Performed end to end testing on the matcha recipe (#1797)

* minor fixes to the `ljspeech/matcha` recipe

.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/inference.py \
+  ./matcha/infer.py \
     --epoch 1 \
     --exp-dir ./matcha/exp \
     --tokens data/tokens.txt \

egs/ljspeech/TTS/README.md

@@ -131,12 +131,12 @@ To inference, use:
 
 wget https://github.com/csukuangfj/models/raw/refs/heads/master/hifigan/generator_v1
 
-./matcha/inference \
+./matcha/synth.py \
   --exp-dir ./matcha/exp-new-3 \
   --epoch 4000 \
   --tokens ./data/tokens.txt \
   --vocoder ./generator_v1 \
-  --input-text "how are you doing?"
+  --input-text "how are you doing?" \
   --output-wav ./generated.wav
 ```
 

egs/ljspeech/TTS/local/audio.py

@@ -0,0 +1 @@
+../matcha/audio.py

egs/ljspeech/TTS/local/compute_fbank_ljspeech.py

@@ -27,102 +27,17 @@ The generated fbank features are saved in data/fbank.
 import argparse
 import logging
 import os
-from dataclasses import dataclass
 from pathlib import Path
-from typing import Union
 
-import numpy as np
 import torch
+from fbank import MatchaFbank, MatchaFbankConfig
 from lhotse import CutSet, LilcomChunkyWriter, load_manifest
 from lhotse.audio import RecordingSet
-from lhotse.features.base import FeatureExtractor, register_extractor
 from lhotse.supervision import SupervisionSet
-from lhotse.utils import Seconds, compute_num_frames
-from matcha.audio import mel_spectrogram
 
 from icefall.utils import get_executor
 
 
-@dataclass
-class MyFbankConfig:
-    n_fft: int
-    n_mels: int
-    sampling_rate: int
-    hop_length: int
-    win_length: int
-    f_min: float
-    f_max: float
-
-
-@register_extractor
-class MyFbank(FeatureExtractor):
-
-    name = "MyFbank"
-    config_type = MyFbankConfig
-
-    def __init__(self, config):
-        super().__init__(config=config)
-
-    @property
-    def device(self) -> Union[str, torch.device]:
-        return self.config.device
-
-    def feature_dim(self, sampling_rate: int) -> int:
-        return self.config.n_mels
-
-    def extract(
-        self,
-        samples: np.ndarray,
-        sampling_rate: int,
-    ) -> torch.Tensor:
-        # Check for sampling rate compatibility.
-        expected_sr = self.config.sampling_rate
-        assert sampling_rate == expected_sr, (
-            f"Mismatched sampling rate: extractor expects {expected_sr}, "
-            f"got {sampling_rate}"
-        )
-        samples = torch.from_numpy(samples)
-        assert samples.ndim == 2, samples.shape
-        assert samples.shape[0] == 1, samples.shape
-
-        mel = (
-            mel_spectrogram(
-                samples,
-                self.config.n_fft,
-                self.config.n_mels,
-                self.config.sampling_rate,
-                self.config.hop_length,
-                self.config.win_length,
-                self.config.f_min,
-                self.config.f_max,
-                center=False,
-            )
-            .squeeze()
-            .t()
-        )
-
-        assert mel.ndim == 2, mel.shape
-        assert mel.shape[1] == self.config.n_mels, mel.shape
-
-        num_frames = compute_num_frames(
-            samples.shape[1] / sampling_rate, self.frame_shift, sampling_rate
-        )
-
-        if mel.shape[0] > num_frames:
-            mel = mel[:num_frames]
-        elif mel.shape[0] < num_frames:
-            mel = mel.unsqueeze(0)
-            mel = torch.nn.functional.pad(
-                mel, (0, 0, 0, num_frames - mel.shape[1]), mode="replicate"
-            ).squeeze(0)
-
-        return mel.numpy()
-
-    @property
-    def frame_shift(self) -> Seconds:
-        return self.config.hop_length / self.config.sampling_rate
-
-
 def get_parser():
     parser = argparse.ArgumentParser(
         formatter_class=argparse.ArgumentDefaultsHelpFormatter
@@ -149,7 +64,7 @@ def compute_fbank_ljspeech(num_jobs: int):
     logging.info(f"num_jobs: {num_jobs}")
     logging.info(f"src_dir: {src_dir}")
     logging.info(f"output_dir: {output_dir}")
-    config = MyFbankConfig(
+    config = MatchaFbankConfig(
         n_fft=1024,
         n_mels=80,
         sampling_rate=22050,
@@ -170,7 +85,7 @@ def compute_fbank_ljspeech(num_jobs: int):
         src_dir / f"{prefix}_supervisions_{partition}.{suffix}", SupervisionSet
     )
 
-    extractor = MyFbank(config)
+    extractor = MatchaFbank(config)
 
     with get_executor() as ex:  # Initialize the executor only once.
         cuts_filename = f"{prefix}_cuts_{partition}.{suffix}"

egs/ljspeech/TTS/local/fbank.py

@@ -0,0 +1 @@
+../matcha/fbank.py

egs/ljspeech/TTS/matcha/compute_fbank_ljspeech.py

@@ -1 +0,0 @@
-../local/compute_fbank_ljspeech.py

egs/ljspeech/TTS/matcha/export_onnx_hifigan.py

@@ -7,7 +7,7 @@ from typing import Any, Dict
 
 import onnx
 import torch
-from inference import load_vocoder
+from infer import load_vocoder
 
 
 def add_meta_data(filename: str, meta_data: Dict[str, Any]):

egs/ljspeech/TTS/matcha/fbank.py

@@ -0,0 +1,88 @@
+from dataclasses import dataclass
+from typing import Union
+
+import numpy as np
+import torch
+from audio import mel_spectrogram
+from lhotse.features.base import FeatureExtractor, register_extractor
+from lhotse.utils import Seconds, compute_num_frames
+
+
+@dataclass
+class MatchaFbankConfig:
+    n_fft: int
+    n_mels: int
+    sampling_rate: int
+    hop_length: int
+    win_length: int
+    f_min: float
+    f_max: float
+
+
+@register_extractor
+class MatchaFbank(FeatureExtractor):
+
+    name = "MatchaFbank"
+    config_type = MatchaFbankConfig
+
+    def __init__(self, config):
+        super().__init__(config=config)
+
+    @property
+    def device(self) -> Union[str, torch.device]:
+        return self.config.device
+
+    def feature_dim(self, sampling_rate: int) -> int:
+        return self.config.n_mels
+
+    def extract(
+        self,
+        samples: np.ndarray,
+        sampling_rate: int,
+    ) -> torch.Tensor:
+        # Check for sampling rate compatibility.
+        expected_sr = self.config.sampling_rate
+        assert sampling_rate == expected_sr, (
+            f"Mismatched sampling rate: extractor expects {expected_sr}, "
+            f"got {sampling_rate}"
+        )
+        samples = torch.from_numpy(samples)
+        assert samples.ndim == 2, samples.shape
+        assert samples.shape[0] == 1, samples.shape
+
+        mel = (
+            mel_spectrogram(
+                samples,
+                self.config.n_fft,
+                self.config.n_mels,
+                self.config.sampling_rate,
+                self.config.hop_length,
+                self.config.win_length,
+                self.config.f_min,
+                self.config.f_max,
+                center=False,
+            )
+            .squeeze()
+            .t()
+        )
+
+        assert mel.ndim == 2, mel.shape
+        assert mel.shape[1] == self.config.n_mels, mel.shape
+
+        num_frames = compute_num_frames(
+            samples.shape[1] / sampling_rate, self.frame_shift, sampling_rate
+        )
+
+        if mel.shape[0] > num_frames:
+            mel = mel[:num_frames]
+        elif mel.shape[0] < num_frames:
+            mel = mel.unsqueeze(0)
+            mel = torch.nn.functional.pad(
+                mel, (0, 0, 0, num_frames - mel.shape[1]), mode="replicate"
+            ).squeeze(0)
+
+        return mel.numpy()
+
+    @property
+    def frame_shift(self) -> Seconds:
+        return self.config.hop_length / self.config.sampling_rate

egs/ljspeech/TTS/matcha/infer.py

@@ -0,0 +1,328 @@
+#!/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
+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, 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",
+        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.""",
+    )
+
+    # The following arguments are used for inference on single text
+    parser.add_argument(
+        "--input-text",
+        type=str,
+        required=False,
+        help="The text to generate speech for",
+    )
+
+    parser.add_argument(
+        "--output-wav",
+        type=str,
+        required=False,
+        help="The filename of the wave to save the generated speech",
+    )
+
+    parser.add_argument(
+        "--sampling-rate",
+        type=int,
+        default=22050,
+        help="The sampling rate of the generated speech (default: 22050 for LJSpeech)",
+    )
+
+    return parser
+
+
+def load_vocoder(checkpoint_path: Path) -> nn.Module:
+    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: 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.squeeze()
+
+
+def process_text(text: str, tokenizer: Tokenizer, device: str = "cpu") -> dict:
+    x = tokenizer.texts_to_token_ids([text], add_sos=True, add_eos=True)
+    x = torch.tensor(x, dtype=torch.long, device=device)
+    x_lengths = torch.tensor([x.shape[-1]], dtype=torch.long, device=device)
+    return {"x_orig": text, "x": x, "x_lengths": x_lengths}
+
+
+def synthesize(
+    model: nn.Module,
+    tokenizer: Tokenizer,
+    n_timesteps: int,
+    text: str,
+    length_scale: float,
+    temperature: float,
+    device: str = "cpu",
+    spks=None,
+) -> dict:
+    text_processed = process_text(text=text, tokenizer=tokenizer, device=device)
+    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
+
+
+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 = [c.supervisions[0].normalized_text for c in batch["cut"]]
+
+        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 = synthesize(
+                model=model,
+                tokenizer=tokenizer,
+                n_timesteps=params.n_timesteps,
+                text=texts[i],
+                length_scale=params.length_scale,
+                temperature=params.temperature,
+                device=device,
+            )
+            output["waveform"] = to_waveform(output["mel"], vocoder, denoiser)
+
+            sf.write(
+                file=params.save_wav_dir / f"{cut_ids[i]}_pred.wav",
+                data=output["waveform"],
+                samplerate=params.data_args.sampling_rate,
+                subtype="PCM_16",
+            )
+            sf.write(
+                file=params.save_wav_dir / f"{cut_ids[i]}_gt.wav",
+                data=audio[i].numpy(),
+                samplerate=params.data_args.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()
+    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
+    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"]
+
+    # 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)
+
+    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.to(device)
+
+    denoiser = Denoiser(vocoder, mode="zeros")
+    denoiser.to(device)
+
+    if params.input_text is not None and params.output_wav is not None:
+        logging.info("Synthesizing a single text")
+        output = synthesize(
+            model=model,
+            tokenizer=tokenizer,
+            n_timesteps=params.n_timesteps,
+            text=params.input_text,
+            length_scale=params.length_scale,
+            temperature=params.temperature,
+            device=device,
+        )
+        output["waveform"] = to_waveform(output["mel"], vocoder, denoiser)
+
+        sf.write(
+            file=params.output_wav,
+            data=output["waveform"],
+            samplerate=params.sampling_rate,
+            subtype="PCM_16",
+        )
+    else:
+        logging.info("Decoding the test set")
+        infer_dataset(
+            dl=test_dl,
+            params=params,
+            model=model,
+            vocoder=vocoder,
+            denoiser=denoiser,
+            tokenizer=tokenizer,
+        )
+
+
+if __name__ == "__main__":
+    main()

egs/ljspeech/TTS/matcha/inference.py

@@ -1,199 +0,0 @@
-#!/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
-
-
-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)
-
-    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.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__":
-    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/models/components/decoder.py

@@ -7,7 +7,7 @@ import torch.nn.functional as F
 from conformer import ConformerBlock
 from diffusers.models.activations import get_activation
 from einops import pack, rearrange, repeat
-from matcha.models.components.transformer import BasicTransformerBlock
+from models.components.transformer import BasicTransformerBlock
 
 
 class SinusoidalPosEmb(torch.nn.Module):

egs/ljspeech/TTS/matcha/models/components/flow_matching.py

@@ -2,7 +2,7 @@ from abc import ABC
 
 import torch
 import torch.nn.functional as F
-from matcha.models.components.decoder import Decoder
+from models.components.decoder import Decoder
 
 
 class BASECFM(torch.nn.Module, ABC):

egs/ljspeech/TTS/matcha/models/components/text_encoder.py

@@ -5,7 +5,7 @@ import math
 import torch
 import torch.nn as nn
 from einops import rearrange
-from matcha.model import sequence_mask
+from model import sequence_mask
 
 
 class LayerNorm(nn.Module):

egs/ljspeech/TTS/matcha/models/matcha_tts.py

@@ -2,17 +2,17 @@ import datetime as dt
 import math
 import random
 
-import matcha.monotonic_align as monotonic_align
+import monotonic_align as monotonic_align
 import torch
-from matcha.model import (
+from model import (
     denormalize,
     duration_loss,
     fix_len_compatibility,
     generate_path,
     sequence_mask,
 )
-from matcha.models.components.flow_matching import CFM
-from matcha.models.components.text_encoder import TextEncoder
+from models.components.flow_matching import CFM
+from models.components.text_encoder import TextEncoder
 
 
 class MatchaTTS(torch.nn.Module):  # 🍵

egs/ljspeech/TTS/matcha/monotonic_align/.gitignore

@@ -1,3 +1,3 @@
 build
 core.c
-*.so
+*.so

egs/ljspeech/TTS/matcha/monotonic_align/__init__.py

@@ -1,8 +1,7 @@
-# Copied from
-# https://github.com/shivammehta25/Matcha-TTS/blob/main/matcha/utils/monotonic_align/__init__.py
 import numpy as np
 import torch
-from matcha.monotonic_align.core import maximum_path_c
+
+from .core import maximum_path_c
 
 
 def maximum_path(value, mask):

egs/ljspeech/TTS/matcha/monotonic_align/core.pyx

@@ -1,5 +1,3 @@
-# Copied from
-# https://github.com/shivammehta25/Matcha-TTS/blob/main/matcha/utils/monotonic_align/core.pyx
 import numpy as np
 
 cimport cython

egs/ljspeech/TTS/matcha/monotonic_align/setup.py

@@ -1,12 +1,30 @@
-# Copied from
+# Modified from
 # https://github.com/shivammehta25/Matcha-TTS/blob/main/matcha/utils/monotonic_align/setup.py
-from distutils.core import setup
-
-import numpy
 from Cython.Build import cythonize
+from setuptools import Extension, setup
+from setuptools.command.build_ext import build_ext as _build_ext
 
+
+class build_ext(_build_ext):
+    """Overwrite build_ext."""
+
+    def finalize_options(self):
+        """Prevent numpy from thinking it is still in its setup process."""
+        _build_ext.finalize_options(self)
+        __builtins__.__NUMPY_SETUP__ = False
+        import numpy
+
+        self.include_dirs.append(numpy.get_include())
+
+
+exts = [
+    Extension(
+        name="core",
+        sources=["core.pyx"],
+    )
+]
 setup(
     name="monotonic_align",
-    ext_modules=cythonize("core.pyx"),
-    include_dirs=[numpy.get_include()],
+    ext_modules=cythonize(exts, language_level=3),
+    cmdclass={"build_ext": build_ext},
 )

egs/ljspeech/TTS/matcha/requirements.txt

@@ -1,3 +1,4 @@
 conformer==0.3.2
 diffusers # developed using version ==0.25.0
 librosa
+einops

egs/ljspeech/TTS/matcha/train.py

@@ -14,9 +14,9 @@ import torch
 import torch.multiprocessing as mp
 import torch.nn as nn
 from lhotse.utils import fix_random_seed
-from matcha.model import fix_len_compatibility
-from matcha.models.matcha_tts import MatchaTTS
-from matcha.tokenizer import Tokenizer
+from model import fix_len_compatibility
+from models.matcha_tts import MatchaTTS
+from tokenizer import Tokenizer
 from torch.cuda.amp import GradScaler, autocast
 from torch.nn.parallel import DistributedDataParallel as DDP
 from torch.optim import Optimizer
@@ -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/matcha/tts_datamodule.py

@@ -24,7 +24,7 @@ from pathlib import Path
 from typing import Any, Dict, Optional
 
 import torch
-from compute_fbank_ljspeech import MyFbank, MyFbankConfig
+from fbank import MatchaFbank, MatchaFbankConfig
 from lhotse import CutSet, load_manifest_lazy
 from lhotse.dataset import (  # noqa F401 for PrecomputedFeatures
     CutConcatenate,
@@ -32,7 +32,6 @@ from lhotse.dataset import (  # noqa F401 for PrecomputedFeatures
     DynamicBucketingSampler,
     PrecomputedFeatures,
     SimpleCutSampler,
-    SpecAugment,
     SpeechSynthesisDataset,
 )
 from lhotse.dataset.input_strategies import (  # noqa F401 For AudioSamples
@@ -177,7 +176,7 @@ class LJSpeechTtsDataModule:
 
         if self.args.on_the_fly_feats:
             sampling_rate = 22050
-            config = MyFbankConfig(
+            config = MatchaFbankConfig(
                 n_fft=1024,
                 n_mels=80,
                 sampling_rate=sampling_rate,
@@ -189,7 +188,7 @@ class LJSpeechTtsDataModule:
             train = SpeechSynthesisDataset(
                 return_text=False,
                 return_tokens=True,
-                feature_input_strategy=OnTheFlyFeatures(MyFbank(config)),
+                feature_input_strategy=OnTheFlyFeatures(MatchaFbank(config)),
                 return_cuts=self.args.return_cuts,
             )
 
@@ -238,7 +237,7 @@ class LJSpeechTtsDataModule:
         logging.info("About to create dev dataset")
         if self.args.on_the_fly_feats:
             sampling_rate = 22050
-            config = MyFbankConfig(
+            config = MatchaFbankConfig(
                 n_fft=1024,
                 n_mels=80,
                 sampling_rate=sampling_rate,
@@ -250,7 +249,7 @@ class LJSpeechTtsDataModule:
             validate = SpeechSynthesisDataset(
                 return_text=False,
                 return_tokens=True,
-                feature_input_strategy=OnTheFlyFeatures(MyFbank(config)),
+                feature_input_strategy=OnTheFlyFeatures(MatchaFbank(config)),
                 return_cuts=self.args.return_cuts,
             )
         else:
@@ -282,7 +281,7 @@ class LJSpeechTtsDataModule:
         logging.info("About to create test dataset")
         if self.args.on_the_fly_feats:
             sampling_rate = 22050
-            config = MyFbankConfig(
+            config = MatchaFbankConfig(
                 n_fft=1024,
                 n_mels=80,
                 sampling_rate=sampling_rate,
@@ -294,7 +293,7 @@ class LJSpeechTtsDataModule:
             test = SpeechSynthesisDataset(
                 return_text=False,
                 return_tokens=True,
-                feature_input_strategy=OnTheFlyFeatures(MyFbank(config)),
+                feature_input_strategy=OnTheFlyFeatures(MatchaFbank(config)),
                 return_cuts=self.args.return_cuts,
             )
         else:

egs/ljspeech/TTS/prepare.sh

@@ -25,26 +25,16 @@ log() {
 log "dl_dir: $dl_dir"
 
 if [ $stage -le -1 ] && [ $stop_stage -ge -1 ]; then
-  log "Stage -1: build monotonic_align lib"
-  if [ ! -d vits/monotonic_align/build ]; then
-    cd vits/monotonic_align
-    python3 setup.py build_ext --inplace
-    cd ../../
-  else
-    log "monotonic_align lib for vits already built"
-  fi
-
-  if [ ! -f ./matcha/monotonic_align/core.cpython-38-x86_64-linux-gnu.so  ]; then
-    pushd matcha/monotonic_align
-    python3 setup.py build
-    mv -v build/lib.*/matcha/monotonic_align/core.*.so .
-    rm -rf build
-    rm core.c
-    ls -lh
-    popd
-  else
-    log "monotonic_align lib for matcha-tts already built"
-  fi
+  log "Stage -1: build monotonic_align lib (used by vits and matcha recipes)"
+  for recipe in vits matcha; do
+    if [ ! -d $recipe/monotonic_align/build ]; then
+      cd $recipe/monotonic_align
+      python3 setup.py build_ext --inplace
+      cd ../../
+    else
+      log "monotonic_align lib for $recipe already built"
+    fi
+  done
 fi
 
 if [ $stage -le 0 ] && [ $stop_stage -ge 0 ]; then

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/monotonic_align/.gitignore

@@ -0,0 +1,3 @@
+build
+core.c
+*.so

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):