Merge branch 'k2-fsa:master' into dev/k2ssl

egs/fluent_speech_commands/SLU/shared

@@ -1 +1 @@
-../../../icefall/shared/
+../../../icefall/shared

egs/ljspeech/TTS/matcha/fbank.py

@@ -17,6 +17,7 @@ class MatchaFbankConfig:
     win_length: int
     f_min: float
     f_max: float
+    device: str = "cuda"
 
 
 @register_extractor
@@ -46,7 +47,7 @@ class MatchaFbank(FeatureExtractor):
             f"Mismatched sampling rate: extractor expects {expected_sr}, "
             f"got {sampling_rate}"
         )
-        samples = torch.from_numpy(samples)
+        samples = torch.from_numpy(samples).to(self.device)
         assert samples.ndim == 2, samples.shape
         assert samples.shape[0] == 1, samples.shape
 
@@ -81,7 +82,7 @@ class MatchaFbank(FeatureExtractor):
                 mel, (0, 0, 0, num_frames - mel.shape[1]), mode="replicate"
             ).squeeze(0)
 
-        return mel.numpy()
+        return mel.cpu().numpy()
 
     @property
     def frame_shift(self) -> Seconds:

egs/wenetspeech4tts/TTS/README.md

@@ -68,5 +68,69 @@ python3 valle/infer.py --output-dir demos_epoch_${epoch}_avg_${avg}_top_p_${top_
         --text-extractor pypinyin_initials_finals --top-p ${top_p}
 ```
 
+# [F5-TTS](https://arxiv.org/abs/2410.06885)
+
+./f5-tts contains the code for training F5-TTS model.
+
+Generated samples and training logs of wenetspeech basic 7k hours data can be found [here](https://huggingface.co/yuekai/f5-tts-small-wenetspeech4tts-basic/tensorboard).
+
+Preparation:
+
+```
+bash prepare.sh --stage 5 --stop_stage 6
+```
+(Note: To compatiable with F5-TTS official checkpoint, we direclty use `vocab.txt` from [here.](https://github.com/SWivid/F5-TTS/blob/129014c5b43f135b0100d49a0c6804dd4cf673e1/data/Emilia_ZH_EN_pinyin/vocab.txt) To generate your own `vocab.txt`, you may refer to [the script](https://github.com/SWivid/F5-TTS/blob/main/src/f5_tts/train/datasets/prepare_emilia.py).)
+
+The training command is given below:
+
+```
+# docker: ghcr.io/swivid/f5-tts:main
+# pip install k2==1.24.4.dev20241030+cuda12.4.torch2.4.0 -f https://k2-fsa.github.io/k2/cuda.html
+# pip install kaldialign lhotse tensorboard bigvganinference sentencepiece
+
+world_size=8
+exp_dir=exp/f5-tts-small
+python3 f5-tts/train.py --max-duration 700 --filter-min-duration 0.5 --filter-max-duration 20  \
+      --num-buckets 6 --dtype "bfloat16" --save-every-n 5000 --valid-interval 10000 \
+      --base-lr 7.5e-5 --warmup-steps 20000 --num-epochs 60  \
+      --num-decoder-layers 18 --nhead 12 --decoder-dim 768 \
+      --exp-dir ${exp_dir} --world-size ${world_size}
+```
+
+To inference with Icefall Wenetspeech4TTS trained F5-Small, use:
+```
+huggingface-cli login
+huggingface-cli download --local-dir seed_tts_eval yuekai/seed_tts_eval --repo-type dataset
+huggingface-cli download --local-dir ${exp_dir} yuekai/f5-tts-small-wenetspeech4tts-basic
+huggingface-cli download nvidia/bigvgan_v2_24khz_100band_256x --local-dir bigvgan_v2_24khz_100band_256x
+
+manifest=./seed_tts_eval/seedtts_testset/zh/meta.lst
+model_path=f5-tts-small-wenetspeech4tts-basic/epoch-56-avg-14.pt
+# skip
+python3 f5-tts/generate_averaged_model.py \
+    --epoch 56 \
+    --avg 14 --decoder-dim 768 --nhead 12 --num-decoder-layers 18 \
+    --exp-dir exp/f5_small
+
+
+accelerate launch f5-tts/infer.py --nfe 16 --model-path $model_path --manifest-file $manifest --output-dir $output_dir --decoder-dim 768 --nhead 12 --num-decoder-layers 18
+bash local/compute_wer.sh $output_dir $manifest
+```
+
+To inference with official Emilia trained F5-Base, use:
+```
+huggingface-cli login
+huggingface-cli download --local-dir seed_tts_eval yuekai/seed_tts_eval --repo-type dataset
+huggingface-cli download --local-dir F5-TTS SWivid/F5-TTS
+huggingface-cli download nvidia/bigvgan_v2_24khz_100band_256x --local-dir bigvgan_v2_24khz_100band_256x
+
+manifest=./seed_tts_eval/seedtts_testset/zh/meta.lst
+model_path=./F5-TTS/F5TTS_Base_bigvgan/model_1250000.pt
+
+accelerate launch f5-tts/infer.py --nfe 16 --model-path $model_path --manifest-file $manifest --output-dir $output_dir
+bash local/compute_wer.sh $output_dir $manifest
+```
+
 # Credits
-- [vall-e](https://github.com/lifeiteng/vall-e)
+- [VALL-E](https://github.com/lifeiteng/vall-e)
+- [F5-TTS](https://github.com/SWivid/F5-TTS)

egs/wenetspeech4tts/TTS/f5-tts/generate_averaged_model.py

@@ -0,0 +1,173 @@
+#!/usr/bin/env python3
+#
+# Copyright 2021-2022 Xiaomi Corporation (Author: Yifan Yang)
+# Copyright 2024                                 Yuekai Zhang
+#
+# See ../../../../LICENSE for clarification regarding multiple authors
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+"""
+Usage:
+(1) use the checkpoint exp_dir/epoch-xxx.pt
+python3 bin/generate_averaged_model.py \
+    --epoch 40 \
+    --avg 5 \
+    --exp-dir ${exp_dir}
+
+It will generate a file `epoch-28-avg-15.pt` in the given `exp_dir`.
+You can later load it by `torch.load("epoch-28-avg-15.pt")`.
+"""
+
+
+import argparse
+from pathlib import Path
+
+import k2
+import torch
+from train import add_model_arguments, get_model
+
+from icefall.checkpoint import (
+    average_checkpoints,
+    average_checkpoints_with_averaged_model,
+    find_checkpoints,
+)
+from icefall.utils import AttributeDict
+
+
+def get_parser():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--epoch",
+        type=int,
+        default=30,
+        help="""It specifies the checkpoint to use for decoding.
+        Note: Epoch counts from 1.
+        You can specify --avg to use more checkpoints for model averaging.""",
+    )
+
+    parser.add_argument(
+        "--iter",
+        type=int,
+        default=0,
+        help="""If positive, --epoch is ignored and it
+        will use the checkpoint exp_dir/checkpoint-iter.pt.
+        You can specify --avg to use more checkpoints for model averaging.
+        """,
+    )
+
+    parser.add_argument(
+        "--avg",
+        type=int,
+        default=9,
+        help="Number of checkpoints to average. Automatically select "
+        "consecutive checkpoints before the checkpoint specified by "
+        "'--epoch' and '--iter'",
+    )
+
+    parser.add_argument(
+        "--exp-dir",
+        type=str,
+        default="zipformer/exp",
+        help="The experiment dir",
+    )
+    add_model_arguments(parser)
+    return parser
+
+
+@torch.no_grad()
+def main():
+    parser = get_parser()
+
+    args = parser.parse_args()
+    args.exp_dir = Path(args.exp_dir)
+
+    params = AttributeDict()
+    params.update(vars(args))
+
+    if params.iter > 0:
+        params.suffix = f"checkpoint-{params.iter}-avg-{params.avg}"
+    else:
+        params.suffix = f"epoch-{params.epoch}-avg-{params.avg}"
+
+    print("Script started")
+
+    device = torch.device("cpu")
+    print(f"Device: {device}")
+
+    print("About to create model")
+    filename = f"{params.exp_dir}/epoch-{params.epoch}.pt"
+    checkpoint = torch.load(filename, map_location=device)
+    args = AttributeDict(checkpoint)
+    model = get_model(args)
+
+    if params.iter > 0:
+        # TODO FIX ME
+        filenames = find_checkpoints(params.exp_dir, iteration=-params.iter)[
+            : params.avg + 1
+        ]
+        if len(filenames) == 0:
+            raise ValueError(
+                f"No checkpoints found for --iter {params.iter}, --avg {params.avg}"
+            )
+        elif len(filenames) < params.avg + 1:
+            raise ValueError(
+                f"Not enough checkpoints ({len(filenames)}) found for"
+                f" --iter {params.iter}, --avg {params.avg}"
+            )
+        filename_start = filenames[-1]
+        filename_end = filenames[0]
+        print(
+            "Calculating the averaged model over iteration checkpoints"
+            f" from {filename_start} (excluded) to {filename_end}"
+        )
+        model.to(device)
+        model.load_state_dict(
+            average_checkpoints_with_averaged_model(
+                filename_start=filename_start,
+                filename_end=filename_end,
+                device=device,
+            )
+        )
+        filename = params.exp_dir / f"checkpoint-{params.iter}-avg-{params.avg}.pt"
+        torch.save({"model": model.state_dict()}, filename)
+    else:
+        assert params.avg > 0, params.avg
+        start = params.epoch - params.avg
+        assert start >= 1, start
+        filename_start = f"{params.exp_dir}/epoch-{start}.pt"
+        filename_end = f"{params.exp_dir}/epoch-{params.epoch}.pt"
+        print(
+            f"Calculating the averaged model over epoch range from "
+            f"{start} (excluded) to {params.epoch}"
+        )
+        filenames = [
+            f"{params.exp_dir}/epoch-{i}.pt" for i in range(start, params.epoch + 1)
+        ]
+        model.to(device)
+        model.load_state_dict(average_checkpoints(filenames, device=device))
+
+        filename = params.exp_dir / f"epoch-{params.epoch}-avg-{params.avg}.pt"
+        checkpoint["model"] = model.state_dict()
+        torch.save(checkpoint, filename)
+
+    num_param = sum([p.numel() for p in model.parameters()])
+    print(f"Number of model parameters: {num_param}")
+
+    print("Done!")
+
+
+if __name__ == "__main__":
+    main()

egs/wenetspeech4tts/TTS/f5-tts/infer.py

@@ -0,0 +1,364 @@
+#!/usr/bin/env python3
+# Modified from https://github.com/SWivid/F5-TTS/blob/main/src/f5_tts/eval/eval_infer_batch.py
+"""
+Usage:
+# docker: ghcr.io/swivid/f5-tts:main
+# pip install k2==1.24.4.dev20241030+cuda12.4.torch2.4.0 -f https://k2-fsa.github.io/k2/cuda.html
+# pip install kaldialign lhotse tensorboard bigvganinference sentencepiece sherpa-onnx
+# huggingface-cli download nvidia/bigvgan_v2_24khz_100band_256x --local-dir bigvgan_v2_24khz_100band_256x
+manifest=/path/seed_tts_eval/seedtts_testset/zh/meta.lst
+python3 f5-tts/generate_averaged_model.py \
+    --epoch 56 \
+    --avg 14 --decoder-dim 768 --nhead 12 --num-decoder-layers 18 \
+    --exp-dir exp/f5_small
+accelerate launch f5-tts/infer.py --nfe 16 --model-path $model_path --manifest-file $manifest --output-dir $output_dir --decoder-dim 768 --nhead 12 --num-decoder-layers 18
+bash local/compute_wer.sh $output_dir $manifest
+"""
+import argparse
+import logging
+import math
+import os
+import random
+import time
+from pathlib import Path
+
+import torch
+import torch.nn.functional as F
+import torchaudio
+from accelerate import Accelerator
+from bigvganinference import BigVGANInference
+from model.cfm import CFM
+from model.dit import DiT
+from model.modules import MelSpec
+from model.utils import convert_char_to_pinyin
+from tqdm import tqdm
+from train import (
+    add_model_arguments,
+    get_model,
+    get_tokenizer,
+    load_F5_TTS_pretrained_checkpoint,
+)
+
+from icefall.checkpoint import load_checkpoint
+
+
+def get_parser():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--tokens",
+        type=str,
+        default="f5-tts/vocab.txt",
+        help="Path to the unique text tokens file",
+    )
+
+    parser.add_argument(
+        "--model-path",
+        type=str,
+        default="/home/yuekaiz/HF/F5-TTS/F5TTS_Base_bigvgan/model_1250000.pt",
+        help="Path to the unique text tokens file",
+    )
+
+    parser.add_argument(
+        "--seed",
+        type=int,
+        default=0,
+        help="The seed for random generators intended for reproducibility",
+    )
+
+    parser.add_argument(
+        "--nfe",
+        type=int,
+        default=16,
+        help="The number of steps for the neural ODE",
+    )
+
+    parser.add_argument(
+        "--manifest-file",
+        type=str,
+        default="/path/seed_tts_eval/seedtts_testset/zh/meta.lst",
+        help="The manifest file in seed_tts_eval format",
+    )
+
+    parser.add_argument(
+        "--output-dir",
+        type=Path,
+        default="results",
+        help="The output directory to save the generated wavs",
+    )
+
+    parser.add_argument("-ss", "--swaysampling", default=-1, type=float)
+    add_model_arguments(parser)
+    return parser.parse_args()
+
+
+def get_inference_prompt(
+    metainfo,
+    speed=1.0,
+    tokenizer="pinyin",
+    polyphone=True,
+    target_sample_rate=24000,
+    n_fft=1024,
+    win_length=1024,
+    n_mel_channels=100,
+    hop_length=256,
+    mel_spec_type="bigvgan",
+    target_rms=0.1,
+    use_truth_duration=False,
+    infer_batch_size=1,
+    num_buckets=200,
+    min_secs=3,
+    max_secs=40,
+):
+    prompts_all = []
+
+    min_tokens = min_secs * target_sample_rate // hop_length
+    max_tokens = max_secs * target_sample_rate // hop_length
+
+    batch_accum = [0] * num_buckets
+    utts, ref_rms_list, ref_mels, ref_mel_lens, total_mel_lens, final_text_list = (
+        [[] for _ in range(num_buckets)] for _ in range(6)
+    )
+
+    mel_spectrogram = MelSpec(
+        n_fft=n_fft,
+        hop_length=hop_length,
+        win_length=win_length,
+        n_mel_channels=n_mel_channels,
+        target_sample_rate=target_sample_rate,
+        mel_spec_type=mel_spec_type,
+    )
+
+    for utt, prompt_text, prompt_wav, gt_text, gt_wav in tqdm(
+        metainfo, desc="Processing prompts..."
+    ):
+        # Audio
+        ref_audio, ref_sr = torchaudio.load(prompt_wav)
+        ref_rms = torch.sqrt(torch.mean(torch.square(ref_audio)))
+        if ref_rms < target_rms:
+            ref_audio = ref_audio * target_rms / ref_rms
+        assert (
+            ref_audio.shape[-1] > 5000
+        ), f"Empty prompt wav: {prompt_wav}, or torchaudio backend issue."
+        if ref_sr != target_sample_rate:
+            resampler = torchaudio.transforms.Resample(ref_sr, target_sample_rate)
+            ref_audio = resampler(ref_audio)
+
+        # Text
+        if len(prompt_text[-1].encode("utf-8")) == 1:
+            prompt_text = prompt_text + " "
+        text = [prompt_text + gt_text]
+        if tokenizer == "pinyin":
+            text_list = convert_char_to_pinyin(text, polyphone=polyphone)
+        else:
+            text_list = text
+
+        # Duration, mel frame length
+        ref_mel_len = ref_audio.shape[-1] // hop_length
+        if use_truth_duration:
+            gt_audio, gt_sr = torchaudio.load(gt_wav)
+            if gt_sr != target_sample_rate:
+                resampler = torchaudio.transforms.Resample(gt_sr, target_sample_rate)
+                gt_audio = resampler(gt_audio)
+            total_mel_len = ref_mel_len + int(gt_audio.shape[-1] / hop_length / speed)
+
+            # # test vocoder resynthesis
+            # ref_audio = gt_audio
+        else:
+            ref_text_len = len(prompt_text.encode("utf-8"))
+            gen_text_len = len(gt_text.encode("utf-8"))
+            total_mel_len = ref_mel_len + int(
+                ref_mel_len / ref_text_len * gen_text_len / speed
+            )
+
+        # to mel spectrogram
+        ref_mel = mel_spectrogram(ref_audio)
+        ref_mel = ref_mel.squeeze(0)
+
+        # deal with batch
+        assert infer_batch_size > 0, "infer_batch_size should be greater than 0."
+        assert (
+            min_tokens <= total_mel_len <= max_tokens
+        ), f"Audio {utt} has duration {total_mel_len*hop_length//target_sample_rate}s out of range [{min_secs}, {max_secs}]."
+        bucket_i = math.floor(
+            (total_mel_len - min_tokens) / (max_tokens - min_tokens + 1) * num_buckets
+        )
+
+        utts[bucket_i].append(utt)
+        ref_rms_list[bucket_i].append(ref_rms)
+        ref_mels[bucket_i].append(ref_mel)
+        ref_mel_lens[bucket_i].append(ref_mel_len)
+        total_mel_lens[bucket_i].append(total_mel_len)
+        final_text_list[bucket_i].extend(text_list)
+
+        batch_accum[bucket_i] += total_mel_len
+
+        if batch_accum[bucket_i] >= infer_batch_size:
+            prompts_all.append(
+                (
+                    utts[bucket_i],
+                    ref_rms_list[bucket_i],
+                    padded_mel_batch(ref_mels[bucket_i]),
+                    ref_mel_lens[bucket_i],
+                    total_mel_lens[bucket_i],
+                    final_text_list[bucket_i],
+                )
+            )
+            batch_accum[bucket_i] = 0
+            (
+                utts[bucket_i],
+                ref_rms_list[bucket_i],
+                ref_mels[bucket_i],
+                ref_mel_lens[bucket_i],
+                total_mel_lens[bucket_i],
+                final_text_list[bucket_i],
+            ) = (
+                [],
+                [],
+                [],
+                [],
+                [],
+                [],
+            )
+
+    # add residual
+    for bucket_i, bucket_frames in enumerate(batch_accum):
+        if bucket_frames > 0:
+            prompts_all.append(
+                (
+                    utts[bucket_i],
+                    ref_rms_list[bucket_i],
+                    padded_mel_batch(ref_mels[bucket_i]),
+                    ref_mel_lens[bucket_i],
+                    total_mel_lens[bucket_i],
+                    final_text_list[bucket_i],
+                )
+            )
+    # not only leave easy work for last workers
+    random.seed(666)
+    random.shuffle(prompts_all)
+
+    return prompts_all
+
+
+def padded_mel_batch(ref_mels):
+    max_mel_length = torch.LongTensor([mel.shape[-1] for mel in ref_mels]).amax()
+    padded_ref_mels = []
+    for mel in ref_mels:
+        padded_ref_mel = F.pad(mel, (0, max_mel_length - mel.shape[-1]), value=0)
+        padded_ref_mels.append(padded_ref_mel)
+    padded_ref_mels = torch.stack(padded_ref_mels)
+    padded_ref_mels = padded_ref_mels.permute(0, 2, 1)
+    return padded_ref_mels
+
+
+def get_seedtts_testset_metainfo(metalst):
+    f = open(metalst)
+    lines = f.readlines()
+    f.close()
+    metainfo = []
+    for line in lines:
+        assert len(line.strip().split("|")) == 4
+        utt, prompt_text, prompt_wav, gt_text = line.strip().split("|")
+        utt = Path(utt).stem
+        gt_wav = os.path.join(os.path.dirname(metalst), "wavs", utt + ".wav")
+        if not os.path.isabs(prompt_wav):
+            prompt_wav = os.path.join(os.path.dirname(metalst), prompt_wav)
+        metainfo.append((utt, prompt_text, prompt_wav, gt_text, gt_wav))
+    return metainfo
+
+
+def main():
+    args = get_parser()
+
+    accelerator = Accelerator()
+    device = f"cuda:{accelerator.process_index}"
+
+    metainfo = get_seedtts_testset_metainfo(args.manifest_file)
+    prompts_all = get_inference_prompt(
+        metainfo,
+        speed=1.0,
+        tokenizer="pinyin",
+        target_sample_rate=24_000,
+        n_mel_channels=100,
+        hop_length=256,
+        mel_spec_type="bigvgan",
+        target_rms=0.1,
+        use_truth_duration=False,
+        infer_batch_size=1,
+    )
+
+    vocoder = BigVGANInference.from_pretrained(
+        "./bigvgan_v2_24khz_100band_256x", use_cuda_kernel=False
+    )
+    vocoder = vocoder.eval().to(device)
+
+    model = get_model(args).eval().to(device)
+    checkpoint = torch.load(args.model_path, map_location="cpu")
+    if "ema_model_state_dict" in checkpoint or "model_state_dict" in checkpoint:
+        model = load_F5_TTS_pretrained_checkpoint(model, args.model_path)
+    else:
+        _ = load_checkpoint(
+            args.model_path,
+            model=model,
+        )
+
+    os.makedirs(args.output_dir, exist_ok=True)
+
+    accelerator.wait_for_everyone()
+    start = time.time()
+
+    with accelerator.split_between_processes(prompts_all) as prompts:
+        for prompt in tqdm(prompts, disable=not accelerator.is_local_main_process):
+            (
+                utts,
+                ref_rms_list,
+                ref_mels,
+                ref_mel_lens,
+                total_mel_lens,
+                final_text_list,
+            ) = prompt
+            ref_mels = ref_mels.to(device)
+            ref_mel_lens = torch.tensor(ref_mel_lens, dtype=torch.long).to(device)
+            total_mel_lens = torch.tensor(total_mel_lens, dtype=torch.long).to(device)
+
+            # Inference
+            with torch.inference_mode():
+                generated, _ = model.sample(
+                    cond=ref_mels,
+                    text=final_text_list,
+                    duration=total_mel_lens,
+                    lens=ref_mel_lens,
+                    steps=args.nfe,
+                    cfg_strength=2.0,
+                    sway_sampling_coef=args.swaysampling,
+                    no_ref_audio=False,
+                    seed=args.seed,
+                )
+                for i, gen in enumerate(generated):
+                    gen = gen[ref_mel_lens[i] : total_mel_lens[i], :].unsqueeze(0)
+                    gen_mel_spec = gen.permute(0, 2, 1).to(torch.float32)
+
+                    generated_wave = vocoder(gen_mel_spec).squeeze(0).cpu()
+                    target_rms = 0.1
+                    target_sample_rate = 24_000
+                    if ref_rms_list[i] < target_rms:
+                        generated_wave = generated_wave * ref_rms_list[i] / target_rms
+                    torchaudio.save(
+                        f"{args.output_dir}/{utts[i]}.wav",
+                        generated_wave,
+                        target_sample_rate,
+                    )
+
+    accelerator.wait_for_everyone()
+    if accelerator.is_main_process:
+        timediff = time.time() - start
+        print(f"Done batch inference in {timediff / 60 :.2f} minutes.")
+
+
+if __name__ == "__main__":
+    formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
+    logging.basicConfig(format=formatter, level=logging.INFO)
+    main()

egs/wenetspeech4tts/TTS/f5-tts/model/README.md

@@ -0,0 +1,3 @@
+# Introduction
+Files in this folder are copied from
+https://github.com/SWivid/F5-TTS/tree/main/src/f5_tts/model

egs/wenetspeech4tts/TTS/f5-tts/model/cfm.py

@@ -0,0 +1,326 @@
+"""
+ein notation:
+b - batch
+n - sequence
+nt - text sequence
+nw - raw wave length
+d - dimension
+"""
+
+from __future__ import annotations
+
+from random import random
+from typing import Callable
+
+import torch
+import torch.nn.functional as F
+from model.modules import MelSpec
+from model.utils import (
+    default,
+    exists,
+    lens_to_mask,
+    list_str_to_idx,
+    list_str_to_tensor,
+    mask_from_frac_lengths,
+)
+from torch import nn
+from torch.nn.utils.rnn import pad_sequence
+from torchdiffeq import odeint
+
+
+class CFM(nn.Module):
+    def __init__(
+        self,
+        transformer: nn.Module,
+        sigma=0.0,
+        odeint_kwargs: dict = dict(
+            # atol = 1e-5,
+            # rtol = 1e-5,
+            method="euler"  # 'midpoint'
+        ),
+        audio_drop_prob=0.3,
+        cond_drop_prob=0.2,
+        num_channels=None,
+        mel_spec_module: nn.Module | None = None,
+        mel_spec_kwargs: dict = dict(),
+        frac_lengths_mask: tuple[float, float] = (0.7, 1.0),
+        vocab_char_map: dict[str:int] | None = None,
+    ):
+        super().__init__()
+
+        self.frac_lengths_mask = frac_lengths_mask
+
+        # mel spec
+        self.mel_spec = default(mel_spec_module, MelSpec(**mel_spec_kwargs))
+        num_channels = default(num_channels, self.mel_spec.n_mel_channels)
+        self.num_channels = num_channels
+
+        # classifier-free guidance
+        self.audio_drop_prob = audio_drop_prob
+        self.cond_drop_prob = cond_drop_prob
+
+        # transformer
+        self.transformer = transformer
+        dim = transformer.dim
+        self.dim = dim
+
+        # conditional flow related
+        self.sigma = sigma
+
+        # sampling related
+        self.odeint_kwargs = odeint_kwargs
+
+        # vocab map for tokenization
+        self.vocab_char_map = vocab_char_map
+
+    @property
+    def device(self):
+        return next(self.parameters()).device
+
+    @torch.no_grad()
+    def sample(
+        self,
+        cond: float["b n d"] | float["b nw"],  # noqa: F722
+        text: int["b nt"] | list[str],  # noqa: F722
+        duration: int | int["b"],  # noqa: F821
+        *,
+        lens: int["b"] | None = None,  # noqa: F821
+        steps=32,
+        cfg_strength=1.0,
+        sway_sampling_coef=None,
+        seed: int | None = None,
+        max_duration=4096,
+        vocoder: Callable[[float["b d n"]], float["b nw"]] | None = None,  # noqa: F722
+        no_ref_audio=False,
+        duplicate_test=False,
+        t_inter=0.1,
+        edit_mask=None,
+    ):
+        self.eval()
+        # raw wave
+
+        if cond.ndim == 2:
+            cond = self.mel_spec(cond)
+            cond = cond.permute(0, 2, 1)
+            assert cond.shape[-1] == self.num_channels
+
+        cond = cond.to(next(self.parameters()).dtype)
+
+        batch, cond_seq_len, device = *cond.shape[:2], cond.device
+        if not exists(lens):
+            lens = torch.full((batch,), cond_seq_len, device=device, dtype=torch.long)
+
+        # text
+
+        if isinstance(text, list):
+            if exists(self.vocab_char_map):
+                text = list_str_to_idx(text, self.vocab_char_map).to(device)
+            else:
+                text = list_str_to_tensor(text).to(device)
+            assert text.shape[0] == batch
+
+        if exists(text):
+            text_lens = (text != -1).sum(dim=-1)
+            lens = torch.maximum(
+                text_lens, lens
+            )  # make sure lengths are at least those of the text characters
+
+        # duration
+
+        cond_mask = lens_to_mask(lens)
+        if edit_mask is not None:
+            cond_mask = cond_mask & edit_mask
+
+        if isinstance(duration, int):
+            duration = torch.full((batch,), duration, device=device, dtype=torch.long)
+
+        duration = torch.maximum(
+            lens + 1, duration
+        )  # just add one token so something is generated
+        duration = duration.clamp(max=max_duration)
+        max_duration = duration.amax()
+
+        # duplicate test corner for inner time step oberservation
+        if duplicate_test:
+            test_cond = F.pad(
+                cond, (0, 0, cond_seq_len, max_duration - 2 * cond_seq_len), value=0.0
+            )
+
+        cond = F.pad(cond, (0, 0, 0, max_duration - cond_seq_len), value=0.0)
+        cond_mask = F.pad(
+            cond_mask, (0, max_duration - cond_mask.shape[-1]), value=False
+        )
+        cond_mask = cond_mask.unsqueeze(-1)
+        step_cond = torch.where(
+            cond_mask, cond, torch.zeros_like(cond)
+        )  # allow direct control (cut cond audio) with lens passed in
+
+        if batch > 1:
+            mask = lens_to_mask(duration)
+        else:  # save memory and speed up, as single inference need no mask currently
+            mask = None
+
+        # test for no ref audio
+        if no_ref_audio:
+            cond = torch.zeros_like(cond)
+
+        # neural ode
+
+        def fn(t, x):
+            # at each step, conditioning is fixed
+            # step_cond = torch.where(cond_mask, cond, torch.zeros_like(cond))
+
+            # predict flow
+            pred = self.transformer(
+                x=x,
+                cond=step_cond,
+                text=text,
+                time=t,
+                mask=mask,
+                drop_audio_cond=False,
+                drop_text=False,
+            )
+            if cfg_strength < 1e-5:
+                return pred
+
+            null_pred = self.transformer(
+                x=x,
+                cond=step_cond,
+                text=text,
+                time=t,
+                mask=mask,
+                drop_audio_cond=True,
+                drop_text=True,
+            )
+            return pred + (pred - null_pred) * cfg_strength
+
+        # noise input
+        # to make sure batch inference result is same with different batch size, and for sure single inference
+        # still some difference maybe due to convolutional layers
+        y0 = []
+        for dur in duration:
+            if exists(seed):
+                torch.manual_seed(seed)
+            y0.append(
+                torch.randn(
+                    dur, self.num_channels, device=self.device, dtype=step_cond.dtype
+                )
+            )
+        y0 = pad_sequence(y0, padding_value=0, batch_first=True)
+
+        t_start = 0
+
+        # duplicate test corner for inner time step oberservation
+        if duplicate_test:
+            t_start = t_inter
+            y0 = (1 - t_start) * y0 + t_start * test_cond
+            steps = int(steps * (1 - t_start))
+
+        t = torch.linspace(
+            t_start, 1, steps + 1, device=self.device, dtype=step_cond.dtype
+        )
+        if sway_sampling_coef is not None:
+            t = t + sway_sampling_coef * (torch.cos(torch.pi / 2 * t) - 1 + t)
+
+        trajectory = odeint(fn, y0, t, **self.odeint_kwargs)
+
+        sampled = trajectory[-1]
+        out = sampled
+        out = torch.where(cond_mask, cond, out)
+
+        if exists(vocoder):
+            out = out.permute(0, 2, 1)
+            out = vocoder(out)
+
+        return out, trajectory
+
+    def forward(
+        self,
+        inp: float["b n d"] | float["b nw"],  # mel or raw wave  # noqa: F722
+        text: int["b nt"] | list[str],  # noqa: F722
+        *,
+        lens: int["b"] | None = None,  # noqa: F821
+        noise_scheduler: str | None = None,
+    ):
+        # handle raw wave
+        if inp.ndim == 2:
+            inp = self.mel_spec(inp)
+            inp = inp.permute(0, 2, 1)
+            assert inp.shape[-1] == self.num_channels
+
+        batch, seq_len, dtype, device, _σ1 = (
+            *inp.shape[:2],
+            inp.dtype,
+            self.device,
+            self.sigma,
+        )
+
+        # handle text as string
+        if isinstance(text, list):
+            if exists(self.vocab_char_map):
+                text = list_str_to_idx(text, self.vocab_char_map).to(device)
+            else:
+                text = list_str_to_tensor(text).to(device)
+            assert text.shape[0] == batch
+
+        # lens and mask
+        if not exists(lens):
+            lens = torch.full((batch,), seq_len, device=device)
+
+        mask = lens_to_mask(
+            lens, length=seq_len
+        )  # useless here, as collate_fn will pad to max length in batch
+
+        # get a random span to mask out for training conditionally
+        frac_lengths = (
+            torch.zeros((batch,), device=self.device)
+            .float()
+            .uniform_(*self.frac_lengths_mask)
+        )
+        rand_span_mask = mask_from_frac_lengths(lens, frac_lengths)
+
+        if exists(mask):
+            rand_span_mask &= mask
+
+        # mel is x1
+        x1 = inp
+
+        # x0 is gaussian noise
+        x0 = torch.randn_like(x1)
+
+        # time step
+        time = torch.rand((batch,), dtype=dtype, device=self.device)
+        # TODO. noise_scheduler
+
+        # sample xt (φ_t(x) in the paper)
+        t = time.unsqueeze(-1).unsqueeze(-1)
+        φ = (1 - t) * x0 + t * x1
+        flow = x1 - x0
+
+        # only predict what is within the random mask span for infilling
+        cond = torch.where(rand_span_mask[..., None], torch.zeros_like(x1), x1)
+
+        # transformer and cfg training with a drop rate
+        drop_audio_cond = random() < self.audio_drop_prob  # p_drop in voicebox paper
+        if random() < self.cond_drop_prob:  # p_uncond in voicebox paper
+            drop_audio_cond = True
+            drop_text = True
+        else:
+            drop_text = False
+
+        # if want rigourously mask out padding, record in collate_fn in dataset.py, and pass in here
+        # adding mask will use more memory, thus also need to adjust batchsampler with scaled down threshold for long sequences
+        pred = self.transformer(
+            x=φ,
+            cond=cond,
+            text=text,
+            time=time,
+            drop_audio_cond=drop_audio_cond,
+            drop_text=drop_text,
+        )
+
+        # flow matching loss
+        loss = F.mse_loss(pred, flow, reduction="none")
+        loss = loss[rand_span_mask]
+
+        return loss.mean(), cond, pred

egs/wenetspeech4tts/TTS/f5-tts/model/dit.py

@@ -0,0 +1,210 @@
+"""
+ein notation:
+b - batch
+n - sequence
+nt - text sequence
+nw - raw wave length
+d - dimension
+"""
+
+from __future__ import annotations
+
+import torch
+import torch.nn.functional as F
+from model.modules import (
+    AdaLayerNormZero_Final,
+    ConvNeXtV2Block,
+    ConvPositionEmbedding,
+    DiTBlock,
+    TimestepEmbedding,
+    get_pos_embed_indices,
+    precompute_freqs_cis,
+)
+from torch import nn
+from x_transformers.x_transformers import RotaryEmbedding
+
+# Text embedding
+
+
+class TextEmbedding(nn.Module):
+    def __init__(self, text_num_embeds, text_dim, conv_layers=0, conv_mult=2):
+        super().__init__()
+        self.text_embed = nn.Embedding(
+            text_num_embeds + 1, text_dim
+        )  # use 0 as filler token
+
+        if conv_layers > 0:
+            self.extra_modeling = True
+            self.precompute_max_pos = 4096  # ~44s of 24khz audio
+            self.register_buffer(
+                "freqs_cis",
+                precompute_freqs_cis(text_dim, self.precompute_max_pos),
+                persistent=False,
+            )
+            self.text_blocks = nn.Sequential(
+                *[
+                    ConvNeXtV2Block(text_dim, text_dim * conv_mult)
+                    for _ in range(conv_layers)
+                ]
+            )
+        else:
+            self.extra_modeling = False
+
+    def forward(self, text: int["b nt"], seq_len, drop_text=False):  # noqa: F722
+        text = (
+            text + 1
+        )  # use 0 as filler token. preprocess of batch pad -1, see list_str_to_idx()
+        text = text[
+            :, :seq_len
+        ]  # curtail if character tokens are more than the mel spec tokens
+        batch, text_len = text.shape[0], text.shape[1]
+        text = F.pad(text, (0, seq_len - text_len), value=0)
+
+        if drop_text:  # cfg for text
+            text = torch.zeros_like(text)
+
+        text = self.text_embed(text)  # b n -> b n d
+
+        # possible extra modeling
+        if self.extra_modeling:
+            # sinus pos emb
+            batch_start = torch.zeros((batch,), dtype=torch.long)
+            pos_idx = get_pos_embed_indices(
+                batch_start, seq_len, max_pos=self.precompute_max_pos
+            )
+            text_pos_embed = self.freqs_cis[pos_idx]
+            text = text + text_pos_embed
+
+            # convnextv2 blocks
+            text = self.text_blocks(text)
+
+        return text
+
+
+# noised input audio and context mixing embedding
+
+
+class InputEmbedding(nn.Module):
+    def __init__(self, mel_dim, text_dim, out_dim):
+        super().__init__()
+        self.proj = nn.Linear(mel_dim * 2 + text_dim, out_dim)
+        self.conv_pos_embed = ConvPositionEmbedding(dim=out_dim)
+
+    def forward(
+        self,
+        x: float["b n d"],  # noqa: F722
+        cond: float["b n d"],  # noqa: F722
+        text_embed: float["b n d"],  # noqa: F722
+        drop_audio_cond=False,
+    ):
+        if drop_audio_cond:  # cfg for cond audio
+            cond = torch.zeros_like(cond)
+
+        x = self.proj(torch.cat((x, cond, text_embed), dim=-1))
+        x = self.conv_pos_embed(x) + x
+        return x
+
+
+# Transformer backbone using DiT blocks
+
+
+class DiT(nn.Module):
+    def __init__(
+        self,
+        *,
+        dim,
+        depth=8,
+        heads=8,
+        dim_head=64,
+        dropout=0.1,
+        ff_mult=4,
+        mel_dim=100,
+        text_num_embeds=256,
+        text_dim=None,
+        conv_layers=0,
+        long_skip_connection=False,
+        checkpoint_activations=False,
+    ):
+        super().__init__()
+
+        self.time_embed = TimestepEmbedding(dim)
+        if text_dim is None:
+            text_dim = mel_dim
+        self.text_embed = TextEmbedding(
+            text_num_embeds, text_dim, conv_layers=conv_layers
+        )
+        self.input_embed = InputEmbedding(mel_dim, text_dim, dim)
+
+        self.rotary_embed = RotaryEmbedding(dim_head)
+
+        self.dim = dim
+        self.depth = depth
+
+        self.transformer_blocks = nn.ModuleList(
+            [
+                DiTBlock(
+                    dim=dim,
+                    heads=heads,
+                    dim_head=dim_head,
+                    ff_mult=ff_mult,
+                    dropout=dropout,
+                )
+                for _ in range(depth)
+            ]
+        )
+        self.long_skip_connection = (
+            nn.Linear(dim * 2, dim, bias=False) if long_skip_connection else None
+        )
+
+        self.norm_out = AdaLayerNormZero_Final(dim)  # final modulation
+        self.proj_out = nn.Linear(dim, mel_dim)
+
+        self.checkpoint_activations = checkpoint_activations
+
+    def ckpt_wrapper(self, module):
+        # https://github.com/chuanyangjin/fast-DiT/blob/main/models.py
+        def ckpt_forward(*inputs):
+            outputs = module(*inputs)
+            return outputs
+
+        return ckpt_forward
+
+    def forward(
+        self,
+        x: float["b n d"],  # nosied input audio  # noqa: F722
+        cond: float["b n d"],  # masked cond audio  # noqa: F722
+        text: int["b nt"],  # text  # noqa: F722
+        time: float["b"] | float[""],  # time step  # noqa: F821 F722
+        drop_audio_cond,  # cfg for cond audio
+        drop_text,  # cfg for text
+        mask: bool["b n"] | None = None,  # noqa: F722
+    ):
+        batch, seq_len = x.shape[0], x.shape[1]
+        if time.ndim == 0:
+            time = time.repeat(batch)
+
+        # t: conditioning time, c: context (text + masked cond audio), x: noised input audio
+        t = self.time_embed(time)
+        text_embed = self.text_embed(text, seq_len, drop_text=drop_text)
+        x = self.input_embed(x, cond, text_embed, drop_audio_cond=drop_audio_cond)
+
+        rope = self.rotary_embed.forward_from_seq_len(seq_len)
+
+        if self.long_skip_connection is not None:
+            residual = x
+
+        for block in self.transformer_blocks:
+            if self.checkpoint_activations:
+                x = torch.utils.checkpoint.checkpoint(
+                    self.ckpt_wrapper(block), x, t, mask, rope
+                )
+            else:
+                x = block(x, t, mask=mask, rope=rope)
+
+        if self.long_skip_connection is not None:
+            x = self.long_skip_connection(torch.cat((x, residual), dim=-1))
+
+        x = self.norm_out(x, t)
+        output = self.proj_out(x)
+
+        return output

egs/wenetspeech4tts/TTS/f5-tts/model/modules.py

@@ -0,0 +1,728 @@
+"""
+ein notation:
+b - batch
+n - sequence
+nt - text sequence
+nw - raw wave length
+d - dimension
+"""
+
+from __future__ import annotations
+
+import math
+from typing import Optional
+
+import torch
+import torch.nn.functional as F
+import torchaudio
+from librosa.filters import mel as librosa_mel_fn
+from torch import nn
+from x_transformers.x_transformers import apply_rotary_pos_emb
+
+# raw wav to mel spec
+
+
+mel_basis_cache = {}
+hann_window_cache = {}
+
+
+def get_bigvgan_mel_spectrogram(
+    waveform,
+    n_fft=1024,
+    n_mel_channels=100,
+    target_sample_rate=24000,
+    hop_length=256,
+    win_length=1024,
+    fmin=0,
+    fmax=None,
+    center=False,
+):  # Copy from https://github.com/NVIDIA/BigVGAN/tree/main
+    device = waveform.device
+    key = f"{n_fft}_{n_mel_channels}_{target_sample_rate}_{hop_length}_{win_length}_{fmin}_{fmax}_{device}"
+
+    if key not in mel_basis_cache:
+        mel = librosa_mel_fn(
+            sr=target_sample_rate,
+            n_fft=n_fft,
+            n_mels=n_mel_channels,
+            fmin=fmin,
+            fmax=fmax,
+        )
+        mel_basis_cache[key] = (
+            torch.from_numpy(mel).float().to(device)
+        )  # TODO: why they need .float()?
+        hann_window_cache[key] = torch.hann_window(win_length).to(device)
+
+    mel_basis = mel_basis_cache[key]
+    hann_window = hann_window_cache[key]
+
+    padding = (n_fft - hop_length) // 2
+    waveform = torch.nn.functional.pad(
+        waveform.unsqueeze(1), (padding, padding), mode="reflect"
+    ).squeeze(1)
+
+    spec = torch.stft(
+        waveform,
+        n_fft,
+        hop_length=hop_length,
+        win_length=win_length,
+        window=hann_window,
+        center=center,
+        pad_mode="reflect",
+        normalized=False,
+        onesided=True,
+        return_complex=True,
+    )
+    spec = torch.sqrt(torch.view_as_real(spec).pow(2).sum(-1) + 1e-9)
+
+    mel_spec = torch.matmul(mel_basis, spec)
+    mel_spec = torch.log(torch.clamp(mel_spec, min=1e-5))
+
+    return mel_spec
+
+
+def get_vocos_mel_spectrogram(
+    waveform,
+    n_fft=1024,
+    n_mel_channels=100,
+    target_sample_rate=24000,
+    hop_length=256,
+    win_length=1024,
+):
+    mel_stft = torchaudio.transforms.MelSpectrogram(
+        sample_rate=target_sample_rate,
+        n_fft=n_fft,
+        win_length=win_length,
+        hop_length=hop_length,
+        n_mels=n_mel_channels,
+        power=1,
+        center=True,
+        normalized=False,
+        norm=None,
+    ).to(waveform.device)
+    if len(waveform.shape) == 3:
+        waveform = waveform.squeeze(1)  # 'b 1 nw -> b nw'
+
+    assert len(waveform.shape) == 2
+
+    mel = mel_stft(waveform)
+    mel = mel.clamp(min=1e-5).log()
+    return mel
+
+
+class MelSpec(nn.Module):
+    def __init__(
+        self,
+        n_fft=1024,
+        hop_length=256,
+        win_length=1024,
+        n_mel_channels=100,
+        target_sample_rate=24_000,
+        mel_spec_type="vocos",
+    ):
+        super().__init__()
+        assert mel_spec_type in ["vocos", "bigvgan"], print(
+            "We only support two extract mel backend: vocos or bigvgan"
+        )
+
+        self.n_fft = n_fft
+        self.hop_length = hop_length
+        self.win_length = win_length
+        self.n_mel_channels = n_mel_channels
+        self.target_sample_rate = target_sample_rate
+
+        if mel_spec_type == "vocos":
+            self.extractor = get_vocos_mel_spectrogram
+        elif mel_spec_type == "bigvgan":
+            self.extractor = get_bigvgan_mel_spectrogram
+
+        self.register_buffer("dummy", torch.tensor(0), persistent=False)
+
+    def forward(self, wav):
+        if self.dummy.device != wav.device:
+            self.to(wav.device)
+
+        mel = self.extractor(
+            waveform=wav,
+            n_fft=self.n_fft,
+            n_mel_channels=self.n_mel_channels,
+            target_sample_rate=self.target_sample_rate,
+            hop_length=self.hop_length,
+            win_length=self.win_length,
+        )
+
+        return mel
+
+
+# sinusoidal position embedding
+
+
+class SinusPositionEmbedding(nn.Module):
+    def __init__(self, dim):
+        super().__init__()
+        self.dim = dim
+
+    def forward(self, x, scale=1000):
+        device = x.device
+        half_dim = self.dim // 2
+        emb = math.log(10000) / (half_dim - 1)
+        emb = torch.exp(torch.arange(half_dim, device=device).float() * -emb)
+        emb = scale * x.unsqueeze(1) * emb.unsqueeze(0)
+        emb = torch.cat((emb.sin(), emb.cos()), dim=-1)
+        return emb
+
+
+# convolutional position embedding
+
+
+class ConvPositionEmbedding(nn.Module):
+    def __init__(self, dim, kernel_size=31, groups=16):
+        super().__init__()
+        assert kernel_size % 2 != 0
+        self.conv1d = nn.Sequential(
+            nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=kernel_size // 2),
+            nn.Mish(),
+            nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=kernel_size // 2),
+            nn.Mish(),
+        )
+
+    def forward(self, x: float["b n d"], mask: bool["b n"] | None = None):  # noqa: F722
+        if mask is not None:
+            mask = mask[..., None]
+            x = x.masked_fill(~mask, 0.0)
+
+        x = x.permute(0, 2, 1)
+        x = self.conv1d(x)
+        out = x.permute(0, 2, 1)
+
+        if mask is not None:
+            out = out.masked_fill(~mask, 0.0)
+
+        return out
+
+
+# rotary positional embedding related
+
+
+def precompute_freqs_cis(
+    dim: int, end: int, theta: float = 10000.0, theta_rescale_factor=1.0
+):
+    # proposed by reddit user bloc97, to rescale rotary embeddings to longer sequence length without fine-tuning
+    # has some connection to NTK literature
+    # https://www.reddit.com/r/LocalLLaMA/comments/14lz7j5/ntkaware_scaled_rope_allows_llama_models_to_have/
+    # https://github.com/lucidrains/rotary-embedding-torch/blob/main/rotary_embedding_torch/rotary_embedding_torch.py
+    theta *= theta_rescale_factor ** (dim / (dim - 2))
+    freqs = 1.0 / (theta ** (torch.arange(0, dim, 2)[: (dim // 2)].float() / dim))
+    t = torch.arange(end, device=freqs.device)  # type: ignore
+    freqs = torch.outer(t, freqs).float()  # type: ignore
+    freqs_cos = torch.cos(freqs)  # real part
+    freqs_sin = torch.sin(freqs)  # imaginary part
+    return torch.cat([freqs_cos, freqs_sin], dim=-1)
+
+
+def get_pos_embed_indices(start, length, max_pos, scale=1.0):
+    # length = length if isinstance(length, int) else length.max()
+    scale = scale * torch.ones_like(
+        start, dtype=torch.float32
+    )  # in case scale is a scalar
+    pos = (
+        start.unsqueeze(1)
+        + (
+            torch.arange(length, device=start.device, dtype=torch.float32).unsqueeze(0)
+            * scale.unsqueeze(1)
+        ).long()
+    )
+    # avoid extra long error.
+    pos = torch.where(pos < max_pos, pos, max_pos - 1)
+    return pos
+
+
+# Global Response Normalization layer (Instance Normalization ?)
+
+
+class GRN(nn.Module):
+    def __init__(self, dim):
+        super().__init__()
+        self.gamma = nn.Parameter(torch.zeros(1, 1, dim))
+        self.beta = nn.Parameter(torch.zeros(1, 1, dim))
+
+    def forward(self, x):
+        Gx = torch.norm(x, p=2, dim=1, keepdim=True)
+        Nx = Gx / (Gx.mean(dim=-1, keepdim=True) + 1e-6)
+        return self.gamma * (x * Nx) + self.beta + x
+
+
+# ConvNeXt-V2 Block https://github.com/facebookresearch/ConvNeXt-V2/blob/main/models/convnextv2.py
+# ref: https://github.com/bfs18/e2_tts/blob/main/rfwave/modules.py#L108
+
+
+class ConvNeXtV2Block(nn.Module):
+    def __init__(
+        self,
+        dim: int,
+        intermediate_dim: int,
+        dilation: int = 1,
+    ):
+        super().__init__()
+        padding = (dilation * (7 - 1)) // 2
+        self.dwconv = nn.Conv1d(
+            dim, dim, kernel_size=7, padding=padding, groups=dim, dilation=dilation
+        )  # depthwise conv
+        self.norm = nn.LayerNorm(dim, eps=1e-6)
+        self.pwconv1 = nn.Linear(
+            dim, intermediate_dim
+        )  # pointwise/1x1 convs, implemented with linear layers
+        self.act = nn.GELU()
+        self.grn = GRN(intermediate_dim)
+        self.pwconv2 = nn.Linear(intermediate_dim, dim)
+
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
+        residual = x
+        x = x.transpose(1, 2)  # b n d -> b d n
+        x = self.dwconv(x)
+        x = x.transpose(1, 2)  # b d n -> b n d
+        x = self.norm(x)
+        x = self.pwconv1(x)
+        x = self.act(x)
+        x = self.grn(x)
+        x = self.pwconv2(x)
+        return residual + x
+
+
+# AdaLayerNormZero
+# return with modulated x for attn input, and params for later mlp modulation
+
+
+class AdaLayerNormZero(nn.Module):
+    def __init__(self, dim):
+        super().__init__()
+
+        self.silu = nn.SiLU()
+        self.linear = nn.Linear(dim, dim * 6)
+
+        self.norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+
+    def forward(self, x, emb=None):
+        emb = self.linear(self.silu(emb))
+        shift_msa, scale_msa, gate_msa, shift_mlp, scale_mlp, gate_mlp = torch.chunk(
+            emb, 6, dim=1
+        )
+
+        x = self.norm(x) * (1 + scale_msa[:, None]) + shift_msa[:, None]
+        return x, gate_msa, shift_mlp, scale_mlp, gate_mlp
+
+
+# AdaLayerNormZero for final layer
+# return only with modulated x for attn input, cuz no more mlp modulation
+
+
+class AdaLayerNormZero_Final(nn.Module):
+    def __init__(self, dim):
+        super().__init__()
+
+        self.silu = nn.SiLU()
+        self.linear = nn.Linear(dim, dim * 2)
+
+        self.norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+
+    def forward(self, x, emb):
+        emb = self.linear(self.silu(emb))
+        scale, shift = torch.chunk(emb, 2, dim=1)
+
+        x = self.norm(x) * (1 + scale)[:, None, :] + shift[:, None, :]
+        return x
+
+
+# FeedForward
+
+
+class FeedForward(nn.Module):
+    def __init__(
+        self, dim, dim_out=None, mult=4, dropout=0.0, approximate: str = "none"
+    ):
+        super().__init__()
+        inner_dim = int(dim * mult)
+        dim_out = dim_out if dim_out is not None else dim
+
+        activation = nn.GELU(approximate=approximate)
+        project_in = nn.Sequential(nn.Linear(dim, inner_dim), activation)
+        self.ff = nn.Sequential(
+            project_in, nn.Dropout(dropout), nn.Linear(inner_dim, dim_out)
+        )
+
+    def forward(self, x):
+        return self.ff(x)
+
+
+# Attention with possible joint part
+# modified from diffusers/src/diffusers/models/attention_processor.py
+
+
+class Attention(nn.Module):
+    def __init__(
+        self,
+        processor: JointAttnProcessor | AttnProcessor,
+        dim: int,
+        heads: int = 8,
+        dim_head: int = 64,
+        dropout: float = 0.0,
+        context_dim: Optional[int] = None,  # if not None -> joint attention
+        context_pre_only=None,
+    ):
+        super().__init__()
+
+        if not hasattr(F, "scaled_dot_product_attention"):
+            raise ImportError(
+                "Attention equires PyTorch 2.0, to use it, please upgrade PyTorch to 2.0."
+            )
+
+        self.processor = processor
+
+        self.dim = dim
+        self.heads = heads
+        self.inner_dim = dim_head * heads
+        self.dropout = dropout
+
+        self.context_dim = context_dim
+        self.context_pre_only = context_pre_only
+
+        self.to_q = nn.Linear(dim, self.inner_dim)
+        self.to_k = nn.Linear(dim, self.inner_dim)
+        self.to_v = nn.Linear(dim, self.inner_dim)
+
+        if self.context_dim is not None:
+            self.to_k_c = nn.Linear(context_dim, self.inner_dim)
+            self.to_v_c = nn.Linear(context_dim, self.inner_dim)
+            if self.context_pre_only is not None:
+                self.to_q_c = nn.Linear(context_dim, self.inner_dim)
+
+        self.to_out = nn.ModuleList([])
+        self.to_out.append(nn.Linear(self.inner_dim, dim))
+        self.to_out.append(nn.Dropout(dropout))
+
+        if self.context_pre_only is not None and not self.context_pre_only:
+            self.to_out_c = nn.Linear(self.inner_dim, dim)
+
+    def forward(
+        self,
+        x: float["b n d"],  # noised input x  # noqa: F722
+        c: float["b n d"] = None,  # context c  # noqa: F722
+        mask: bool["b n"] | None = None,  # noqa: F722
+        rope=None,  # rotary position embedding for x
+        c_rope=None,  # rotary position embedding for c
+    ) -> torch.Tensor:
+        if c is not None:
+            return self.processor(self, x, c=c, mask=mask, rope=rope, c_rope=c_rope)
+        else:
+            return self.processor(self, x, mask=mask, rope=rope)
+
+
+# Attention processor
+
+
+class AttnProcessor:
+    def __init__(self):
+        pass
+
+    def __call__(
+        self,
+        attn: Attention,
+        x: float["b n d"],  # noised input x  # noqa: F722
+        mask: bool["b n"] | None = None,  # noqa: F722
+        rope=None,  # rotary position embedding
+    ) -> torch.FloatTensor:
+        batch_size = x.shape[0]
+
+        # `sample` projections.
+        query = attn.to_q(x)
+        key = attn.to_k(x)
+        value = attn.to_v(x)
+
+        # apply rotary position embedding
+        if rope is not None:
+            freqs, xpos_scale = rope
+            q_xpos_scale, k_xpos_scale = (
+                (xpos_scale, xpos_scale**-1.0)
+                if xpos_scale is not None
+                else (1.0, 1.0)
+            )
+
+            query = apply_rotary_pos_emb(query, freqs, q_xpos_scale)
+            key = apply_rotary_pos_emb(key, freqs, k_xpos_scale)
+
+        # attention
+        inner_dim = key.shape[-1]
+        head_dim = inner_dim // attn.heads
+        query = query.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+        key = key.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+        value = value.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+
+        # mask. e.g. inference got a batch with different target durations, mask out the padding
+        if mask is not None:
+            attn_mask = mask
+            attn_mask = attn_mask.unsqueeze(1).unsqueeze(1)  # 'b n -> b 1 1 n'
+            attn_mask = attn_mask.expand(
+                batch_size, attn.heads, query.shape[-2], key.shape[-2]
+            )
+        else:
+            attn_mask = None
+
+        x = F.scaled_dot_product_attention(
+            query, key, value, attn_mask=attn_mask, dropout_p=0.0, is_causal=False
+        )
+        x = x.transpose(1, 2).reshape(batch_size, -1, attn.heads * head_dim)
+        x = x.to(query.dtype)
+
+        # linear proj
+        x = attn.to_out[0](x)
+        # dropout
+        x = attn.to_out[1](x)
+
+        if mask is not None:
+            mask = mask.unsqueeze(-1)
+            x = x.masked_fill(~mask, 0.0)
+
+        return x
+
+
+# Joint Attention processor for MM-DiT
+# modified from diffusers/src/diffusers/models/attention_processor.py
+
+
+class JointAttnProcessor:
+    def __init__(self):
+        pass
+
+    def __call__(
+        self,
+        attn: Attention,
+        x: float["b n d"],  # noised input x  # noqa: F722
+        c: float["b nt d"] = None,  # context c, here text # noqa: F722
+        mask: bool["b n"] | None = None,  # noqa: F722
+        rope=None,  # rotary position embedding for x
+        c_rope=None,  # rotary position embedding for c
+    ) -> torch.FloatTensor:
+        residual = x
+
+        batch_size = c.shape[0]
+
+        # `sample` projections.
+        query = attn.to_q(x)
+        key = attn.to_k(x)
+        value = attn.to_v(x)
+
+        # `context` projections.
+        c_query = attn.to_q_c(c)
+        c_key = attn.to_k_c(c)
+        c_value = attn.to_v_c(c)
+
+        # apply rope for context and noised input independently
+        if rope is not None:
+            freqs, xpos_scale = rope
+            q_xpos_scale, k_xpos_scale = (
+                (xpos_scale, xpos_scale**-1.0)
+                if xpos_scale is not None
+                else (1.0, 1.0)
+            )
+            query = apply_rotary_pos_emb(query, freqs, q_xpos_scale)
+            key = apply_rotary_pos_emb(key, freqs, k_xpos_scale)
+        if c_rope is not None:
+            freqs, xpos_scale = c_rope
+            q_xpos_scale, k_xpos_scale = (
+                (xpos_scale, xpos_scale**-1.0)
+                if xpos_scale is not None
+                else (1.0, 1.0)
+            )
+            c_query = apply_rotary_pos_emb(c_query, freqs, q_xpos_scale)
+            c_key = apply_rotary_pos_emb(c_key, freqs, k_xpos_scale)
+
+        # attention
+        query = torch.cat([query, c_query], dim=1)
+        key = torch.cat([key, c_key], dim=1)
+        value = torch.cat([value, c_value], dim=1)
+
+        inner_dim = key.shape[-1]
+        head_dim = inner_dim // attn.heads
+        query = query.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+        key = key.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+        value = value.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+
+        # mask. e.g. inference got a batch with different target durations, mask out the padding
+        if mask is not None:
+            attn_mask = F.pad(mask, (0, c.shape[1]), value=True)  # no mask for c (text)
+            attn_mask = attn_mask.unsqueeze(1).unsqueeze(1)  # 'b n -> b 1 1 n'
+            attn_mask = attn_mask.expand(
+                batch_size, attn.heads, query.shape[-2], key.shape[-2]
+            )
+        else:
+            attn_mask = None
+
+        x = F.scaled_dot_product_attention(
+            query, key, value, attn_mask=attn_mask, dropout_p=0.0, is_causal=False
+        )
+        x = x.transpose(1, 2).reshape(batch_size, -1, attn.heads * head_dim)
+        x = x.to(query.dtype)
+
+        # Split the attention outputs.
+        x, c = (
+            x[:, : residual.shape[1]],
+            x[:, residual.shape[1] :],
+        )
+
+        # linear proj
+        x = attn.to_out[0](x)
+        # dropout
+        x = attn.to_out[1](x)
+        if not attn.context_pre_only:
+            c = attn.to_out_c(c)
+
+        if mask is not None:
+            mask = mask.unsqueeze(-1)
+            x = x.masked_fill(~mask, 0.0)
+            # c = c.masked_fill(~mask, 0.)  # no mask for c (text)
+
+        return x, c
+
+
+# DiT Block
+
+
+class DiTBlock(nn.Module):
+    def __init__(self, dim, heads, dim_head, ff_mult=4, dropout=0.1):
+        super().__init__()
+
+        self.attn_norm = AdaLayerNormZero(dim)
+        self.attn = Attention(
+            processor=AttnProcessor(),
+            dim=dim,
+            heads=heads,
+            dim_head=dim_head,
+            dropout=dropout,
+        )
+
+        self.ff_norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+        self.ff = FeedForward(
+            dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh"
+        )
+
+    def forward(self, x, t, mask=None, rope=None):  # x: noised input, t: time embedding
+        # pre-norm & modulation for attention input
+        norm, gate_msa, shift_mlp, scale_mlp, gate_mlp = self.attn_norm(x, emb=t)
+
+        # attention
+        attn_output = self.attn(x=norm, mask=mask, rope=rope)
+
+        # process attention output for input x
+        x = x + gate_msa.unsqueeze(1) * attn_output
+
+        norm = self.ff_norm(x) * (1 + scale_mlp[:, None]) + shift_mlp[:, None]
+        ff_output = self.ff(norm)
+        x = x + gate_mlp.unsqueeze(1) * ff_output
+
+        return x
+
+
+# MMDiT Block https://arxiv.org/abs/2403.03206
+
+
+class MMDiTBlock(nn.Module):
+    r"""
+    modified from diffusers/src/diffusers/models/attention.py
+
+    notes.
+    _c: context related. text, cond, etc. (left part in sd3 fig2.b)
+    _x: noised input related. (right part)
+    context_pre_only: last layer only do prenorm + modulation cuz no more ffn
+    """
+
+    def __init__(
+        self, dim, heads, dim_head, ff_mult=4, dropout=0.1, context_pre_only=False
+    ):
+        super().__init__()
+
+        self.context_pre_only = context_pre_only
+
+        self.attn_norm_c = (
+            AdaLayerNormZero_Final(dim) if context_pre_only else AdaLayerNormZero(dim)
+        )
+        self.attn_norm_x = AdaLayerNormZero(dim)
+        self.attn = Attention(
+            processor=JointAttnProcessor(),
+            dim=dim,
+            heads=heads,
+            dim_head=dim_head,
+            dropout=dropout,
+            context_dim=dim,
+            context_pre_only=context_pre_only,
+        )
+
+        if not context_pre_only:
+            self.ff_norm_c = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+            self.ff_c = FeedForward(
+                dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh"
+            )
+        else:
+            self.ff_norm_c = None
+            self.ff_c = None
+        self.ff_norm_x = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+        self.ff_x = FeedForward(
+            dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh"
+        )
+
+    def forward(
+        self, x, c, t, mask=None, rope=None, c_rope=None
+    ):  # x: noised input, c: context, t: time embedding
+        # pre-norm & modulation for attention input
+        if self.context_pre_only:
+            norm_c = self.attn_norm_c(c, t)
+        else:
+            norm_c, c_gate_msa, c_shift_mlp, c_scale_mlp, c_gate_mlp = self.attn_norm_c(
+                c, emb=t
+            )
+        norm_x, x_gate_msa, x_shift_mlp, x_scale_mlp, x_gate_mlp = self.attn_norm_x(
+            x, emb=t
+        )
+
+        # attention
+        x_attn_output, c_attn_output = self.attn(
+            x=norm_x, c=norm_c, mask=mask, rope=rope, c_rope=c_rope
+        )
+
+        # process attention output for context c
+        if self.context_pre_only:
+            c = None
+        else:  # if not last layer
+            c = c + c_gate_msa.unsqueeze(1) * c_attn_output
+
+            norm_c = (
+                self.ff_norm_c(c) * (1 + c_scale_mlp[:, None]) + c_shift_mlp[:, None]
+            )
+            c_ff_output = self.ff_c(norm_c)
+            c = c + c_gate_mlp.unsqueeze(1) * c_ff_output
+
+        # process attention output for input x
+        x = x + x_gate_msa.unsqueeze(1) * x_attn_output
+
+        norm_x = self.ff_norm_x(x) * (1 + x_scale_mlp[:, None]) + x_shift_mlp[:, None]
+        x_ff_output = self.ff_x(norm_x)
+        x = x + x_gate_mlp.unsqueeze(1) * x_ff_output
+
+        return c, x
+
+
+# time step conditioning embedding
+
+
+class TimestepEmbedding(nn.Module):
+    def __init__(self, dim, freq_embed_dim=256):
+        super().__init__()
+        self.time_embed = SinusPositionEmbedding(freq_embed_dim)
+        self.time_mlp = nn.Sequential(
+            nn.Linear(freq_embed_dim, dim), nn.SiLU(), nn.Linear(dim, dim)
+        )
+
+    def forward(self, timestep: float["b"]):  # noqa: F821
+        time_hidden = self.time_embed(timestep)
+        time_hidden = time_hidden.to(timestep.dtype)
+        time = self.time_mlp(time_hidden)  # b d
+        return time

egs/wenetspeech4tts/TTS/f5-tts/model/utils.py

@@ -0,0 +1,206 @@
+from __future__ import annotations
+
+import os
+import random
+from collections import defaultdict
+from importlib.resources import files
+
+import jieba
+import torch
+from pypinyin import Style, lazy_pinyin
+from torch.nn.utils.rnn import pad_sequence
+
+# seed everything
+
+
+def seed_everything(seed=0):
+    random.seed(seed)
+    os.environ["PYTHONHASHSEED"] = str(seed)
+    torch.manual_seed(seed)
+    torch.cuda.manual_seed(seed)
+    torch.cuda.manual_seed_all(seed)
+    torch.backends.cudnn.deterministic = True
+    torch.backends.cudnn.benchmark = False
+
+
+# helpers
+
+
+def exists(v):
+    return v is not None
+
+
+def default(v, d):
+    return v if exists(v) else d
+
+
+# tensor helpers
+
+
+def lens_to_mask(
+    t: int["b"], length: int | None = None  # noqa: F722 F821
+) -> bool["b n"]:  # noqa: F722 F821
+    if not exists(length):
+        length = t.amax()
+
+    seq = torch.arange(length, device=t.device)
+    return seq[None, :] < t[:, None]
+
+
+def mask_from_start_end_indices(
+    seq_len: int["b"], start: int["b"], end: int["b"]  # noqa: F722 F821
+):
+    max_seq_len = seq_len.max().item()
+    seq = torch.arange(max_seq_len, device=start.device).long()
+    start_mask = seq[None, :] >= start[:, None]
+    end_mask = seq[None, :] < end[:, None]
+    return start_mask & end_mask
+
+
+def mask_from_frac_lengths(
+    seq_len: int["b"], frac_lengths: float["b"]  # noqa: F722 F821
+):
+    lengths = (frac_lengths * seq_len).long()
+    max_start = seq_len - lengths
+
+    rand = torch.rand_like(frac_lengths)
+    start = (max_start * rand).long().clamp(min=0)
+    end = start + lengths
+
+    return mask_from_start_end_indices(seq_len, start, end)
+
+
+def maybe_masked_mean(
+    t: float["b n d"], mask: bool["b n"] = None  # noqa: F722 F821
+) -> float["b d"]:  # noqa: F722 F821
+    if not exists(mask):
+        return t.mean(dim=1)
+
+    t = torch.where(mask[:, :, None], t, torch.tensor(0.0, device=t.device))
+    num = t.sum(dim=1)
+    den = mask.float().sum(dim=1)
+
+    return num / den.clamp(min=1.0)
+
+
+# simple utf-8 tokenizer, since paper went character based
+def list_str_to_tensor(text: list[str], padding_value=-1) -> int["b nt"]:  # noqa: F722
+    list_tensors = [torch.tensor([*bytes(t, "UTF-8")]) for t in text]  # ByT5 style
+    text = pad_sequence(list_tensors, padding_value=padding_value, batch_first=True)
+    return text
+
+
+# char tokenizer, based on custom dataset's extracted .txt file
+def list_str_to_idx(
+    text: list[str] | list[list[str]],
+    vocab_char_map: dict[str, int],  # {char: idx}
+    padding_value=-1,
+) -> int["b nt"]:  # noqa: F722
+    list_idx_tensors = [
+        torch.tensor([vocab_char_map.get(c, 0) for c in t]) for t in text
+    ]  # pinyin or char style
+    text = pad_sequence(list_idx_tensors, padding_value=padding_value, batch_first=True)
+    return text
+
+
+# Get tokenizer
+
+
+def get_tokenizer(dataset_name, tokenizer: str = "pinyin"):
+    """
+    tokenizer   - "pinyin" do g2p for only chinese characters, need .txt vocab_file
+                - "char" for char-wise tokenizer, need .txt vocab_file
+                - "byte" for utf-8 tokenizer
+                - "custom" if you're directly passing in a path to the vocab.txt you want to use
+    vocab_size  - if use "pinyin", all available pinyin types, common alphabets (also those with accent) and symbols
+                - if use "char", derived from unfiltered character & symbol counts of custom dataset
+                - if use "byte", set to 256 (unicode byte range)
+    """
+    if tokenizer in ["pinyin", "char"]:
+        tokenizer_path = os.path.join(
+            files("f5_tts").joinpath("../../data"),
+            f"{dataset_name}_{tokenizer}/vocab.txt",
+        )
+        with open(tokenizer_path, "r", encoding="utf-8") as f:
+            vocab_char_map = {}
+            for i, char in enumerate(f):
+                vocab_char_map[char[:-1]] = i
+        vocab_size = len(vocab_char_map)
+        assert (
+            vocab_char_map[" "] == 0
+        ), "make sure space is of idx 0 in vocab.txt, cuz 0 is used for unknown char"
+
+    elif tokenizer == "byte":
+        vocab_char_map = None
+        vocab_size = 256
+
+    elif tokenizer == "custom":
+        with open(dataset_name, "r", encoding="utf-8") as f:
+            vocab_char_map = {}
+            for i, char in enumerate(f):
+                vocab_char_map[char[:-1]] = i
+        vocab_size = len(vocab_char_map)
+
+    return vocab_char_map, vocab_size
+
+
+# convert char to pinyin
+
+jieba.initialize()
+print("Word segmentation module jieba initialized.\n")
+
+
+def convert_char_to_pinyin(text_list, polyphone=True):
+    final_text_list = []
+    custom_trans = str.maketrans(
+        {";": ",", "“": '"', "”": '"', "‘": "'", "’": "'"}
+    )  # add custom trans here, to address oov
+
+    def is_chinese(c):
+        return "\u3100" <= c <= "\u9fff"  # common chinese characters
+
+    for text in text_list:
+        char_list = []
+        text = text.translate(custom_trans)
+        for seg in jieba.cut(text):
+            seg_byte_len = len(bytes(seg, "UTF-8"))
+            if seg_byte_len == len(seg):  # if pure alphabets and symbols
+                if char_list and seg_byte_len > 1 and char_list[-1] not in " :'\"":
+                    char_list.append(" ")
+                char_list.extend(seg)
+            elif polyphone and seg_byte_len == 3 * len(
+                seg
+            ):  # if pure east asian characters
+                seg_ = lazy_pinyin(seg, style=Style.TONE3, tone_sandhi=True)
+                for i, c in enumerate(seg):
+                    if is_chinese(c):
+                        char_list.append(" ")
+                    char_list.append(seg_[i])
+            else:  # if mixed characters, alphabets and symbols
+                for c in seg:
+                    if ord(c) < 256:
+                        char_list.extend(c)
+                    elif is_chinese(c):
+                        char_list.append(" ")
+                        char_list.extend(
+                            lazy_pinyin(c, style=Style.TONE3, tone_sandhi=True)
+                        )
+                    else:
+                        char_list.append(c)
+        final_text_list.append(char_list)
+
+    return final_text_list
+
+
+# filter func for dirty data with many repetitions
+
+
+def repetition_found(text, length=2, tolerance=10):
+    pattern_count = defaultdict(int)
+    for i in range(len(text) - length + 1):
+        pattern = text[i : i + length]
+        pattern_count[pattern] += 1
+    for pattern, count in pattern_count.items():
+        if count > tolerance:
+            return True
+    return False

egs/wenetspeech4tts/TTS/f5-tts/speech_synthesis.py

@@ -0,0 +1,104 @@
+from typing import Callable, Dict, List, Sequence, Union
+
+import torch
+from lhotse import validate
+from lhotse.cut import CutSet
+from lhotse.dataset.collation import collate_audio
+from lhotse.dataset.input_strategies import BatchIO, PrecomputedFeatures
+from lhotse.utils import ifnone
+
+
+class SpeechSynthesisDataset(torch.utils.data.Dataset):
+    """
+    The PyTorch Dataset for the speech synthesis task.
+    Each item in this dataset is a dict of:
+
+    .. code-block::
+
+        {
+            'audio': (B x NumSamples) float tensor
+            'features': (B x NumFrames x NumFeatures) float tensor
+            'audio_lens': (B, ) int tensor
+            'features_lens': (B, ) int tensor
+            'text': List[str] of len B  # when return_text=True
+            'tokens': List[List[str]]  # when return_tokens=True
+            'speakers': List[str] of len B  # when return_spk_ids=True
+            'cut': List of Cuts  # when return_cuts=True
+        }
+    """
+
+    def __init__(
+        self,
+        cut_transforms: List[Callable[[CutSet], CutSet]] = None,
+        feature_input_strategy: BatchIO = PrecomputedFeatures(),
+        feature_transforms: Union[Sequence[Callable], Callable] = None,
+        return_text: bool = True,
+        return_tokens: bool = False,
+        return_spk_ids: bool = False,
+        return_cuts: bool = False,
+    ) -> None:
+        super().__init__()
+
+        self.cut_transforms = ifnone(cut_transforms, [])
+        self.feature_input_strategy = feature_input_strategy
+
+        self.return_text = return_text
+        self.return_tokens = return_tokens
+        self.return_spk_ids = return_spk_ids
+        self.return_cuts = return_cuts
+
+        if feature_transforms is None:
+            feature_transforms = []
+        elif not isinstance(feature_transforms, Sequence):
+            feature_transforms = [feature_transforms]
+
+        assert all(
+            isinstance(transform, Callable) for transform in feature_transforms
+        ), "Feature transforms must be Callable"
+        self.feature_transforms = feature_transforms
+
+    def __getitem__(self, cuts: CutSet) -> Dict[str, torch.Tensor]:
+        validate_for_tts(cuts)
+
+        for transform in self.cut_transforms:
+            cuts = transform(cuts)
+
+        # audio, audio_lens = collate_audio(cuts)
+        features, features_lens = self.feature_input_strategy(cuts)
+
+        for transform in self.feature_transforms:
+            features = transform(features)
+
+        batch = {
+            #     "audio": audio,
+            "features": features,
+            #    "audio_lens": audio_lens,
+            "features_lens": features_lens,
+        }
+
+        if self.return_text:
+            # use normalized text
+            # text = [cut.supervisions[0].normalized_text for cut in cuts]
+            text = [cut.supervisions[0].text for cut in cuts]
+            batch["text"] = text
+
+        if self.return_tokens:
+            # tokens = [cut.tokens for cut in cuts]
+            tokens = [cut.supervisions[0].custom["tokens"]["text"] for cut in cuts]
+            batch["tokens"] = tokens
+
+        if self.return_spk_ids:
+            batch["speakers"] = [cut.supervisions[0].speaker for cut in cuts]
+
+        if self.return_cuts:
+            batch["cut"] = [cut for cut in cuts]
+
+        return batch
+
+
+def validate_for_tts(cuts: CutSet) -> None:
+    validate(cuts)
+    for cut in cuts:
+        assert (
+            len(cut.supervisions) == 1
+        ), "Only the Cuts with single supervision are supported."

egs/wenetspeech4tts/TTS/f5-tts/tts_datamodule.py

@@ -0,0 +1,306 @@
+# Copyright      2021  Piotr Żelasko
+# Copyright      2022-2023  Xiaomi Corporation     (Authors: Mingshuang Luo,
+#                                                            Zengwei Yao)
+#
+# See ../../../../LICENSE for clarification regarding multiple authors
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+
+import argparse
+import logging
+from functools import lru_cache
+from pathlib import Path
+from typing import Any, Dict, Optional
+
+import torch
+from lhotse import CutSet, load_manifest_lazy
+from lhotse.dataset import (  # noqa F401 for PrecomputedFeatures; SpeechSynthesisDataset,
+    CutConcatenate,
+    CutMix,
+    DynamicBucketingSampler,
+    PrecomputedFeatures,
+    SimpleCutSampler,
+)
+from lhotse.dataset.input_strategies import (  # noqa F401 For AudioSamples
+    AudioSamples,
+    OnTheFlyFeatures,
+)
+from lhotse.utils import fix_random_seed
+from speech_synthesis import SpeechSynthesisDataset  # noqa F401
+from torch.utils.data import DataLoader
+
+from icefall.utils import str2bool
+
+
+class _SeedWorkers:
+    def __init__(self, seed: int):
+        self.seed = seed
+
+    def __call__(self, worker_id: int):
+        fix_random_seed(self.seed + worker_id)
+
+
+class TtsDataModule:
+    """
+    DataModule for tts experiments.
+    It assumes there is always one train and valid dataloader,
+    but there can be multiple test dataloaders (e.g. LibriSpeech test-clean
+    and test-other).
+
+    It contains all the common data pipeline modules used in ASR
+    experiments, e.g.:
+    - dynamic batch size,
+    - bucketing samplers,
+    - cut concatenation,
+    - on-the-fly feature extraction
+
+    This class should be derived for specific corpora used in ASR tasks.
+    """
+
+    def __init__(self, args: argparse.Namespace):
+        self.args = args
+
+    @classmethod
+    def add_arguments(cls, parser: argparse.ArgumentParser):
+        group = parser.add_argument_group(
+            title="TTS data related options",
+            description="These options are used for the preparation of "
+            "PyTorch DataLoaders from Lhotse CutSet's -- they control the "
+            "effective batch sizes, sampling strategies, applied data "
+            "augmentations, etc.",
+        )
+
+        group.add_argument(
+            "--manifest-dir",
+            type=Path,
+            default=Path("data/fbank"),
+            help="Path to directory with train/valid/test cuts.",
+        )
+        group.add_argument(
+            "--max-duration",
+            type=int,
+            default=200.0,
+            help="Maximum pooled recordings duration (seconds) in a "
+            "single batch. You can reduce it if it causes CUDA OOM.",
+        )
+        group.add_argument(
+            "--bucketing-sampler",
+            type=str2bool,
+            default=True,
+            help="When enabled, the batches will come from buckets of "
+            "similar duration (saves padding frames).",
+        )
+        group.add_argument(
+            "--num-buckets",
+            type=int,
+            default=30,
+            help="The number of buckets for the DynamicBucketingSampler"
+            "(you might want to increase it for larger datasets).",
+        )
+
+        group.add_argument(
+            "--on-the-fly-feats",
+            type=str2bool,
+            default=False,
+            help="When enabled, use on-the-fly cut mixing and feature "
+            "extraction. Will drop existing precomputed feature manifests "
+            "if available.",
+        )
+        group.add_argument(
+            "--shuffle",
+            type=str2bool,
+            default=True,
+            help="When enabled (=default), the examples will be "
+            "shuffled for each epoch.",
+        )
+        group.add_argument(
+            "--drop-last",
+            type=str2bool,
+            default=True,
+            help="Whether to drop last batch. Used by sampler.",
+        )
+        group.add_argument(
+            "--return-cuts",
+            type=str2bool,
+            default=False,
+            help="When enabled, each batch will have the "
+            "field: batch['cut'] with the cuts that "
+            "were used to construct it.",
+        )
+        group.add_argument(
+            "--num-workers",
+            type=int,
+            default=2,
+            help="The number of training dataloader workers that "
+            "collect the batches.",
+        )
+
+        group.add_argument(
+            "--input-strategy",
+            type=str,
+            default="PrecomputedFeatures",
+            help="AudioSamples or PrecomputedFeatures",
+        )
+        parser.add_argument(
+            "--prefix",
+            type=str,
+            default="wenetspeech4tts",
+            help="prefix of the manifest file",
+        )
+
+    def train_dataloaders(
+        self,
+        cuts_train: CutSet,
+        sampler_state_dict: Optional[Dict[str, Any]] = None,
+    ) -> DataLoader:
+        """
+        Args:
+          cuts_train:
+            CutSet for training.
+          sampler_state_dict:
+            The state dict for the training sampler.
+        """
+        logging.info("About to create train dataset")
+        train = SpeechSynthesisDataset(
+            return_text=True,
+            return_tokens=False,
+            feature_input_strategy=eval(self.args.input_strategy)(),
+            return_cuts=self.args.return_cuts,
+        )
+
+        if self.args.on_the_fly_feats:
+            raise NotImplementedError(
+                "On-the-fly feature extraction is not implemented yet."
+            )
+
+        if self.args.bucketing_sampler:
+            logging.info("Using DynamicBucketingSampler.")
+            train_sampler = DynamicBucketingSampler(
+                cuts_train,
+                max_duration=self.args.max_duration,
+                shuffle=self.args.shuffle,
+                num_buckets=self.args.num_buckets,
+                buffer_size=self.args.num_buckets * 2000,
+                shuffle_buffer_size=self.args.num_buckets * 5000,
+                drop_last=self.args.drop_last,
+            )
+        else:
+            logging.info("Using SimpleCutSampler.")
+            train_sampler = SimpleCutSampler(
+                cuts_train,
+                max_duration=self.args.max_duration,
+                shuffle=self.args.shuffle,
+            )
+        logging.info("About to create train dataloader")
+
+        if sampler_state_dict is not None:
+            logging.info("Loading sampler state dict")
+            train_sampler.load_state_dict(sampler_state_dict)
+
+        # 'seed' is derived from the current random state, which will have
+        # previously been set in the main process.
+        seed = torch.randint(0, 100000, ()).item()
+        worker_init_fn = _SeedWorkers(seed)
+
+        train_dl = DataLoader(
+            train,
+            sampler=train_sampler,
+            batch_size=None,
+            num_workers=self.args.num_workers,
+            persistent_workers=True,
+            pin_memory=True,
+            worker_init_fn=worker_init_fn,
+        )
+
+        return train_dl
+
+    def valid_dataloaders(self, cuts_valid: CutSet) -> DataLoader:
+        logging.info("About to create dev dataset")
+        if self.args.on_the_fly_feats:
+            raise NotImplementedError(
+                "On-the-fly feature extraction is not implemented yet."
+            )
+        else:
+            validate = SpeechSynthesisDataset(
+                return_text=True,
+                return_tokens=False,
+                feature_input_strategy=eval(self.args.input_strategy)(),
+                return_cuts=self.args.return_cuts,
+            )
+        valid_sampler = DynamicBucketingSampler(
+            cuts_valid,
+            max_duration=self.args.max_duration,
+            num_buckets=self.args.num_buckets,
+            shuffle=False,
+        )
+        logging.info("About to create valid dataloader")
+        valid_dl = DataLoader(
+            validate,
+            sampler=valid_sampler,
+            batch_size=None,
+            num_workers=2,
+            persistent_workers=True,
+            pin_memory=True,
+        )
+
+        return valid_dl
+
+    def test_dataloaders(self, cuts: CutSet) -> DataLoader:
+        logging.info("About to create test dataset")
+        if self.args.on_the_fly_feats:
+            raise NotImplementedError(
+                "On-the-fly feature extraction is not implemented yet."
+            )
+        else:
+            test = SpeechSynthesisDataset(
+                return_text=True,
+                return_tokens=False,
+                feature_input_strategy=eval(self.args.input_strategy)(),
+                return_cuts=self.args.return_cuts,
+            )
+        test_sampler = DynamicBucketingSampler(
+            cuts,
+            max_duration=self.args.max_duration,
+            num_buckets=self.args.num_buckets,
+            shuffle=False,
+        )
+        logging.info("About to create test dataloader")
+        test_dl = DataLoader(
+            test,
+            batch_size=None,
+            sampler=test_sampler,
+            num_workers=self.args.num_workers,
+        )
+        return test_dl
+
+    @lru_cache()
+    def train_cuts(self) -> CutSet:
+        logging.info("About to get train cuts")
+        return load_manifest_lazy(
+            self.args.manifest_dir / f"{self.args.prefix}_cuts_train.jsonl.gz"
+        )
+
+    @lru_cache()
+    def valid_cuts(self) -> CutSet:
+        logging.info("About to get validation cuts")
+        return load_manifest_lazy(
+            self.args.manifest_dir / f"{self.args.prefix}_cuts_valid.jsonl.gz"
+        )
+
+    @lru_cache()
+    def test_cuts(self) -> CutSet:
+        logging.info("About to get test cuts")
+        return load_manifest_lazy(
+            self.args.manifest_dir / f"{self.args.prefix}_cuts_test.jsonl.gz"
+        )

egs/wenetspeech4tts/TTS/f5-tts/utils.py

@@ -0,0 +1 @@
+../../../ljspeech/TTS/matcha/utils.py

egs/wenetspeech4tts/TTS/local/audio.py

@@ -0,0 +1,122 @@
+# Copyright (c) 2024 NVIDIA CORPORATION.
+#   Licensed under the MIT license.
+
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+#   LICENSE is in incl_licenses directory.
+
+import math
+import os
+import pathlib
+import random
+from typing import List, Optional, Tuple
+
+import librosa
+import numpy as np
+import torch
+import torch.utils.data
+from librosa.filters import mel as librosa_mel_fn
+from tqdm import tqdm
+
+# from env import AttrDict
+
+MAX_WAV_VALUE = 32767.0  # NOTE: 32768.0 -1 to prevent int16 overflow (results in popping sound in corner cases)
+
+
+def dynamic_range_compression(x, C=1, clip_val=1e-5):
+    return np.log(np.clip(x, a_min=clip_val, a_max=None) * C)
+
+
+def dynamic_range_decompression(x, C=1):
+    return np.exp(x) / C
+
+
+def dynamic_range_compression_torch(x, C=1, clip_val=1e-5):
+    return torch.log(torch.clamp(x, min=clip_val) * C)
+
+
+def dynamic_range_decompression_torch(x, C=1):
+    return torch.exp(x) / C
+
+
+def spectral_normalize_torch(magnitudes):
+    return dynamic_range_compression_torch(magnitudes)
+
+
+def spectral_de_normalize_torch(magnitudes):
+    return dynamic_range_decompression_torch(magnitudes)
+
+
+mel_basis_cache = {}
+hann_window_cache = {}
+
+
+def mel_spectrogram(
+    y: torch.Tensor,
+    n_fft: int = 1024,
+    num_mels: int = 100,
+    sampling_rate: int = 24_000,
+    hop_size: int = 256,
+    win_size: int = 1024,
+    fmin: int = 0,
+    fmax: int = None,
+    center: bool = False,
+) -> torch.Tensor:
+    """
+    Calculate the mel spectrogram of an input signal.
+    This function uses slaney norm for the librosa mel filterbank (using librosa.filters.mel) and uses Hann window for STFT (using torch.stft).
+
+    Args:
+        y (torch.Tensor): Input signal.
+        n_fft (int): FFT size.
+        num_mels (int): Number of mel bins.
+        sampling_rate (int): Sampling rate of the input signal.
+        hop_size (int): Hop size for STFT.
+        win_size (int): Window size for STFT.
+        fmin (int): Minimum frequency for mel filterbank.
+        fmax (int): Maximum frequency for mel filterbank. If None, defaults to half the sampling rate (fmax = sr / 2.0) inside librosa_mel_fn
+        center (bool): Whether to pad the input to center the frames. Default is False.
+
+    Returns:
+        torch.Tensor: Mel spectrogram.
+    """
+    if torch.min(y) < -1.0:
+        print(f"[WARNING] Min value of input waveform signal is {torch.min(y)}")
+    if torch.max(y) > 1.0:
+        print(f"[WARNING] Max value of input waveform signal is {torch.max(y)}")
+
+    device = y.device
+    key = f"{n_fft}_{num_mels}_{sampling_rate}_{hop_size}_{win_size}_{fmin}_{fmax}_{device}"
+
+    if key not in mel_basis_cache:
+        mel = librosa_mel_fn(
+            sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax
+        )
+        mel_basis_cache[key] = torch.from_numpy(mel).float().to(device)
+        hann_window_cache[key] = torch.hann_window(win_size).to(device)
+
+    mel_basis = mel_basis_cache[key]
+    hann_window = hann_window_cache[key]
+
+    padding = (n_fft - hop_size) // 2
+    y = torch.nn.functional.pad(
+        y.unsqueeze(1), (padding, padding), mode="reflect"
+    ).squeeze(1)
+
+    spec = torch.stft(
+        y,
+        n_fft,
+        hop_length=hop_size,
+        win_length=win_size,
+        window=hann_window,
+        center=center,
+        pad_mode="reflect",
+        normalized=False,
+        onesided=True,
+        return_complex=True,
+    )
+    spec = torch.sqrt(torch.view_as_real(spec).pow(2).sum(-1) + 1e-9)
+
+    mel_spec = torch.matmul(mel_basis, spec)
+    mel_spec = spectral_normalize_torch(mel_spec)
+
+    return mel_spec

egs/wenetspeech4tts/TTS/local/compute_mel_feat.py

@@ -0,0 +1,218 @@
+#!/usr/bin/env python3
+# Copyright    2021-2023  Xiaomi Corp.        (authors: Fangjun Kuang,
+#                                                       Zengwei Yao)
+#
+# See ../../../../LICENSE for clarification regarding multiple authors
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+
+"""
+This file computes fbank features of the LJSpeech dataset.
+It looks for manifests in the directory data/manifests.
+
+The generated fbank features are saved in data/fbank.
+"""
+
+import argparse
+import logging
+import os
+from pathlib import Path
+
+import torch
+from fbank import MatchaFbank, MatchaFbankConfig
+from lhotse import CutSet, LilcomChunkyWriter
+from lhotse.recipes.utils import read_manifests_if_cached
+
+from icefall.utils import get_executor
+
+
+def get_parser():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--num-jobs",
+        type=int,
+        default=1,
+        help="""It specifies the checkpoint to use for decoding.
+        Note: Epoch counts from 1.
+        """,
+    )
+
+    parser.add_argument(
+        "--src-dir",
+        type=Path,
+        default=Path("data/manifests"),
+        help="Path to the manifest files",
+    )
+
+    parser.add_argument(
+        "--output-dir",
+        type=Path,
+        default=Path("data/fbank"),
+        help="Path to the tokenized files",
+    )
+
+    parser.add_argument(
+        "--dataset-parts",
+        type=str,
+        default="Basic",
+        help="Space separated dataset parts",
+    )
+
+    parser.add_argument(
+        "--prefix",
+        type=str,
+        default="wenetspeech4tts",
+        help="prefix of the manifest file",
+    )
+
+    parser.add_argument(
+        "--suffix",
+        type=str,
+        default="jsonl.gz",
+        help="suffix of the manifest file",
+    )
+
+    parser.add_argument(
+        "--split",
+        type=int,
+        default=100,
+        help="Split the cut_set into multiple parts",
+    )
+
+    parser.add_argument(
+        "--resample-to-24kHz",
+        default=True,
+        help="Resample the audio to 24kHz",
+    )
+
+    parser.add_argument(
+        "--extractor",
+        type=str,
+        choices=["bigvgan", "hifigan"],
+        default="bigvgan",
+        help="The type of extractor to use",
+    )
+    return parser
+
+
+def compute_fbank(args):
+    src_dir = Path(args.src_dir)
+    output_dir = Path(args.output_dir)
+    Path(args.output_dir).mkdir(parents=True, exist_ok=True)
+
+    num_jobs = min(args.num_jobs, os.cpu_count())
+    dataset_parts = args.dataset_parts.replace("--dataset-parts", "").strip().split(" ")
+
+    logging.info(f"num_jobs: {num_jobs}")
+    logging.info(f"src_dir: {src_dir}")
+    logging.info(f"output_dir: {output_dir}")
+    logging.info(f"dataset_parts: {dataset_parts}")
+    if args.extractor == "bigvgan":
+        config = MatchaFbankConfig(
+            n_fft=1024,
+            n_mels=100,
+            sampling_rate=24_000,
+            hop_length=256,
+            win_length=1024,
+            f_min=0,
+            f_max=None,
+        )
+    elif args.extractor == "hifigan":
+        config = MatchaFbankConfig(
+            n_fft=1024,
+            n_mels=80,
+            sampling_rate=22050,
+            hop_length=256,
+            win_length=1024,
+            f_min=0,
+            f_max=8000,
+        )
+    else:
+        raise NotImplementedError(f"Extractor {args.extractor} is not implemented")
+
+    extractor = MatchaFbank(config)
+
+    manifests = read_manifests_if_cached(
+        dataset_parts=dataset_parts,
+        output_dir=args.src_dir,
+        prefix=args.prefix,
+        suffix=args.suffix,
+        types=["recordings", "supervisions", "cuts"],
+    )
+
+    with get_executor() as ex:
+        for partition, m in manifests.items():
+            logging.info(
+                f"Processing partition: {partition} CUDA: {torch.cuda.is_available()}"
+            )
+            try:
+                cut_set = CutSet.from_manifests(
+                    recordings=m["recordings"],
+                    supervisions=m["supervisions"],
+                )
+            except Exception:
+                cut_set = m["cuts"]
+
+            if args.split > 1:
+                cut_sets = cut_set.split(args.split)
+            else:
+                cut_sets = [cut_set]
+
+            for idx, part in enumerate(cut_sets):
+                if args.split > 1:
+                    storage_path = f"{args.output_dir}/{args.prefix}_{args.extractor}_{partition}_{idx}"
+                else:
+                    storage_path = (
+                        f"{args.output_dir}/{args.prefix}_{args.extractor}_{partition}"
+                    )
+
+                if args.resample_to_24kHz:
+                    part = part.resample(24000)
+
+                with torch.no_grad():
+                    part = part.compute_and_store_features(
+                        extractor=extractor,
+                        storage_path=storage_path,
+                        num_jobs=num_jobs if ex is None else 64,
+                        executor=ex,
+                        storage_type=LilcomChunkyWriter,
+                    )
+
+                if args.split > 1:
+                    cuts_filename = (
+                        f"{args.prefix}_cuts_{partition}.{idx}.{args.suffix}"
+                    )
+                else:
+                    cuts_filename = f"{args.prefix}_cuts_{partition}.{args.suffix}"
+
+                part.to_file(f"{args.output_dir}/{cuts_filename}")
+                logging.info(f"Saved {cuts_filename}")
+
+
+if __name__ == "__main__":
+    # Torch's multithreaded behavior needs to be disabled or
+    # it wastes a lot of CPU and slow things down.
+    # Do this outside of main() in case it needs to take effect
+    # even when we are not invoking the main (e.g. when spawning subprocesses).
+    torch.set_num_threads(1)
+    torch.set_num_interop_threads(1)
+    formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
+
+    logging.basicConfig(format=formatter, level=logging.INFO)
+
+    args = get_parser().parse_args()
+    compute_fbank(args)

egs/wenetspeech4tts/TTS/local/compute_wer.sh

@@ -0,0 +1,26 @@
+wav_dir=$1
+wav_files=$(ls $wav_dir/*.wav)
+# if wav_files is empty, then exit
+if [ -z "$wav_files" ]; then
+    exit 1
+fi
+label_file=$2
+model_path=local/sherpa-onnx-paraformer-zh-2023-09-14
+
+if [ ! -d $model_path ]; then
+    pip install sherpa-onnx
+    wget -nc https://github.com/k2-fsa/sherpa-onnx/releases/download/asr-models/sherpa-onnx-paraformer-zh-2023-09-14.tar.bz2
+    tar xvf sherpa-onnx-paraformer-zh-2023-09-14.tar.bz2 -C local
+fi
+
+python3 local/offline-decode-files.py  \
+    --tokens=$model_path/tokens.txt \
+    --paraformer=$model_path/model.int8.onnx \
+    --num-threads=2 \
+    --decoding-method=greedy_search \
+    --debug=false \
+    --sample-rate=24000 \
+    --log-dir $wav_dir \
+    --feature-dim=80 \
+    --label $label_file \
+    $wav_files

egs/wenetspeech4tts/TTS/local/fbank.py

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

egs/wenetspeech4tts/TTS/local/offline-decode-files.py

@@ -0,0 +1,495 @@
+#!/usr/bin/env python3
+#
+# Copyright (c)  2023 by manyeyes
+# Copyright (c)  2023  Xiaomi Corporation
+
+"""
+This file demonstrates how to use sherpa-onnx Python API to transcribe
+file(s) with a non-streaming model.
+
+(1) For paraformer
+
+    ./python-api-examples/offline-decode-files.py  \
+      --tokens=/path/to/tokens.txt \
+      --paraformer=/path/to/paraformer.onnx \
+      --num-threads=2 \
+      --decoding-method=greedy_search \
+      --debug=false \
+      --sample-rate=16000 \
+      --feature-dim=80 \
+      /path/to/0.wav \
+      /path/to/1.wav
+
+(2) For transducer models from icefall
+
+    ./python-api-examples/offline-decode-files.py  \
+      --tokens=/path/to/tokens.txt \
+      --encoder=/path/to/encoder.onnx \
+      --decoder=/path/to/decoder.onnx \
+      --joiner=/path/to/joiner.onnx \
+      --num-threads=2 \
+      --decoding-method=greedy_search \
+      --debug=false \
+      --sample-rate=16000 \
+      --feature-dim=80 \
+      /path/to/0.wav \
+      /path/to/1.wav
+
+(3) For CTC models from NeMo
+
+python3 ./python-api-examples/offline-decode-files.py \
+  --tokens=./sherpa-onnx-nemo-ctc-en-citrinet-512/tokens.txt \
+  --nemo-ctc=./sherpa-onnx-nemo-ctc-en-citrinet-512/model.onnx \
+  --num-threads=2 \
+  --decoding-method=greedy_search \
+  --debug=false \
+  ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/0.wav \
+  ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/1.wav \
+  ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/8k.wav
+
+(4) For Whisper models
+
+python3 ./python-api-examples/offline-decode-files.py \
+  --whisper-encoder=./sherpa-onnx-whisper-base.en/base.en-encoder.int8.onnx \
+  --whisper-decoder=./sherpa-onnx-whisper-base.en/base.en-decoder.int8.onnx \
+  --tokens=./sherpa-onnx-whisper-base.en/base.en-tokens.txt \
+  --whisper-task=transcribe \
+  --num-threads=1 \
+  ./sherpa-onnx-whisper-base.en/test_wavs/0.wav \
+  ./sherpa-onnx-whisper-base.en/test_wavs/1.wav \
+  ./sherpa-onnx-whisper-base.en/test_wavs/8k.wav
+
+(5) For CTC models from WeNet
+
+python3 ./python-api-examples/offline-decode-files.py \
+  --wenet-ctc=./sherpa-onnx-zh-wenet-wenetspeech/model.onnx \
+  --tokens=./sherpa-onnx-zh-wenet-wenetspeech/tokens.txt \
+  ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/0.wav \
+  ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/1.wav \
+  ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/8k.wav
+
+(6) For tdnn models of the yesno recipe from icefall
+
+python3 ./python-api-examples/offline-decode-files.py \
+  --sample-rate=8000 \
+  --feature-dim=23 \
+  --tdnn-model=./sherpa-onnx-tdnn-yesno/model-epoch-14-avg-2.onnx \
+  --tokens=./sherpa-onnx-tdnn-yesno/tokens.txt \
+  ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_0_1_0_0_0_1.wav \
+  ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_1_0_0_0_1_0.wav \
+  ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_1_0_0_1_1_1.wav
+
+Please refer to
+https://k2-fsa.github.io/sherpa/onnx/index.html
+to install sherpa-onnx and to download non-streaming pre-trained models
+used in this file.
+"""
+import argparse
+import time
+import wave
+from pathlib import Path
+from typing import List, Tuple
+
+import numpy as np
+import sherpa_onnx
+import soundfile as sf
+
+
+def get_args():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--tokens",
+        type=str,
+        help="Path to tokens.txt",
+    )
+
+    parser.add_argument(
+        "--hotwords-file",
+        type=str,
+        default="",
+        help="""
+        The file containing hotwords, one words/phrases per line, like
+        HELLO WORLD
+        你好世界
+        """,
+    )
+
+    parser.add_argument(
+        "--hotwords-score",
+        type=float,
+        default=1.5,
+        help="""
+        The hotword score of each token for biasing word/phrase. Used only if
+        --hotwords-file is given.
+        """,
+    )
+
+    parser.add_argument(
+        "--modeling-unit",
+        type=str,
+        default="",
+        help="""
+        The modeling unit of the model, valid values are cjkchar, bpe, cjkchar+bpe.
+        Used only when hotwords-file is given.
+        """,
+    )
+
+    parser.add_argument(
+        "--bpe-vocab",
+        type=str,
+        default="",
+        help="""
+        The path to the bpe vocabulary, the bpe vocabulary is generated by
+        sentencepiece, you can also export the bpe vocabulary through a bpe model
+        by `scripts/export_bpe_vocab.py`. Used only when hotwords-file is given
+        and modeling-unit is bpe or cjkchar+bpe.
+        """,
+    )
+
+    parser.add_argument(
+        "--encoder",
+        default="",
+        type=str,
+        help="Path to the encoder model",
+    )
+
+    parser.add_argument(
+        "--decoder",
+        default="",
+        type=str,
+        help="Path to the decoder model",
+    )
+
+    parser.add_argument(
+        "--joiner",
+        default="",
+        type=str,
+        help="Path to the joiner model",
+    )
+
+    parser.add_argument(
+        "--paraformer",
+        default="",
+        type=str,
+        help="Path to the model.onnx from Paraformer",
+    )
+
+    parser.add_argument(
+        "--nemo-ctc",
+        default="",
+        type=str,
+        help="Path to the model.onnx from NeMo CTC",
+    )
+
+    parser.add_argument(
+        "--wenet-ctc",
+        default="",
+        type=str,
+        help="Path to the model.onnx from WeNet CTC",
+    )
+
+    parser.add_argument(
+        "--tdnn-model",
+        default="",
+        type=str,
+        help="Path to the model.onnx for the tdnn model of the yesno recipe",
+    )
+
+    parser.add_argument(
+        "--num-threads",
+        type=int,
+        default=1,
+        help="Number of threads for neural network computation",
+    )
+
+    parser.add_argument(
+        "--whisper-encoder",
+        default="",
+        type=str,
+        help="Path to whisper encoder model",
+    )
+
+    parser.add_argument(
+        "--whisper-decoder",
+        default="",
+        type=str,
+        help="Path to whisper decoder model",
+    )
+
+    parser.add_argument(
+        "--whisper-language",
+        default="",
+        type=str,
+        help="""It specifies the spoken language in the input audio file.
+        Example values: en, fr, de, zh, jp.
+        Available languages for multilingual models can be found at
+        https://github.com/openai/whisper/blob/main/whisper/tokenizer.py#L10
+        If not specified, we infer the language from the input audio file.
+        """,
+    )
+
+    parser.add_argument(
+        "--whisper-task",
+        default="transcribe",
+        choices=["transcribe", "translate"],
+        type=str,
+        help="""For multilingual models, if you specify translate, the output
+        will be in English.
+        """,
+    )
+
+    parser.add_argument(
+        "--whisper-tail-paddings",
+        default=-1,
+        type=int,
+        help="""Number of tail padding frames.
+        We have removed the 30-second constraint from whisper, so you need to
+        choose the amount of tail padding frames by yourself.
+        Use -1 to use a default value for tail padding.
+        """,
+    )
+
+    parser.add_argument(
+        "--blank-penalty",
+        type=float,
+        default=0.0,
+        help="""
+        The penalty applied on blank symbol during decoding.
+        Note: It is a positive value that would be applied to logits like
+        this `logits[:, 0] -= blank_penalty` (suppose logits.shape is
+        [batch_size, vocab] and blank id is 0).
+        """,
+    )
+
+    parser.add_argument(
+        "--decoding-method",
+        type=str,
+        default="greedy_search",
+        help="Valid values are greedy_search and modified_beam_search",
+    )
+    parser.add_argument(
+        "--debug",
+        type=bool,
+        default=False,
+        help="True to show debug messages",
+    )
+
+    parser.add_argument(
+        "--sample-rate",
+        type=int,
+        default=16000,
+        help="""Sample rate of the feature extractor. Must match the one
+        expected  by the model. Note: The input sound files can have a
+        different sample rate from this argument.""",
+    )
+
+    parser.add_argument(
+        "--feature-dim",
+        type=int,
+        default=80,
+        help="Feature dimension. Must match the one expected by the model",
+    )
+
+    parser.add_argument(
+        "sound_files",
+        type=str,
+        nargs="+",
+        help="The input sound file(s) to decode. Each file must be of WAVE"
+        "format with a single channel, and each sample has 16-bit, "
+        "i.e., int16_t. "
+        "The sample rate of the file can be arbitrary and does not need to "
+        "be 16 kHz",
+    )
+
+    parser.add_argument(
+        "--name",
+        type=str,
+        default="",
+        help="The directory containing the input sound files to decode",
+    )
+
+    parser.add_argument(
+        "--log-dir",
+        type=str,
+        default="",
+        help="The directory containing the input sound files to decode",
+    )
+
+    parser.add_argument(
+        "--label",
+        type=str,
+        default=None,
+        help="wav_base_name label",
+    )
+    return parser.parse_args()
+
+
+def assert_file_exists(filename: str):
+    assert Path(filename).is_file(), (
+        f"{filename} does not exist!\n"
+        "Please refer to "
+        "https://k2-fsa.github.io/sherpa/onnx/pretrained_models/index.html to download it"
+    )
+
+
+def read_wave(wave_filename: str) -> Tuple[np.ndarray, int]:
+    """
+    Args:
+      wave_filename:
+        Path to a wave file. It should be single channel and can be of type
+        32-bit floating point PCM. Its sample rate does not need to be 24kHz.
+
+    Returns:
+      Return a tuple containing:
+       - A 1-D array of dtype np.float32 containing the samples,
+         which are normalized to the range [-1, 1].
+       - Sample rate of the wave file.
+    """
+
+    samples, sample_rate = sf.read(wave_filename, dtype="float32")
+    assert (
+        samples.ndim == 1
+    ), f"Expected single channel, but got {samples.ndim} channels."
+
+    samples_float32 = samples.astype(np.float32)
+
+    return samples_float32, sample_rate
+
+
+def normalize_text_alimeeting(text: str) -> str:
+    """
+    Text normalization similar to M2MeT challenge baseline.
+    See: https://github.com/yufan-aslp/AliMeeting/blob/main/asr/local/text_normalize.pl
+    """
+    import re
+
+    text = text.replace(" ", "")
+    text = text.replace("<sil>", "")
+    text = text.replace("<%>", "")
+    text = text.replace("<->", "")
+    text = text.replace("<$>", "")
+    text = text.replace("<#>", "")
+    text = text.replace("<_>", "")
+    text = text.replace("<space>", "")
+    text = text.replace("`", "")
+    text = text.replace("&", "")
+    text = text.replace(",", "")
+    if re.search("[a-zA-Z]", text):
+        text = text.upper()
+    text = text.replace("Ａ", "A")
+    text = text.replace("ａ", "A")
+    text = text.replace("ｂ", "B")
+    text = text.replace("ｃ", "C")
+    text = text.replace("ｋ", "K")
+    text = text.replace("ｔ", "T")
+    text = text.replace("，", "")
+    text = text.replace("丶", "")
+    text = text.replace("。", "")
+    text = text.replace("、", "")
+    text = text.replace("？", "")
+    return text
+
+
+def main():
+    args = get_args()
+    assert_file_exists(args.tokens)
+    assert args.num_threads > 0, args.num_threads
+
+    assert len(args.nemo_ctc) == 0, args.nemo_ctc
+    assert len(args.wenet_ctc) == 0, args.wenet_ctc
+    assert len(args.whisper_encoder) == 0, args.whisper_encoder
+    assert len(args.whisper_decoder) == 0, args.whisper_decoder
+    assert len(args.tdnn_model) == 0, args.tdnn_model
+
+    assert_file_exists(args.paraformer)
+
+    recognizer = sherpa_onnx.OfflineRecognizer.from_paraformer(
+        paraformer=args.paraformer,
+        tokens=args.tokens,
+        num_threads=args.num_threads,
+        sample_rate=args.sample_rate,
+        feature_dim=args.feature_dim,
+        decoding_method=args.decoding_method,
+        debug=args.debug,
+    )
+
+    print("Started!")
+    start_time = time.time()
+
+    streams, results = [], []
+    total_duration = 0
+
+    for i, wave_filename in enumerate(args.sound_files):
+        assert_file_exists(wave_filename)
+        samples, sample_rate = read_wave(wave_filename)
+        duration = len(samples) / sample_rate
+        total_duration += duration
+        s = recognizer.create_stream()
+        s.accept_waveform(sample_rate, samples)
+
+        streams.append(s)
+        if i % 10 == 0:
+            recognizer.decode_streams(streams)
+            results += [s.result.text for s in streams]
+            streams = []
+            print(f"Processed {i} files")
+        # process the last batch
+    if streams:
+        recognizer.decode_streams(streams)
+        results += [s.result.text for s in streams]
+    end_time = time.time()
+    print("Done!")
+
+    results_dict = {}
+    for wave_filename, result in zip(args.sound_files, results):
+        print(f"{wave_filename}\n{result}")
+        print("-" * 10)
+        wave_basename = Path(wave_filename).stem
+        results_dict[wave_basename] = result
+
+    elapsed_seconds = end_time - start_time
+    rtf = elapsed_seconds / total_duration
+    print(f"num_threads: {args.num_threads}")
+    print(f"decoding_method: {args.decoding_method}")
+    print(f"Wave duration: {total_duration:.3f} s")
+    print(f"Elapsed time: {elapsed_seconds:.3f} s")
+    print(
+        f"Real time factor (RTF): {elapsed_seconds:.3f}/{total_duration:.3f} = {rtf:.3f}"
+    )
+    if args.label:
+        from icefall.utils import store_transcripts, write_error_stats
+
+        labels_dict = {}
+        with open(args.label, "r") as f:
+            for line in f:
+                # fields = line.strip().split(" ")
+                # fields = [item for item in fields if item]
+                # assert len(fields) == 4
+                # prompt_text, prompt_audio, text, audio_path = fields
+
+                fields = line.strip().split("|")
+                fields = [item for item in fields if item]
+                assert len(fields) == 4
+                audio_path, prompt_text, prompt_audio, text = fields
+                labels_dict[Path(audio_path).stem] = normalize_text_alimeeting(text)
+
+        final_results = []
+        for key, value in results_dict.items():
+            final_results.append((key, labels_dict[key], value))
+
+        store_transcripts(
+            filename=f"{args.log_dir}/recogs-{args.name}.txt", texts=final_results
+        )
+        with open(f"{args.log_dir}/errs-{args.name}.txt", "w") as f:
+            write_error_stats(f, "test-set", final_results, enable_log=True)
+
+        with open(f"{args.log_dir}/errs-{args.name}.txt", "r") as f:
+            print(f.readline())  # WER
+            print(f.readline())  # Detailed errors
+
+
+if __name__ == "__main__":
+    main()

egs/wenetspeech4tts/TTS/prepare.sh

@@ -98,3 +98,44 @@ if [ $stage -le 4 ] && [ $stop_stage -ge 4 ]; then
   fi
   python3 ./local/display_manifest_statistics.py --manifest-dir ${audio_feats_dir}
 fi
+
+subset="Basic"
+prefix="wenetspeech4tts"
+if [ $stage -le 5 ] && [ $stop_stage -ge 5 ]; then
+  log "Stage 5: Generate fbank (used by ./f5-tts)"
+  mkdir -p data/fbank
+  if [ ! -e data/fbank/.${prefix}.done ]; then
+    ./local/compute_mel_feat.py --dataset-parts $subset --split 100
+    touch data/fbank/.${prefix}.done
+  fi
+fi
+
+if [ $stage -le 6 ] && [ $stop_stage -ge 6 ]; then
+  log "Stage 7: Split the ${prefix} cuts into train, valid and test sets (used by ./f5-tts)"
+  if [ ! -f data/fbank/${prefix}_cuts_${subset}.jsonl.gz ]; then
+    echo "Combining ${prefix} cuts"
+    pieces=$(find data/fbank/ -name "${prefix}_cuts_${subset}.*.jsonl.gz")
+    lhotse combine $pieces data/fbank/${prefix}_cuts_${subset}.jsonl.gz
+  fi
+  if [ ! -e data/fbank/.${prefix}_split.done ]; then
+    echo "Splitting ${prefix} cuts into train, valid and test sets"
+
+    lhotse subset --last 800 \
+      data/fbank/${prefix}_cuts_${subset}.jsonl.gz \
+      data/fbank/${prefix}_cuts_validtest.jsonl.gz
+    lhotse subset --first 400 \
+      data/fbank/${prefix}_cuts_validtest.jsonl.gz \
+      data/fbank/${prefix}_cuts_valid.jsonl.gz
+    lhotse subset --last 400 \
+      data/fbank/${prefix}_cuts_validtest.jsonl.gz \
+      data/fbank/${prefix}_cuts_test.jsonl.gz
+
+    rm data/fbank/${prefix}_cuts_validtest.jsonl.gz
+
+    n=$(( $(gunzip -c data/fbank/${prefix}_cuts_${subset}.jsonl.gz | wc -l) - 800 ))
+    lhotse subset --first $n  \
+      data/fbank/${prefix}_cuts_${subset}.jsonl.gz \
+      data/fbank/${prefix}_cuts_train.jsonl.gz
+      touch data/fbank/.${prefix}_split.done
+  fi
+fi