_results.append(
420 {
421 "wav": Path(gen_wav).stem,
422 "sim": sim,
423 }
424 )
425
426 return sim_results
427
427 lines PYTHON