minor updates

.github/scripts/ljspeech/TTS/run-matcha.sh

@@ -56,7 +56,7 @@ function infer() {
 
   curl -SL -O https://github.com/csukuangfj/models/raw/refs/heads/master/hifigan/generator_v1
 
-  ./matcha/infer.py \
+  ./matcha/synth.py \
     --epoch 1 \
     --exp-dir ./matcha/exp \
     --tokens data/tokens.txt \

egs/ljspeech/TTS/matcha/infer.py

@@ -9,14 +9,16 @@ from pathlib import Path
 
 import soundfile as sf
 import torch
-from matcha.hifigan.config import v1, v2, v3
-from matcha.hifigan.denoiser import Denoiser
-from matcha.hifigan.models import Generator as HiFiGAN
+import torch.nn as nn
+from hifigan.config import v1, v2, v3
+from hifigan.denoiser import Denoiser
+from hifigan.models import Generator as HiFiGAN
 from tokenizer import Tokenizer
 from train import get_model, get_params
+from tts_datamodule import LJSpeechTtsDataModule
 
 from icefall.checkpoint import load_checkpoint
-from icefall.utils import AttributeDict
+from icefall.utils import AttributeDict, setup_logger
 
 
 def get_parser():
@@ -63,24 +65,10 @@ def get_parser():
         help="""Path to vocabulary.""",
     )
 
-    parser.add_argument(
-        "--input-text",
-        type=str,
-        required=True,
-        help="The text to generate speech for",
-    )
-
-    parser.add_argument(
-        "--output-wav",
-        type=str,
-        required=True,
-        help="The filename of the wave to save the generated speech",
-    )
-
     return parser
 
 
-def load_vocoder(checkpoint_path):
+def load_vocoder(checkpoint_path: Path) -> nn.Module:
     checkpoint_path = str(checkpoint_path)
     if checkpoint_path.endswith("v1"):
         h = AttributeDict(v1)
@@ -100,13 +88,15 @@ def load_vocoder(checkpoint_path):
     return hifigan
 
 
-def to_waveform(mel, vocoder, denoiser):
+def to_waveform(
+    mel: torch.Tensor, vocoder: nn.Module, denoiser: nn.Module
+) -> torch.Tensor:
     audio = vocoder(mel).clamp(-1, 1)
     audio = denoiser(audio.squeeze(0), strength=0.00025).cpu().squeeze()
     return audio.cpu().squeeze()
 
 
-def process_text(text: str, tokenizer):
+def process_text(text: str, tokenizer: Tokenizer) -> dict:
     x = tokenizer.texts_to_token_ids([text], add_sos=True, add_eos=True)
     x = torch.tensor(x, dtype=torch.long)
     x_lengths = torch.tensor([x.shape[-1]], dtype=torch.long, device="cpu")
@@ -114,8 +104,14 @@ def process_text(text: str, tokenizer):
 
 
 def synthesise(
-    model, tokenizer, n_timesteps, text, length_scale, temperature, spks=None
-):
+    model: nn.Module,
+    tokenizer: Tokenizer,
+    n_timesteps: int,
+    text: str,
+    length_scale: float,
+    temperature: float,
+    spks=None,
+) -> dict:
     text_processed = process_text(text, tokenizer)
     start_t = dt.datetime.now()
     output = model.synthesise(
@@ -131,14 +127,102 @@ def synthesise(
     return output
 
 
+def infer_dataset(
+    dl: torch.utils.data.DataLoader,
+    params: AttributeDict,
+    model: nn.Module,
+    vocoder: nn.Module,
+    denoiser: nn.Module,
+    tokenizer: Tokenizer,
+) -> None:
+    """Decode dataset.
+    The ground-truth and generated audio pairs will be saved to `params.save_wav_dir`.
+
+    Args:
+      dl:
+        PyTorch's dataloader containing the dataset to decode.
+      params:
+        It is returned by :func:`get_params`.
+      model:
+        The neural model.
+      tokenizer:
+        Used to convert text to phonemes.
+    """
+
+    device = next(model.parameters()).device
+    num_cuts = 0
+    log_interval = 5
+
+    try:
+        num_batches = len(dl)
+    except TypeError:
+        num_batches = "?"
+
+    for batch_idx, batch in enumerate(dl):
+        batch_size = len(batch["tokens"])
+
+        texts = batch["supervisions"]["text"]
+
+        audio = batch["audio"]
+        audio_lens = batch["audio_lens"].tolist()
+        cut_ids = [cut.id for cut in batch["cut"]]
+
+        for i in range(batch_size):
+            output = synthesise(
+                model=model,
+                tokenizer=tokenizer,
+                n_timesteps=params.n_timesteps,
+                text=texts[i],
+                length_scale=params.length_scale,
+                temperature=params.temperature,
+            )
+            output["waveform"] = to_waveform(output["mel"], vocoder, denoiser)
+
+            sf.write(
+                file=params.save_wave_dir / f"{cut_ids[i]}_pred.wav", 
+                data=output["waveform"], 
+                samplerate=params.sampling_rate, 
+                subtype="PCM_16"
+            )
+            sf.write(
+                file=params.save_wave_dir / f"{cut_ids[i]}_gt.wav", 
+                data=audio[i].numpy(), 
+                samplerate=params.sampling_rate, 
+                subtype="PCM_16"
+            )
+
+        num_cuts += batch_size
+
+        if batch_idx % log_interval == 0:
+            batch_str = f"{batch_idx}/{num_batches}"
+
+            logging.info(f"batch {batch_str}, cuts processed until now is {num_cuts}")
+
+
 @torch.inference_mode()
 def main():
     parser = get_parser()
+    LJSpeechTtsDataModule.add_arguments(parser)
     args = parser.parse_args()
-    params = get_params()
+    args.exp_dir = Path(args.exp_dir)
 
+    params = get_params()
     params.update(vars(args))
 
+    params.suffix = f"epoch-{params.epoch}"
+
+    params.res_dir = params.exp_dir / "infer" / params.suffix
+    params.save_wav_dir = params.res_dir / "wav"
+    params.save_wav_dir.mkdir(parents=True, exist_ok=True)
+
+    setup_logger(f"{params.res_dir}/log-infer-{params.suffix}")
+    logging.info("Infer started")
+
+    device = torch.device("cpu")
+    if torch.cuda.is_available():
+        device = torch.device("cuda", 0)
+    logging.info(f"Device: {device}")
+
     tokenizer = Tokenizer(params.tokens)
     params.blank_id = tokenizer.pad_id
     params.vocab_size = tokenizer.vocab_size
@@ -151,49 +235,49 @@ def main():
 
         params.model_args.data_statistics.mel_mean = stats["fbank_mean"]
         params.model_args.data_statistics.mel_std = stats["fbank_std"]
+
+    # Number of ODE Solver steps
+    params.n_timesteps = 2
+
+    # Changes to the speaking rate
+    params.length_scale = 1.0
+
+    # Sampling temperature
+    params.temperature = 0.667
     logging.info(params)
 
     logging.info("About to create model")
     model = get_model(params)
 
-    if not Path(f"{params.exp_dir}/epoch-{params.epoch}.pt").is_file():
-        raise ValueError("{params.exp_dir}/epoch-{params.epoch}.pt does not exist")
-
     load_checkpoint(f"{params.exp_dir}/epoch-{params.epoch}.pt", model)
+    model.to(device)
     model.eval()
 
+    # we need cut ids to organize tts results.
+    args.return_cuts = True
+    ljspeech = LJSpeechTtsDataModule(args)
+
+    test_cuts = ljspeech.test_cuts()
+    test_dl = ljspeech.test_dataloaders(test_cuts)
+
     if not Path(params.vocoder).is_file():
         raise ValueError(f"{params.vocoder} does not exist")
 
     vocoder = load_vocoder(params.vocoder)
+    vocoder = vocoder.to(device)
+
     denoiser = Denoiser(vocoder, mode="zeros")
+    denoiser = denoiser.to(device)
 
-    # Number of ODE Solver steps
-    n_timesteps = 2
-
-    # Changes to the speaking rate
-    length_scale = 1.0
-
-    # Sampling temperature
-    temperature = 0.667
-
-    output = synthesise(
+    infer_dataset(
+        dl=test_dl,
+        params=params,
         model=model,
+        vocoder=vocoder,
+        denoiser=denoiser,
         tokenizer=tokenizer,
-        n_timesteps=n_timesteps,
-        text=params.input_text,
-        length_scale=length_scale,
-        temperature=temperature,
     )
-    output["waveform"] = to_waveform(output["mel"], vocoder, denoiser)
-
-    sf.write(params.output_wav, output["waveform"], 22050, "PCM_16")
 
 
 if __name__ == "__main__":
-    formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
-
-    logging.basicConfig(format=formatter, level=logging.INFO)
-    torch.set_num_threads(1)
-    torch.set_num_interop_threads(1)
     main()

egs/ljspeech/TTS/matcha/synth.py

@@ -0,0 +1,193 @@
+#!/usr/bin/env python3
+# Copyright         2024  Xiaomi Corp.        (authors: Fangjun Kuang)
+
+import argparse
+import datetime as dt
+import json
+import logging
+from pathlib import Path
+
+import soundfile as sf
+import torch
+from matcha.hifigan.config import v1, v2, v3
+from matcha.hifigan.denoiser import Denoiser
+from matcha.hifigan.models import Generator as HiFiGAN
+from tokenizer import Tokenizer
+from train import get_model, get_params
+
+from icefall.checkpoint import load_checkpoint
+from icefall.utils import AttributeDict, setup_logger
+
+
+def get_parser():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--epoch",
+        type=int,
+        default=4000,
+        help="""It specifies the checkpoint to use for decoding.
+        Note: Epoch counts from 1.
+        """,
+    )
+
+    parser.add_argument(
+        "--exp-dir",
+        type=Path,
+        default="matcha/exp-new-3",
+        help="""The experiment dir.
+        It specifies the directory where all training related
+        files, e.g., checkpoints, log, etc, are saved
+        """,
+    )
+
+    parser.add_argument(
+        "--vocoder",
+        type=Path,
+        default="./generator_v1",
+        help="Path to the vocoder",
+    )
+
+    parser.add_argument(
+        "--tokens",
+        type=Path,
+        default="data/tokens.txt",
+    )
+
+    parser.add_argument(
+        "--cmvn",
+        type=str,
+        default="data/fbank/cmvn.json",
+        help="""Path to vocabulary.""",
+    )
+
+    parser.add_argument(
+        "--input-text",
+        type=str,
+        required=True,
+        help="The text to generate speech for",
+    )
+
+    parser.add_argument(
+        "--output-wav",
+        type=str,
+        required=True,
+        help="The filename of the wave to save the generated speech",
+    )
+
+    return parser
+
+
+def load_vocoder(checkpoint_path):
+    checkpoint_path = str(checkpoint_path)
+    if checkpoint_path.endswith("v1"):
+        h = AttributeDict(v1)
+    elif checkpoint_path.endswith("v2"):
+        h = AttributeDict(v2)
+    elif checkpoint_path.endswith("v3"):
+        h = AttributeDict(v3)
+    else:
+        raise ValueError(f"supports only v1, v2, and v3, given {checkpoint_path}")
+
+    hifigan = HiFiGAN(h).to("cpu")
+    hifigan.load_state_dict(
+        torch.load(checkpoint_path, map_location="cpu")["generator"]
+    )
+    _ = hifigan.eval()
+    hifigan.remove_weight_norm()
+    return hifigan
+
+
+def to_waveform(mel, vocoder, denoiser):
+    audio = vocoder(mel).clamp(-1, 1)
+    audio = denoiser(audio.squeeze(0), strength=0.00025).cpu().squeeze()
+    return audio.cpu().squeeze()
+
+
+def process_text(text: str, tokenizer):
+    x = tokenizer.texts_to_token_ids([text], add_sos=True, add_eos=True)
+    x = torch.tensor(x, dtype=torch.long)
+    x_lengths = torch.tensor([x.shape[-1]], dtype=torch.long, device="cpu")
+    return {"x_orig": text, "x": x, "x_lengths": x_lengths}
+
+
+def synthesise(
+    model, tokenizer, n_timesteps, text, length_scale, temperature, spks=None
+):
+    text_processed = process_text(text, tokenizer)
+    start_t = dt.datetime.now()
+    output = model.synthesise(
+        text_processed["x"],
+        text_processed["x_lengths"],
+        n_timesteps=n_timesteps,
+        temperature=temperature,
+        spks=spks,
+        length_scale=length_scale,
+    )
+    # merge everything to one dict
+    output.update({"start_t": start_t, **text_processed})
+    return output
+
+
+@torch.inference_mode()
+def main():
+    parser = get_parser()
+    args = parser.parse_args()
+    params = get_params()
+
+    params.update(vars(args))
+
+    tokenizer = Tokenizer(params.tokens)
+    params.blank_id = tokenizer.pad_id
+    params.vocab_size = tokenizer.vocab_size
+    params.model_args.n_vocab = params.vocab_size
+
+    with open(params.cmvn) as f:
+        stats = json.load(f)
+        params.data_args.data_statistics.mel_mean = stats["fbank_mean"]
+        params.data_args.data_statistics.mel_std = stats["fbank_std"]
+
+        params.model_args.data_statistics.mel_mean = stats["fbank_mean"]
+        params.model_args.data_statistics.mel_std = stats["fbank_std"]
+    logging.info(params)
+
+    logging.info("About to create model")
+    model = get_model(params)
+
+    load_checkpoint(f"{params.exp_dir}/epoch-{params.epoch}.pt", model)
+    model.eval()
+
+    if not Path(params.vocoder).is_file():
+        raise ValueError(f"{params.vocoder} does not exist")
+
+    vocoder = load_vocoder(params.vocoder)
+    denoiser = Denoiser(vocoder, mode="zeros")
+
+    # Number of ODE Solver steps
+    n_timesteps = 2
+
+    # Changes to the speaking rate
+    length_scale = 1.0
+
+    # Sampling temperature
+    temperature = 0.667
+
+    output = synthesise(
+        model=model,
+        tokenizer=tokenizer,
+        n_timesteps=n_timesteps,
+        text=params.input_text,
+        length_scale=length_scale,
+        temperature=temperature,
+    )
+    output["waveform"] = to_waveform(output["mel"], vocoder, denoiser)
+
+    sf.write(params.output_wav, output["waveform"], 22050, "PCM_16")
+
+
+if __name__ == "__main__":
+    torch.set_num_threads(1)
+    torch.set_num_interop_threads(1)
+    main()

egs/ljspeech/TTS/matcha/train.py

@@ -150,7 +150,7 @@ def _get_data_params() -> AttributeDict:
             "n_spks": 1,
             "n_fft": 1024,
             "n_feats": 80,
-            "sample_rate": 22050,
+            "sampling_rate": 22050,
             "hop_length": 256,
             "win_length": 1024,
             "f_min": 0,
@@ -445,11 +445,6 @@ def train_one_epoch(
 
     saved_bad_model = False
 
-    # used to track the stats over iterations in one epoch
-    tot_loss = MetricsTracker()
-
-    saved_bad_model = False
-
     def save_bad_model(suffix: str = ""):
         save_checkpoint(
             filename=params.exp_dir / f"bad-model{suffix}-{rank}.pt",

egs/ljspeech/TTS/vits/infer.py

@@ -234,7 +234,7 @@ def main():
     logging.info(f"Number of parameters in discriminator: {num_param_d}")
     logging.info(f"Total number of parameters: {num_param_g + num_param_d}")
 
-    # we need cut ids to display recognition results.
+    # we need cut ids to organize tts results.
     args.return_cuts = True
     ljspeech = LJSpeechTtsDataModule(args)
 

egs/ljspeech/TTS/vits/test_model.py

@@ -18,7 +18,6 @@
 
 from tokenizer import Tokenizer
 from train import get_model, get_params
-from vits import VITS
 
 
 def test_model_type(model_type):